diff --git a/.editorconfig b/.editorconfig
index 9aac58c1..70c960be 100644
--- a/.editorconfig
+++ b/.editorconfig
@@ -11,3 +11,7 @@ insert_final_newline = true
 
 [*.md]
 trim_trailing_whitespace = false
+
+[*.{*rc,json}]
+indent_style = space
+indent_size = 2
diff --git a/.eslintrc b/.eslintrc
new file mode 100644
index 00000000..4367b995
--- /dev/null
+++ b/.eslintrc
@@ -0,0 +1,19 @@
+{
+  "extends": [
+    "plugin:prettier/recommended"
+  ],
+  "ignorePatterns": ["www/**/*.js"],
+  "env": {
+    "es6": true
+  },
+  "rules": {
+    "indent": [
+      2,
+      "tab",
+      {
+        "SwitchCase": 1
+      }
+    ],
+    "no-with": 2
+  }
+}
diff --git a/.github/FUNDING.yml b/.github/FUNDING.yml
new file mode 100644
index 00000000..17cf87e9
--- /dev/null
+++ b/.github/FUNDING.yml
@@ -0,0 +1,2 @@
+github: [hthetiot]
+custom: ["https://www.paypal.me/hthetiot", sylaps.biz]
diff --git a/.github/ISSUE_TEMPLATE.md b/.github/ISSUE_TEMPLATE.md
new file mode 100644
index 00000000..75d57160
--- /dev/null
+++ b/.github/ISSUE_TEMPLATE.md
@@ -0,0 +1,29 @@
+# YOU MUST read first!
+Please use [Community Forum](https://github.com/cordova-rtc/cordova-plugin-iosrtc/discussions) for general technical discussions and questions.
+
+- [ ] **I have used Google with the error message** or bug in association with the library and Cordova words to make sure the issue I'm reporting is only related to iOSRTC.
+- [ ] **I have provided steps to reproduce** (e.g. using sample app code https://github.com/cordova-rtc/cordova-plugin-iosrtc-sample or updated `extra/renderer-and-libwebrtc-tests.js` file).
+- [ ] **I have provided versions** of third party library name, ios, Xcode and plugin version and adapter.js version if used.
+
+**Note: If the checkboxes above are not checked (which you do after the issue is posted), the issue will be closed, removing this checkbox will result in automatic closed issue.**
+
+#### Versions affected
+
+* **Cordova version (e.g 7.1.0)**:
+* **Cordova iOS version (e.g 5.1.0)**:
+* **Plugin version (e.g 6.0.12)**:
+* **iOS version (e.g 10.2)**: 
+* **Xcode version (e.g 11.1 - 11A1027)**:
+* **WebRTC-adapter version  (e.g. 7.4.0)**:
+* **WebRTC Framework version (e.g. JSSip 3.1.2)**:
+
+## Description
+
+
+## Steps to reproduce 
+
+
+## Expected results
+
+
+## Actual results
diff --git a/.github/workflows/codeql-analysis.yml b/.github/workflows/codeql-analysis.yml
new file mode 100644
index 00000000..3e024a94
--- /dev/null
+++ b/.github/workflows/codeql-analysis.yml
@@ -0,0 +1,70 @@
+# For most projects, this workflow file will not need changing; you simply need
+# to commit it to your repository.
+#
+# You may wish to alter this file to override the set of languages analyzed,
+# or to provide custom queries or build logic.
+#
+# ******** NOTE ********
+# We have attempted to detect the languages in your repository. Please check
+# the `language` matrix defined below to confirm you have the correct set of
+# supported CodeQL languages.
+#
+name: "CodeQL"
+
+on:
+  push:
+    branches: [ master ]
+  pull_request:
+    # The branches below must be a subset of the branches above
+    branches: [ master ]
+  schedule:
+    - cron: '29 4 * * 5'
+
+jobs:
+  analyze:
+    name: Analyze
+    runs-on: ubuntu-latest
+    permissions:
+      actions: read
+      contents: read
+      security-events: write
+
+    strategy:
+      fail-fast: false
+      matrix:
+        language: [ 'javascript' ]
+        # CodeQL supports [ 'cpp', 'csharp', 'go', 'java', 'javascript', 'python', 'ruby' ]
+        # Learn more about CodeQL language support at https://aka.ms/codeql-docs/language-support
+
+    steps:
+    - name: Checkout repository
+      uses: actions/checkout@v3
+
+    # Initializes the CodeQL tools for scanning.
+    - name: Initialize CodeQL
+      uses: github/codeql-action/init@v2
+      with:
+        languages: ${{ matrix.language }}
+        # If you wish to specify custom queries, you can do so here or in a config file.
+        # By default, queries listed here will override any specified in a config file.
+        # Prefix the list here with "+" to use these queries and those in the config file.
+        # queries: ./path/to/local/query, your-org/your-repo/queries@main
+
+    # Autobuild attempts to build any compiled languages  (C/C++, C#, or Java).
+    # If this step fails, then you should remove it and run the build manually (see below)
+    - name: Autobuild
+      uses: github/codeql-action/autobuild@v2
+
+    # ℹ️ Command-line programs to run using the OS shell.
+    # 📚 See https://docs.github.com/en/actions/using-workflows/workflow-syntax-for-github-actions#jobsjob_idstepsrun
+
+    # ✏️ If the Autobuild fails above, remove it and uncomment the following three lines
+    #    and modify them (or add more) to build your code if your project
+    #    uses a compiled language
+
+    #- run: |
+    #   make bootstrap
+    #   make release
+
+    - name: Perform CodeQL Analysis
+      uses: github/codeql-action/analyze@v2
diff --git a/.github/workflows/main.yml b/.github/workflows/main.yml
new file mode 100644
index 00000000..f36d17c1
--- /dev/null
+++ b/.github/workflows/main.yml
@@ -0,0 +1,66 @@
+# This is a basic workflow to help you get started with Actions
+
+name: Cordova GitHub CI #
+
+# Controls when the workflow will run
+on:
+  # Triggers the workflow on push or pull request events but only for the master branch
+  push:
+    branches: [ master ]
+  pull_request:
+    branches: [ master ]
+
+  # Allows you to run this workflow manually from the Actions tab
+  workflow_dispatch:
+
+# A workflow run is made up of one or more jobs that can run sequentially or in parallel
+jobs:
+  # This workflow contains a single job called "build"
+  test:
+    # The type of runner that the job will run on
+    runs-on: macOS-12
+
+    # Steps represent a sequence of tasks that will be executed as part of the job
+    steps:
+      # Checks-out your repository under $GITHUB_WORKSPACE, so your job can access it
+      - uses: actions/checkout@v3
+
+      - name: Setup NodeJS
+        uses: actions/setup-node@v3
+        with:
+          node-version: "20"
+
+      - name: Cache Node.js modules
+        uses: actions/cache@v3
+        with:
+          # npm cache files are stored in `~/.npm` on Linux/macOS
+          path: ~/.npm
+          key: ${{ runner.OS }}-node-${{ hashFiles('**/package-lock.json') }}
+          restore-keys: |
+            ${{ runner.OS }}-node-
+            ${{ runner.OS }}-
+
+      - name: Install Dependencies
+        run: npm ci
+
+      - name: Lint Plugin
+        run: npm run lint
+
+      - name: Build Plugin
+        run: npm run build
+
+      - name: Install Cordova CLI
+        run: |
+          mkdir ~/.npm-global
+          npm config set prefix '~/.npm-global'
+          export PATH=~/.npm-global/bin:$PATH
+          npm install cordova@10 -g
+
+      - name: Create Cordova app, install plugin then build app
+        run: |
+          npm config set prefix '~/.npm-global'
+          export PATH=~/.npm-global/bin:$PATH
+          cd ..; cordova create myApp org.apache.cordova.myApp myApp; cd myApp;
+          cordova platform add ios@latest --verbose
+          cordova plugin add ../cordova-plugin-iosrtc --verbose
+          cordova build ios --verbose
diff --git a/.gitignore b/.gitignore
index 805609be..b23414db 100644
--- a/.gitignore
+++ b/.gitignore
@@ -1,3 +1,6 @@
 /node_modules/
 /TODO
 /NO_GIT/
+.DS_Store
+.idea
+lib/WebRTC.framework/
\ No newline at end of file
diff --git a/.jscsrc b/.jscsrc
deleted file mode 100644
index 40e1e8d0..00000000
--- a/.jscsrc
+++ /dev/null
@@ -1,7 +0,0 @@
-{
-	"preset": "crockford",
-	"validateIndentation": "\t",
-	"disallowKeywords": ["with"],
-	"disallowDanglingUnderscores": null,
-	"requireVarDeclFirst": null
-}
diff --git a/.jshintrc b/.jshintrc
index 9d32ec17..b32ec80d 100644
--- a/.jshintrc
+++ b/.jshintrc
@@ -1,32 +1,34 @@
 {
-	"bitwise": false,
-	"curly": true,
-	"eqeqeq": true,
-	"forin": true,
-	"freeze": true,
-	"latedef": "function",
-	"noarg": true,
-	"nonbsp": true,
-	"nonew": true,
-	"plusplus": false,
-	"undef": true,
-	"unused": true,
-	"strict": false,
-	"maxparams": 6,
-	"maxdepth": 4,
-	"maxstatements": false,
-	"maxlen": 210,
+  "bitwise": false,
+  "curly": true,
+  "eqeqeq": true,
+  "forin": true,
+  "freeze": true,
+  "latedef": "function",
+  "noarg": true,
+  "nonbsp": true,
+  "nonew": true,
+  "plusplus": false,
+  "undef": true,
+  "unused": true,
+  "strict": false,
+  "maxparams": 6,
+  "maxdepth": 4,
+  "maxstatements": false,
+  "maxlen": 210,
+  "laxbreak": true,
 
-	"browser": true,
-	"browserify": true,
-	"devel": false,
-	"jquery": false,
-	"mocha": false,
-	"node": true,
-	"shelljs": false,
-	"worker": false,
+  "browser": true,
+  "browserify": true,
+  "devel": false,
+  "jquery": false,
+  "mocha": false,
+  "node": true,
+  "shelljs": false,
+  "worker": false,
+  "esversion": 6,
 
-	"globals": {
-		"Promise": false
-	}
+  "globals": {
+    "Promise": false
+  }
 }
diff --git a/.prettierrc b/.prettierrc
new file mode 100644
index 00000000..ff2677ef
--- /dev/null
+++ b/.prettierrc
@@ -0,0 +1,6 @@
+{
+	"useTabs": true,
+	"singleQuote": true,
+	"trailingComma": "none",
+	"printWidth": 100
+}
diff --git a/.travis.yml b/.travis.yml
new file mode 100644
index 00000000..0ee5d3b5
--- /dev/null
+++ b/.travis.yml
@@ -0,0 +1,118 @@
+sudo: required
+branches:
+  only:
+    - master
+git:
+  depth: 1
+env:
+  global:
+    - CORDOVA_PLATFORM=browser
+    - NODE_VERSION=10
+    - CORDOVA_VERSION=10
+    - ANGULAR_CLI_VERSION=8
+    - IONIC_VERSION=6
+    - ANDROID_API_LEVEL=30
+    - ANDROID_SDK_VERSION=30
+    - ANDROID_BUILD_TOOLS_VERSION=30.0.3
+    - ANDROID_ABI=armeabi-v7a
+    - JAVA_HOME=/usr/lib/jvm/java-8-oracle
+install:
+  - "nvm install $NODE_VERSION"
+  - "npm install -g gulp-cli"
+  - "npm install -g cordova@$CORDOVA_VERSION"
+  - "npm install -g @angular/cli@$ANGULAR_CLI_VERSION"
+  - "npm install -g @ionic/cli@$IONIC_VERSION"
+  - "npm install -g cordova-res"
+  - "npm ci"
+matrix:
+  include:
+    - language: node_js
+      env: CORDOVA_PLATFORM=browser
+      node_js:
+        - "12"
+      cache:
+        directories:
+          - node_modules
+          - plugins
+          - platforms
+    - language: android
+      os: linux
+      env: CORDOVA_PLATFORM=android
+      jdk: oraclejdk8
+      dist: xenial
+      addons:
+        apt:
+          packages:
+            - oracle-java8-installer
+            - oracle-java8-set-default
+            - jq
+      android:
+        components:
+          # Uncomment the lines below if you want to
+          # use the latest revision of Android SDK Tools
+          - tools
+          - platform-tools
+          # The BuildTools version used
+          - build-tools-29.0.2
+          # The SDK version used to compile
+          - android-29
+          # Additional components
+          - extra-google-google_play_services
+          - extra-google-m2repository
+          - extra-android-m2repository
+          # Specify at least one system image,
+          # if you need to run emulator(s) during your tests
+        licenses:
+          - android-sdk-preview-license-.+
+          - android-sdk-license-.+
+          - android-sdk-license-29
+          - google-gdk-license-.+
+      before_install:
+          - export LANG=en_US.UTF-8
+          - touch $HOME/.android/repositories.cfg
+          - yes | sdkmanager "platforms;android-$ANDROID_API_LEVEL"
+          - yes | sdkmanager "build-tools;$ANDROID_BUILD_TOOLS_VERSION"
+      before_cache:
+        - "rm -f  $HOME/.gradle/caches/modules-2/modules-2.lock"
+        - "rm -fr $HOME/.gradle/caches/*/plugin-resolution/"
+      cache:
+        directories:
+          - $HOME/.gradle/caches/
+          - $HOME/.gradle/wrapper/
+          - $HOME/.android/build-cache
+          - node_modules
+    - language: objective-c
+      os: osx
+      osx_image: xcode11
+      env: CORDOVA_PLATFORM=ios
+      before_install:
+        - rvm use system
+        #- sudo gem install bundler
+        #- sudo gem install xcpretty --no-rdoc --no-ri --no-document --quiet
+        #- sudo gem install xcpretty-travis-formatter --no-rdoc --no-ri --no-document --quiet
+        #- sudo gem install cocoapods --pre # Since Travis is not always on latest version
+      cache:
+        directories:
+          - node_modules
+      after_script:
+        # Test ios_arch
+        - "cordova platform remove $CORDOVA_PLATFORM --verbose"
+        - "cordova platform add $CORDOVA_PLATFORM@latest --verbose"
+        #- "cordova build $CORDOVA_PLATFORM --verbose --device"
+script:
+  # Get current branch and repository
+  - export BRANCH=$(if [ "$TRAVIS_PULL_REQUEST" == "false" ]; then echo $TRAVIS_BRANCH; else echo $TRAVIS_PULL_REQUEST_BRANCH; fi)
+  - export REPO_SLUG=$(if [ "$TRAVIS_PULL_REQUEST" == "false" ]; then echo $TRAVIS_REPO_SLUG; else echo $TRAVIS_PULL_REQUEST_SLUG; fi)
+  # Lint and Build plugin
+  - "npm run lint"
+  - "gulp browserify"
+  # Test Plugin platform build with ionic
+  - "ionic start myApp blank; cd myApp;"
+  - "ionic cordova platform add $CORDOVA_PLATFORM@latest --verbose"
+  - "ionic cordova plugin add https://github.com/$REPO_SLUG.git#$BRANCH --verbose"
+  - "ionic cordova build $CORDOVA_PLATFORM --verbose"
+  # Test Plugin platform build with cordova only
+  - "cordova create myApp org.apache.cordova.myApp myApp; cd myApp;"
+  - "cordova platform add $CORDOVA_PLATFORM@latest --verbose"
+  - "cordova plugin add https://github.com/$REPO_SLUG.git#$BRANCH --verbose"
+  - "cordova build $CORDOVA_PLATFORM --verbose"
diff --git a/CHANGELOG.md b/CHANGELOG.md
index f82903d8..13c081b1 100644
--- a/CHANGELOG.md
+++ b/CHANGELOG.md
@@ -1,3 +1,285 @@
+#### Version 8.0.4
+* fix: spec compliant RTCStatsReport #769 via @jerry2013
+* fix: add setDefaultAudioOutput to source #768 via @mjcctech and @jerry2013
+
+#### Version 8.0.3
+* fix: PluginRTCPeerConnection: cyclic reference leak fixed #754 via @RSATom
+* fix: update npm packages with audit security warnings
+
+#### Version 8.0.2
+* feat: add github action CI build
+* fix: update npm packages with audit security warnings
+* feat: Added createDTMFSender for RTCRtpSender via @buckun
+
+#### Version 8.0.1
+* fix: RTCPeerConnection.removeTrack support for sender without track
+* feat: add support for local description rollback #704 via @slavchev
+
+#### Version 8.0.0
+* feat: implement RTCPeerConnection.addTransceiver #589 via @agelito
+* fix: RtpSenders inconsistency after removeTrack call #702 via @RSATom
+* fix: RTCRtpSender MediaStreamTrack replace #699 via @RSATom
+* fix: requestPermission returns true if AVAuthorizationStatus is notDetermined #692
+* fix: crash when used with Janus Audiobridge #691 via @RSATom
+* fix: Warning [LayoutConstraints] Unable to simultaneously satisfy constraints. #422 via @slavchev
+* fix: crash when used with Janus Audiobridge #691 via @RSATom
+* feat: extend transceiver/sender/receiver functionality #664 via @slavchev
+* fix: updateTransceiversState call #661 @ducile
+* fix: update to WebRTC.framework M87
+* fix: update to WebRTC.framework M84
+
+#### Version 7.0.0
+* fix: update to WebRTC.framework M79
+* fix: update to WebRTC.framework M75
+
+#### Version 6.0.21
+* fix: canvas drawImage memory leak fix #681
+* fix: warning [LayoutConstraints] Unable to simultaneously satisfy constraints.
+
+#### Version 6.0.20
+* fix: Remove extmap-allow-mixed sdp header on RTCPeerConnection.prototype.setRemoteDescription to handle chrome 89
+
+#### Version 6.0.19
+* fix: MediaStreamRenderer doesn't call PluginMediaStreamTrack.unregisterRender = stuck reference. #649
+* fix: ios not build after cordova-plugin-iosrtc #642
+
+#### Version 6.0.18
+* fix: update ios_arch.js to generate WebRTC.xcframework, Include WebRTC.xcframework instead of WebRTC.framework
+* fix: ignore empty IceCandidate, fix RTCPeerConnection_addIceCandidate pluginRTCPeerConnection argument is NIL
+* fix: allow registerGlobals to be called multiple time without error: Attempting to change value of a readonly property.
+
+#### Version 6.0.17
+* fix: fix possible duplicate pluginMediaStream and pluginMediaTrack causing no video
+* fix: fix plugin media stream renderer save leak #609
+* fix: fix navigator.mediaDevices.GetUserMedia webrtc-adapter causing overide of navigator.mediaDevices on ios 14.3 #618
+* fix: fix gUM crash with invalid audio deviceId #630
+* fix: fix event dispatch in replace track in rtcrtpsender #619
+* fix: fix MediaStreamTrack: added getCapabilities().deviceId support for cam device media streams #612
+* fix: fix iosrtcPlugin#dump()
+* fix: Implement MediaStreamTrack.getConstraints|applyConstraints SHAM
+* fix: fix iOS 14.3 getUserMedia fails #618
+
+#### Version 6.0.16
+* fix: correct canvas drawImage SHIM for VideoElement
+* fix: ensure proper camera orientation for first launch
+* fix: fix hook possible Cannot read property '1' of null at getProjectName #597
+* fix: fix possible error on icecandidate event on RTCPeerConnection
+* fix: fix RTCIceCandidate typo
+* fix: fix PluginMediaStreamTrack.eventListener possible Fatal error #602
+* fix: fix MediaStreamTrack lifecycle, clone and make GetStats logs silent. #605 via @samgabriel
+* fix: remove calling checkActive for local streams connected to PeerConnection #607
+* fix: lint codestyle fixed #608 via @RSATom
+* fix: fix fatal error Unexpectedly found nil while unwrapping an Optional value: file PluginMediaStreamTrack.swift, line 123
+* feat: use background-color from video element css #586 via @calebboyd
+* feat: prettier for JS styling #598 via @dgreif
+
+#### Version 6.0.15
+* fix: refactor pluginMediaTracks lifecyle in PluginRTCPeerConnection
+* feat: add canvas drawImage SHIM for VideoElement
+* feat: add (MediaStream|MediaStreamTrack).prototype.clone
+
+#### Version 6.0.14
+* Move dist/cordova-plugin-iosrtc.js to www/cordova-plugin-iosrtc.js to match other cordova plugins practice.
+* fix: Use dispatchAsyncOnType:RTCDispatcherTypeMain instead of dispatchAsyncOnType:RTCDispatcherTypeCaptureSession to fix ios 14.0.1 crash
+* fix: eventListenerForAddStream causing black video on remote stream when using addtrack #576
+* fix: undefined references #563
+* fix: use video view for correct screencap framing #570
+* fix: cleanup old streams #570
+* implement basic MediaDevices.prototype.getSupportedConstraints SHIM #564
+* fix: RTCPeerConnection.prototype.addTrack missing return RTCRtpSender #572
+
+#### Version 6.0.13
+* Bump lodash from 4.17.15 to 4.17.19 #543
+* Bump elliptic from 6.5.0 to 6.5.3 #553
+* Add build doc for CocoaPods and Capacitor users #538
+* Add SWIFT_VERSION and IPHONEOS_DEPLOYMENT_TARGET and DISABLE_IOSRTC_HOOK env options for extra/hooks/iosrtc-swift-support.js #512
+* Improve PluginEnumerateDevices.getAllVideoDevices with builtInTrueDepthCamera and better ios 10.0 support #548
+* Improve cleanup RTCPeerConnections, StreamRenderers, MediaStream, MediaStreamTracks when onReset and onAppTerminate is trigger
+* Implement MediaDevices SHAM
+* Fix crash in PluginEnumerateDevices when audioSession.availableInputs is empty
+* Fix MediaDevices.prototype Object.create(EventTarget.prototype)
+* Fix onaddtrack wihout stream crash during call initializing #532
+* Fix Video Element Redundancy on Refresh / Re-Navigation on same page #535
+* Fix Bridging header is getting added into the Widget/Extension as well causing build process to fail #504 via #513
+* Fix iceRestart constraint doesnt work #530 ([PR #531](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/531) by @andrewvmail)
+* Add Plugin option MANUAL_INIT_AUDIO_DEVICE default to false  ([PR #503](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/503) by @andrewvmail)
+* Fix getUserMedia compatiblity with Twilio Video #497
+* Fix attach stream to ontrack events only when available.
+* Fix issue with pluginMediaStream creation causing black screen due duplicate rtcMediaStream.streamId
+
+#### Version 6.0.12
+* Implement RTCPeerConnection track event on PluginRTCPeerConnection and RTCPeerConnection SHIM #508
+* Fix WebRTC-adapter <= 7.5.0 track SHIM failure
+* Fix WebRTC-adapter >= 7.6.0 track missing SHIM
+* Add SHAM for MediaStreamTrack.getSettings|getCapabilities
+* Add Throw Error Not implemented for MediaStreamTrack.getConstraints|applyConstraints
+* Fix RTCPeerConnection.prototype.getStats.length to match features detection #511
+* Fix Error: Callbacks are not supported by "RTCPeerConnection.prototype.getStats" anymore, use Promise instead. #510
+* Fix Regex in iosrtc-swift-support.js returns wrong value #502
+* Fix Backgroundcolor is now "clear" instead of "black" #514
+* Fix gulp-util is deprecated - replace it #428
+* Add support for MediaStreamTrack.clone() method #474
+* Add basic RTCRtpTransceiver|RTCRtpSender|RTCRtpReceiver shim #423
+* Fix getReceivers method doesn't return RTCRtpReceiver array #442
+* Fix Blob only support for iOS 10.x that does not know MediaStream #495
+* Incompatibility with Janus WebRTC gateway when using WebRTC-adapter >= 7.6.0 #505
+* Improve getUserMedia compatiblity with Twilio Video #497
+
+#### Version 6.0.11
+* Fix possible duplicate remote streamId/trackId with janus/kurento/freeswitch or short duplicate name #493
+* Fix Calling removeTrack/addTrack does not update renderer #492
+
+#### Version 6.0.10
+* Fix ios 10.2+ issue with loading cordova when iosrtc plugin present. #488
+* Fix TypeError: undefined is not an object(evaluating 'originaMediaStream.prototype') #485
+* Handle ios 10.x.x that does not have MediaStream Native Prototype and fallback on Blob with EventTarget shim #489
+
+#### Version 6.0.9
+* fix: possible TypeError: null is not an object (evaluating 'iceCandidateFields.foundation') due fail match candidateToJson #473
+* fix: getStats typo report.timestamp #472
+* Fix getMediaDevice audioConstraints to allowing audio devices change #470
+
+#### Version 6.0.8
+* Add Known Issues > iOS >= 13.3.1 Device support to README.md
+* Implement candidateToJson SDP candidate parser into RTCIceCandidate to populate foundation, component, priority, type, address, ip, protocol, port, relatedAddress and relatedPort RTCIceCandidate values #468
+* Fix PeerConnection.addStream|addTrack by using UUID().uuidString preffix for PluginMediaStream and PluginMediaStreamTrack only for Janus #467
+
+#### Version 6.0.7
+* Missing Event.target value on iosRTC Events to fix Datachannel for Janus.js #447
+* Update extras/renderer-and-libwebrtc-tests.js
+
+#### Version 6.0.6
+* Update ios_arch.js script link ([PR #457](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/457) by @onerinas)
+* Fix removeStream and add real Ids with UUID suffix to PluginMediaStream and PluginMediaStreamTrack ([PR #460](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/460))
+* handle webrtc-adapter multiple video.optional constraints values add support for maxHeight and maxWidth
+* Update README with sdpSemantics, bundlePolicy, rtcpMuxPolicy sample
+
+#### Version 6.0.5
+* Fix missing typeof on constraints.video.facingMode.ideal
+
+#### Version 6.0.4
+* Improve video constraints (width|height|aspectRatio|frameRate).(ideal|exact) and add (aspectRatio|frameRate|facingMode).(ideal|exact) support enhancement #451
+* Generate unique PluginMediaStream and PluginMediaStreamTrack, keep original streamId/trackId for internal calls #447
+* Add English facingLabel suffix if localizedName does not match for facing detection using label, fix PluginEnumerateDevices ordered array #446
+* Implement PluginRTCAudioController setCategory and use audioMode AVAudioSession.Mode.voiceChat and make initAudioDevices non-static #448
+* Fix Callbacks support for getUserMedia ([PR #453](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/453) by @onerinas)
+
+#### Version 6.0.3
+* make second argument MediaStream of RTCPeerConnection.prototype.addTrack optional #437
+* remove redundant PluginRTCAudioController.initAudioDevices call in getAllAudioDevices causing AVAudioSession to reset to its default settings #439
+
+#### Version 6.0.2
+* implement M69 Native RTCPeerConnection.(addTrack|removeTrack) and sdpSemantics unified-plan support #407
+* fix: insertDtmf #431
+
+#### Version 6.0.1
+* fix: stopCapture while RTCCameraVideoCapturer is not capturing causing crash due Assertion failure in -[FBSSerialQueue assertOnQueue] #426
+
+#### Version 6.0.0
+* Use WebRTC M69
+* Use WebRTC.framework
+* implement PluginRTCVideoCaptureController with MediaTrackConstraintSet
+* Replace the libwebrtc static lib with the dynamic library from the WebRTC.framework build process using M69 ([PR #399](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/399) by @hthetiot)
+* Be able to use minAspectRatio/maxAspectRatio #287
+* webrtc/adapter breaks deviceId constraint #282
+* Support for facingMode as a video constraint #315
+* Capturing a static image from the local MediaStream #116
+* Added Speakerphone funcionality #379
+* Restore PluginRTCAudioController including selectAudioOutput and move EnumerateDevice. setPreferredInput| saveAudioDevice into PluginRTCAudioController
+* Fix closing and re-opening local stream several times #247
+* H264 issues #170
+* implement RTCPeerConnectionFactory(encoderFactory, decoderFactory) using getSupportedVideoEncoder to enable VP8 and VP9 #416
+* Add Script to manipulate WebRTC binary architectures #421
+* Use clean WebRTC.framework build with x86 and ARM support #412
+* Decode XML entities in project name in iosrtc-swift-support.js hook #413
+* Update travis build to use Xcode 11.0 (11A420a) and iOS 13 support #376
+* Generic RTCPeerConnection constraint handling #119 via #394
+* Video renders in landscape, but not portrait orientation #360
+* Adapter JS change frameRate constraints #286
+* Regression switch camera fail (stop stream, remove stream from peer, get stream, add stream to peer, renegociate, fail on m69 but not master)
+*  Fix GSM call interrupts the current WebRTC call (Note: Use cordova.plugins.backgroundMode.enable(); on local media Start/Stop).
+
+#### Version 5.0.6
+* Decode XML entities in project name in iosrtc-swift-support.js hook  ([PR #413](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/413) by @GProst).
+
+#### Version 5.0.5
+* Fix RTCPeerConnection.addTrack when providing stream argument and when multiple tracks are added.
+* Handle webrtc-adapter getUserMedia constraints mandatory / optional syntax ([PR #405](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/405) by @hthetiot).
+* Restore Callbacks Support on registerGlobals getUserMedia|enumerateDevices RTCPeerConnection.prototype.createAnswer|createOffer|setRemoteDescription|setLocalDescription|addIceCandidate support for JsSIP, SIP.js and sipML5  ([PR #404](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/404) by @hthetiot).
+* Update extra/hooks/iosrtc-swift-support.js to be trigger on after_prepare and handle existing SWIFT_OBJC_BRIDGING_HEADER values instead of overiding them ([PR #402](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/402) by @hthetiot).
+* Update README.md usages typo and add Requirements about swift-version add npm version badge and travis status badge.
+
+#### Version 5.0.4
+
+* Fix MediaStream.create false positive "ERROR: video track not added" and "ERROR: audio track not added" cause the rtcMediaStream already has them internaly trigger by getUserMedia.
+* Detect deprecated callbacks usage and throw Error instead of been silent to assist 5.0.1 to 5.0.2 migration from WebRTC callback RTCPeerConnection.(createOffer|createAnswer|setLocalDescription|setRemoteDescription|addIceCandidate|getStats) and getUserMedia API.
+* Reset enumerateDevices videoinput order with front camera first to keep legacy behavior.
+* Use safeAreaLayoutGuide for ios 11 and NSLayoutConstraint for ios 10 to fix elementView bad position depending status bar visibility ([PR #367](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/367) by @hthetiot).
+* Recognize when a video receives a new srcObject and re-render it ([PR #367](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/395) by @SejH).
+
+#### Version 5.0.3
+
+* Fix MediaStream.active getter issue.
+* Fix cordova.plugins.iosrtc.observeVideo MutationObserver issue with srcObject using loadstart and emptied events that does get triggered.
+* Add NSBluetoothAlwaysUsageDescription to Info.plist for wireless headphones and microphone consent.
+* Deprecate usage of `video.src = URL.createObjectURL(stream)` in favor of `video.srcObject = stream` only MediaStream are not Blob anymore. ([PR #388](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/388) by @hthetiot).
+* Update audio input priority to Wired microphone > Wireless microphone > built-in microphone. ([PR #387](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/387) by @CSantosM).
+* Add possible missing argument for parameter 'mode' in call on audioSession.setCategory.
+
+#### Version 5.0.2
+
+* Remove WebRTC callback based API for RTCPeerConnection.(createOffer|createAnswer|setLocalDescription|setRemoteDescription|addIceCandidate|getStats) and getUserMedia.
+* Set default deployment target to 10.2
+* Implement partial RTCPeerConnection.(getSenders|getReceivers|addTrack|removeTrack)
+* Fix webrtc-adapter external library support
+* Fix Blob prototype pollution
+* Extend native MediaStream instead of using Blob
+* Fix RTCPeerConnection.setLocalDescription() and other methods which take SDP as input now directly accept an object
+* Upgrade packages debug to ^4.1.1 and yaeti to ^1.0.2
+* Add cordova.plugins.iosrtc.getUserMedia MediaTrackConstraints.(video|audio).deviceId.(exact|ideal) support ([PR #374](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/374) by @CSantosM).
+* Add cordova.plugins.iosrtc.getMediaDevices bluetooth and wired audio devices support ([PR #374](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/374) by @CSantosM).
+* fix: TypeError: undefined is not an object (evaluating 'stream.id') when removing stream [PR #383](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/383) by @hthetiot via @l7s).
+
+#### Version 5.0.1
+
+* fix: typo on iosrtcPlugin.swift
+
+#### Version 5.0.0
+
+* fix: README.md
+* Convert syntax to Swift 4.2
+* Uncomment, and fix, onGetStatsCallback closure
+* Update NPM dependencies
+* Add Travis build (Ionic, Cordova, Browser, Android, iOS Xcode 10.2)
+* Fix gulp browserify caused by old vinyl package version
+* Migrate from jscs to eslint to fix vulnerabilities reported by npm audit
+
+#### Version 4.0.2
+
+* `getUserMedia` constraints: Allow `sourceId` (rather than just `deviceId`) to make adapter.js happy (#282).
+
+
+#### Version 4.0.1
+
+* Let `addIceCandidate()` be called with a `RTCIceCandidateInit` object as argument (as per the latest WebRTC spec) rather than mandating a `RTCIceCandidate` instance.
+
+
+#### Version 4.0.0
+
+* Moved the repository over to its new home with the Basque VoIP Mafia
+* Fix compatibility with "--browserify" cordova option
+* Convert syntax to Swift 3
+* Remove rtcninja integration
+* Remove selectAudioOutput function
+* Add convenience Makefile
+* Update documentation
+
+
+#### Version 3.2.2
+
+* Fix promise implementation of RTCPeerConnection.getStats
+
+
 #### Version 3.2.1
 
 * Fix emitting "connected" stream event for local streams when using getUserMedia with promises.
@@ -5,7 +287,7 @@
 
 #### Version 3.2.0
 
-* Add support for RTCPeerConnection.getStats ([PR #163](https://github.com/eface2face/cordova-plugin-iosrtc/pull/163) by @oNaiPs)
+* Add support for RTCPeerConnection.getStats ([PR #163](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/163) by @oNaiPs)
 
 * Set default deployment target to 9.0
 
@@ -20,34 +302,34 @@
 
 #### Version 3.1.0
 
-* Implement `RTCPeerConnection.createDTMFSender()` ([PR #189](https://github.com/eface2face/cordova-plugin-iosrtc/pull/189) by @saghul).
+* Implement `RTCPeerConnection.createDTMFSender()` ([PR #189](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/189) by @saghul).
 
 * Make `ios-websocket-hack.js` more reliable.
 
 
 #### Version 3.0.1
 
-* Fix positioning video elements using `z-index` and allow pure HTML on top of `<video>` elements ([PR #179](https://github.com/eface2face/cordova-plugin-iosrtc/issues/179) by @saghul and @1N50MN14).
- 
-* Improve `ios-websocket-hack.js` ([PR #138](https://github.com/eface2face/cordova-plugin-iosrtc/issues/138) by @apparition47).
+* Fix positioning video elements using `z-index` and allow pure HTML on top of `<video>` elements ([PR #179](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/179) by @saghul and @1N50MN14).
+
+* Improve `ios-websocket-hack.js` ([PR #138](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/138) by @apparition47).
 
 
 #### Version 3.0.0
 
-* Upgrade to `cordova-ios` 4 ([PR #159](https://github.com/eface2face/cordova-plugin-iosrtc/issues/159) by @apparition47).
+* Upgrade to `cordova-ios` 4 ([PR #159](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/159) by @apparition47).
 
-* Swift: Use closure syntax for weak and unowned vars ([PR #160](https://github.com/eface2face/cordova-plugin-iosrtc/issues/160) by @oNaiPs).
+* Swift: Use closure syntax for weak and unowned vars ([PR #160](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/160) by @oNaiPs).
 
-* Swift: Sanitize arguments given to `NSLog()` ([issue #157](https://github.com/eface2face/cordova-plugin-iosrtc/issues/157)).
+* Swift: Sanitize arguments given to `NSLog()` ([issue #157](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/157)).
 
-* `MediaDeviceInfo`: Add deprecated `facing` property ("front", "back" or "unknown") and update `kind` ("audio"/"video" become "audioinput"/"videoinput") ([issue #155](https://github.com/eface2face/cordova-plugin-iosrtc/issues/155)).
+* `MediaDeviceInfo`: Add deprecated `facing` property ("front", "back" or "unknown") and update `kind` ("audio"/"video" become "audioinput"/"videoinput") ([issue #155](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/155)).
 
-* Update `libwebrtc` to revision 12558 ([issue #169](https://github.com/eface2face/cordova-plugin-iosrtc/issues/169)).
+* Update `libwebrtc` to revision 12558 ([issue #169](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/169)).
 
 
 #### Version 2.2.4
 
-* Fix crash ([issue #144](https://github.com/eface2face/cordova-plugin-iosrtc/issues/144)).
+* Fix crash ([issue #144](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/144)).
 
 * Update NPM dependencies.
 
@@ -56,18 +338,18 @@
 
 * Enable iOS native H.264 encoder/decoder.
 
-* `RTCDataChannel`: Allow empty `label` ([issue #124](https://github.com/eface2face/cordova-plugin-iosrtc/issues/124)).
+* `RTCDataChannel`: Allow empty `label` ([issue #124](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/124)).
 
-* Update [yaeti](https://www.npmjs.com/package/yaeti) dependency ([issue #123](https://github.com/eface2face/cordova-plugin-iosrtc/issues/123)).
+* Update [yaeti](https://www.npmjs.com/package/yaeti) dependency ([issue #123](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/123)).
 
-* Fix retain on `pluginMediaStreamTrack` does not allow camera/mic to be freed ([PR #126](https://github.com/eface2face/cordova-plugin-iosrtc/pull/126)). Credits to @oNaiPs.
+* Fix retain on `pluginMediaStreamTrack` does not allow camera/mic to be freed ([PR #126](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/126)). Credits to @oNaiPs.
 
-* Allow a handled video element to be removed from the DOM and added again later ([PR #127](https://github.com/eface2face/cordova-plugin-iosrtc/pull/127)). Credits to @oNaiPs.
+* Allow a handled video element to be removed from the DOM and added again later ([PR #127](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/127)). Credits to @oNaiPs.
 
 
 #### Version 2.2.2
 
-* Update `libwebrtc` to revision 11063 so `MediaStream` events (`onaddtrack` and `onremovetrack`) work again ([issue #95](https://github.com/eface2face/cordova-plugin-iosrtc/issues/95)).
+* Update `libwebrtc` to revision 11063 so `MediaStream` events (`onaddtrack` and `onremovetrack`) work again ([issue #95](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/95)).
 
 
 #### Version 2.2.1
@@ -90,23 +372,23 @@
 
 #### Version 2.0.2
 
-* Enable CSS padding (thanks to @saghul) ([pull request #89](https://github.com/eface2face/cordova-plugin-iosrtc/issues/89)).
+* Enable CSS padding (thanks to @saghul) ([pull request #89](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/89)).
 
 
 #### Version 2.0.1
 
-* Don't crash if user or iOS settings deny access lo local audio/video devices ([issue #73](https://github.com/eface2face/cordova-plugin-iosrtc/issues/73)).
+* Don't crash if user or iOS settings deny access lo local audio/video devices ([issue #73](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/73)).
 
 
 #### Version 2.0.0
 
-* Swift 2.0 is here! Credits to @saghul for his [pull request](https://github.com/eface2face/cordova-plugin-iosrtc/pull/70).
+* Swift 2.0 is here! Credits to @saghul for his [pull request](https://github.com/cordova-rtc/cordova-plugin-iosrtc/pull/70).
 * `extra/hooks/iosrtc-swift-support.js`: Set `BUILD_VERSION` to 7.0.
 
 
 #### Version 1.4.5
 
-* Add `cordova.plugins.iosrtc.observeVideo(video)` API for the plugin to handle `<video>` elements not yet in the DOM ([issue #49](https://github.com/eface2face/cordova-plugin-iosrtc/issues/49)).
+* Add `cordova.plugins.iosrtc.observeVideo(video)` API for the plugin to handle `<video>` elements not yet in the DOM ([issue #49](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/49)).
 
 
 #### Version 1.4.4
@@ -116,7 +398,7 @@
 
 #### Version 1.4.3
 
-* Make private properties more private ([issue #34](https://github.com/eface2face/cordova-plugin-iosrtc/issues/34)).
+* Make private properties more private ([issue #34](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/34)).
 
 
 #### Version 1.4.2
@@ -126,12 +408,12 @@
 
 #### Version 1.4.1
 
-* Release `MediaStreamRenderer` and revert `<video>` properties when the attached `MediaStream` emits "inactive" ([issue #27](https://github.com/eface2face/cordova-plugin-iosrtc/issues/27)).
+* Release `MediaStreamRenderer` and revert `<video>` properties when the attached `MediaStream` emits "inactive" ([issue #27](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/27)).
 
 
 #### Version 1.4.0
 
-* Implemented some `<video>` properties such as `readyState`, `videoWidth` and `videoHeight` ([issue #25](https://github.com/eface2face/cordova-plugin-iosrtc/issues/25)).
+* Implemented some `<video>` properties such as `readyState`, `videoWidth` and `videoHeight` ([issue #25](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/25)).
 * Building simplified for both Cordova CLI and Xcode by providing a single ["hook"](extra/hooks/iosrtc-swift-support.js) the user must add into his Cordova application (check the [Building](docs/Building.md) documentation for further details).
 
 
@@ -142,12 +424,12 @@
 
 #### Version 1.3.2
 
-* CSS [object-fit](https://css-tricks.com/almanac/properties/o/object-fit/) property implemented in `<video>` elements ([issue #23](https://github.com/eface2face/cordova-plugin-iosrtc/issues/23)).
+* CSS [object-fit](https://css-tricks.com/almanac/properties/o/object-fit/) property implemented in `<video>` elements ([issue #23](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/23)).
 
 
 #### Version 1.3.1
 
-* Stop "error" event propagation in `<video>` element when attaching a `MediaStream` to it ([issue #22](https://github.com/eface2face/cordova-plugin-iosrtc/issues/22)).
+* Stop "error" event propagation in `<video>` element when attaching a `MediaStream` to it ([issue #22](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/22)).
 
 
 #### Version 1.3.0
diff --git a/FAQ.md b/FAQ.md
new file mode 100644
index 00000000..d01bde1f
--- /dev/null
+++ b/FAQ.md
@@ -0,0 +1,37 @@
+# FAQ
+
+**Q:** What about `<video>` elements and `video.srcObject = stream`? do I need custom HTML tags or functions to display WebRTC videos?
+
+**A:** No. Just use an HTML video element as usual, really. The plugin will properly place a native *UIView* layer on top of it by respecting (most of) its [CSS properties](docs/videoCSS.md).
+
+**Q:** Can I place HTML elements (buttons and so on) on top of active `<video>` elements?
+
+**A:** Yes. See the [<video> CSS](docs/videoCSS.md) documentation.
+
+**Q:** What about [HTML5 video events](https://developer.mozilla.org/en-US/docs/Web/Guide/Events/Media_events)? Can I rely on `video.oncanplay`?
+
+**A:** I see what you mean. As there is no real video attached to the `<video>` element, media events are artificially emitted by the plugin. The following events are emitted when the `MediaStream` attached to a video element is ready to render video: `onloadedmetadata`, `onloadeddata`, `oncanplay`, `oncanplaythrough`. So yes, you can rely on them.
+
+**Q:** Can I read `<video>` properties such as `readyState`, `videoWidth`, etc?
+
+**A:** Again, there is no real video attached to the `<video>` element so some peroperties are artificially set by the plugin. These are `readyState`, `videoWidth` and `videoHeight`.
+
+**Q:** Do I need to call special methods to release/free native WebRTC objects? How are they garbage collected?
+
+**A:** Good question. An `RTCPeerConnection` is released when `close()` is called on it, a `MediaStream` is released when all its tracks end, and other elements are garbage collected when no longer needed. Basically the same behavior as in a WebRTC capable browser.
+
+**Q:** What about Android? Why just iOS?
+
+**A:** In modern versions of Android the *WebView* component is based on the Chromium open source project which already includes WebRTC ([more info](https://developer.chrome.com/multidevice/webview/overview)). For older versions of Android the [CrossWalk](https://crosswalk-project.org) project provides new *WebView* versions with WebRTC support as well.
+
+**Q:** Why does a "play button" show up on top of the video?
+
+**A:** That's automatically done by Safari, you can disable it by using the following CSS rules:
+```css
+video::-webkit-media-controls {
+    display: none !important;
+}
+video::-webkit-media-controls-start-playback-button {
+    display: none !important;
+}
+```
diff --git a/ISSUE_TEMPLATE.md b/ISSUE_TEMPLATE.md
deleted file mode 100644
index 116e3b0c..00000000
--- a/ISSUE_TEMPLATE.md
+++ /dev/null
@@ -1,3 +0,0 @@
-> This is a issue tracker, not a forum for questions regarding how to use the plugin nor a support forum for issues when integrating this plugin with third party applications or libraries.
-> 
-> If you have found a specific issue or bug in this software, or want to propose a new feature or specific improvement, please delete this text and continue. Otherwise, please place your question or doubt in the public Google Group: https://groups.google.com/forum/#!forum/cordova-plugin-iosrtc
diff --git a/LICENSE b/LICENSE
index cb47f03c..ba66798f 100644
--- a/LICENSE
+++ b/LICENSE
@@ -1,6 +1,8 @@
 The MIT License (MIT)
 
-Copyright (c) 2015 eFace2Face, Inc.
+Copyright (c) 2015-2017 eFace2Face, Inc. (https://eface2face.com)
+Copyright (c) 2017-2019 BasqueVoIPMafia (https://github.com/BasqueVoIPMafia)
+Copyright (c) 2019 Cordova-RTC (https://github.com/cordova-rtc)
 
 Permission is hereby granted, free of charge, to any person obtaining a copy
 of this software and associated documentation files (the "Software"), to deal
diff --git a/Makefile b/Makefile
new file mode 100644
index 00000000..d7eff284
--- /dev/null
+++ b/Makefile
@@ -0,0 +1,11 @@
+GULP := ./node_modules/.bin/gulp
+
+all: build
+
+build:
+	$(GULP)
+
+clean:
+	rm -rf node_modules
+
+.PHONY: build clean
diff --git a/README.md b/README.md
index 49329d26..5d1545b0 100644
--- a/README.md
+++ b/README.md
@@ -1,31 +1,52 @@
+![cordova-rtc-logo](https://raw.githubusercontent.com/cordova-rtc/cordova-plugin-iosrtc/master/art/cordova-rtc-logo.png)
+
 # cordova-plugin-iosrtc
 
-[Cordova](http://cordova.apache.org/) iOS plugin exposing the full [WebRTC W3C JavaScript APIs](http://www.w3.org/TR/webrtc/).
+[![npm version](https://img.shields.io/npm/v/cordova-plugin-iosrtc.svg?style=flat)](https://www.npmjs.com/package/cordova-plugin-iosrtc)
+[![Build Status](https://github.com/cordova-rtc/cordova-plugin-iosrtc/actions/workflows/main.yml/badge.svg)](https://github.com/cordova-rtc/cordova-plugin-iosrtc/actions/workflows/main.yml)
+
+[![NPM](https://nodei.co/npm/cordova-plugin-iosrtc.png)](https://npmjs.org/package/cordova-plugin-iosrtc)
 
+[Cordova](http://cordova.apache.org/) iOS plugin exposing the  ̶f̶u̶l̶l̶ [WebRTC W3C JavaScript APIs](http://www.w3.org/TR/webrtc/).
 
-* [Public Google Group (mailing list)](https://groups.google.com/forum/#!forum/cordova-plugin-iosrtc) for questions and discussions about *cordova-plugin-iosrtc*.
-* [Bug Tracker](https://github.com/eface2face/cordova-plugin-iosrtc/issues) for reporting issues and requesting new features (**please** don't use the bug tracker for questions or problems, use the mailing list instead).
+* [Community Forum](https://github.com/cordova-rtc/cordova-plugin-iosrtc/discussions) for questions and discussions about *cordova-plugin-iosrtc*.
+* [Bug Tracker](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues) for reporting issues and requesting new features (**please** don't use the bug tracker for questions or problems, use the mailing list instead).
 * [NPM package](https://www.npmjs.com/package/cordova-plugin-iosrtc).
 
 
 **Yet another WebRTC SDK for iOS?**
 
-Absolutely **not**. This plugin exposes the WebRTC W3C API for Cordova iOS apps (you know there is no WebRTC in iOS, right?), which means no need to learn "yet another WebRTC API" and no need to use a specific service/product/provider. 
+Absolutely **not**. This plugin exposes the WebRTC W3C API for Cordova iOS apps (you know there is no WebRTC in iOS, right?), which means no need to learn "yet another WebRTC API" and no need to use a specific service/product/provider.
+
 
-**Why?**
+**Who?**
 
-Check the [release announcement](https://eface2face.com/blog/cordova-plugin-iosrtc.html) at the [eFace2Face](https://eface2face.com) site.
+This plugin was initially developed at eFace2Face, and later maintained by the community, specially by [Saúl Ibarra Corretgé](http://bettercallsaghul.com) (_The OpenSource Warrior Who Does Not Burn_).
 
 
 ## Requirements
 
 In order to make this Cordova plugin run into a iOS application some requirements must be satisfied in both development computer and target devices:
 
-* Xcode >= 7.2.1
-* iOS >= 9 (run on lower versions at your own risk, but don't report issues)
-* `cordova-ios` 4.X
-* No bitcode (built-in *libwebrtc* does not contain bitcode so you need to disable it in your Xcode project settings)
-
+* Xcode >= 13.2.1 (13C100)
+* iOS >= 14.3 (run on lower versions at your own risk, don't report issues)
+* `swift-version` => 4.2
+* `cordova` >= 11.0.0
+* `cordova-ios` >= 6.2.0
+
+### Third-Party Supported Library
+
+* WebRTC W3C v1.0.0
+* WebRTC.framework => M87
+* Janus => 0.7.4
+* JSSip => 3.1.2
+* Sip.js => 0.15.6
+* OpenEasyrtc => 2.0.3
+* openvidu => 2.11.0
+* Ionic => v8
+* Jitsi ~ 3229
+* Apizee => 2.6.11
+* Twillio => 2.4.0
 
 ## Installation
 
@@ -37,96 +58,300 @@ $ cordova plugin add cordova-plugin-iosrtc
 
 (or add it into a `<plugin>` entry in the `config.xml` of your app).
 
-
 ## Building
 
+* Last [Tested WebRTC.framework](./lib/WebRTC.framework/) version: M87 on cordova-plugin-iosrtc version 8.0.0+
 * [Building](docs/Building.md): Guidelines for building a Cordova iOS application including the *cordova-plugin-iosrtc* plugin.
-* [Building `libwebrtc`](docs/BuildingLibWebRTC.md): Guidelines for building Google's *libwebrtc* with modifications needed by the *cordova-plugin-iosrtc* plugin (just in case you want to use a different version of *libwebrtc* or aplpy your own changes to it).
+* [Building `libwebrtc`](docs/BuildingLibWebRTC.md): Guidelines for building Google's *libwebrtc* with modifications needed by the *cordova-plugin-iosrtc* plugin (just in case you want to use a different version of *libwebrtc* or apply your own changes to it).
+
+## Sample Application
 
+The `cordova-plugin-iosrtc-sample` include mutiple example for using `cordova-plugin-iosrtc` with JsSip, Janus, EasyRTC, and basic WebSocket Signaling. It's is used to test `cordova-plugin-iosrtc` new release and reproduce reported issues.
+
+- https://github.com/cordova-rtc/cordova-plugin-iosrtc-sample
 
 ## Usage
 
 The plugin exposes the `cordova.plugins.iosrtc` JavaScript namespace which contains all the WebRTC classes and functions.
 
 ```javascript
-var pc = new cordova.plugins.iosrtc.RTCPeerConnection({
-  iceServers: []
-});
-
-cordova.plugins.iosrtc.getUserMedia(
-  // constraints
-  { audio: true, video: true },
-  // success callback
-  function (stream) {
-    console.log('got local MediaStream: ', stream);
-
-    pc.addStream(stream);
-  },
-  // failure callback
-  function (error) {
-    console.error('getUserMedia failed: ', error);
+/* global RTCPeerConnection */
+
+
+//
+// Container for this sample
+//
+
+var appContainer = document.body;
+appContainer.innerHTML = "";
+
+//
+// Sample getUserMedia
+//
+
+//
+var localStream, localVideoEl;
+function TestGetUserMedia() {
+  localVideoEl = document.createElement('video');
+  localVideoEl.style.height = "50vh";
+  localVideoEl.setAttribute('autoplay', 'autoplay');
+  localVideoEl.setAttribute('playsinline', 'playsinline');
+  appContainer.appendChild(localVideoEl);
+
+  return navigator.mediaDevices.getUserMedia({
+    video: true,
+    audio: true
+    // Note: Use navigator.mediaDevices.enumerateDevices() Promise to get deviceIds
+    /*
+    video: {
+      // Test Back Camera
+      //deviceId: 'com.apple.avfoundation.avcapturedevice.built-in_video:0'
+      //sourceId: 'com.apple.avfoundation.avcapturedevice.built-in_video:0'
+      deviceId: {
+        exact: 'com.apple.avfoundation.avcapturedevice.built-in_video:0'
+      }
+      // Test FrameRate
+      frameRate:{ min: 15.0, max: 30.0 } // Note: Back camera may only support max 30 fps
+    },
+    audio: {
+      deviceId: {
+        exact: 'Built-In Microphone'
+      }
+    }*/
+  }).then(function (stream) {
+
+    console.log('getUserMedia.stream', stream);
+    console.log('getUserMedia.stream.getTracks', stream.getTracks());
+
+    // Note: Expose for debug
+    localStream = stream;
+
+    // Attach local stream to video element
+    localVideoEl.srcObject = localStream;
+
+    return localStream;
+
+  }).catch(function (err) {
+    console.log('getUserMedia.error', err, err.stack);
+  });
+}
+
+//
+// Sample RTCPeerConnection
+//
+
+var pc1, pc2;
+
+var peerConnectionConfig = {
+    offerToReceiveVideo: true,
+    offerToReceiveAudio: true,
+    //iceTransportPolicy: 'relay',
+    sdpSemantics: 'unified-plan',
+    //sdpSemantics: 'plan-b',
+    bundlePolicy: 'max-compat',
+    rtcpMuxPolicy: 'negotiate',
+    iceServers: [
+      {
+         urls: ["stun:stun.stunprotocol.org"]
+      }
+    ]
+};
+
+// This plugin handle 'addstream' and 'track' event for MediaStream creation.
+var useTrackEvent = Object.getOwnPropertyDescriptors(RTCPeerConnection.prototype).ontrack;
+
+var peerVideoEl, peerStream;
+function TestRTCPeerConnection(localStream) {
+
+  pc1 = new RTCPeerConnection(peerConnectionConfig);
+  pc2 = new RTCPeerConnection(peerConnectionConfig);
+
+  if (useTrackEvent) {
+
+    // Add local stream tracks to RTCPeerConnection
+    var localPeerStream = new MediaStream();
+    localStream.getTracks().forEach(function (track) {
+      console.log('pc1.addTrack', track, localPeerStream);
+      pc1.addTrack(track, localPeerStream);
+    });
+
+  // Note: Deprecated but supported
+  } else {
+     pc1.addStream(localStream);
+
+     // Note: Deprecated Test removeStream
+     // pc1.removeStream(pc1.getLocalStreams()[0]);<
   }
-);
-```
 
-**Q:** But... wait! Does it mean that there is no `window.RTCPeerConnection` nor `navigator.getUserMedia`?
+  // Basic RTCPeerConnection Local WebRTC Signaling follow.
+  function onAddIceCandidate(pc, can) {
+    console.log('addIceCandidate', pc, can);
+    return can && pc.addIceCandidate(can).catch(function (err) {
+      console.log('addIceCandidateError', err);
+    });
+  }
 
-**R:** A Cordova plugin is supposed to expose its JavaScript stuff in a specific namespace and, personally, I just hate those libraries that pollute the global namespace. Said that, the plugin provides a `registerGlobals()` method, so you just need the following extra-code in your existing WebRTC app (assuming that [cordova-plugin-device](https://www.npmjs.com/package/cordova-plugin-device) is installed):
+  pc1.addEventListener('icecandidate', function (e) {
+    onAddIceCandidate(pc2, e.candidate);
+  });
 
-```javascript
-// Just for Cordova apps.
-document.addEventListener('deviceready', function () {
-  // Just for iOS devices.
-  if (window.device.platform === 'iOS') {
-    cordova.plugins.iosrtc.registerGlobals();
-  }
-});
-```
+  pc2.addEventListener('icecandidate', function (e) {
+    onAddIceCandidate(pc1, e.candidate);
+  });
 
-And that's all. Now you have `window.RTCPeerConnection`, `navigator.getUserMedia`, etc.
+  function setPeerVideoStream(stream) {
 
-**Q:** What about `<video>` elements and `video.src = URL.createObjectURL(stream)`? do I need custom HTML tags or functions to display WebRTC videos?
+    // Create peer video element
+    peerVideoEl = document.createElement('video');
+    peerVideoEl.style.height = "50vh";
+    peerVideoEl.setAttribute('autoplay', 'autoplay');
+    peerVideoEl.setAttribute('playsinline', 'playsinline');
+    appContainer.appendChild(peerVideoEl);
 
-**R:** No. Just use an HTML video element as usual, really. The plugin will properly place a native *UIView* layer on top of it by respecting (most of) its [CSS properties](docs/videoCSS.md).
+    // Note: Expose for debug
+    peerStream = stream;
 
-**Q:** Can I place HTML elements (buttons and so on) on top of active `<video>` elements?
+    // Attach peer stream to video element
+    peerVideoEl.srcObject = peerStream;
+  }
 
-**R:** Yes. See the [<video> CSS](docs/videoCSS.md) documentation.
+  if (useTrackEvent) {
+    var newPeerStream;
+    pc2.addEventListener('track', function(e) {
+      console.log('pc2.track', e);
+      newPeerStream = e.streams[0] || newPeerStream || new MediaStream();
+      setPeerVideoStream(newPeerStream);
+      newPeerStream.addTrack(e.track);
+    });
+
+  // Note: Deprecated but supported
+  } else {
+    pc2.addEventListener('addstream', function(e) {
+      console.log('pc2.addStream', e);
+      setPeerVideoStream(e.stream);
+    });
+  }
 
-**Q:** What about [HTML5 video events](https://developer.mozilla.org/en-US/docs/Web/Guide/Events/Media_events)? Can I rely on `video.oncanplay`?
+  pc1.addEventListener('iceconnectionstatechange', function (e) {
+    console.log('pc1.iceConnectionState', e, pc1.iceConnectionState);
+
+    if (pc1.iceConnectionState === 'completed') {
+      console.log('pc1.getSenders', pc1.getSenders());
+      console.log('pc2.getReceivers', pc2.getReceivers());
+    }
+  });
+
+  pc1.addEventListener('icegatheringstatechange', function (e) {
+    console.log('pc1.iceGatheringStateChange', e);
+  });
+
+  pc1.addEventListener('negotiationneeded', function (e) {
+    console.log('pc1.negotiatioNeeded', e);
+
+    return pc1.createOffer().then(function (d) {
+      var desc = {
+        type: d.type,
+        sdp: d.sdp
+      };
+      console.log('pc1.setLocalDescription', desc);
+      return pc1.setLocalDescription(desc);
+    }).then(function () {
+      var desc = {
+        type: pc1.localDescription.type,
+        sdp: pc1.localDescription.sdp
+      };
+      console.log('pc2.setLocalDescription', desc);
+      return pc2.setRemoteDescription(desc);
+    }).then(function () {
+      console.log('pc2.createAnswer');
+      return pc2.createAnswer();
+    }).then(function (d) {
+      var desc = {
+        type: d.type,
+        sdp: d.sdp
+      };
+      console.log('pc2.setLocalDescription', desc);
+      return pc2.setLocalDescription(d);
+    }).then(function () {
+      var desc = {
+        type: pc2.localDescription.type,
+        sdp: pc2.localDescription.sdp
+      };
+      console.log('pc1.setRemoteDescription', desc);
+      return pc1.setRemoteDescription(desc);
+    }).catch(function (err) {
+      console.log('pc1.createOffer.error', err);
+    });
+  });
+}
+
+function TestRTCPeerConnectionLocal() {
+
+  // Note: This allow this sample to run on any Browser
+  var cordova = window.cordova;
+  if (cordova && cordova.plugins && cordova.plugins.iosrtc) {
+
+    // Expose WebRTC and GetUserMedia SHIM as Globals (Optional)
+    // Alternatively WebRTC API will be inside cordova.plugins.iosrtc namespace
+    cordova.plugins.iosrtc.registerGlobals();
 
-**R:** I see what you mean. As there is no real video attached to the `<video>` element, media events are artificially emitted by the plugin. The following events are emitted when the `MediaStream` attached to a video element is ready to render video: `onloadedmetadata`, `onloadeddata`, `oncanplay`, `oncanplaythrough`. So yes, you can rely on them.
+    // Enable iosrtc debug (Optional)
+    cordova.plugins.iosrtc.debug.enable('*', true);
+  }
 
-**Q:** Can I read `<video>` properties such as `readyState`, `videoWidth`, etc?
+  // Run sample
+  TestGetUserMedia().then(function (localStream) {
+    TestRTCPeerConnection(localStream);
+  });
+}
 
-Again, there is no real video attached to the `<video>` element so some peroperties are artificially set by the plugin. These are `readyState`, `videoWidth` and `videoHeight`.
+if (document.readyState === "complete" || document.readyState === "loaded") {
+  TestRTCPeerConnectionLocal();
+} else {
+  window.addEventListener("DOMContentLoaded", TestRTCPeerConnectionLocal);
+}
 
-**Q:** Do I need to call special methods to release/free native WebRTC objects? How are they garbage collected?
+// See ./extra/renderer-and-libwebrtc-tests.js for more samples usage.
+```
 
-**R:** Good question. An `RTCPeerConnection` is released when `close()` is called on it, a `MediaStream` is released when all its tracks end, and other elements are garbage collected when no longer needed. Basically the same behavior as in a WebRTC capable browser.
+In case you'd like to expose the API in the global namespace like regular browsers you can do the following:
 
-**Q:** What about Android? Why just iOS?
+```javascript
+// Just for Cordova apps.
+document.addEventListener('deviceready', function () {
+  // Just for iOS devices.
+  if (window.device.platform === 'iOS') {
+    cordova.plugins.iosrtc.registerGlobals();
 
-**R:** In modern versions of Android the *WebView* component is based on the Chromium open source project which already includes WebRTC ([more info](https://developer.chrome.com/multidevice/webview/overview)). For older versions of Android the [CrossWalk](https://crosswalk-project.org) project provides new *WebView* versions with WebRTC support as well.
+    // load adapter.js
+    var adapterVersion = 'latest';
+    var script = document.createElement("script");
+    script.type = "text/javascript";
+    script.src = "https://webrtc.github.io/adapter/adapter-" + adapterVersion + ".js";
+    script.async = false;
+    document.getElementsByTagName("head")[0].appendChild(script);
+  }
+});
+```
 
+And that's all. Now you have `window.RTCPeerConnection`, `navigator.getUserMedia`, etc.
 
-## Documentation
 
-Read the full [documentation](docs/index.md) in the *docs* folder.
+## FAQ
 
+See [the FAQ](FAQ.md).
 
-## Demo Application
 
-Check our [iOSRTCApp](https://github.com/eface2face/iOSRTCApp) (Google's [AppRTC](https://apprtc.appspot.com/) adapted to Cordova iOS with pure HTML5/JavaScript and *cordova-plugin-iosrtc*).
+## Documentation
 
-*NOTE:* The demo app is currently unmaintained and it may just fail.
+Read the full [documentation](docs/index.md) in the *docs* folder.
 
 
 ## Who Uses It
 
 [People and companies](WHO_USES_IT.md) using *cordova-plugin-iosrtc*.
 
-If you are using the plugin we would love to [heard back from you](WHO_USES_IT.md)!
+If you are using the plugin we would love to [hear back from you](WHO_USES_IT.md)!
 
 
 ## Known Issues
@@ -134,7 +359,7 @@ If you are using the plugin we would love to [heard back from you](WHO_USES_IT.m
 
 #### iOS Safari and crash on WebSocket events
 
-Don't call plugin methods within WebSocket events (`onopen`, `onmessage`, etc). There is an issue in iOS Safari (see [issue #12](https://github.com/eface2face/cordova-plugin-iosrtc/issues/12)). Instead run a `setTimeout()` within the WebSocket event if you need to call plugin methods on it.
+Don't call plugin methods within WebSocket events (`onopen`, `onmessage`, etc). There is an issue in iOS Safari (see [issue #12](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/12)). Instead run a `setTimeout()` within the WebSocket event if you need to call plugin methods on it.
 
 Or better yet, include the provided [ios-websocket-hack.js](extra/ios-websocket-hack.js) in your app and load into your `index.html` as follows:
 
@@ -145,24 +370,35 @@ Or better yet, include the provided [ios-websocket-hack.js](extra/ios-websocket-
 
 #### HTML5 video API
 
-As explained above, there is no real media source attached to the `<video>` element so some [HTML5 video events](https://developer.mozilla.org/en-US/docs/Web/Guide/Events/Media_events) and properties are artificially emitted/set by the plugin on behalf of the video element.
+There is no real media source attached to the `<video>` element so some [HTML5 video events](https://developer.mozilla.org/en-US/docs/Web/Guide/Events/Media_events) and properties are artificially emitted/set by the plugin on behalf of the video element.
 
 Methods such as `play()`, `pause()` are not implemented. In order to pause a video just set `enabled = false` on the associated `MediaStreamTrack`.
 
+#### iOS >= 13.3.1 Device support
+
+To run on Devices with iOS >= 13.3.1, you need a valid Apple Developer account to sign the WebRTC dynamic frameworks for more info see https://stackoverflow.com/a/60090629/8691951
 
 ## Changelog
 
 See [CHANGELOG.md](./CHANGELOG.md).
 
-## Author
 
-Iñaki Baz Castillo at [eFace2Face, inc.](https://eface2face.com)
+## Authors
+
+* [Iñaki Baz Castillo](https://inakibaz.me) (no longer active maintainer)
+* [Saúl Ibarra Corretgé](http://bettercallsaghul.com) (no longer active maintainer)
+* [Harold Thetiot](https://sylaps.com)
+
+
+### Maintainer
 
-### Maintainers
+* [Harold Thetiot](https://sylaps.com)
 
-* Iñaki Baz Castillo at [eFace2Face, inc.](https://eface2face.com)
-* Saúl Ibarra Corretgé ([Better Call Saghul](http://bettercallsaghul.com)
+**If you like this project you can support me.**
 
+<a href="https://www.buymeacoffee.com/hthetiot" target="_blank">
+	<img src="https://cdn.buymeacoffee.com/buttons/default-white.png" alt="Buy Me A Coffee" style="height: 51px !important;width: 217px !important;" >
+</a>
 
 ## License
 
diff --git a/SECURITY.md b/SECURITY.md
new file mode 100644
index 00000000..add2ec1f
--- /dev/null
+++ b/SECURITY.md
@@ -0,0 +1,20 @@
+# Security Policy
+
+## Supported Versions
+
+Use this section to tell people about which versions of your project are
+currently being supported with security updates.
+
+| Version | Supported          |
+| ------- | ------------------ |
+| 8.x   | :white_check_mark: |
+| 6.x   | :white_check_mark: |
+| < 5.x   | :x:                |
+
+## Reporting a Vulnerability
+
+Use this section to tell people how to report a vulnerability.
+
+Tell them where to go, how often they can expect to get an update on a
+reported vulnerability, what to expect if the vulnerability is accepted or
+declined, etc.
diff --git a/WHO_USES_IT.md b/WHO_USES_IT.md
index 35bdc3f2..de427918 100644
--- a/WHO_USES_IT.md
+++ b/WHO_USES_IT.md
@@ -4,16 +4,47 @@ If your company is using *cordova-plugin-iosrtc* we would love to heard back fro
 
 Below a list of companies using *cordova-plugin-iostc*:
 
+### [263视频会议](https://www.263.net/263/videoConference/)
+
+![263视频会议-logo](https://www.263.net/r/cms/www/web2018/img/download-icon-videomeet.png)
+
+At [263视频会议](https://apps.apple.com/cn/app/263%E8%A7%86%E9%A2%91%E4%BC%9A%E8%AE%AE/id1361487536) we use **cordova-plugin-iosrtc** for the iOS version of our application, becoming a web/mobile/tablet HTML5 hybrid solution.
+
+### [ConnectyCube](https://connectycube.com)
+
+![ConnectyCube Logo](https://connectycube.com/imgs/connectycube_logo_with_text.png)
+
+*A Swiss Army knife for chat & video chat apps.*
+
+Using **cordova-plugin-iosrtc** in our [Cordova сode samples](https://developers.connectycube.com/cordova/code-samples) for P2P and Conference calling.
+
+### [OpenVidu](https://openvidu.io)
+
+![openvidu-picture](https://openvidu.io/img/logos/openvidu_vert_grey_bg_transp_cropped.png)
+
+*Easy videoconference in your web or mobile app*
+
+Using **cordova-plugin-iosrtc** in our Ionic iOS [openvidu-ionic](https://openvidu.io/docs/tutorials/openvidu-ionic/) and [openvidu-call-ionic](https://openvidu.io/docs/demos/openvidu-call-ionic/) apps.
+
+### [Sylaps](https://sylaps.com)
+
+![sylaps-picture](https://raw.githubusercontent.com/cordova-rtc/cordova-plugin-iosrtc/master/art/sylaps-picture.png)
+
+*Instantaneous, Audio and Video meeting collaboration*
+
+At [Sylaps](https://sylaps.com) we use **cordova-plugin-iosrtc** for the iOS version of our application, becoming a web/mobile/tablet HTML5 hybrid solution.
+
+You can test the Sylaps iOS application that use iosrtc here:
+- https://apps.apple.com/us/app/sylaps/id954016392
 
 ### [eFace2Face](https://eface2face.com)
 
-![eface2face-picture](https://raw.githubusercontent.com/eface2face/cordova-plugin-iosrtc/master/art/eface2face-picture.jpg)
+![eface2face-picture](https://raw.githubusercontent.com/cordova-rtc/cordova-plugin-iosrtc/master/art/eface2face-picture.jpg)
 
 *The world's only complete web-based video calling and e-sign solution.*
 
 At [eFace2Face](https://eface2face.com) we use **cordova-plugin-iosrtc** for the iOS version of our application, becoming a web/mobile/tablet HTML5 hybrid solution.
 
-
 ### [OpenTokJS Cordova Plugin](https://github.com/aoberoi/cordova-plugin-opentokjs)
 
 ![opentok-logo](https://static.opentok.com/img/press/logo_opentok_registered.png)
@@ -31,15 +62,28 @@ This is a community supported, open source project for enabling developers to us
 
 [ApiRTC](http://apirtc.com) use WebRTC technology to add real-time communication to your websites & applications.
 
+
 ### [Free](https://assistance.free.fr/facetofree/)
 
 ![free-logo](http://free.fr/assistance/img/logo.png)
 
 [Free](https://assistance.free.fr/facetofree/) (french major ISP) use WebRTC technology to provide video support for our customers on iPhone & iPad.
 
+
 ### [Blink](http://icanblink.com)
 
 <img src="https://blink.sipthor.net/blink-ipad.jpg" width="400"/>
 
 [Blink for Mobile](https://itunes.apple.com/us/app/blink-for-mobile/id1100356948?ls=1&mt=8) uses WebRTC technology to provide a rich audio and video
 experience on mobile devices.
+
+
+### [VEEDEEO](http://www.veedeeo.me)
+
+<img src="https://www.veedeeo.me/resource/1599641653000/vdo_website_css6_2020/images/logo.png" />
+
+VEEDEEO is a enterprise video communications platform and we use iosrtc - by far the best WebRTC plugin for mobile - to power our [mobile apps](https://itunes.apple.com/us/app/veedeeo/id1337029218) on iOS.
+
+
+
+
diff --git a/art/cordova-plugin-iosrtc-sample-landscape.png b/art/cordova-plugin-iosrtc-sample-landscape.png
new file mode 100644
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diff --git a/art/cordova-plugin-iosrtc-sample-portrait.png b/art/cordova-plugin-iosrtc-sample-portrait.png
new file mode 100644
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diff --git a/art/cordova-rtc-logo.png b/art/cordova-rtc-logo.png
new file mode 100644
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diff --git a/art/sylaps-picture.png b/art/sylaps-picture.png
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diff --git a/banner.txt b/banner.txt
index 779c5d31..14738825 100644
--- a/banner.txt
+++ b/banner.txt
@@ -1,7 +1,9 @@
 /*
  * <%= pkg.name %> v<%= pkg.version %>
  * <%= pkg.description %>
- * Copyright 2015<%= currentYear > 2015 ? '-' + currentYear : '' %> <%= pkg.author %>
+ * Copyright 2015-2017 eFace2Face, Inc. (https://eface2face.com)
+ * Copyright 2015-2019 BasqueVoIPMafia (https://github.com/BasqueVoIPMafia)
+ * Copyright 2017<%= currentYear > 2017 ? '-' + currentYear : '' %> Cordova-RTC (https://github.com/cordova-rtc)
  * License <%= pkg.license %>
  */
 
diff --git a/dist/cordova-plugin-iosrtc.js b/dist/cordova-plugin-iosrtc.js
deleted file mode 100644
index 561dbca9..00000000
--- a/dist/cordova-plugin-iosrtc.js
+++ /dev/null
@@ -1,3871 +0,0 @@
-/*
- * cordova-plugin-iosrtc v3.2.2
- * Cordova iOS plugin exposing the full WebRTC W3C JavaScript APIs
- * Copyright 2015-2016 Iñaki Baz Castillo at eFace2Face, inc. (https://eface2face.com)
- * License MIT
- */
-
-(function(f){if(typeof exports==="object"&&typeof module!=="undefined"){module.exports=f()}else if(typeof define==="function"&&define.amd){define([],f)}else{var g;if(typeof window!=="undefined"){g=window}else if(typeof global!=="undefined"){g=global}else if(typeof self!=="undefined"){g=self}else{g=this}g.iosrtc = f()}})(function(){var define,module,exports;return (function e(t,n,r){function s(o,u){if(!n[o]){if(!t[o]){var a=typeof require=="function"&&require;if(!u&&a)return a(o,!0);if(i)return i(o,!0);var f=new Error("Cannot find module '"+o+"'");throw f.code="MODULE_NOT_FOUND",f}var l=n[o]={exports:{}};t[o][0].call(l.exports,function(e){var n=t[o][1][e];return s(n?n:e)},l,l.exports,e,t,n,r)}return n[o].exports}var i=typeof require=="function"&&require;for(var o=0;o<r.length;o++)s(r[o]);return s})({1:[function(require,module,exports){
-/**
- * Expose an object with WebRTC Errors.
- */
-var Errors = module.exports = {},
-
-
-/**
- * Local variables.
- */
-	IntermediateInheritor = function () {};
-
-
-IntermediateInheritor.prototype = Error.prototype;
-
-
-/**
- * Create error classes.
- */
-addError('InvalidStateError');
-addError('InvalidSessionDescriptionError');
-addError('InternalError');
-addError('MediaStreamError');
-
-
-function addError(name) {
-	Errors[name] = function () {
-		var tmp = Error.apply(this, arguments);
-
-		this.name = tmp.name = name;
-		this.message = tmp.message;
-
-		Object.defineProperty(this, 'stack', {
-			get: function () {
-				return tmp.stack;
-			}
-		});
-
-		return this;
-	};
-
-	Errors[name].prototype = new IntermediateInheritor();
-}
-
-},{}],2:[function(require,module,exports){
-/**
- * Expose the MediaDeviceInfo class.
- */
-module.exports = MediaDeviceInfo;
-
-
-function MediaDeviceInfo(data) {
-	data = data || {};
-
-	Object.defineProperties(this, {
-		// MediaDeviceInfo spec.
-		deviceId: {
-			value: data.deviceId
-		},
-		kind: {
-			value: data.kind
-		},
-		label: {
-			value: data.label
-		},
-		groupId: {
-			value: data.groupId || ''
-		},
-		// SourceInfo old spec.
-		id: {
-			value: data.deviceId
-		},
-		// Deprecated, but useful until there is an alternative
-		facing: {
-			value: ''
-		}
-	});
-}
-
-},{}],3:[function(require,module,exports){
-/**
- * Expose the MediaStream class.
- * Make MediaStream be a Blob so it can be consumed by URL.createObjectURL().
- */
-var MediaStream = module.exports = window.Blob,
-
-
-/**
- * Spec: http://w3c.github.io/mediacapture-main/#mediastream
- */
-
-
-/**
- * Dependencies.
- */
-	debug = require('debug')('iosrtc:MediaStream'),
-	exec = require('cordova/exec'),
-	EventTarget = require('yaeti').EventTarget,
-	MediaStreamTrack = require('./MediaStreamTrack'),
-
-
-/**
- * Local variables.
- */
-
-	// Dictionary of MediaStreams (provided via setMediaStreams() class method).
-	// - key: MediaStream blobId.
-	// - value: MediaStream.
-	mediaStreams;
-
-
-/**
- * Class methods.
- */
-
-
-MediaStream.setMediaStreams = function (_mediaStreams) {
-	mediaStreams = _mediaStreams;
-};
-
-
-MediaStream.create = function (dataFromEvent) {
-	debug('create() | [dataFromEvent:%o]', dataFromEvent);
-
-	var stream,
-		blobId = 'MediaStream_' + dataFromEvent.id,
-		trackId,
-		track;
-
-	// Note that this is the Blob constructor.
-	stream = new MediaStream([blobId], {
-		type: 'stream'
-	});
-
-	// Store the stream into the dictionary.
-	mediaStreams[blobId] = stream;
-
-	// Make it an EventTarget.
-	EventTarget.call(stream);
-
-	// Public atributes.
-	stream.id = dataFromEvent.id;
-	stream.label = dataFromEvent.id;  // Backwards compatibility.
-	stream.active = true;
-
-	// Public but internal attributes.
-	stream.connected = false;
-
-	// Private attributes.
-	stream._blobId = blobId;
-	stream._audioTracks = {};
-	stream._videoTracks = {};
-
-	for (trackId in dataFromEvent.audioTracks) {
-		if (dataFromEvent.audioTracks.hasOwnProperty(trackId)) {
-			track = new MediaStreamTrack(dataFromEvent.audioTracks[trackId]);
-
-			stream._audioTracks[track.id] = track;
-
-			addListenerForTrackEnded.call(stream, track);
-		}
-	}
-
-	for (trackId in dataFromEvent.videoTracks) {
-		if (dataFromEvent.videoTracks.hasOwnProperty(trackId)) {
-			track = new MediaStreamTrack(dataFromEvent.videoTracks[trackId]);
-
-			stream._videoTracks[track.id] = track;
-
-			addListenerForTrackEnded.call(stream, track);
-		}
-	}
-
-	function onResultOK(data) {
-		onEvent.call(stream, data);
-	}
-
-	exec(onResultOK, null, 'iosrtcPlugin', 'MediaStream_setListener', [stream.id]);
-
-	return stream;
-};
-
-
-MediaStream.prototype.getAudioTracks = function () {
-	debug('getAudioTracks()');
-
-	var tracks = [],
-		id;
-
-	for (id in this._audioTracks) {
-		if (this._audioTracks.hasOwnProperty(id)) {
-			tracks.push(this._audioTracks[id]);
-		}
-	}
-
-	return tracks;
-};
-
-
-MediaStream.prototype.getVideoTracks = function () {
-	debug('getVideoTracks()');
-
-	var tracks = [],
-		id;
-
-	for (id in this._videoTracks) {
-		if (this._videoTracks.hasOwnProperty(id)) {
-			tracks.push(this._videoTracks[id]);
-		}
-	}
-
-	return tracks;
-};
-
-
-MediaStream.prototype.getTracks = function () {
-	debug('getTracks()');
-
-	var tracks = [],
-		id;
-
-	for (id in this._audioTracks) {
-		if (this._audioTracks.hasOwnProperty(id)) {
-			tracks.push(this._audioTracks[id]);
-		}
-	}
-
-	for (id in this._videoTracks) {
-		if (this._videoTracks.hasOwnProperty(id)) {
-			tracks.push(this._videoTracks[id]);
-		}
-	}
-
-	return tracks;
-};
-
-
-MediaStream.prototype.getTrackById = function (id) {
-	debug('getTrackById()');
-
-	return this._audioTracks[id] || this._videoTracks[id] || null;
-};
-
-
-MediaStream.prototype.addTrack = function (track) {
-	debug('addTrack() [track:%o]', track);
-
-	if (!(track instanceof MediaStreamTrack)) {
-		throw new Error('argument must be an instance of MediaStreamTrack');
-	}
-
-	if (this._audioTracks[track.id] || this._videoTracks[track.id]) {
-		return;
-	}
-
-	if (track.kind === 'audio') {
-		this._audioTracks[track.id] = track;
-	} else if (track.kind === 'video') {
-		this._videoTracks[track.id] = track;
-	} else {
-		throw new Error('unknown kind attribute: ' + track.kind);
-	}
-
-	addListenerForTrackEnded.call(this, track);
-
-	exec(null, null, 'iosrtcPlugin', 'MediaStream_addTrack', [this.id, track.id]);
-};
-
-
-MediaStream.prototype.removeTrack = function (track) {
-	debug('removeTrack() [track:%o]', track);
-
-	if (!(track instanceof MediaStreamTrack)) {
-		throw new Error('argument must be an instance of MediaStreamTrack');
-	}
-
-	if (!this._audioTracks[track.id] && !this._videoTracks[track.id]) {
-		return;
-	}
-
-	if (track.kind === 'audio') {
-		delete this._audioTracks[track.id];
-	} else if (track.kind === 'video') {
-		delete this._videoTracks[track.id];
-	} else {
-		throw new Error('unknown kind attribute: ' + track.kind);
-	}
-
-	exec(null, null, 'iosrtcPlugin', 'MediaStream_removeTrack', [this.id, track.id]);
-
-	checkActive.call(this);
-};
-
-
-// Backwards compatible API.
-MediaStream.prototype.stop = function () {
-	debug('stop()');
-
-	var trackId;
-
-	for (trackId in this._audioTracks) {
-		if (this._audioTracks.hasOwnProperty(trackId)) {
-			this._audioTracks[trackId].stop();
-		}
-	}
-
-	for (trackId in this._videoTracks) {
-		if (this._videoTracks.hasOwnProperty(trackId)) {
-			this._videoTracks[trackId].stop();
-		}
-	}
-};
-
-
-// TODO: API methods and events.
-
-
-/**
- * Private API.
- */
-
-
-MediaStream.prototype.emitConnected = function () {
-	debug('emitConnected()');
-
-	var self = this;
-
-	if (this.connected) {
-		return;
-	}
-	this.connected = true;
-
-	setTimeout(function () {
-		self.dispatchEvent(new Event('connected'));
-	});
-};
-
-
-function addListenerForTrackEnded(track) {
-	var self = this;
-
-	track.addEventListener('ended', function () {
-		if (track.kind === 'audio' && !self._audioTracks[track.id]) {
-			return;
-		} else if (track.kind === 'video' && !self._videoTracks[track.id]) {
-			return;
-		}
-
-		checkActive.call(self);
-	});
-}
-
-
-function checkActive() {
-	// A MediaStream object is said to be active when it has at least one MediaStreamTrack
-	// that has not ended. A MediaStream that does not have any tracks or only has tracks
-	// that are ended is inactive.
-
-	var self = this,
-		trackId;
-
-	if (!this.active) {
-		return;
-	}
-
-	if (Object.keys(this._audioTracks).length === 0 && Object.keys(this._videoTracks).length === 0) {
-		debug('no tracks, releasing MediaStream');
-
-		release();
-		return;
-	}
-
-	for (trackId in this._audioTracks) {
-		if (this._audioTracks.hasOwnProperty(trackId)) {
-			if (this._audioTracks[trackId].readyState !== 'ended') {
-				return;
-			}
-		}
-	}
-
-	for (trackId in this._videoTracks) {
-		if (this._videoTracks.hasOwnProperty(trackId)) {
-			if (this._videoTracks[trackId].readyState !== 'ended') {
-				return;
-			}
-		}
-	}
-
-	debug('all tracks are ended, releasing MediaStream');
-	release();
-
-	function release() {
-		self.active = false;
-		self.dispatchEvent(new Event('inactive'));
-
-		// Remove the stream from the dictionary.
-		delete mediaStreams[self._blobId];
-
-		exec(null, null, 'iosrtcPlugin', 'MediaStream_release', [self.id]);
-	}
-}
-
-
-function onEvent(data) {
-	var type = data.type,
-		event,
-		track;
-
-	debug('onEvent() | [type:%s, data:%o]', type, data);
-
-	switch (type) {
-		case 'addtrack':
-			track = new MediaStreamTrack(data.track);
-
-			if (track.kind === 'audio') {
-				this._audioTracks[track.id] = track;
-			} else if (track.kind === 'video') {
-				this._videoTracks[track.id] = track;
-			}
-			addListenerForTrackEnded.call(this, track);
-
-			event = new Event('addtrack');
-			event.track = track;
-
-			this.dispatchEvent(event);
-
-			// Also emit 'update' for the MediaStreamRenderer.
-			this.dispatchEvent(new Event('update'));
-			break;
-
-		case 'removetrack':
-			if (data.track.kind === 'audio') {
-				track = this._audioTracks[data.track.id];
-				delete this._audioTracks[data.track.id];
-			} else if (data.track.kind === 'video') {
-				track = this._videoTracks[data.track.id];
-				delete this._videoTracks[data.track.id];
-			}
-
-			if (!track) {
-				throw new Error('"removetrack" event fired on MediaStream for a non existing MediaStreamTrack');
-			}
-
-			event = new Event('removetrack');
-			event.track = track;
-
-			this.dispatchEvent(event);
-			// Also emit 'update' for the MediaStreamRenderer.
-			this.dispatchEvent(new Event('update'));
-
-			// Check whether the MediaStream still is active.
-			checkActive.call(this);
-			break;
-	}
-}
-
-},{"./MediaStreamTrack":5,"cordova/exec":undefined,"debug":18,"yaeti":24}],4:[function(require,module,exports){
-/**
- * Expose the MediaStreamRenderer class.
- */
-module.exports = MediaStreamRenderer;
-
-
-/**
- * Dependencies.
- */
-var
-	debug = require('debug')('iosrtc:MediaStreamRenderer'),
-	exec = require('cordova/exec'),
-	randomNumber = require('random-number').generator({min: 10000, max: 99999, integer: true}),
-	EventTarget = require('yaeti').EventTarget,
-	MediaStream = require('./MediaStream');
-
-
-function MediaStreamRenderer(element) {
-	debug('new() | [element:"%s"]', element);
-
-	var self = this;
-
-	// Make this an EventTarget.
-	EventTarget.call(this);
-
-	if (!(element instanceof HTMLElement)) {
-		throw new Error('a valid HTMLElement is required');
-	}
-
-	// Public atributes.
-	this.element = element;
-	this.stream = undefined;
-	this.videoWidth = undefined;
-	this.videoHeight = undefined;
-
-	// Private attributes.
-	this.id = randomNumber();
-
-	function onResultOK(data) {
-		onEvent.call(self, data);
-	}
-
-	exec(onResultOK, null, 'iosrtcPlugin', 'new_MediaStreamRenderer', [this.id]);
-
-	this.refresh(this);
-}
-
-
-MediaStreamRenderer.prototype.render = function (stream) {
-	debug('render() [stream:%o]', stream);
-
-	var self = this;
-
-	if (!(stream instanceof MediaStream)) {
-		throw new Error('render() requires a MediaStream instance as argument');
-	}
-
-	this.stream = stream;
-
-	exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_render', [this.id, stream.id]);
-
-	// Subscribe to 'update' event so we call native mediaStreamChanged() on it.
-	stream.addEventListener('update', function () {
-		if (self.stream !== stream) {
-			return;
-		}
-
-		debug('MediaStream emits "update", calling native mediaStreamChanged()');
-
-		exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_mediaStreamChanged', [self.id]);
-	});
-
-	// Subscribe to 'inactive' event and emit "close" so the video element can react.
-	stream.addEventListener('inactive', function () {
-		if (self.stream !== stream) {
-			return;
-		}
-
-		debug('MediaStream emits "inactive", emiting "close" and closing this MediaStreamRenderer');
-
-		self.dispatchEvent(new Event('close'));
-		self.close();
-	});
-
-	if (stream.connected) {
-		connected();
-	// Otherwise subscribe to 'connected' event to emulate video elements events.
-	} else {
-		stream.addEventListener('connected', function () {
-			if (self.stream !== stream) {
-				return;
-			}
-
-			connected();
-		});
-	}
-
-	function connected() {
-		// Emit video events.
-		self.element.dispatchEvent(new Event('loadedmetadata'));
-		self.element.dispatchEvent(new Event('loadeddata'));
-		self.element.dispatchEvent(new Event('canplay'));
-		self.element.dispatchEvent(new Event('canplaythrough'));
-	}
-};
-
-
-MediaStreamRenderer.prototype.refresh = function () {
-	debug('refresh()');
-
-	var elementPositionAndSize = getElementPositionAndSize.call(this),
-		computedStyle,
-		videoRatio,
-		elementRatio,
-		elementLeft = elementPositionAndSize.left,
-		elementTop = elementPositionAndSize.top,
-		elementWidth = elementPositionAndSize.width,
-		elementHeight = elementPositionAndSize.height,
-		videoViewWidth,
-		videoViewHeight,
-		visible,
-		opacity,
-		zIndex,
-		mirrored,
-		objectFit,
-		clip,
-		borderRadius,
-		paddingTop,
-		paddingBottom,
-		paddingLeft,
-		paddingRight;
-
-	computedStyle = window.getComputedStyle(this.element);
-
-	// get padding values
-	paddingTop = parseInt(computedStyle.paddingTop) | 0;
-	paddingBottom = parseInt(computedStyle.paddingBottom) | 0;
-	paddingLeft = parseInt(computedStyle.paddingLeft) | 0;
-	paddingRight = parseInt(computedStyle.paddingRight) | 0;
-
-	// fix position according to padding
-	elementLeft += paddingLeft;
-	elementTop += paddingTop;
-
-	// fix width and height according to padding
-	elementWidth -= (paddingLeft + paddingRight);
-	elementHeight -= (paddingTop + paddingBottom);
-
-	videoViewWidth = elementWidth;
-	videoViewHeight = elementHeight;
-
-	// visible
-	if (computedStyle.visibility === 'hidden') {
-		visible = false;
-	} else {
-		visible = !!this.element.offsetHeight;  // Returns 0 if element or any parent is hidden.
-	}
-
-	// opacity
-	opacity = parseFloat(computedStyle.opacity);
-
-	// zIndex
-	zIndex = parseFloat(computedStyle.zIndex) || parseFloat(this.element.style.zIndex) || 0;
-
-	// mirrored (detect "-webkit-transform: scaleX(-1);" or equivalent)
-	if (computedStyle.transform === 'matrix(-1, 0, 0, 1, 0, 0)' ||
-		computedStyle['-webkit-transform'] === 'matrix(-1, 0, 0, 1, 0, 0)') {
-		mirrored = true;
-	} else {
-		mirrored = false;
-	}
-
-	// objectFit ('contain' is set as default value)
-	objectFit = computedStyle.objectFit || 'contain';
-
-	// clip
-	if (objectFit === 'none') {
-		clip = false;
-	} else {
-		clip = true;
-	}
-
-	// borderRadius
-	borderRadius = parseFloat(computedStyle.borderRadius);
-	if (/%$/.test(borderRadius)) {
-		borderRadius = Math.min(elementHeight, elementWidth) * borderRadius;
-	}
-
-	/**
-	 * No video yet, so just update the UIView with the element settings.
-	 */
-
-	if (!this.videoWidth || !this.videoHeight) {
-		debug('refresh() | no video track yet');
-
-		nativeRefresh.call(this);
-		return;
-	}
-
-	videoRatio = this.videoWidth / this.videoHeight;
-
-	/**
-	 * Element has no width and/or no height.
-	 */
-
-	if (!elementWidth || !elementHeight) {
-		debug('refresh() | video element has 0 width and/or 0 height');
-
-		nativeRefresh.call(this);
-		return;
-	}
-
-	/**
-	 * Set video view position and size.
-	 */
-
-	elementRatio = elementWidth / elementHeight;
-
-	switch (objectFit) {
-		case 'cover':
-			// The element has higher or equal width/height ratio than the video.
-			if (elementRatio >= videoRatio) {
-				videoViewWidth = elementWidth;
-				videoViewHeight = videoViewWidth / videoRatio;
-			// The element has lower width/height ratio than the video.
-			} else if (elementRatio < videoRatio) {
-				videoViewHeight = elementHeight;
-				videoViewWidth = videoViewHeight * videoRatio;
-			}
-			break;
-
-		case 'fill':
-			videoViewHeight = elementHeight;
-			videoViewWidth = elementWidth;
-			break;
-
-		case 'none':
-			videoViewHeight = this.videoHeight;
-			videoViewWidth = this.videoWidth;
-			break;
-
-		case 'scale-down':
-			// Same as 'none'.
-			if (this.videoWidth <= elementWidth && this.videoHeight <= elementHeight) {
-				videoViewHeight = this.videoHeight;
-				videoViewWidth = this.videoWidth;
-			// Same as 'contain'.
-			} else {
-				// The element has higher or equal width/height ratio than the video.
-				if (elementRatio >= videoRatio) {
-					videoViewHeight = elementHeight;
-					videoViewWidth = videoViewHeight * videoRatio;
-				// The element has lower width/height ratio than the video.
-				} else if (elementRatio < videoRatio) {
-					videoViewWidth = elementWidth;
-					videoViewHeight = videoViewWidth / videoRatio;
-				}
-			}
-			break;
-
-		// 'contain'.
-		default:
-			objectFit = 'contain';
-			// The element has higher or equal width/height ratio than the video.
-			if (elementRatio >= videoRatio) {
-				videoViewHeight = elementHeight;
-				videoViewWidth = videoViewHeight * videoRatio;
-			// The element has lower width/height ratio than the video.
-			} else if (elementRatio < videoRatio) {
-				videoViewWidth = elementWidth;
-				videoViewHeight = videoViewWidth / videoRatio;
-			}
-			break;
-	}
-
-	nativeRefresh.call(this);
-
-	function nativeRefresh() {
-		var data = {
-			elementLeft: elementLeft,
-			elementTop: elementTop,
-			elementWidth: elementWidth,
-			elementHeight: elementHeight,
-			videoViewWidth: videoViewWidth,
-			videoViewHeight: videoViewHeight,
-			visible: visible,
-			opacity: opacity,
-			zIndex: zIndex,
-			mirrored: mirrored,
-			objectFit: objectFit,
-			clip: clip,
-			borderRadius: borderRadius
-		};
-
-		debug('refresh() | [data:%o]', data);
-
-		exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_refresh', [this.id, data]);
-	}
-};
-
-
-MediaStreamRenderer.prototype.close = function () {
-	debug('close()');
-
-	if (!this.stream) {
-		return;
-	}
-	this.stream = undefined;
-
-	exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_close', [this.id]);
-};
-
-
-/**
- * Private API.
- */
-
-
-function onEvent(data) {
-	var type = data.type,
-		event;
-
-	debug('onEvent() | [type:%s, data:%o]', type, data);
-
-	switch (type) {
-		case 'videoresize':
-			this.videoWidth = data.size.width;
-			this.videoHeight = data.size.height;
-			this.refresh(this);
-
-			event = new Event(type);
-			event.videoWidth = data.size.width;
-			event.videoHeight = data.size.height;
-			this.dispatchEvent(event);
-
-			break;
-	}
-}
-
-
-function getElementPositionAndSize() {
-	var rect = this.element.getBoundingClientRect();
-
-	return {
-		left:   rect.left + this.element.clientLeft,
-		top:    rect.top + this.element.clientTop,
-		width:  this.element.clientWidth,
-		height: this.element.clientHeight
-	};
-}
-
-},{"./MediaStream":3,"cordova/exec":undefined,"debug":18,"random-number":23,"yaeti":24}],5:[function(require,module,exports){
-/**
- * Expose the MediaStreamTrack class.
- */
-module.exports = MediaStreamTrack;
-
-
-/**
- * Spec: http://w3c.github.io/mediacapture-main/#mediastreamtrack
- */
-
-
-/**
- * Dependencies.
- */
-var
-	debug = require('debug')('iosrtc:MediaStreamTrack'),
-	exec = require('cordova/exec'),
-	enumerateDevices = require('./enumerateDevices'),
-	EventTarget = require('yaeti').EventTarget;
-
-
-function MediaStreamTrack(dataFromEvent) {
-	debug('new() | [dataFromEvent:%o]', dataFromEvent);
-
-	var self = this;
-
-	// Make this an EventTarget.
-	EventTarget.call(this);
-
-	// Public atributes.
-	this.id = dataFromEvent.id;  // NOTE: It's a string.
-	this.kind = dataFromEvent.kind;
-	this.label = dataFromEvent.label;
-	this.muted = false;  // TODO: No "muted" property in ObjC API.
-	this.readyState = dataFromEvent.readyState;
-
-	// Private attributes.
-	this._enabled = dataFromEvent.enabled;
-	this._ended = false;
-
-	function onResultOK(data) {
-		onEvent.call(self, data);
-	}
-
-	exec(onResultOK, null, 'iosrtcPlugin', 'MediaStreamTrack_setListener', [this.id]);
-}
-
-
-// Setters.
-Object.defineProperty(MediaStreamTrack.prototype, 'enabled', {
-	get: function () {
-		return this._enabled;
-	},
-	set: function (value) {
-		debug('enabled = %s', !!value);
-
-		this._enabled = !!value;
-		exec(null, null, 'iosrtcPlugin', 'MediaStreamTrack_setEnabled', [this.id, this._enabled]);
-	}
-});
-
-
-MediaStreamTrack.prototype.stop = function () {
-	debug('stop()');
-
-	if (this._ended) {
-		return;
-	}
-
-	exec(null, null, 'iosrtcPlugin', 'MediaStreamTrack_stop', [this.id]);
-};
-
-
-// TODO: API methods and events.
-
-
-/**
- * Class methods.
- */
-
-
-MediaStreamTrack.getSources = function () {
-	debug('getSources()');
-
-	return enumerateDevices.apply(this, arguments);
-};
-
-
-/**
- * Private API.
- */
-
-
-function onEvent(data) {
-	var type = data.type;
-
-	debug('onEvent() | [type:%s, data:%o]', type, data);
-
-	switch (type) {
-		case 'statechange':
-			this.readyState = data.readyState;
-			this._enabled = data.enabled;
-
-			switch (data.readyState) {
-				case 'initializing':
-					break;
-				case 'live':
-					break;
-				case 'ended':
-					this._ended = true;
-					this.dispatchEvent(new Event('ended'));
-					break;
-				case 'failed':
-					break;
-			}
-			break;
-	}
-}
-
-},{"./enumerateDevices":13,"cordova/exec":undefined,"debug":18,"yaeti":24}],6:[function(require,module,exports){
-/**
- * Expose the RTCDTMFSender class.
- */
-module.exports = RTCDTMFSender;
-
-
-/**
- * Dependencies.
- */
-var
-	debug = require('debug')('iosrtc:RTCDTMFSender'),
-	debugerror = require('debug')('iosrtc:ERROR:RTCDTMFSender'),
-	exec = require('cordova/exec'),
-	randomNumber = require('random-number').generator({min: 10000, max: 99999, integer: true}),
-	EventTarget = require('yaeti').EventTarget;
-
-
-debugerror.log = console.warn.bind(console);
-
-
-function RTCDTMFSender(peerConnection, track) {
-	var self = this;
-
-	// Make this an EventTarget.
-	EventTarget.call(this);
-
-	debug('new() | [track:%o]', track);
-
-	// Public atributes (accessed as read-only properties)
-	this._track = track;
-	// TODO: read these from the properties exposed in Swift?
-	this._duration = 100;
-	this._interToneGap = 70;
-	this._toneBuffer = '';
-
-	// Private attributes.
-	this.peerConnection = peerConnection;
-	this.dsId = randomNumber();
-
-	function onResultOK(data) {
-		onEvent.call(self, data);
-	}
-
-	exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_createDTMFSender', [this.peerConnection.pcId, this.dsId, this._track.id]);
-
-}
-
-
-Object.defineProperty(RTCDTMFSender.prototype, 'canInsertDTMF', {
-	get: function () {
-		// TODO: check if it's muted or stopped?
-		return this._track && this._track.kind === 'audio' && this._track.enabled;
-	}
-});
-
-
-Object.defineProperty(RTCDTMFSender.prototype, 'track', {
-	get: function () {
-		return this._track;
-	}
-});
-
-
-Object.defineProperty(RTCDTMFSender.prototype, 'duration', {
-	get: function () {
-		return this._duration;
-	}
-});
-
-
-Object.defineProperty(RTCDTMFSender.prototype, 'interToneGap', {
-	get: function () {
-		return this._interToneGap;
-	}
-});
-
-
-Object.defineProperty(RTCDTMFSender.prototype, 'toneBuffer', {
-	get: function () {
-		return this._toneBuffer;
-	}
-});
-
-
-RTCDTMFSender.prototype.insertDTMF = function (tones, duration, interToneGap) {
-	if (isClosed.call(this)) {
-		return;
-	}
-
-	debug('insertDTMF() | [tones:%o, duration:%o, interToneGap:%o]', tones, duration, interToneGap);
-
-	if (!tones) {
-		return;
-	}
-
-	this._duration = duration || 100;
-	this._interToneGap = interToneGap || 70;
-
-	var self = this;
-
-	function onResultOK(data) {
-		onEvent.call(self, data);
-	}
-
-	exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDTMFSender_insertDTMF', [this.peerConnection.pcId, this.dsId, tones, this._duration, this._interToneGap]);
-};
-
-
-/**
- * Private API.
- */
-
-
-function isClosed() {
-	return this.peerConnection.signalingState === 'closed';
-}
-
-
-function onEvent(data) {
-	var type = data.type,
-		event;
-
-	debug('onEvent() | [type:%s, data:%o]', type, data);
-
-	if (type === 'tonechange') {
-		event = new Event('tonechange');
-		event.tone = data.tone;
-		this.dispatchEvent(event);
-	}
-}
-
-},{"cordova/exec":undefined,"debug":18,"random-number":23,"yaeti":24}],7:[function(require,module,exports){
-/**
- * Expose the RTCDataChannel class.
- */
-module.exports = RTCDataChannel;
-
-
-/**
- * Dependencies.
- */
-var
-	debug = require('debug')('iosrtc:RTCDataChannel'),
-	debugerror = require('debug')('iosrtc:ERROR:RTCDataChannel'),
-	exec = require('cordova/exec'),
-	randomNumber = require('random-number').generator({min: 10000, max: 99999, integer: true}),
-	EventTarget = require('yaeti').EventTarget;
-
-
-debugerror.log = console.warn.bind(console);
-
-
-function RTCDataChannel(peerConnection, label, options, dataFromEvent) {
-	var self = this;
-
-	// Make this an EventTarget.
-	EventTarget.call(this);
-
-	// Created via pc.createDataChannel().
-	if (!dataFromEvent) {
-		debug('new() | [label:%o, options:%o]', label, options);
-
-		if (typeof label !== 'string') {
-			label = '';
-		}
-
-		options = options || {};
-
-		if (options.hasOwnProperty('maxPacketLifeTime') && options.hasOwnProperty('maxRetransmits')) {
-			throw new SyntaxError('both maxPacketLifeTime and maxRetransmits can not be present');
-		}
-
-		if (options.hasOwnProperty('id')) {
-			if (typeof options.id !== 'number' || isNaN(options.id) || options.id < 0) {
-				throw new SyntaxError('id must be a number');
-			}
-			// TODO:
-			//   https://code.google.com/p/webrtc/issues/detail?id=4618
-			if (options.id > 1023) {
-				throw new SyntaxError('id cannot be greater than 1023 (https://code.google.com/p/webrtc/issues/detail?id=4614)');
-			}
-		}
-
-		// Public atributes.
-		this.label = label;
-		this.ordered = options.hasOwnProperty('ordered') ? !!options.ordered : true;
-		this.maxPacketLifeTime = options.hasOwnProperty('maxPacketLifeTime') ? Number(options.maxPacketLifeTime) : null;
-		this.maxRetransmits = options.hasOwnProperty('maxRetransmits') ? Number(options.maxRetransmits) : null;
-		this.protocol = options.hasOwnProperty('protocol') ? String(options.protocol) : '';
-		this.negotiated = options.hasOwnProperty('negotiated') ? !!options.negotiated : false;
-		this.id = options.hasOwnProperty('id') ? Number(options.id) : undefined;
-		this.readyState = 'connecting';
-		this.bufferedAmount = 0;
-		this.bufferedAmountLowThreshold = 0;
-
-		// Private attributes.
-		this.peerConnection = peerConnection;
-		this.dcId = randomNumber();
-
-		exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_createDataChannel', [this.peerConnection.pcId, this.dcId, label, options]);
-	// Created via pc.ondatachannel.
-	} else {
-		debug('new() | [dataFromEvent:%o]', dataFromEvent);
-
-		// Public atributes.
-		this.label = dataFromEvent.label;
-		this.ordered = dataFromEvent.ordered;
-		this.maxPacketLifeTime = dataFromEvent.maxPacketLifeTime;
-		this.maxRetransmits = dataFromEvent.maxRetransmits;
-		this.protocol = dataFromEvent.protocol;
-		this.negotiated = dataFromEvent.negotiated;
-		this.id = dataFromEvent.id;
-		this.readyState = dataFromEvent.readyState;
-		this.bufferedAmount = dataFromEvent.bufferedAmount;
-		this.bufferedAmountLowThreshold = dataFromEvent.bufferedAmountLowThreshold;
-
-		// Private attributes.
-		this.peerConnection = peerConnection;
-		this.dcId = dataFromEvent.dcId;
-
-		exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_setListener', [this.peerConnection.pcId, this.dcId]);
-	}
-
-	function onResultOK(data) {
-		if (data.type) {
-			onEvent.call(self, data);
-		// Special handler for received binary mesage.
-		} else {
-			onEvent.call(self, {
-				type: 'message',
-				message: data
-			});
-		}
-	}
-}
-
-
-// Just 'arraybuffer' binaryType is implemented in Chromium.
-Object.defineProperty(RTCDataChannel.prototype, 'binaryType', {
-	get: function () {
-		return 'arraybuffer';
-	},
-	set: function (type) {
-		if (type !== 'arraybuffer') {
-			throw new Error('just "arraybuffer" is implemented for binaryType');
-		}
-	}
-});
-
-
-RTCDataChannel.prototype.send = function (data) {
-	if (isClosed.call(this) || this.readyState !== 'open') {
-		return;
-	}
-
-	debug('send() | [data:%o]', data);
-
-	if (!data) {
-		return;
-	}
-
-	if (typeof data === 'string' || data instanceof String) {
-		exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendString', [this.peerConnection.pcId, this.dcId, data]);
-	} else if (window.ArrayBuffer && data instanceof window.ArrayBuffer) {
-		exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendBinary', [this.peerConnection.pcId, this.dcId, data]);
-	} else if (
-		(window.Int8Array && data instanceof window.Int8Array) ||
-		(window.Uint8Array && data instanceof window.Uint8Array) ||
-		(window.Uint8ClampedArray && data instanceof window.Uint8ClampedArray) ||
-		(window.Int16Array && data instanceof window.Int16Array) ||
-		(window.Uint16Array && data instanceof window.Uint16Array) ||
-		(window.Int32Array && data instanceof window.Int32Array) ||
-		(window.Uint32Array && data instanceof window.Uint32Array) ||
-		(window.Float32Array && data instanceof window.Float32Array) ||
-		(window.Float64Array && data instanceof window.Float64Array) ||
-		(window.DataView && data instanceof window.DataView)
-	) {
-		exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendBinary', [this.peerConnection.pcId, this.dcId, data.buffer]);
-	} else {
-		throw new Error('invalid data type');
-	}
-};
-
-
-RTCDataChannel.prototype.close = function () {
-	if (isClosed.call(this)) {
-		return;
-	}
-
-	debug('close()');
-
-	this.readyState = 'closing';
-
-	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_close', [this.peerConnection.pcId, this.dcId]);
-};
-
-
-/**
- * Private API.
- */
-
-
-function isClosed() {
-	return this.readyState === 'closed' || this.readyState === 'closing' || this.peerConnection.signalingState === 'closed';
-}
-
-
-function onEvent(data) {
-	var type = data.type,
-		event;
-
-	debug('onEvent() | [type:%s, data:%o]', type, data);
-
-	switch (type) {
-		case 'new':
-			// Update properties and exit without firing the event.
-			this.ordered = data.channel.ordered;
-			this.maxPacketLifeTime = data.channel.maxPacketLifeTime;
-			this.maxRetransmits = data.channel.maxRetransmits;
-			this.protocol = data.channel.protocol;
-			this.negotiated = data.channel.negotiated;
-			this.id = data.channel.id;
-			this.readyState = data.channel.readyState;
-			this.bufferedAmount = data.channel.bufferedAmount;
-			break;
-
-		case 'statechange':
-			this.readyState = data.readyState;
-
-			switch (data.readyState) {
-				case 'connecting':
-					break;
-				case 'open':
-					this.dispatchEvent(new Event('open'));
-					break;
-				case 'closing':
-					break;
-				case 'closed':
-					this.dispatchEvent(new Event('close'));
-					break;
-			}
-			break;
-
-		case 'message':
-			event = new Event('message');
-			event.data = data.message;
-			this.dispatchEvent(event);
-			break;
-
-		case 'bufferedamount':
-			this.bufferedAmount = data.bufferedAmount;
-
-			if (this.bufferedAmountLowThreshold > 0 && this.bufferedAmountLowThreshold > this.bufferedAmount) {
-				event = new Event('bufferedamountlow');
-				event.bufferedAmount = this.bufferedAmount;
-				this.dispatchEvent(event);
-			}
-
-			break;
-	}
-}
-
-},{"cordova/exec":undefined,"debug":18,"random-number":23,"yaeti":24}],8:[function(require,module,exports){
-/**
- * Expose the RTCIceCandidate class.
- */
-module.exports = RTCIceCandidate;
-
-
-function RTCIceCandidate(data) {
-	data = data || {};
-
-	// Public atributes.
-	this.sdpMid = data.sdpMid;
-	this.sdpMLineIndex = data.sdpMLineIndex;
-	this.candidate = data.candidate;
-}
-
-},{}],9:[function(require,module,exports){
-(function (global){
-/**
- * Expose the RTCPeerConnection class.
- */
-module.exports = RTCPeerConnection;
-
-
-/**
- * Dependencies.
- */
-var
-	debug = require('debug')('iosrtc:RTCPeerConnection'),
-	debugerror = require('debug')('iosrtc:ERROR:RTCPeerConnection'),
-	exec = require('cordova/exec'),
-	randomNumber = require('random-number').generator({min: 10000, max: 99999, integer: true}),
-	EventTarget = require('yaeti').EventTarget,
-	RTCSessionDescription = require('./RTCSessionDescription'),
-	RTCIceCandidate = require('./RTCIceCandidate'),
-	RTCDataChannel = require('./RTCDataChannel'),
-	RTCDTMFSender = require('./RTCDTMFSender'),
-	RTCStatsResponse = require('./RTCStatsResponse'),
-	RTCStatsReport = require('./RTCStatsReport'),
-	MediaStream = require('./MediaStream'),
-	MediaStreamTrack = require('./MediaStreamTrack'),
-	Errors = require('./Errors');
-
-
-debugerror.log = console.warn.bind(console);
-
-
-function RTCPeerConnection(pcConfig, pcConstraints) {
-	debug('new() | [pcConfig:%o, pcConstraints:%o]', pcConfig, pcConstraints);
-
-	var self = this;
-
-	// Make this an EventTarget.
-	EventTarget.call(this);
-
-	// Public atributes.
-	this.localDescription = null;
-	this.remoteDescription = null;
-	this.signalingState = 'stable';
-	this.iceGatheringState = 'new';
-	this.iceConnectionState = 'new';
-	this.pcConfig = fixPcConfig(pcConfig);
-
-	// Private attributes.
-	this.pcId = randomNumber();
-	this.localStreams = {};
-	this.remoteStreams = {};
-
-	function onResultOK(data) {
-		onEvent.call(self, data);
-	}
-
-	exec(onResultOK, null, 'iosrtcPlugin', 'new_RTCPeerConnection', [this.pcId, this.pcConfig, pcConstraints]);
-}
-
-
-RTCPeerConnection.prototype.createOffer = function () {
-	var self = this,
-		isPromise,
-		options,
-		callback, errback;
-
-	if (typeof arguments[0] !== 'function') {
-		isPromise = true;
-		options = arguments[0];
-	} else {
-		isPromise = false;
-		callback = arguments[0];
-		errback = arguments[1];
-		options = arguments[2];
-	}
-
-	if (isClosed.call(this)) {
-		return;
-	}
-
-	debug('createOffer() [options:%o]', options);
-
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				var desc = new RTCSessionDescription(data);
-
-				debug('createOffer() | success [desc:%o]', desc);
-				resolve(desc);
-			}
-
-			function onResultError(error) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('createOffer() | failure: %s', error);
-				reject(new global.DOMError(error));
-			}
-
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createOffer', [self.pcId, options]);
-		});
-	}
-
-	function onResultOK(data) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		var desc = new RTCSessionDescription(data);
-
-		debug('createOffer() | success [desc:%o]', desc);
-		callback(desc);
-	}
-
-	function onResultError(error) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		debugerror('createOffer() | failure: %s', error);
-		if (typeof errback === 'function') {
-			errback(new global.DOMError(error));
-		}
-	}
-
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createOffer', [this.pcId, options]);
-};
-
-
-RTCPeerConnection.prototype.createAnswer = function () {
-	var self = this,
-		isPromise,
-		options,
-		callback, errback;
-
-	if (typeof arguments[0] !== 'function') {
-		isPromise = true;
-		options = arguments[0];
-	} else {
-		isPromise = false;
-		callback = arguments[0];
-		errback = arguments[1];
-		options = arguments[2];
-	}
-
-	if (isClosed.call(this)) {
-		return;
-	}
-
-	debug('createAnswer() [options:%o]', options);
-
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				var desc = new RTCSessionDescription(data);
-
-				debug('createAnswer() | success [desc:%o]', desc);
-				resolve(desc);
-			}
-
-			function onResultError(error) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('createAnswer() | failure: %s', error);
-				reject(new global.DOMError(error));
-			}
-
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createAnswer', [self.pcId, options]);
-		});
-	}
-
-	function onResultOK(data) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		var desc = new RTCSessionDescription(data);
-
-		debug('createAnswer() | success [desc:%o]', desc);
-		callback(desc);
-	}
-
-	function onResultError(error) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		debugerror('createAnswer() | failure: %s', error);
-		if (typeof errback === 'function') {
-			errback(new global.DOMError(error));
-		}
-	}
-
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createAnswer', [this.pcId, options]);
-};
-
-
-
-RTCPeerConnection.prototype.setLocalDescription = function (desc) {
-	var self = this,
-		isPromise,
-		callback, errback;
-
-	if (typeof arguments[1] !== 'function') {
-		isPromise = true;
-	} else {
-		isPromise = false;
-		callback = arguments[1];
-		errback = arguments[2];
-	}
-
-	if (isClosed.call(this)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.InvalidStateError('peerconnection is closed'));
-			});
-		} else {
-			throw new Errors.InvalidStateError('peerconnection is closed');
-		}
-	}
-
-	debug('setLocalDescription() [desc:%o]', desc);
-
-	if (!(desc instanceof RTCSessionDescription)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.InvalidSessionDescriptionError('setLocalDescription() must be called with a RTCSessionDescription instance as first argument'));
-			});
-		} else {
-			if (typeof errback === 'function') {
-				errback(new Errors.InvalidSessionDescriptionError('setLocalDescription() must be called with a RTCSessionDescription instance as first argument'));
-			}
-			return;
-		}
-	}
-
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debug('setLocalDescription() | success');
-				// Update localDescription.
-				self.localDescription = new RTCSessionDescription(data);
-				resolve();
-			}
-
-			function onResultError(error) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('setLocalDescription() | failure: %s', error);
-				reject(new Errors.InvalidSessionDescriptionError('setLocalDescription() failed: ' + error));
-			}
-
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setLocalDescription', [self.pcId, desc]);
-		});
-	}
-
-	function onResultOK(data) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		debug('setLocalDescription() | success');
-		// Update localDescription.
-		self.localDescription = new RTCSessionDescription(data);
-		callback();
-	}
-
-	function onResultError(error) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		debugerror('setLocalDescription() | failure: %s', error);
-
-		if (typeof errback === 'function') {
-			errback(new Errors.InvalidSessionDescriptionError('setLocalDescription() failed: ' + error));
-		}
-	}
-
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setLocalDescription', [this.pcId, desc]);
-};
-
-
-RTCPeerConnection.prototype.setRemoteDescription = function (desc) {
-	var self = this,
-		isPromise,
-		callback, errback;
-
-	if (typeof arguments[1] !== 'function') {
-		isPromise = true;
-	} else {
-		isPromise = false;
-		callback = arguments[1];
-		errback = arguments[2];
-	}
-
-	if (isClosed.call(this)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.InvalidStateError('peerconnection is closed'));
-			});
-		} else {
-			throw new Errors.InvalidStateError('peerconnection is closed');
-		}
-	}
-
-	debug('setRemoteDescription() [desc:%o]', desc);
-
-	if (!(desc instanceof RTCSessionDescription)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.InvalidSessionDescriptionError('setRemoteDescription() must be called with a RTCSessionDescription instance as first argument'));
-			});
-		} else {
-			if (typeof errback === 'function') {
-				errback(new Errors.InvalidSessionDescriptionError('setRemoteDescription() must be called with a RTCSessionDescription instance as first argument'));
-			}
-			return;
-		}
-	}
-
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debug('setRemoteDescription() | success');
-				// Update remoteDescription.
-				self.remoteDescription = new RTCSessionDescription(data);
-				resolve();
-			}
-
-			function onResultError(error) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('setRemoteDescription() | failure: %s', error);
-				reject(new Errors.InvalidSessionDescriptionError('setRemoteDescription() failed: ' + error));
-			}
-
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setRemoteDescription', [self.pcId, desc]);
-		});
-	}
-
-	function onResultOK(data) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		debug('setRemoteDescription() | success');
-		// Update remoteDescription.
-		self.remoteDescription = new RTCSessionDescription(data);
-		callback();
-	}
-
-	function onResultError(error) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		debugerror('setRemoteDescription() | failure: %s', error);
-
-		if (typeof errback === 'function') {
-			errback(new Errors.InvalidSessionDescriptionError('setRemoteDescription() failed: ' + error));
-		}
-	}
-
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setRemoteDescription', [this.pcId, desc]);
-};
-
-
-
-RTCPeerConnection.prototype.addIceCandidate = function (candidate) {
-	var self = this,
-		isPromise,
-		callback, errback;
-
-	if (typeof arguments[1] !== 'function') {
-		isPromise = true;
-	} else {
-		isPromise = false;
-		callback = arguments[1];
-		errback = arguments[2];
-	}
-
-	if (isClosed.call(this)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.InvalidStateError('peerconnection is closed'));
-			});
-		} else {
-			throw new Errors.InvalidStateError('peerconnection is closed');
-		}
-	}
-
-	debug('addIceCandidate() | [candidate:%o]', candidate);
-
-	if (!(candidate instanceof RTCIceCandidate)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new global.DOMError('addIceCandidate() must be called with a RTCIceCandidate instance as first argument'));
-			});
-		} else {
-			if (typeof errback === 'function') {
-				errback(new global.DOMError('addIceCandidate() must be called with a RTCIceCandidate instance as first argument'));
-			}
-			return;
-		}
-	}
-
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debug('addIceCandidate() | success');
-				// Update remoteDescription.
-				if (self.remoteDescription && data.remoteDescription) {
-					self.remoteDescription.type = data.remoteDescription.type;
-					self.remoteDescription.sdp = data.remoteDescription.sdp;
-				} else if (data.remoteDescription) {
-					self.remoteDescription = new RTCSessionDescription(data.remoteDescription);
-				}
-				resolve();
-			}
-
-			function onResultError() {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('addIceCandidate() | failure');
-				reject(new global.DOMError('addIceCandidate() failed'));
-			}
-
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_addIceCandidate', [self.pcId, candidate]);
-		});
-	}
-
-	function onResultOK(data) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		debug('addIceCandidate() | success');
-		// Update remoteDescription.
-		if (self.remoteDescription && data.remoteDescription) {
-			self.remoteDescription.type = data.remoteDescription.type;
-			self.remoteDescription.sdp = data.remoteDescription.sdp;
-		} else if (data.remoteDescription) {
-			self.remoteDescription = new RTCSessionDescription(data.remoteDescription);
-		}
-		callback();
-	}
-
-	function onResultError() {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		debugerror('addIceCandidate() | failure');
-		if (typeof errback === 'function') {
-			errback(new global.DOMError('addIceCandidate() failed'));
-		}
-	}
-
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_addIceCandidate', [this.pcId, candidate]);
-};
-
-
-RTCPeerConnection.prototype.getConfiguration = function () {
-	debug('getConfiguration()');
-
-	return this.pcConfig;
-};
-
-
-RTCPeerConnection.prototype.getLocalStreams = function () {
-	debug('getLocalStreams()');
-
-	var streams = [],
-		id;
-
-	for (id in this.localStreams) {
-		if (this.localStreams.hasOwnProperty(id)) {
-			streams.push(this.localStreams[id]);
-		}
-	}
-
-	return streams;
-};
-
-
-RTCPeerConnection.prototype.getRemoteStreams = function () {
-	debug('getRemoteStreams()');
-
-	var streams = [],
-		id;
-
-	for (id in this.remoteStreams) {
-		if (this.remoteStreams.hasOwnProperty(id)) {
-			streams.push(this.remoteStreams[id]);
-		}
-	}
-
-	return streams;
-};
-
-
-RTCPeerConnection.prototype.getStreamById = function (id) {
-	debug('getStreamById()');
-
-	return this.localStreams[id] || this.remoteStreams[id] || null;
-};
-
-
-RTCPeerConnection.prototype.addStream = function (stream) {
-	if (isClosed.call(this)) {
-		throw new Errors.InvalidStateError('peerconnection is closed');
-	}
-
-	debug('addStream()');
-
-	if (!(stream instanceof MediaStream)) {
-		throw new Error('addStream() must be called with a MediaStream instance as argument');
-	}
-
-	if (this.localStreams[stream.id]) {
-		debugerror('addStream() | given stream already in present in local streams');
-		return;
-	}
-
-	this.localStreams[stream.id] = stream;
-
-	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_addStream', [this.pcId, stream.id]);
-};
-
-
-RTCPeerConnection.prototype.removeStream = function (stream) {
-	if (isClosed.call(this)) {
-		throw new Errors.InvalidStateError('peerconnection is closed');
-	}
-
-	debug('removeStream()');
-
-	if (!(stream instanceof MediaStream)) {
-		throw new Error('removeStream() must be called with a MediaStream instance as argument');
-	}
-
-	if (!this.localStreams[stream.id]) {
-		debugerror('removeStream() | given stream not present in local streams');
-		return;
-	}
-
-	delete this.localStreams[stream.id];
-
-	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_removeStream', [this.pcId, stream.id]);
-};
-
-
-RTCPeerConnection.prototype.createDataChannel = function (label, options) {
-	if (isClosed.call(this)) {
-		throw new Errors.InvalidStateError('peerconnection is closed');
-	}
-
-	debug('createDataChannel() [label:%s, options:%o]', label, options);
-
-	return new RTCDataChannel(this, label, options);
-};
-
-
-RTCPeerConnection.prototype.createDTMFSender = function (track) {
-	if (isClosed.call(this)) {
-		throw new Errors.InvalidStateError('peerconnection is closed');
-	}
-
-	debug('createDTMFSender() [track:%o]', track);
-
-	return new RTCDTMFSender(this, track);
-};
-
-RTCPeerConnection.prototype.getStats = function () {
-	var self = this,
-		isPromise,
-		selector,
-		callback, errback;
-
-	if (typeof arguments[0] !== 'function') {
-		isPromise = true;
-		selector = arguments[0];
-	} else {
-		isPromise = false;
-		callback = arguments[0];
-		selector = arguments[1];
-		errback = arguments[2];
-	}
-
-	if (selector && !(selector instanceof MediaStreamTrack)) {
-		throw new Error('getStats() must be called with null or a valid MediaStreamTrack instance as argument');
-	}
-
-	if (isClosed.call(this)) {
-		throw new Errors.InvalidStateError('peerconnection is closed');
-	}
-
-	debug('getStats() [selector:%o]', selector);
-
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(array) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				var res = [];
-				array.forEach(function (stat) {
-					res.push(new RTCStatsReport(stat));
-				});
-				resolve(new RTCStatsResponse(res));
-			}
-
-			function onResultError(error) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('getStats() | failure: %s', error);
-				reject(new global.DOMError(error));
-			}
-
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_getStats', [self.pcId, selector ? selector.id : null]);
-		});
-	}
-
-	function onResultOK(array) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		var res = [];
-		array.forEach(function (stat) {
-			res.push(new RTCStatsReport(stat));
-		});
-		callback(new RTCStatsResponse(res));
-	}
-
-	function onResultError(error) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		debugerror('getStats() | failure: %s', error);
-		if (typeof errback === 'function') {
-			errback(new global.DOMError(error));
-		}
-	}
-
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_getStats', [this.pcId, selector ? selector.id : null]);
-};
-
-RTCPeerConnection.prototype.close = function () {
-	if (isClosed.call(this)) {
-		return;
-	}
-
-	debug('close()');
-
-	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_close', [this.pcId]);
-};
-
-
-/**
- * Private API.
- */
-
-
-function fixPcConfig(pcConfig) {
-	if (!pcConfig) {
-		return {
-			iceServers: []
-		};
-	}
-
-	var iceServers = pcConfig.iceServers,
-		i, len, iceServer;
-
-	if (!Array.isArray(iceServers)) {
-		pcConfig.iceServers = [];
-		return pcConfig;
-	}
-
-	for (i = 0, len = iceServers.length; i < len; i++) {
-		iceServer = iceServers[i];
-
-		// THe Objective-C wrapper of WebRTC is old and does not implement .urls.
-		if (iceServer.url) {
-			continue;
-		} else if (Array.isArray(iceServer.urls)) {
-			iceServer.url = iceServer.urls[0];
-		} else if (typeof iceServer.urls === 'string') {
-			iceServer.url = iceServer.urls;
-		}
-	}
-
-	return pcConfig;
-}
-
-
-function isClosed() {
-	return this.signalingState === 'closed';
-}
-
-
-function onEvent(data) {
-	var type = data.type,
-		event = new Event(type),
-		stream,
-		dataChannel,
-		id;
-
-	debug('onEvent() | [type:%s, data:%o]', type, data);
-
-	switch (type) {
-		case 'signalingstatechange':
-			this.signalingState = data.signalingState;
-			break;
-
-		case 'icegatheringstatechange':
-			this.iceGatheringState = data.iceGatheringState;
-			break;
-
-		case 'iceconnectionstatechange':
-			this.iceConnectionState = data.iceConnectionState;
-
-			// Emit "connected" on remote streams if ICE connected.
-			if (data.iceConnectionState === 'connected') {
-				for (id in this.remoteStreams) {
-					if (this.remoteStreams.hasOwnProperty(id)) {
-						this.remoteStreams[id].emitConnected();
-					}
-				}
-			}
-			break;
-
-		case 'icecandidate':
-			if (data.candidate) {
-				event.candidate = new RTCIceCandidate(data.candidate);
-			} else {
-				event.candidate = null;
-			}
-			// Update localDescription.
-			if (this.localDescription) {
-				this.localDescription.type = data.localDescription.type;
-				this.localDescription.sdp = data.localDescription.sdp;
-			} else {
-				this.localDescription = new RTCSessionDescription(data);
-			}
-			break;
-
-		case 'negotiationneeded':
-			break;
-
-		case 'addstream':
-			stream = MediaStream.create(data.stream);
-			event.stream = stream;
-
-			// Append to the remote streams.
-			this.remoteStreams[stream.id] = stream;
-
-			// Emit "connected" on the stream if ICE connected.
-			if (this.iceConnectionState === 'connected' || this.iceConnectionState === 'completed') {
-				stream.emitConnected();
-			}
-			break;
-
-		case 'removestream':
-			stream = this.remoteStreams[data.streamId];
-			event.stream = stream;
-
-			// Remove from the remote streams.
-			delete this.remoteStreams[stream.id];
-			break;
-
-		case 'datachannel':
-			dataChannel = new RTCDataChannel(this, null, null, data.channel);
-			event.channel = dataChannel;
-			break;
-	}
-
-	this.dispatchEvent(event);
-}
-
-}).call(this,typeof global !== "undefined" ? global : typeof self !== "undefined" ? self : typeof window !== "undefined" ? window : {})
-},{"./Errors":1,"./MediaStream":3,"./MediaStreamTrack":5,"./RTCDTMFSender":6,"./RTCDataChannel":7,"./RTCIceCandidate":8,"./RTCSessionDescription":10,"./RTCStatsReport":11,"./RTCStatsResponse":12,"cordova/exec":undefined,"debug":18,"random-number":23,"yaeti":24}],10:[function(require,module,exports){
-/**
- * Expose the RTCSessionDescription class.
- */
-module.exports = RTCSessionDescription;
-
-
-function RTCSessionDescription(data) {
-	data = data || {};
-
-	// Public atributes.
-	this.type = data.type;
-	this.sdp = data.sdp;
-}
-
-},{}],11:[function(require,module,exports){
-/**
- * Expose the RTCStatsReport class.
- */
-module.exports = RTCStatsReport;
-
-function RTCStatsReport(data) {
-	data = data || [];
-
-	this.id = data.reportId;
-	this.timestamp = data.timestamp;
-	this.type = data.type;
-
-	this.names = function () {
-		return Object.keys(data.values);
-	};
-
-	this.stat = function (key) {
-		return data.values[key] || '';
-	};
-}
-
-},{}],12:[function(require,module,exports){
-/**
- * Expose the RTCStatsResponse class.
- */
-module.exports = RTCStatsResponse;
-
-function RTCStatsResponse(data) {
-	data = data || [];
-
-	this.result = function () {
-		return data;
-	};
-
-	this.namedItem = function () {
-		return null;
-	};
-}
-
-},{}],13:[function(require,module,exports){
-/**
- * Expose the enumerateDevices function.
- */
-module.exports = enumerateDevices;
-
-
-/**
- * Dependencies.
- */
-var
-	debug = require('debug')('iosrtc:enumerateDevices'),
-	exec = require('cordova/exec'),
-	MediaDeviceInfo = require('./MediaDeviceInfo');
-
-
-function enumerateDevices() {
-	debug('');
-
-	var isPromise,
-		callback;
-
-	if (typeof arguments[0] !== 'function') {
-		isPromise = true;
-	} else {
-		isPromise = false;
-		callback = arguments[0];
-	}
-
-	if (isPromise) {
-		return new Promise(function (resolve) {
-			function onResultOK(data) {
-				debug('enumerateDevices() | success');
-				resolve(getMediaDeviceInfos(data.devices));
-			}
-
-			exec(onResultOK, null, 'iosrtcPlugin', 'enumerateDevices', []);
-		});
-	}
-
-	function onResultOK(data) {
-		debug('enumerateDevices() | success');
-		callback(getMediaDeviceInfos(data.devices));
-	}
-
-	exec(onResultOK, null, 'iosrtcPlugin', 'enumerateDevices', []);
-}
-
-
-/**
- * Private API.
- */
-
-
-function getMediaDeviceInfos(devices) {
-	debug('getMediaDeviceInfos() | [devices:%o]', devices);
-
-	var id,
-		mediaDeviceInfos = [];
-
-	for (id in devices) {
-		if (devices.hasOwnProperty(id)) {
-			mediaDeviceInfos.push(new MediaDeviceInfo(devices[id]));
-		}
-	}
-
-	return mediaDeviceInfos;
-}
-
-},{"./MediaDeviceInfo":2,"cordova/exec":undefined,"debug":18}],14:[function(require,module,exports){
-/**
- * Expose the getUserMedia function.
- */
-module.exports = getUserMedia;
-
-
-/**
- * Dependencies.
- */
-var
-	debug = require('debug')('iosrtc:getUserMedia'),
-	debugerror = require('debug')('iosrtc:ERROR:getUserMedia'),
-	exec = require('cordova/exec'),
-	MediaStream = require('./MediaStream'),
-	Errors = require('./Errors');
-
-debugerror.log = console.warn.bind(console);
-
-
-function isPositiveInteger(number) {
-	return typeof number === 'number' && number >= 0 && number % 1 === 0;
-}
-
-function isPositiveFloat(number) {
-	return typeof number === 'number' && number >= 0;
-}
-
-
-function getUserMedia(constraints) {
-	debug('[original constraints:%o]', constraints);
-
-	var
-		isPromise,
-		callback, errback,
-		audioRequested = false,
-		videoRequested = false,
-		newConstraints = {
-			audio: false,
-			video: false
-		};
-
-	if (typeof arguments[1] !== 'function') {
-		isPromise = true;
-	} else {
-		isPromise = false;
-		callback = arguments[1];
-		errback = arguments[2];
-	}
-
-	if (
-		typeof constraints !== 'object' ||
-		(!constraints.hasOwnProperty('audio') && !constraints.hasOwnProperty('video'))
-	) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.MediaStreamError('constraints must be an object with at least "audio" or "video" keys'));
-			});
-		} else {
-			if (typeof errback === 'function') {
-				errback(new Errors.MediaStreamError('constraints must be an object with at least "audio" or "video" keys'));
-			}
-			return;
-		}
-	}
-
-	if (constraints.audio) {
-		audioRequested = true;
-		newConstraints.audio = true;
-	}
-	if (constraints.video) {
-		videoRequested = true;
-		newConstraints.video = true;
-	}
-
-	// Example:
-	//
-	// getUserMedia({
-	//  audio: true,
-	//  video: {
-	//  	deviceId: 'qwer-asdf-zxcv',
-	//  	width: {
-	//  		min: 400,
-	//  		max: 600
-	//  	},
-	//  	frameRate: {
-	//  		min: 1.0,
-	//  		max: 60.0
-	//  	}
-	//  }
-	// });
-
-	// Get video constraints
-	if (videoRequested) {
-		// Get requested video deviceId.
-		if (typeof constraints.video.deviceId === 'string') {
-			newConstraints.videoDeviceId = constraints.video.deviceId;
-		}
-
-		// Get requested min/max width.
-		if (typeof constraints.video.width === 'object') {
-			if (isPositiveInteger(constraints.video.width.min)) {
-				newConstraints.videoMinWidth = constraints.video.width.min;
-			}
-			if (isPositiveInteger(constraints.video.width.max)) {
-				newConstraints.videoMaxWidth = constraints.video.width.max;
-			}
-		}
-		// Get requested min/max height.
-		if (typeof constraints.video.height === 'object') {
-			if (isPositiveInteger(constraints.video.height.min)) {
-				newConstraints.videoMinHeight = constraints.video.height.min;
-			}
-			if (isPositiveInteger(constraints.video.height.max)) {
-				newConstraints.videoMaxHeight = constraints.video.height.max;
-			}
-		}
-		// Get requested min/max frame rate.
-		if (typeof constraints.video.frameRate === 'object') {
-			if (isPositiveFloat(constraints.video.frameRate.min)) {
-				newConstraints.videoMinFrameRate = constraints.video.frameRate.min;
-			}
-			if (isPositiveFloat(constraints.video.frameRate.max)) {
-				newConstraints.videoMaxFrameRate = constraints.video.frameRate.max;
-			}
-		} else if (isPositiveFloat(constraints.video.frameRate)) {
-			newConstraints.videoMinFrameRate = constraints.video.frameRate;
-			newConstraints.videoMaxFrameRate = constraints.video.frameRate;
-		}
-	}
-
-	debug('[computed constraints:%o]', newConstraints);
-
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				debug('getUserMedia() | success');
-				var stream = MediaStream.create(data.stream);
-				resolve(stream);
-				// Emit "connected" on the stream.
-				stream.emitConnected();
-			}
-
-			function onResultError(error) {
-				debugerror('getUserMedia() | failure: %s', error);
-				reject(new Errors.MediaStreamError('getUserMedia() failed: ' + error));
-			}
-
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'getUserMedia', [newConstraints]);
-		});
-	}
-
-	function onResultOK(data) {
-		debug('getUserMedia() | success');
-
-		var stream = MediaStream.create(data.stream);
-
-		callback(stream);
-
-		// Emit "connected" on the stream.
-		stream.emitConnected();
-	}
-
-	function onResultError(error) {
-		debugerror('getUserMedia() | failure: %s', error);
-
-		if (typeof errback === 'function') {
-			errback(new Errors.MediaStreamError('getUserMedia() failed: ' + error));
-		}
-	}
-
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'getUserMedia', [newConstraints]);
-}
-
-},{"./Errors":1,"./MediaStream":3,"cordova/exec":undefined,"debug":18}],15:[function(require,module,exports){
-(function (global){
-/**
- * Variables.
- */
-
-var
-	// Dictionary of MediaStreamRenderers.
-	// - key: MediaStreamRenderer id.
-	// - value: MediaStreamRenderer.
-	mediaStreamRenderers = {},
-
-	// Dictionary of MediaStreams.
-	// - key: MediaStream blobId.
-	// - value: MediaStream.
-	mediaStreams = {},
-
-
-/**
- * Dependencies.
- */
-	debug                  = require('debug')('iosrtc'),
-	exec                   = require('cordova/exec'),
-	domready               = require('domready'),
-
-	getUserMedia           = require('./getUserMedia'),
-	enumerateDevices       = require('./enumerateDevices'),
-	RTCPeerConnection      = require('./RTCPeerConnection'),
-	RTCSessionDescription  = require('./RTCSessionDescription'),
-	RTCIceCandidate        = require('./RTCIceCandidate'),
-	MediaStream            = require('./MediaStream'),
-	MediaStreamTrack       = require('./MediaStreamTrack'),
-	videoElementsHandler   = require('./videoElementsHandler'),
-	rtcninjaPlugin         = require('./rtcninjaPlugin');
-
-
-/**
- * Expose the iosrtc object.
- */
-module.exports = {
-	// Expose WebRTC classes and functions.
-	getUserMedia:          getUserMedia,
-	enumerateDevices:      enumerateDevices,
-	getMediaDevices:       enumerateDevices,  // TMP
-	RTCPeerConnection:     RTCPeerConnection,
-	RTCSessionDescription: RTCSessionDescription,
-	RTCIceCandidate:       RTCIceCandidate,
-	MediaStream:           MediaStream,
-	MediaStreamTrack:      MediaStreamTrack,
-
-	// Expose a function to refresh current videos rendering a MediaStream.
-	refreshVideos:         refreshVideos,
-
-	// Expose a function to handle a video not yet inserted in the DOM.
-	observeVideo:          videoElementsHandler.observeVideo,
-
-	// Select audio output (earpiece or speaker).
-	selectAudioOutput:     selectAudioOutput,
-
-	// Expose a function to pollute window and naigator namespaces.
-	registerGlobals:       registerGlobals,
-
-	// Expose the rtcninjaPlugin module.
-	rtcninjaPlugin:        rtcninjaPlugin,
-
-	// Expose the debug module.
-	debug:                 require('debug'),
-
-	// Debug function to see what happens internally.
-	dump:                  dump
-};
-
-
-domready(function () {
-	// Let the MediaStream class and the videoElementsHandler share same MediaStreams container.
-	MediaStream.setMediaStreams(mediaStreams);
-	videoElementsHandler(mediaStreams, mediaStreamRenderers);
-});
-
-
-function refreshVideos() {
-	debug('refreshVideos()');
-
-	var id;
-
-	for (id in mediaStreamRenderers) {
-		if (mediaStreamRenderers.hasOwnProperty(id)) {
-			mediaStreamRenderers[id].refresh();
-		}
-	}
-}
-
-
-function selectAudioOutput(output) {
-	debug('selectAudioOutput() | [output:"%s"]', output);
-
-	switch (output) {
-		case 'earpiece':
-			exec(null, null, 'iosrtcPlugin', 'selectAudioOutputEarpiece', []);
-			break;
-		case 'speaker':
-			exec(null, null, 'iosrtcPlugin', 'selectAudioOutputSpeaker', []);
-			break;
-		default:
-			throw new Error('output must be "earpiece" or "speaker"');
-	}
-}
-
-
-function registerGlobals() {
-	if (!global.navigator) {
-		global.navigator = {};
-	}
-
-	if (!navigator.mediaDevices) {
-		navigator.mediaDevices = {};
-	}
-
-	navigator.getUserMedia                  = getUserMedia;
-	navigator.webkitGetUserMedia            = getUserMedia;
-	navigator.mediaDevices.getUserMedia     = getUserMedia;
-	navigator.mediaDevices.enumerateDevices = enumerateDevices;
-	window.RTCPeerConnection                = RTCPeerConnection;
-	window.webkitRTCPeerConnection          = RTCPeerConnection;
-	window.RTCSessionDescription            = RTCSessionDescription;
-	window.RTCIceCandidate                  = RTCIceCandidate;
-	window.MediaStream                      = MediaStream;
-	window.webkitMediaStream                = MediaStream;
-	window.MediaStreamTrack                 = MediaStreamTrack;
-}
-
-
-function dump() {
-	exec(null, null, 'iosrtcPlugin', 'dump', []);
-}
-
-}).call(this,typeof global !== "undefined" ? global : typeof self !== "undefined" ? self : typeof window !== "undefined" ? window : {})
-},{"./MediaStream":3,"./MediaStreamTrack":5,"./RTCIceCandidate":8,"./RTCPeerConnection":9,"./RTCSessionDescription":10,"./enumerateDevices":13,"./getUserMedia":14,"./rtcninjaPlugin":16,"./videoElementsHandler":17,"cordova/exec":undefined,"debug":18,"domready":20}],16:[function(require,module,exports){
-/**
- * Expose the rtcninjaPlugin object.
- */
-module.exports = {
-	isRequired: function () {
-		return true;
-	},
-
-	isInstalled: function () {
-		return true;
-	},
-
-	interface: {
-		getUserMedia:          require('./getUserMedia'),
-		enumerateDevices:      require('./enumerateDevices'),
-		getMediaDevices:       require('./enumerateDevices'),  // TMP
-		RTCPeerConnection:     require('./RTCPeerConnection'),
-		RTCSessionDescription: require('./RTCSessionDescription'),
-		RTCIceCandidate:       require('./RTCIceCandidate'),
-		MediaStreamTrack:      require('./MediaStreamTrack'),
-		attachMediaStream:     attachMediaStream,
-		canRenegotiate:        true
-	}
-};
-
-
-function attachMediaStream(element, stream) {
-	element.src = URL.createObjectURL(stream);
-	return element;
-}
-
-},{"./MediaStreamTrack":5,"./RTCIceCandidate":8,"./RTCPeerConnection":9,"./RTCSessionDescription":10,"./enumerateDevices":13,"./getUserMedia":14}],17:[function(require,module,exports){
-(function (global){
-/**
- * Expose a function that must be called when the library is loaded.
- * And also a helper function.
- */
-module.exports = videoElementsHandler;
-module.exports.observeVideo = observeVideo;
-
-
-/**
- * Dependencies.
- */
-var debug = require('debug')('iosrtc:videoElementsHandler'),
-	MediaStreamRenderer = require('./MediaStreamRenderer'),
-
-
-/**
- * Local variables.
- */
-
-	// RegExp for MediaStream blobId.
-	MEDIASTREAM_ID_REGEXP = new RegExp(/^MediaStream_/),
-
-	// RegExp for Blob URI.
-	BLOB_URI_REGEX = new RegExp(/^blob:/),
-
-	// Dictionary of MediaStreamRenderers (provided via module argument).
-	// - key: MediaStreamRenderer id.
-	// - value: MediaStreamRenderer.
-	mediaStreamRenderers,
-
-	// Dictionary of MediaStreams (provided via module argument).
-	// - key: MediaStream blobId.
-	// - value: MediaStream.
-	mediaStreams,
-
-	// Video element mutation observer.
-	videoObserver = new MutationObserver(function (mutations) {
-		var i, numMutations, mutation,
-			video;
-
-		for (i = 0, numMutations = mutations.length; i < numMutations; i++) {
-			mutation = mutations[i];
-
-			// HTML video element.
-			video = mutation.target;
-
-			// .src removed.
-			if (!video.src) {
-				// If this video element was previously handling a MediaStreamRenderer, release it.
-				releaseMediaStreamRenderer(video);
-				continue;
-			}
-
-			handleVideo(video);
-		}
-	}),
-
-	// DOM mutation observer.
-	domObserver = new MutationObserver(function (mutations) {
-		var i, numMutations, mutation,
-			j, numNodes, node;
-
-		for (i = 0, numMutations = mutations.length; i < numMutations; i++) {
-			mutation = mutations[i];
-
-			// Check if there has been addition or deletion of nodes.
-			if (mutation.type !== 'childList') {
-				continue;
-			}
-
-			// Check added nodes.
-			for (j = 0, numNodes = mutation.addedNodes.length; j < numNodes; j++) {
-				node = mutation.addedNodes[j];
-
-				checkNewNode(node);
-			}
-
-			// Check removed nodes.
-			for (j = 0, numNodes = mutation.removedNodes.length; j < numNodes; j++) {
-				node = mutation.removedNodes[j];
-
-				checkRemovedNode(node);
-			}
-		}
-
-		function checkNewNode(node) {
-			var j, childNode;
-
-			if (node.nodeName === 'VIDEO') {
-				debug('new video element added');
-
-				// Avoid same node firing more than once (really, may happen in some cases).
-				if (node._iosrtcVideoHandled) {
-					return;
-				}
-				node._iosrtcVideoHandled = true;
-
-				// Observe changes in the video element.
-				observeVideo(node);
-			} else {
-				for (j = 0; j < node.childNodes.length; j++) {
-					childNode = node.childNodes.item(j);
-
-					checkNewNode(childNode);
-				}
-			}
-		}
-
-		function checkRemovedNode(node) {
-			var j, childNode;
-
-			if (node.nodeName === 'VIDEO') {
-				debug('video element removed');
-
-				// If this video element was previously handling a MediaStreamRenderer, release it.
-				releaseMediaStreamRenderer(node);
-				delete node._iosrtcVideoHandled;
-			} else {
-				for (j = 0; j < node.childNodes.length; j++) {
-					childNode = node.childNodes.item(j);
-
-					checkRemovedNode(childNode);
-				}
-			}
-		}
-	});
-
-
-function videoElementsHandler(_mediaStreams, _mediaStreamRenderers) {
-	var existingVideos = document.querySelectorAll('video'),
-		i, len,
-		video;
-
-	mediaStreams = _mediaStreams;
-	mediaStreamRenderers = _mediaStreamRenderers;
-
-	// Search the whole document for already existing HTML video elements and observe them.
-	for (i = 0, len = existingVideos.length; i < len; i++) {
-		video = existingVideos.item(i);
-
-		debug('video element found');
-
-		observeVideo(video);
-	}
-
-	// Observe the whole document for additions of new HTML video elements and observe them.
-	domObserver.observe(document, {
-		// Set to true if additions and removals of the target node's child elements (including text nodes) are to
-		// be observed.
-		childList: true,
-		// Set to true if mutations to target's attributes are to be observed.
-		attributes: false,
-		// Set to true if mutations to target's data are to be observed.
-		characterData: false,
-		// Set to true if mutations to not just target, but also target's descendants are to be observed.
-		subtree: true,
-		// Set to true if attributes is set to true and target's attribute value before the mutation needs to be
-		// recorded.
-		attributeOldValue: false,
-		// Set to true if characterData is set to true and target's data before the mutation needs to be recorded.
-		characterDataOldValue: false
-		// Set to an array of attribute local names (without namespace) if not all attribute mutations need to be
-		// observed.
-		// attributeFilter:
-	});
-}
-
-
-function observeVideo(video) {
-	debug('observeVideo()');
-
-	// If the video already has a src property but is not yet handled by the plugin
-	// then handle it now.
-	if (video.src && !video._iosrtcMediaStreamRendererId) {
-		handleVideo(video);
-	}
-
-	// Add .src observer to the video element.
-	videoObserver.observe(video, {
-		// Set to true if additions and removals of the target node's child elements (including text
-		// nodes) are to be observed.
-		childList: false,
-		// Set to true if mutations to target's attributes are to be observed.
-		attributes: true,
-		// Set to true if mutations to target's data are to be observed.
-		characterData: false,
-		// Set to true if mutations to not just target, but also target's descendants are to be observed.
-		subtree: false,
-		// Set to true if attributes is set to true and target's attribute value before the mutation
-		// needs to be recorded.
-		attributeOldValue: false,
-		// Set to true if characterData is set to true and target's data before the mutation needs to be
-		// recorded.
-		characterDataOldValue: false,
-		// Set to an array of attribute local names (without namespace) if not all attribute mutations
-		// need to be observed.
-		// TODO: Add srcObject, mozSrcObject
-		attributeFilter: ['src']
-	});
-
-	// Intercept video 'error' events if it's due to the attached MediaStream.
-	video.addEventListener('error', function (event) {
-		if (video.error.code === global.MediaError.MEDIA_ERR_SRC_NOT_SUPPORTED && BLOB_URI_REGEX.test(video.src)) {
-			debug('stopping "error" event propagation for video element');
-
-			event.stopImmediatePropagation();
-		}
-	});
-}
-
-
-/**
- * Private API.
- */
-
-function handleVideo(video) {
-	var xhr = new XMLHttpRequest();
-
-	xhr.open('GET', video.src, true);
-	xhr.responseType = 'blob';
-	xhr.onload = function () {
-		if (xhr.status !== 200) {
-			// If this video element was previously handling a MediaStreamRenderer, release it.
-			releaseMediaStreamRenderer(video);
-
-			return;
-		}
-
-		var reader = new FileReader();
-
-		// Some versions of Safari fail to set onloadend property, some others do not react
-		// on 'loadend' event. Try everything here.
-		try {
-			reader.onloadend = onloadend;
-		} catch (error) {
-			reader.addEventListener('loadend', onloadend);
-		}
-		reader.readAsText(xhr.response);
-
-		function onloadend() {
-			var mediaStreamBlobId = reader.result;
-
-			// The retrieved URL does not point to a MediaStream.
-			if (!mediaStreamBlobId || typeof mediaStreamBlobId !== 'string' || !MEDIASTREAM_ID_REGEXP.test(mediaStreamBlobId)) {
-				// If this video element was previously handling a MediaStreamRenderer, release it.
-				releaseMediaStreamRenderer(video);
-
-				return;
-			}
-
-			provideMediaStreamRenderer(video, mediaStreamBlobId);
-		}
-	};
-	xhr.send();
-}
-
-
-function provideMediaStreamRenderer(video, mediaStreamBlobId) {
-	var mediaStream = mediaStreams[mediaStreamBlobId],
-		mediaStreamRenderer = mediaStreamRenderers[video._iosrtcMediaStreamRendererId];
-
-	if (!mediaStream) {
-		releaseMediaStreamRenderer(video);
-
-		return;
-	}
-
-	if (mediaStreamRenderer) {
-		mediaStreamRenderer.render(mediaStream);
-	} else {
-		mediaStreamRenderer = new MediaStreamRenderer(video);
-		mediaStreamRenderer.render(mediaStream);
-
-		mediaStreamRenderers[mediaStreamRenderer.id] = mediaStreamRenderer;
-		video._iosrtcMediaStreamRendererId = mediaStreamRenderer.id;
-	}
-
-	// Close the MediaStreamRenderer of this video if it emits "close" event.
-	mediaStreamRenderer.addEventListener('close', function () {
-		if (mediaStreamRenderers[video._iosrtcMediaStreamRendererId] !== mediaStreamRenderer) {
-			return;
-		}
-
-		releaseMediaStreamRenderer(video);
-	});
-
-	// Override some <video> properties.
-	// NOTE: This is a terrible hack but it works.
-	Object.defineProperties(video, {
-		videoWidth: {
-			configurable: true,
-			get: function () {
-				return mediaStreamRenderer.videoWidth || 0;
-			}
-		},
-		videoHeight: {
-			configurable: true,
-			get: function () {
-				return mediaStreamRenderer.videoHeight || 0;
-			}
-		},
-		readyState: {
-			configurable: true,
-			get: function () {
-				if (mediaStreamRenderer && mediaStreamRenderer.stream && mediaStreamRenderer.stream.connected) {
-					return video.HAVE_ENOUGH_DATA;
-				} else {
-					return video.HAVE_NOTHING;
-				}
-			}
-		}
-	});
-}
-
-
-function releaseMediaStreamRenderer(video) {
-	if (!video._iosrtcMediaStreamRendererId) {
-		return;
-	}
-
-	var mediaStreamRenderer = mediaStreamRenderers[video._iosrtcMediaStreamRendererId];
-
-	if (mediaStreamRenderer) {
-		delete mediaStreamRenderers[video._iosrtcMediaStreamRendererId];
-		mediaStreamRenderer.close();
-	}
-
-	delete video._iosrtcMediaStreamRendererId;
-
-	// Remove overrided <video> properties.
-	delete video.videoWidth;
-	delete video.videoHeight;
-	delete video.readyState;
-}
-
-}).call(this,typeof global !== "undefined" ? global : typeof self !== "undefined" ? self : typeof window !== "undefined" ? window : {})
-},{"./MediaStreamRenderer":4,"debug":18}],18:[function(require,module,exports){
-(function (process){
-/**
- * This is the web browser implementation of `debug()`.
- *
- * Expose `debug()` as the module.
- */
-
-exports = module.exports = require('./debug');
-exports.log = log;
-exports.formatArgs = formatArgs;
-exports.save = save;
-exports.load = load;
-exports.useColors = useColors;
-exports.storage = 'undefined' != typeof chrome
-               && 'undefined' != typeof chrome.storage
-                  ? chrome.storage.local
-                  : localstorage();
-
-/**
- * Colors.
- */
-
-exports.colors = [
-  'lightseagreen',
-  'forestgreen',
-  'goldenrod',
-  'dodgerblue',
-  'darkorchid',
-  'crimson'
-];
-
-/**
- * Currently only WebKit-based Web Inspectors, Firefox >= v31,
- * and the Firebug extension (any Firefox version) are known
- * to support "%c" CSS customizations.
- *
- * TODO: add a `localStorage` variable to explicitly enable/disable colors
- */
-
-function useColors() {
-  // NB: In an Electron preload script, document will be defined but not fully
-  // initialized. Since we know we're in Chrome, we'll just detect this case
-  // explicitly
-  if (typeof window !== 'undefined' && 'process' in window && window.process.type === 'renderer') {
-    return true;
-  }
-
-  // is webkit? http://stackoverflow.com/a/16459606/376773
-  // document is undefined in react-native: https://github.com/facebook/react-native/pull/1632
-  return (typeof document !== 'undefined' && 'WebkitAppearance' in document.documentElement.style) ||
-    // is firebug? http://stackoverflow.com/a/398120/376773
-    (typeof window !== 'undefined' && window.console && (console.firebug || (console.exception && console.table))) ||
-    // is firefox >= v31?
-    // https://developer.mozilla.org/en-US/docs/Tools/Web_Console#Styling_messages
-    (navigator.userAgent && navigator.userAgent.toLowerCase().match(/firefox\/(\d+)/) && parseInt(RegExp.$1, 10) >= 31) ||
-    // double check webkit in userAgent just in case we are in a worker
-    (navigator.userAgent && navigator.userAgent.toLowerCase().match(/applewebkit\/(\d+)/));
-}
-
-/**
- * Map %j to `JSON.stringify()`, since no Web Inspectors do that by default.
- */
-
-exports.formatters.j = function(v) {
-  try {
-    return JSON.stringify(v);
-  } catch (err) {
-    return '[UnexpectedJSONParseError]: ' + err.message;
-  }
-};
-
-
-/**
- * Colorize log arguments if enabled.
- *
- * @api public
- */
-
-function formatArgs(args) {
-  var useColors = this.useColors;
-
-  args[0] = (useColors ? '%c' : '')
-    + this.namespace
-    + (useColors ? ' %c' : ' ')
-    + args[0]
-    + (useColors ? '%c ' : ' ')
-    + '+' + exports.humanize(this.diff);
-
-  if (!useColors) return;
-
-  var c = 'color: ' + this.color;
-  args.splice(1, 0, c, 'color: inherit')
-
-  // the final "%c" is somewhat tricky, because there could be other
-  // arguments passed either before or after the %c, so we need to
-  // figure out the correct index to insert the CSS into
-  var index = 0;
-  var lastC = 0;
-  args[0].replace(/%[a-z%]/g, function(match) {
-    if ('%%' === match) return;
-    index++;
-    if ('%c' === match) {
-      // we only are interested in the *last* %c
-      // (the user may have provided their own)
-      lastC = index;
-    }
-  });
-
-  args.splice(lastC, 0, c);
-}
-
-/**
- * Invokes `console.log()` when available.
- * No-op when `console.log` is not a "function".
- *
- * @api public
- */
-
-function log() {
-  // this hackery is required for IE8/9, where
-  // the `console.log` function doesn't have 'apply'
-  return 'object' === typeof console
-    && console.log
-    && Function.prototype.apply.call(console.log, console, arguments);
-}
-
-/**
- * Save `namespaces`.
- *
- * @param {String} namespaces
- * @api private
- */
-
-function save(namespaces) {
-  try {
-    if (null == namespaces) {
-      exports.storage.removeItem('debug');
-    } else {
-      exports.storage.debug = namespaces;
-    }
-  } catch(e) {}
-}
-
-/**
- * Load `namespaces`.
- *
- * @return {String} returns the previously persisted debug modes
- * @api private
- */
-
-function load() {
-  var r;
-  try {
-    return exports.storage.debug;
-  } catch(e) {}
-
-  // If debug isn't set in LS, and we're in Electron, try to load $DEBUG
-  if (typeof process !== 'undefined' && 'env' in process) {
-    return process.env.DEBUG;
-  }
-}
-
-/**
- * Enable namespaces listed in `localStorage.debug` initially.
- */
-
-exports.enable(load());
-
-/**
- * Localstorage attempts to return the localstorage.
- *
- * This is necessary because safari throws
- * when a user disables cookies/localstorage
- * and you attempt to access it.
- *
- * @return {LocalStorage}
- * @api private
- */
-
-function localstorage() {
-  try {
-    return window.localStorage;
-  } catch (e) {}
-}
-
-}).call(this,require('_process'))
-},{"./debug":19,"_process":22}],19:[function(require,module,exports){
-
-/**
- * This is the common logic for both the Node.js and web browser
- * implementations of `debug()`.
- *
- * Expose `debug()` as the module.
- */
-
-exports = module.exports = createDebug.debug = createDebug.default = createDebug;
-exports.coerce = coerce;
-exports.disable = disable;
-exports.enable = enable;
-exports.enabled = enabled;
-exports.humanize = require('ms');
-
-/**
- * The currently active debug mode names, and names to skip.
- */
-
-exports.names = [];
-exports.skips = [];
-
-/**
- * Map of special "%n" handling functions, for the debug "format" argument.
- *
- * Valid key names are a single, lower or upper-case letter, i.e. "n" and "N".
- */
-
-exports.formatters = {};
-
-/**
- * Previous log timestamp.
- */
-
-var prevTime;
-
-/**
- * Select a color.
- * @param {String} namespace
- * @return {Number}
- * @api private
- */
-
-function selectColor(namespace) {
-  var hash = 0, i;
-
-  for (i in namespace) {
-    hash  = ((hash << 5) - hash) + namespace.charCodeAt(i);
-    hash |= 0; // Convert to 32bit integer
-  }
-
-  return exports.colors[Math.abs(hash) % exports.colors.length];
-}
-
-/**
- * Create a debugger with the given `namespace`.
- *
- * @param {String} namespace
- * @return {Function}
- * @api public
- */
-
-function createDebug(namespace) {
-
-  function debug() {
-    // disabled?
-    if (!debug.enabled) return;
-
-    var self = debug;
-
-    // set `diff` timestamp
-    var curr = +new Date();
-    var ms = curr - (prevTime || curr);
-    self.diff = ms;
-    self.prev = prevTime;
-    self.curr = curr;
-    prevTime = curr;
-
-    // turn the `arguments` into a proper Array
-    var args = new Array(arguments.length);
-    for (var i = 0; i < args.length; i++) {
-      args[i] = arguments[i];
-    }
-
-    args[0] = exports.coerce(args[0]);
-
-    if ('string' !== typeof args[0]) {
-      // anything else let's inspect with %O
-      args.unshift('%O');
-    }
-
-    // apply any `formatters` transformations
-    var index = 0;
-    args[0] = args[0].replace(/%([a-zA-Z%])/g, function(match, format) {
-      // if we encounter an escaped % then don't increase the array index
-      if (match === '%%') return match;
-      index++;
-      var formatter = exports.formatters[format];
-      if ('function' === typeof formatter) {
-        var val = args[index];
-        match = formatter.call(self, val);
-
-        // now we need to remove `args[index]` since it's inlined in the `format`
-        args.splice(index, 1);
-        index--;
-      }
-      return match;
-    });
-
-    // apply env-specific formatting (colors, etc.)
-    exports.formatArgs.call(self, args);
-
-    var logFn = enabled.log || exports.log || console.log.bind(console);
-    logFn.apply(self, args);
-  }
-
-  debug.namespace = namespace;
-  debug.enabled = exports.enabled(namespace);
-  debug.useColors = exports.useColors();
-  debug.color = selectColor(namespace);
-
-  // env-specific initialization logic for debug instances
-  if ('function' === typeof exports.init) {
-    exports.init(debug);
-  }
-
-  return debug;
-}
-
-/**
- * Enables a debug mode by namespaces. This can include modes
- * separated by a colon and wildcards.
- *
- * @param {String} namespaces
- * @api public
- */
-
-function enable(namespaces) {
-  exports.save(namespaces);
-
-  var split = (namespaces || '').split(/[\s,]+/);
-  var len = split.length;
-
-  for (var i = 0; i < len; i++) {
-    if (!split[i]) continue; // ignore empty strings
-    namespaces = split[i].replace(/\*/g, '.*?');
-    if (namespaces[0] === '-') {
-      exports.skips.push(new RegExp('^' + namespaces.substr(1) + '$'));
-    } else {
-      exports.names.push(new RegExp('^' + namespaces + '$'));
-    }
-  }
-}
-
-/**
- * Disable debug output.
- *
- * @api public
- */
-
-function disable() {
-  exports.enable('');
-}
-
-/**
- * Returns true if the given mode name is enabled, false otherwise.
- *
- * @param {String} name
- * @return {Boolean}
- * @api public
- */
-
-function enabled(name) {
-  var i, len;
-  for (i = 0, len = exports.skips.length; i < len; i++) {
-    if (exports.skips[i].test(name)) {
-      return false;
-    }
-  }
-  for (i = 0, len = exports.names.length; i < len; i++) {
-    if (exports.names[i].test(name)) {
-      return true;
-    }
-  }
-  return false;
-}
-
-/**
- * Coerce `val`.
- *
- * @param {Mixed} val
- * @return {Mixed}
- * @api private
- */
-
-function coerce(val) {
-  if (val instanceof Error) return val.stack || val.message;
-  return val;
-}
-
-},{"ms":21}],20:[function(require,module,exports){
-/*!
-  * domready (c) Dustin Diaz 2014 - License MIT
-  */
-!function (name, definition) {
-
-  if (typeof module != 'undefined') module.exports = definition()
-  else if (typeof define == 'function' && typeof define.amd == 'object') define(definition)
-  else this[name] = definition()
-
-}('domready', function () {
-
-  var fns = [], listener
-    , doc = document
-    , hack = doc.documentElement.doScroll
-    , domContentLoaded = 'DOMContentLoaded'
-    , loaded = (hack ? /^loaded|^c/ : /^loaded|^i|^c/).test(doc.readyState)
-
-
-  if (!loaded)
-  doc.addEventListener(domContentLoaded, listener = function () {
-    doc.removeEventListener(domContentLoaded, listener)
-    loaded = 1
-    while (listener = fns.shift()) listener()
-  })
-
-  return function (fn) {
-    loaded ? setTimeout(fn, 0) : fns.push(fn)
-  }
-
-});
-
-},{}],21:[function(require,module,exports){
-/**
- * Helpers.
- */
-
-var s = 1000
-var m = s * 60
-var h = m * 60
-var d = h * 24
-var y = d * 365.25
-
-/**
- * Parse or format the given `val`.
- *
- * Options:
- *
- *  - `long` verbose formatting [false]
- *
- * @param {String|Number} val
- * @param {Object} options
- * @throws {Error} throw an error if val is not a non-empty string or a number
- * @return {String|Number}
- * @api public
- */
-
-module.exports = function (val, options) {
-  options = options || {}
-  var type = typeof val
-  if (type === 'string' && val.length > 0) {
-    return parse(val)
-  } else if (type === 'number' && isNaN(val) === false) {
-    return options.long ?
-			fmtLong(val) :
-			fmtShort(val)
-  }
-  throw new Error('val is not a non-empty string or a valid number. val=' + JSON.stringify(val))
-}
-
-/**
- * Parse the given `str` and return milliseconds.
- *
- * @param {String} str
- * @return {Number}
- * @api private
- */
-
-function parse(str) {
-  str = String(str)
-  if (str.length > 10000) {
-    return
-  }
-  var match = /^((?:\d+)?\.?\d+) *(milliseconds?|msecs?|ms|seconds?|secs?|s|minutes?|mins?|m|hours?|hrs?|h|days?|d|years?|yrs?|y)?$/i.exec(str)
-  if (!match) {
-    return
-  }
-  var n = parseFloat(match[1])
-  var type = (match[2] || 'ms').toLowerCase()
-  switch (type) {
-    case 'years':
-    case 'year':
-    case 'yrs':
-    case 'yr':
-    case 'y':
-      return n * y
-    case 'days':
-    case 'day':
-    case 'd':
-      return n * d
-    case 'hours':
-    case 'hour':
-    case 'hrs':
-    case 'hr':
-    case 'h':
-      return n * h
-    case 'minutes':
-    case 'minute':
-    case 'mins':
-    case 'min':
-    case 'm':
-      return n * m
-    case 'seconds':
-    case 'second':
-    case 'secs':
-    case 'sec':
-    case 's':
-      return n * s
-    case 'milliseconds':
-    case 'millisecond':
-    case 'msecs':
-    case 'msec':
-    case 'ms':
-      return n
-    default:
-      return undefined
-  }
-}
-
-/**
- * Short format for `ms`.
- *
- * @param {Number} ms
- * @return {String}
- * @api private
- */
-
-function fmtShort(ms) {
-  if (ms >= d) {
-    return Math.round(ms / d) + 'd'
-  }
-  if (ms >= h) {
-    return Math.round(ms / h) + 'h'
-  }
-  if (ms >= m) {
-    return Math.round(ms / m) + 'm'
-  }
-  if (ms >= s) {
-    return Math.round(ms / s) + 's'
-  }
-  return ms + 'ms'
-}
-
-/**
- * Long format for `ms`.
- *
- * @param {Number} ms
- * @return {String}
- * @api private
- */
-
-function fmtLong(ms) {
-  return plural(ms, d, 'day') ||
-    plural(ms, h, 'hour') ||
-    plural(ms, m, 'minute') ||
-    plural(ms, s, 'second') ||
-    ms + ' ms'
-}
-
-/**
- * Pluralization helper.
- */
-
-function plural(ms, n, name) {
-  if (ms < n) {
-    return
-  }
-  if (ms < n * 1.5) {
-    return Math.floor(ms / n) + ' ' + name
-  }
-  return Math.ceil(ms / n) + ' ' + name + 's'
-}
-
-},{}],22:[function(require,module,exports){
-// shim for using process in browser
-var process = module.exports = {};
-
-// cached from whatever global is present so that test runners that stub it
-// don't break things.  But we need to wrap it in a try catch in case it is
-// wrapped in strict mode code which doesn't define any globals.  It's inside a
-// function because try/catches deoptimize in certain engines.
-
-var cachedSetTimeout;
-var cachedClearTimeout;
-
-function defaultSetTimout() {
-    throw new Error('setTimeout has not been defined');
-}
-function defaultClearTimeout () {
-    throw new Error('clearTimeout has not been defined');
-}
-(function () {
-    try {
-        if (typeof setTimeout === 'function') {
-            cachedSetTimeout = setTimeout;
-        } else {
-            cachedSetTimeout = defaultSetTimout;
-        }
-    } catch (e) {
-        cachedSetTimeout = defaultSetTimout;
-    }
-    try {
-        if (typeof clearTimeout === 'function') {
-            cachedClearTimeout = clearTimeout;
-        } else {
-            cachedClearTimeout = defaultClearTimeout;
-        }
-    } catch (e) {
-        cachedClearTimeout = defaultClearTimeout;
-    }
-} ())
-function runTimeout(fun) {
-    if (cachedSetTimeout === setTimeout) {
-        //normal enviroments in sane situations
-        return setTimeout(fun, 0);
-    }
-    // if setTimeout wasn't available but was latter defined
-    if ((cachedSetTimeout === defaultSetTimout || !cachedSetTimeout) && setTimeout) {
-        cachedSetTimeout = setTimeout;
-        return setTimeout(fun, 0);
-    }
-    try {
-        // when when somebody has screwed with setTimeout but no I.E. maddness
-        return cachedSetTimeout(fun, 0);
-    } catch(e){
-        try {
-            // When we are in I.E. but the script has been evaled so I.E. doesn't trust the global object when called normally
-            return cachedSetTimeout.call(null, fun, 0);
-        } catch(e){
-            // same as above but when it's a version of I.E. that must have the global object for 'this', hopfully our context correct otherwise it will throw a global error
-            return cachedSetTimeout.call(this, fun, 0);
-        }
-    }
-
-
-}
-function runClearTimeout(marker) {
-    if (cachedClearTimeout === clearTimeout) {
-        //normal enviroments in sane situations
-        return clearTimeout(marker);
-    }
-    // if clearTimeout wasn't available but was latter defined
-    if ((cachedClearTimeout === defaultClearTimeout || !cachedClearTimeout) && clearTimeout) {
-        cachedClearTimeout = clearTimeout;
-        return clearTimeout(marker);
-    }
-    try {
-        // when when somebody has screwed with setTimeout but no I.E. maddness
-        return cachedClearTimeout(marker);
-    } catch (e){
-        try {
-            // When we are in I.E. but the script has been evaled so I.E. doesn't  trust the global object when called normally
-            return cachedClearTimeout.call(null, marker);
-        } catch (e){
-            // same as above but when it's a version of I.E. that must have the global object for 'this', hopfully our context correct otherwise it will throw a global error.
-            // Some versions of I.E. have different rules for clearTimeout vs setTimeout
-            return cachedClearTimeout.call(this, marker);
-        }
-    }
-
-
-
-}
-var queue = [];
-var draining = false;
-var currentQueue;
-var queueIndex = -1;
-
-function cleanUpNextTick() {
-    if (!draining || !currentQueue) {
-        return;
-    }
-    draining = false;
-    if (currentQueue.length) {
-        queue = currentQueue.concat(queue);
-    } else {
-        queueIndex = -1;
-    }
-    if (queue.length) {
-        drainQueue();
-    }
-}
-
-function drainQueue() {
-    if (draining) {
-        return;
-    }
-    var timeout = runTimeout(cleanUpNextTick);
-    draining = true;
-
-    var len = queue.length;
-    while(len) {
-        currentQueue = queue;
-        queue = [];
-        while (++queueIndex < len) {
-            if (currentQueue) {
-                currentQueue[queueIndex].run();
-            }
-        }
-        queueIndex = -1;
-        len = queue.length;
-    }
-    currentQueue = null;
-    draining = false;
-    runClearTimeout(timeout);
-}
-
-process.nextTick = function (fun) {
-    var args = new Array(arguments.length - 1);
-    if (arguments.length > 1) {
-        for (var i = 1; i < arguments.length; i++) {
-            args[i - 1] = arguments[i];
-        }
-    }
-    queue.push(new Item(fun, args));
-    if (queue.length === 1 && !draining) {
-        runTimeout(drainQueue);
-    }
-};
-
-// v8 likes predictible objects
-function Item(fun, array) {
-    this.fun = fun;
-    this.array = array;
-}
-Item.prototype.run = function () {
-    this.fun.apply(null, this.array);
-};
-process.title = 'browser';
-process.browser = true;
-process.env = {};
-process.argv = [];
-process.version = ''; // empty string to avoid regexp issues
-process.versions = {};
-
-function noop() {}
-
-process.on = noop;
-process.addListener = noop;
-process.once = noop;
-process.off = noop;
-process.removeListener = noop;
-process.removeAllListeners = noop;
-process.emit = noop;
-
-process.binding = function (name) {
-    throw new Error('process.binding is not supported');
-};
-
-process.cwd = function () { return '/' };
-process.chdir = function (dir) {
-    throw new Error('process.chdir is not supported');
-};
-process.umask = function() { return 0; };
-
-},{}],23:[function(require,module,exports){
-void function(root){
-
-  function defaults(options){
-    var options = options || {}
-    var min = options.min
-    var max = options.max
-    var integer = options.integer || false
-    if ( min == null && max == null ) {
-      min = 0
-      max = 1
-    } else if ( min == null ) {
-      min = max - 1
-    } else if ( max == null ) {
-      max = min + 1
-    }
-    if ( max < min ) throw new Error('invalid options, max must be >= min')
-    return {
-      min:     min
-    , max:     max
-    , integer: integer
-    }
-  }
-
-  function random(options){
-    options = defaults(options)
-    if ( options.max === options.min ) return options.min
-    var r = Math.random() * (options.max - options.min + Number(!!options.integer)) + options.min
-    return options.integer ? Math.floor(r) : r
-  }
-
-  function generator(options){
-    options = defaults(options)
-    return function(min, max, integer){
-      options.min     = min != null ? min : options.min
-      options.max     = max != null ? max : options.max
-      options.integer = integer != null ? integer : options.integer
-      return random(options)
-    }
-  }
-
-  module.exports =  random
-  module.exports.generator = generator
-  module.exports.defaults = defaults
-}(this)
-
-},{}],24:[function(require,module,exports){
-module.exports = {
-	EventTarget : require('./lib/EventTarget'),
-	Event       : require('./lib/Event')
-};
-
-},{"./lib/Event":25,"./lib/EventTarget":26}],25:[function(require,module,exports){
-(function (global){
-/**
- * In browsers export the native Event interface.
- */
-
-module.exports = global.Event;
-
-}).call(this,typeof global !== "undefined" ? global : typeof self !== "undefined" ? self : typeof window !== "undefined" ? window : {})
-},{}],26:[function(require,module,exports){
-/**
- * Expose the _EventTarget class.
- */
-module.exports = _EventTarget;
-
-function _EventTarget() {
-	// Do nothing if called for a native EventTarget object..
-	if (typeof this.addEventListener === 'function') {
-		return;
-	}
-
-	this._listeners = {};
-
-	this.addEventListener = _addEventListener;
-	this.removeEventListener = _removeEventListener;
-	this.dispatchEvent = _dispatchEvent;
-}
-
-Object.defineProperties(_EventTarget.prototype, {
-	listeners: {
-		get: function () {
-			return this._listeners;
-		}
-	}
-});
-
-function _addEventListener(type, newListener) {
-	var
-		listenersType,
-		i, listener;
-
-	if (!type || !newListener) {
-		return;
-	}
-
-	listenersType = this._listeners[type];
-	if (listenersType === undefined) {
-		this._listeners[type] = listenersType = [];
-	}
-
-	for (i = 0; !!(listener = listenersType[i]); i++) {
-		if (listener === newListener) {
-			return;
-		}
-	}
-
-	listenersType.push(newListener);
-}
-
-function _removeEventListener(type, oldListener) {
-	var
-		listenersType,
-		i, listener;
-
-	if (!type || !oldListener) {
-		return;
-	}
-
-	listenersType = this._listeners[type];
-	if (listenersType === undefined) {
-		return;
-	}
-
-	for (i = 0; !!(listener = listenersType[i]); i++) {
-		if (listener === oldListener) {
-			listenersType.splice(i, 1);
-			break;
-		}
-	}
-
-	if (listenersType.length === 0) {
-		delete this._listeners[type];
-	}
-}
-
-function _dispatchEvent(event) {
-	var
-		type,
-		listenersType,
-		dummyListener,
-		stopImmediatePropagation = false,
-		i, listener;
-
-	if (!event || typeof event.type !== 'string') {
-		throw new Error('`event` must have a valid `type` property');
-	}
-
-	// Do some stuff to emulate DOM Event behavior (just if this is not a
-	// DOM Event object)
-	if (event._yaeti) {
-		event.target = this;
-		event.cancelable = true;
-	}
-
-	// Attempt to override the stopImmediatePropagation() method
-	try {
-		event.stopImmediatePropagation = function () {
-			stopImmediatePropagation = true;
-		};
-	} catch (error) {}
-
-	type = event.type;
-	listenersType = (this._listeners[type] || []);
-
-	dummyListener = this['on' + type];
-	if (typeof dummyListener === 'function') {
-		dummyListener.call(this, event);
-	}
-
-	for (i = 0; !!(listener = listenersType[i]); i++) {
-		if (stopImmediatePropagation) {
-			break;
-		}
-
-		listener.call(this, event);
-	}
-
-	return !event.defaultPrevented;
-}
-
-},{}]},{},[15])(15)
-});
\ No newline at end of file
diff --git a/docs/Building.md b/docs/Building.md
index b5a8c12c..07cbcda8 100644
--- a/docs/Building.md
+++ b/docs/Building.md
@@ -1,30 +1,11 @@
 # Building
 
-
 ### Building Steps
 
 An iOS Cordova application including the *cordova-plugin-iosrtc* plugin can be built using the [cordova-cli](https://cordova.apache.org/docs/en/edge/guide_cli_index.md.html#The%20Command-Line%20Interface) or Xcode.
 
-The plugin provides a ["hook"](../extra/hooks/iosrtc-swift-support.js) to automate required modifications in both *cordova-cli* and Xcode generated projects.
+The plugin provides a ["hook"](../extra/hooks/iosrtc-swift-support.js) to automate required modifications in both *cordova-cli* and Xcode generated projects. It is no **longer necessary to add the "hook" manually or add and remove the platform again**, it is executed before and after cordova is preparing your application.
 
-* Put the hook script under the "hooks/" folder of your Cordova project (or wherever you prefer) and give it execution permission:
-```bash
-$ chmod +x hooks/iosrtc-swift-support.js
-```
-* Add these lines to you "config.xml" file:
-```xml
-<platform name="ios">
-	<hook type="after_platform_add" src="hooks/iosrtc-swift-support.js" />
-</platform>
-```
-* Make sure you have the NPM [xcode](https://www.npmjs.com/package/xcode) package installed (locally or globally):
-```bash
-$ npm install -g xcode
-```
-* Remove the iOS platform and add it again:
-```bash
-$ cordova platform remove ios
-$ cordova platform add ios
 ```
 * You have two options right now:
   * Open the Xcode project and compile your application.
@@ -33,6 +14,8 @@ $ cordova platform add ios
 $ cordova build ios
 ```
 
+For more details about Cordova hook life cycle see: [Hooks Guide - Apache Cordova](https://cordova.apache.org/docs/en/latest/guide/appdev/hooks/)
+
 
 #### Bridging Header
 
@@ -50,24 +33,28 @@ It may happen that your Cordova application uses more than a single plugin coded
 #import "cordova-plugin-iosrtc-Bridging-Header.h"
 ```
 
-And then set `Unified-Bridging-Header.h` as the value of the "Objective-C Bridging Header" build setting in your Xcode project. For more information check this [issue](https://github.com/eface2face/cordova-plugin-iosrtc/issues/9).
+And then set `Unified-Bridging-Header.h` as the value of the "Objective-C Bridging Header" build setting in your Xcode project. For more information check this [issue](https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/9).
 
 
-#### Xcode
+#### Minimum Xcode Version
+
+You need to use minimum Xcode version 10.2 otherwise the build will fail due Apple Xcode know Bugs that have been fixed only on version above Xcode 10.2.
+
+See: [Xcode 10.2 Release Notes](https://developer.apple.com/documentation/xcode_release_notes/xcode_10_2_release_notes)
+
+#### Configuring Xcode manually
 
 If you still prefer to do it manually open it with Xcode and follow these steps:
 
-* Set "iOS Deployment Target" to `7.0` or higher within your project settings.
-* Set "Deployment Target" to `7.0` or higher within the project target settings.
+* Set "iOS Deployment Target" to `10.2` or higher within your project settings.
+* Set "Deployment Target" to `10.2` or higher within the project target settings.
 * Within the project "Build Settings" add an entry to the "Runpath Search Paths" setting with value `@executable_path/Frameworks`.
 * Within the project "Build Settings" set "Objective-C Bridging Header" to `PROJECT_NAME/Plugins/cordova-plugin-iosrtc/cordova-plugin-iosrtc-Bridging-Header.h` (read more about the "Bridging Header" above).
 * Within the project "Build Settings" set "Enable Bitcode" to "No".
-* If using Xcode 8, select "Use Swift Legacy version" under "Build Settings".
-
 
 #### iOS 10 notes
 
-On iOS 10 each permission requested must be accompanied by a description or the app won't start. Here is an example:
+On iOS 10 each permission requested must be accompanied by a description or the app will crash. Here is an example:
 
 ```xml
 <platform name="ios">
@@ -82,3 +69,23 @@ On iOS 10 each permission requested must be accompanied by a description or the
     </config-file>
 </platform>
 ```
+
+#### CocoaPods
+
+Using CocoaPods will lose the bitcode setting on `pod install`. Adding post_install step to the Podfile can help
+
+```
+# Example post install to disable bitcode for pods
+post_install do |installer|
+  installer.pods_project.targets.each do |target|
+    target.build_configurations.each do |config|
+      # Disable bitcode in order to support cordova-plugin-iosrtc
+      config.build_settings['ENABLE_BITCODE'] = 'NO'
+    end
+  end
+end
+```
+
+#### Capacitor
+
+When adding ios support using `npx cap add ios` the bitcode setting mentioned above will need to be set to 'NO' for the App project. Also see above for CocoaPods concerns since `pod install` will run every time you do `npx cap sync ios`
diff --git a/docs/BuildingLibWebRTC.md b/docs/BuildingLibWebRTC.md
index 6aa32c8e..c7cbb931 100644
--- a/docs/BuildingLibWebRTC.md
+++ b/docs/BuildingLibWebRTC.md
@@ -1,81 +1,18 @@
 # Build *libwebrtc* for cordova-plugin-iosrtc
 
-
 ### Get the *libwebrtc* source code
 
 There are several ways to get and build the library. We choose a project that performs this task nicely:
 
-* Clone the [webrtc-build-scripts](https://github.com/pristineio/webrtc-build-scripts) project and enter into it:
-```bash
-$ git clone https://github.com/pristineio/webrtc-build-scripts
-$ cd webrtc-build-scripts
-```
-
-* Follow the project steps and fetch the *libwebrtc* source code:
-```bash
-$ source ios/build.sh
-$ export WEBRTC_RELEASE=true
-$ get_webrtc
-```
-
-* Set a specific *libwebrtc* version:
-```bash
-$ update2Revision 13311
-```
-
-
-### Apply `libwebrtc-objc-iosrtc.patch`
-
-Google's *libwebrtc* Objective-C code lacks of `addtrack` and `removetrack` events in its `RTCMediaStream` class, making imposible to emulate them at JavaScript level. Thus a patch is needed to enable these W3C WebRTC features.
-
-**eFace2Face** created such a patch and sent it to the Google's *WebRTC* project, but [it was rejected](https://webrtc-codereview.appspot.com/50109004/) by arguing that the future of the WebRTC API is "track based" rather than "stream based".
-
-Since we live in the present we need those two events to properly implement the W3C WebRTC API in *cordova-plugin-iosrtc*, and hence the [patch](../extra/libwebrtc-objc-iosrtc.patch) is provided in the source code.
-
-* Apply the patch provided at `extra/libwebrtc-objc-iosrtc.patch` into the Objective-C source code of *libwebrtc*:
-```bash
-$ cd ios/webrtc/src
-$ patch -p1 < $PATH_TO_CORDOVA_PLUGIN_IOSRTC/extra/libwebrtc-objc-iosrtc.patch
-```
-
-
-### Apply `libwebrtc-objc-iosrtc-dtmf.patch`
-
-Google's *libwebrtc* Objective-C wrapper does not implement a `RTCPeerConnection.createDTMF` function, which makes it impossible to send DTMF from the Objective-C wrapper, and in turn this plugin. Google [will not implement this](https://bugs.chromium.org/p/webrtc/issues/detail?id=4180)
-because a new API was devised. Since we need to use this today, **AG Projects** created this patch exposing an API analogous to that on the C++ layer.
-
-* Apply the patch provided at `extra/libwebrtc-objc-iosrtc-dtmf.patch` into the Objective-C source code of *libwebrtc*:
-```bash
-$ cd ios/webrtc/src
-$ patch -p1 < $PATH_TO_CORDOVA_PLUGIN_IOSRTC/extra/libwebrtc-objc-iosrtc-dtmf.patch
-```
-
-
-### Misc
-
-In order to enable H.264 support and compile libWebRTC properly, apply `libwebrtc-objc-iosrtc-misc.patch`.
-
-```bash
-$ cd ios/webrtc/src
-$ patch -p1 < $PATH_TO_CORDOVA_PLUGIN_IOSRTC/extra/libwebrtc-objc-iosrtc-misc.patch
-
+[react-native-webrtc/blob/master/Documentation/BuildingWebRTC.md](https://github.com/react-native-webrtc/react-native-webrtc/blob/master/Documentation/BuildingWebRTC.md#building-webrtc)
 
-### Build *libwebrtc*
+Once you have WebRTC.framework copy the build into [lib](https://github.com/cordova-rtc/cordova-plugin-iosrtc/blob/master/lib/)
 
-* Go back to the `webrtc-build-scripts` root folder and build *libwebrtc*:
-```bash
-$ build_webrtc
-```
+#### Generate xcframework
 
-* Copy the *libwebrtc* Objective-C headers to *cordova-plugin-iosrtc*:
-```bash
-$ rm -f $PATH_TO_CORDOVA_PLUGIN_IOSRTC/src/webrtc-headers/*
-$ cp ios/webrtc/src/talk/app/webrtc/objc/public/* $PATH_TO_CORDOVA_PLUGIN_IOSRTC/src/webrtc-headers/
-```
+Once you have WebRTC.framework into [lib](https://github.com/cordova-rtc/cordova-plugin-iosrtc/blob/master/lib/), you need to generate WebRTC.xcframework
 
-* Copy the *libwebrtc* static library to *cordova-plugin-iosrtc*:
-```bash
-$ rm -f $PATH_TO_CORDOVA_PLUGIN_IOSRTC/lib/*
-$ cp ios/webrtc/libWebRTC-LATEST-Universal-Release.a $PATH_TO_CORDOVA_PLUGIN_IOSRTC/lib/
-```
+The script is here: https://github.com/cordova-rtc/cordova-plugin-iosrtc/blob/master/extra/ios_arch.js
 
+1. go to `plugins/cordova-plugin-iosrtc/extra` folder
+2. generate xcframework using: `node ios_arch.js --xcframework`
diff --git a/docs/iosrtc.md b/docs/iosrtc.md
index 600973e4..038f2379 100644
--- a/docs/iosrtc.md
+++ b/docs/iosrtc.md
@@ -13,13 +13,13 @@ cordova.plugins.iosrtc.getUserMedia(
   // constraints
   { audio: true, video: true },
   // success callback
-  function (stream) {
+  ).then(function (stream) {
     console.log('got local MediaStream: ', stream);
 
     pc.addStream(stream);
-  },
+  
   // failure callback
-  function (error) {
+  }).catch(function (error) {
     console.error('getUserMedia failed: ', error);
   }
 );
@@ -37,13 +37,28 @@ The function allows both the old/deprecated callbacks based syntax and the new o
 Constraints can be applied to the local video by using the latest W3C specification. Currently just the following constraints are supported:
 
 * `video.deviceId`
+* `video.width`
+* `video.width.exact`
+* `video.width.ideal`
 * `video.width.min`
 * `video.width.max`
+* `video.height`
+* `video.height.exact`
+* `video.height.ideal`
 * `video.height.min`
 * `video.height.max`
 * `video.frameRate`
 * `video.frameRate.min`
 * `video.frameRate.max`
+* `video.facingMode`
+* `video.facingMode.ideal`
+* `video.facingMode.exact`
+* `video.aspectRatio`
+* `video.aspectRatio.exact`
+* `video.aspectRatio.ideal`
+* `video.aspectRatio.max`
+* `video.aspectRatio.min`
+* `audio.deviceId`
 
 ```javascript
 cordova.plugins.iosrtc.getUserMedia({
@@ -55,8 +70,15 @@ cordova.plugins.iosrtc.getUserMedia({
       max: 640
     },
     frameRate: {
-      min: 1.0,
+      min: 2.0,
       max: 60.0
+    },
+    facingMode: 'environment',
+    aspectRatio: 4/3
+  },
+  {
+    audio: {
+      deviceId: 'Built-In Microphone'
     }
   }
 });
@@ -115,13 +137,8 @@ Exposes the  `MediaStream` class as defined in the [spec](http://w3c.github.io/m
 
 *NOTES:*
 
-* For internal reasons the `MediaStream` class points to the [Blob](https://developer.mozilla.org/en-US/docs/Web/API/Blob) class, so the `MediaStream` class constructor is not implemented (this class is exposed to make some WebRTC polyfill libraries happy).
 * `stop()` method implemented for backwards compatibility (it calls `stop()` on all its `MediaStreamTracks`).
 
-*TODO:*
-
-* `clone()` method.
-
 
 ### `iosrtc.MediaStreamTrack`
 
@@ -163,7 +180,7 @@ peerconnection.addEventListener('addstream', function (event) {
   cordova.plugins.iosrtc.observeVideo(video);
 
   // Attach the MediaStream to it.
-  video.src = URL.createObjectURL(event.stream);
+  video.srcObject = event.stream;
 
   // When the stream is ready to be rendered then append the video
   // element to the DOM.
@@ -174,13 +191,6 @@ peerconnection.addEventListener('addstream', function (event) {
 ```
 
 
-### `iosrtc.selectAudioOutput(output)`
-
-Select the audio output device. Given `output` argument must be "earpiece" or "speaker".
-
-*NOTE:* "speaker" output can only be set during a WebRTC session.
-
-
 ### `iosrtc.registerGlobals()`
 
 By calling this method the JavaScript global namespace gets "polluted" with the following additions:
@@ -208,33 +218,69 @@ The [debug](https://github.com/visionmedia/debug) module. Useful to enable verbo
 cordova.plugins.iosrtc.debug.enable('iosrtc*');
 ```
 
+### `iosrtc.selectAudioOutput`
+
+Tell the plugin that it must use `speaker` or `earpiece` as default audio output.
+
+```javascript
+// Use speaker audio output
+cordova.plugins.iosrtc.selectAudioOutput('speaker');
 
-### `iosrtc.rtcninjaPlugin`
+// Use earpiece audio output
+cordova.plugins.iosrtc.selectAudioOutput('earpiece');
+```
 
-A plugin interface for [rtcninja](https://github.com/eface2face/rtcninja.js/). 
+### `iosrtc.setDefaultAudioOutput`
 
-Usage (assuming that [cordova-plugin-device](https://www.npmjs.com/package/cordova-plugin-device) is installed):
+Tell the plugin that it must use `speaker` or `earpiece` as default audio output. You can call this method before establishing audio connection.
 
 ```javascript
-// Just for Cordova apps.
-document.addEventListener('deviceready', function () {
-  // Just for iOS devices.
-  if (window.device.platform === 'iOS') {
-    // Load rtcninja with cordova-plugin-iosrtc.
-    rtcninja({
-      plugin: cordova.plugins.iosrtc.rtcninjaPlugin
-    });
-  }
+// Use speaker audio output
+cordova.plugins.iosrtc.setDefaultAudioOutput('speaker');
+
+// Use earpiece audio output
+cordova.plugins.iosrtc.setDefaultAudioOutput('earpiece');
+```
 
-  console.log('WebRTC supported?: %s', rtcninja.hasWebRTC());
-  // => WebRTC supported?: true
+### `iosrtc.turnOnSpeaker`
 
-  rtcninja.RTCPeerConnection === cordova.plugins.iosrtc.RTCPeerConnection;
-  // => true
-});
+Tell the plugin that it should use `speaker` if no headphones (bluetooth or wire) is plugged or available.
+
+```javascript
+// Use speaker if no headphones available
+cordova.plugins.iosrtc.turnOnSpeaker(true);
+
+// Use default audio output
+cordova.plugins.iosrtc.turnOnSpeaker(false);
+```
+
+
+### `iosrtc.requestPermission`
+
+Request audio and/or camera permission if not requested yet.
+
+```javascript
+var needMic = true;
+var needCamera = true;
+cordova.plugins.iosrtc.requestPermission(needMic, needCamera, function (permissionApproved) {
+  // permissionApproved will be true if user accepted permission otherwise will be false.
+   console.error('requestPermission status: ', permissionApproved ? 'Approved' : 'Rejected');
+})
 ```
 
+### `iosrtc.initAudioDevices`
+
+Initialize the audio session, specifically setting the session's category and activating the session.
+This is to avoid cutting the audio of other applications on app boot.
 
+To use this install plugin using 
+```bash
+cordova plugin add cordova-plugin-iosrtc --variable MANUAL_INIT_AUDIO_DEVICE=TRUE
+```
+
+```javascript
+cordova.plugins.iosrtc.initAudioDevices();
+```
 
 ## Others
 
diff --git a/docs/videoCSS.md b/docs/videoCSS.md
index b72cc054..7fb450df 100644
--- a/docs/videoCSS.md
+++ b/docs/videoCSS.md
@@ -20,15 +20,13 @@ HTML elements on top of the native video elements.
 
 ## Quirks
 
-### Position of the `Webview` element does not match the corresponding `video` element
+### Position of the `WebView` element does not match the corresponding `video` element
 
-In some cases iOS adds offsets to the `Webview` element. As a result the coordinates between the `video` element and it's corresponding `Webview` element are unsync. 
+In some cases iOS adds an offset to the `WebView` element. As a result the coordinates between the `video` element and it's corresponding `WebView` element are out of sync.
 This happens when the status bar is shown. This can be fixed through disabling the statusbar in two ways:
 
-* at runtime through [cordova-plugin-statusbar](https://github.com/apache/cordova-plugin-statusbar)
-
-* permanently through editing the project `.plist` like
-
+* at runtime, using [cordova-plugin-statusbar](https://github.com/apache/cordova-plugin-statusbar)
+* permanently, editing the project `.plist` file:
 ```xml
   <key>UIViewControllerBasedStatusBarAppearance</key>
   <false/>
diff --git a/extra/hooks/iosrtc-swift-support.js b/extra/hooks/iosrtc-swift-support.js
index b392aae5..4c9c94bf 100755
--- a/extra/hooks/iosrtc-swift-support.js
+++ b/extra/hooks/iosrtc-swift-support.js
@@ -3,40 +3,42 @@
 'use strict';
 
 // This hook automates this:
-// https://github.com/eface2face/cordova-plugin-iosrtc/blob/master/docs/Building.md
+// https://github.com/cordova-rtc/cordova-plugin-iosrtc/blob/master/docs/Building.md
 
-var
-	fs = require("fs"),
-	path = require("path"),
+var fs = require('fs'),
+	path = require('path'),
 	xcode = require('xcode'),
-
-	BUILD_VERSION = '9.0',
-	BUILD_VERSION_XCODE = '"' + BUILD_VERSION + '"',
-	SWIFT_VERSION = '2.3',
+	xmlEntities = new (require('html-entities').XmlEntities)(),
+	DISABLE_IOSRTC_HOOK = process.env.DISABLE_IOSRTC_HOOK ? true : false,
+	IPHONEOS_DEPLOYMENT_TARGET = process.env.IPHONEOS_DEPLOYMENT_TARGET || '10.2',
+	IPHONEOS_DEPLOYMENT_TARGET_XCODE = '"' + IPHONEOS_DEPLOYMENT_TARGET + '"',
+	SWIFT_VERSION = process.env.SWIFT_VERSION || '4.2',
 	SWIFT_VERSION_XCODE = '"' + SWIFT_VERSION + '"',
 	RUNPATH_SEARCH_PATHS = '@executable_path/Frameworks',
 	RUNPATH_SEARCH_PATHS_XCODE = '"' + RUNPATH_SEARCH_PATHS + '"',
 	ENABLE_BITCODE = 'NO',
 	ENABLE_BITCODE_XCODE = '"' + ENABLE_BITCODE + '"',
-	BRIDGING_HEADER_END = '/Plugins/cordova-plugin-iosrtc/cordova-plugin-iosrtc-Bridging-Header.h',
-	COMMENT_KEY = /_comment$/;
-
+	UNIFIED_BRIDGING_HEADER = 'Plugins/Unified-Bridging-Header.h',
+	IOSRTC_BRIDGING_HEADER = 'cordova-plugin-iosrtc-Bridging-Header.h',
+	BRIDGING_HEADER_END = '/Plugins/cordova-plugin-iosrtc/' + IOSRTC_BRIDGING_HEADER,
+	TEST_UNIFIED_BRIDGING_HEADER = process.env.TEST_UNIFIED_BRIDGING_HEADER ? true : false; // Set to true to test handling of existing swift bridging header
 
 // Helpers
 
 // Returns the project name
 function getProjectName(protoPath) {
-	var
-		cordovaConfigPath = path.join(protoPath, 'config.xml'),
+	var cordovaConfigPath = path.join(protoPath, 'config.xml'),
 		content = fs.readFileSync(cordovaConfigPath, 'utf-8');
 
-	return /<name>([\s\S]*)<\/name>/mi.exec(content)[1].trim();
+	var name = /<name.*>([\s\S]*)<\/name>/im.exec(content)[1].trim();
+
+	return xmlEntities.decode(name);
 }
 
 // Drops the comments
+var COMMENT_KEY = /_comment$/;
 function nonComments(obj) {
-	var
-		keys = Object.keys(obj),
+	var keys = Object.keys(obj),
 		newObj = {},
 		i = 0;
 
@@ -49,94 +51,384 @@ function nonComments(obj) {
 	return newObj;
 }
 
+function matchBuildSettingsValue(value, expectedValue) {
+	return value === expectedValue;
+}
+
+function hasBuildSettingsValue(value, expectedValue) {
+	return (
+		value &&
+		(matchBuildSettingsValue(value, expectedValue) || value.indexOf(expectedValue) !== -1)
+	);
+}
+
+function convertToFloat(value) {
+	return parseFloat(String(value).replace(/[^\d.-]/g, ''), 10);
+}
+
+function matchMinValue(value, minValue) {
+	//console.log('matchMinValue', value, minValue)
+	return value && convertToFloat(value) >= convertToFloat(minValue);
+}
+
+function matchBuildSettingsMinValue(value, expectedValue) {
+	return (
+		value &&
+		(matchBuildSettingsValue(value, expectedValue) || matchMinValue(value, expectedValue))
+	);
+}
+
+function matchXcconfigPathValue(swiftOptions, path, expectedValue) {
+	return (
+		swiftOptions.filter(function (swiftOption) {
+			return swiftOption === path + ' = ' + expectedValue;
+		}).length > 0
+	);
+}
+
+function getXcconfigPathValue(swiftOptions, path) {
+	var swiftOption = swiftOptions.filter(function (swiftOption) {
+		return swiftOption.indexOf(path + ' = ') === 0;
+	})[0];
+
+	return swiftOption ? swiftOption.split(' = ')[1] : '';
+}
+
+function matchXcconfigMinValue(swiftOptions, path, expectedValue) {
+	var swiftOption = swiftOptions.find(function (swiftOption) {
+		return swiftOption.split(' = ')[0] === path;
+	});
+
+	return swiftOption && matchMinValue(swiftOption.split(' = ')[1], expectedValue);
+}
+
+function getRelativeToProjectRootPath(path, projectRoot) {
+	return path.replace(projectRoot, '.');
+}
+
+function readFileLines(filePath) {
+	return fs.readFileSync(filePath).toString().split('\n');
+}
+
+function debug(msg) {
+	console.log('iosrtc-swift-support.js [INFO] ' + msg);
+}
+
+function debugError(msg) {
+	console.error('iosrtc-swift-support.js [ERROR] ' + msg);
+	process.exit(1);
+}
 
 // Starting here
 
 module.exports = function (context) {
-	var
-		projectRoot = context.opts.projectRoot,
+	// This script has to be executed depending on the command line arguments, not
+	// on the hook execution cycle.
+	/*
+	if (context.hook !== 'before_build' && !context.cmdLine.includes('build')) {
+		return;
+	}
+	*/
+
+	if (DISABLE_IOSRTC_HOOK) {
+		debug('cordova-plugin-iosrtc hook is disabled');
+		return;
+	}
+
+	var projectRoot = context.opts.projectRoot,
 		projectName = getProjectName(projectRoot),
-		xcconfigPath = path.join(projectRoot, '/platforms/ios/cordova/build.xcconfig'),
+		platformPath = path.join(projectRoot, 'platforms', 'ios'),
+		platformProjectPath = path.join(platformPath, projectName),
+		xcconfigPath = path.join(platformPath, '/cordova/build.xcconfig'),
 		xcodeProjectName = projectName + '.xcodeproj',
-		xcodeProjectPath = path.join(projectRoot, 'platforms', 'ios', xcodeProjectName, 'project.pbxproj'),
-		swiftBridgingHead = projectName + BRIDGING_HEADER_END,
-		swiftBridgingHeadXcode = '"' + swiftBridgingHead + '"',
+		xcodeProjectConfigPath = path.join(platformPath, xcodeProjectName, 'project.pbxproj'),
+		swiftBridgingHeaderPath = projectName + BRIDGING_HEADER_END,
+		swiftBridgingHeaderPathXcode = '"' + swiftBridgingHeaderPath + '"',
 		swiftOptions = [''], // <-- begin to file appending AFTER initial newline
 		xcodeProject;
 
+	// Showing info about the tasks to do
+	debug('cordova-plugin-iosrtc hook is checking issues in the generated project files:');
+	debug(
+		'- Minimum "iOS Deployment Target" and "Deployment Target" to: ' +
+			IPHONEOS_DEPLOYMENT_TARGET_XCODE
+	);
+	debug('- "Runpath Search Paths" to: ' + RUNPATH_SEARCH_PATHS_XCODE);
+	if (TEST_UNIFIED_BRIDGING_HEADER) {
+		debug('- "Objective-C Bridging Header" to: ' + UNIFIED_BRIDGING_HEADER);
+	} else {
+		debug('- "Objective-C Bridging Header" to: ' + swiftBridgingHeaderPathXcode);
+	}
+	debug('- "ENABLE_BITCODE" set to: ' + ENABLE_BITCODE_XCODE);
+	debug('- "SWIFT_VERSION" set to: ' + SWIFT_VERSION_XCODE);
+
 	// Checking if the project files are in the right place
-	if (!fs.existsSync(xcodeProjectPath)) {
-		debugerror('an error occurred searching the project file at: "' + xcodeProjectPath + '"');
+	if (!fs.existsSync(xcodeProjectConfigPath)) {
+		debugError(
+			'an error occurred searching the project file at: "' + xcodeProjectConfigPath + '"'
+		);
 
 		return;
 	}
-	debug('".pbxproj" project file found: ' + xcodeProjectPath);
+	debug(
+		'".pbxproj" project file found: ' +
+			getRelativeToProjectRootPath(xcodeProjectConfigPath, projectRoot)
+	);
 
 	if (!fs.existsSync(xcconfigPath)) {
-		debugerror('an error occurred searching the project file at: "' + xcconfigPath + '"');
+		debugError('an error occurred searching the project file at: "' + xcconfigPath + '"');
 
 		return;
 	}
-	debug('".xcconfig" project file found: ' + xcconfigPath);
-
-	xcodeProject = xcode.project(xcodeProjectPath);
-
-	// Showing info about the tasks to do
-	debug('fixing issues in the generated project files:');
-	debug('- "iOS Deployment Target" and "Deployment Target" to: ' + BUILD_VERSION_XCODE);
-	debug('- "Runpath Search Paths" to: ' + RUNPATH_SEARCH_PATHS_XCODE);
-	debug('- "Objective-C Bridging Header" to: ' + swiftBridgingHeadXcode);
-	debug('- "ENABLE_BITCODE" set to: ' + ENABLE_BITCODE_XCODE);
-	debug('- "SWIFT_VERSION" set to: ' + SWIFT_VERSION_XCODE);
+	debug(
+		'".xcconfig" project file found: ' + getRelativeToProjectRootPath(xcconfigPath, projectRoot)
+	);
 
+	xcodeProject = xcode.project(xcodeProjectConfigPath);
 
 	// Massaging the files
 
 	// "build.xcconfig"
-	swiftOptions.push('LD_RUNPATH_SEARCH_PATHS = ' + RUNPATH_SEARCH_PATHS);
-	swiftOptions.push('SWIFT_OBJC_BRIDGING_HEADER = ' + swiftBridgingHead);
-	swiftOptions.push('IPHONEOS_DEPLOYMENT_TARGET = ' + BUILD_VERSION);
-	swiftOptions.push('ENABLE_BITCODE = ' + ENABLE_BITCODE);
+	debug('checking file: ' + getRelativeToProjectRootPath(xcconfigPath, projectRoot));
+	var xcconfigPathChanged = false;
+
+	// Parsing it
+	var currentSwiftOptions = [];
+	if (fs.existsSync(xcconfigPath)) {
+		currentSwiftOptions = readFileLines(xcconfigPath);
+	}
+
+	if (
+		!matchXcconfigPathValue(
+			currentSwiftOptions,
+			'LD_RUNPATH_SEARCH_PATHS',
+			RUNPATH_SEARCH_PATHS
+		)
+	) {
+		swiftOptions.push('LD_RUNPATH_SEARCH_PATHS = ' + RUNPATH_SEARCH_PATHS);
+		xcconfigPathChanged = true;
+	}
+
+	if (
+		!matchXcconfigMinValue(
+			currentSwiftOptions,
+			'IPHONEOS_DEPLOYMENT_TARGET',
+			IPHONEOS_DEPLOYMENT_TARGET
+		)
+	) {
+		swiftOptions.push('IPHONEOS_DEPLOYMENT_TARGET = ' + IPHONEOS_DEPLOYMENT_TARGET);
+		xcconfigPathChanged = true;
+	}
+
+	if (!matchXcconfigPathValue(currentSwiftOptions, 'ENABLE_BITCODE', ENABLE_BITCODE)) {
+		swiftOptions.push('ENABLE_BITCODE = ' + ENABLE_BITCODE);
+		xcconfigPathChanged = true;
+	}
+
+	if (!matchXcconfigPathValue(currentSwiftOptions, 'SWIFT_VERSION', SWIFT_VERSION)) {
+		swiftOptions.push('SWIFT_VERSION = ' + SWIFT_VERSION);
+		xcconfigPathChanged = true;
+	}
+
+	var existingSwiftBridgingHeaderPath;
+	if (TEST_UNIFIED_BRIDGING_HEADER) {
+		existingSwiftBridgingHeaderPath = path.join(platformProjectPath, UNIFIED_BRIDGING_HEADER);
+	} else {
+		if (
+			!matchXcconfigPathValue(
+				currentSwiftOptions,
+				'SWIFT_OBJC_BRIDGING_HEADER',
+				swiftBridgingHeaderPath
+			)
+		) {
+			var currentSwiftBridgingHeader = getXcconfigPathValue(
+				currentSwiftOptions,
+				'SWIFT_OBJC_BRIDGING_HEADER'
+			).replace('$(PROJECT_DIR)/$(PROJECT_NAME)/', '');
+			if (!existingSwiftBridgingHeaderPath) {
+				swiftOptions.push('SWIFT_OBJC_BRIDGING_HEADER = ' + swiftBridgingHeaderPath);
+				xcconfigPathChanged = true;
+			} else {
+				existingSwiftBridgingHeaderPath = path.join(
+					platformProjectPath,
+					currentSwiftBridgingHeader
+				);
+			}
+		}
+	}
+
 	// NOTE: Not needed
 	// swiftOptions.push('EMBEDDED_CONTENT_CONTAINS_SWIFT = YES');
-	fs.appendFileSync(xcconfigPath, swiftOptions.join('\n'));
-	debug('file correctly fixed: ' + xcconfigPath);
+
+	if (xcconfigPathChanged) {
+		fs.appendFileSync(xcconfigPath, swiftOptions.join('\n'));
+		debug('file correctly fixed: ' + getRelativeToProjectRootPath(xcconfigPath, projectRoot));
+	} else {
+		debug('file is correct: ' + getRelativeToProjectRootPath(xcconfigPath, projectRoot));
+	}
 
 	// "project.pbxproj"
+	debug('checking file: ' + getRelativeToProjectRootPath(xcodeProjectConfigPath, projectRoot));
+
 	// Parsing it
 	xcodeProject.parse(function (error) {
-		var configurations, buildSettings;
-
 		if (error) {
-			debugerror('an error occurred during the parsing of the project file');
+			debugError(
+				'an error occurred during the parsing of the project file: ' +
+					xcodeProjectConfigPath
+			);
 
 			return;
 		}
 
-
-		configurations = nonComments(xcodeProject.pbxXCBuildConfigurationSection());
-		// Adding or changing the parameters we need
+		// Adding or changing the parameters only if we need
+		var buildSettingsChanged = false;
+		var hasSwiftBridgingHeaderPathXcodes = [];
+		var configurations = nonComments(xcodeProject.pbxXCBuildConfigurationSection());
 		Object.keys(configurations).forEach(function (config) {
-			buildSettings = configurations[config].buildSettings;
-			buildSettings.LD_RUNPATH_SEARCH_PATHS = RUNPATH_SEARCH_PATHS_XCODE;
-			buildSettings.SWIFT_OBJC_BRIDGING_HEADER = swiftBridgingHeadXcode;
-			buildSettings.IPHONEOS_DEPLOYMENT_TARGET = BUILD_VERSION_XCODE;
-			buildSettings.ENABLE_BITCODE = ENABLE_BITCODE_XCODE;
-			buildSettings.SWIFT_VERSION = SWIFT_VERSION_XCODE;
+			var configuration = configurations[config],
+				buildSettings = configuration.buildSettings;
+
+			// Skip if not SDKROOT iphoneos
+			if (buildSettings.SDKROOT !== 'iphoneos') {
+				return;
+			}
+
+			if (
+				!hasBuildSettingsValue(
+					buildSettings.LD_RUNPATH_SEARCH_PATHS,
+					RUNPATH_SEARCH_PATHS_XCODE
+				)
+			) {
+				buildSettings.LD_RUNPATH_SEARCH_PATHS = RUNPATH_SEARCH_PATHS_XCODE;
+				buildSettingsChanged = true;
+			}
+
+			if (
+				!matchBuildSettingsMinValue(
+					buildSettings.IPHONEOS_DEPLOYMENT_TARGET,
+					IPHONEOS_DEPLOYMENT_TARGET_XCODE
+				)
+			) {
+				buildSettings.IPHONEOS_DEPLOYMENT_TARGET = IPHONEOS_DEPLOYMENT_TARGET_XCODE;
+				buildSettingsChanged = true;
+			}
+
+			if (!matchBuildSettingsValue(buildSettings.ENABLE_BITCODE, ENABLE_BITCODE_XCODE)) {
+				buildSettings.ENABLE_BITCODE = ENABLE_BITCODE_XCODE;
+				buildSettingsChanged = true;
+			}
+
+			if (!matchBuildSettingsValue(buildSettings.SWIFT_VERSION, SWIFT_VERSION_XCODE)) {
+				buildSettings.SWIFT_VERSION = SWIFT_VERSION_XCODE;
+				buildSettingsChanged = true;
+			}
+
+			if (TEST_UNIFIED_BRIDGING_HEADER) {
+				buildSettings.SWIFT_OBJC_BRIDGING_HEADER = '"' + UNIFIED_BRIDGING_HEADER + '"';
+				xcodeProject.addHeaderFile(UNIFIED_BRIDGING_HEADER);
+			}
+
+			if (
+				!hasBuildSettingsValue(
+					buildSettings.SWIFT_OBJC_BRIDGING_HEADER,
+					swiftBridgingHeaderPathXcode
+				)
+			) {
+				// Play nice with existing Swift Bridging Header value
+				if (existingSwiftBridgingHeaderPath) {
+					// Sync SWIFT_OBJC_BRIDGING_HEADER with existingSwiftBridgingHeaderPath if do not match
+					var existingSwiftBridgingHeaderPathXcode =
+						'"' + existingSwiftBridgingHeaderPath + '"';
+					if (
+						buildSettings.SWIFT_OBJC_BRIDGING_HEADER !==
+						existingSwiftBridgingHeaderPathXcode
+					) {
+						buildSettings.SWIFT_OBJC_BRIDGING_HEADER = existingSwiftBridgingHeaderPathXcode;
+						buildSettingsChanged = true;
+					}
+
+					if (
+						hasSwiftBridgingHeaderPathXcodes.indexOf(
+							existingSwiftBridgingHeaderPath
+						) === -1
+					) {
+						// Check if existing existingSwiftBridgingHeaderPath exists and get file lines
+
+						debug(
+							'checking file: ' +
+								getRelativeToProjectRootPath(
+									existingSwiftBridgingHeaderPath,
+									projectRoot
+								)
+						);
+
+						var existingSwiftBridgingHeaderFileLines = [];
+						if (fs.existsSync(existingSwiftBridgingHeaderPath)) {
+							existingSwiftBridgingHeaderFileLines = readFileLines(
+								existingSwiftBridgingHeaderPath
+							);
+						}
+
+						// Check if existing existingSwiftBridgingHeaderFileLines contains swiftBridgingHeaderPath
+						var swiftBridgingHeaderImport = '#import "' + IOSRTC_BRIDGING_HEADER + '"';
+						var hasSwiftBridgingHeaderPathXcode =
+							existingSwiftBridgingHeaderFileLines.filter(function (line) {
+								return line === swiftBridgingHeaderImport;
+							}).length > 0;
+
+						if (!hasSwiftBridgingHeaderPathXcode) {
+							debug(
+								'updating existing swift bridging header file: ' +
+									getRelativeToProjectRootPath(
+										existingSwiftBridgingHeaderPath,
+										projectRoot
+									)
+							);
+							existingSwiftBridgingHeaderFileLines.push(swiftBridgingHeaderImport);
+							fs.writeFileSync(
+								existingSwiftBridgingHeaderPath,
+								existingSwiftBridgingHeaderFileLines.join('\n'),
+								'utf-8'
+							);
+							debug(
+								'file correctly fixed: ' +
+									getRelativeToProjectRootPath(
+										existingSwiftBridgingHeaderPath,
+										projectRoot
+									)
+							);
+						} else {
+							debug(
+								'file is correct: ' +
+									getRelativeToProjectRootPath(
+										existingSwiftBridgingHeaderPath,
+										projectRoot
+									)
+							);
+						}
+
+						hasSwiftBridgingHeaderPathXcodes.push(existingSwiftBridgingHeaderPath);
+					}
+				} else {
+					buildSettings.SWIFT_OBJC_BRIDGING_HEADER = swiftBridgingHeaderPathXcode;
+					buildSettingsChanged = true;
+				}
+			}
 		});
 
-		// Writing the file again
-		fs.writeFileSync(xcodeProjectPath, xcodeProject.writeSync(), 'utf-8');
-		debug('file correctly fixed: ' + xcodeProjectPath);
+		// Writing the file only if changed
+		if (buildSettingsChanged) {
+			fs.writeFileSync(xcodeProjectConfigPath, xcodeProject.writeSync(), 'utf-8');
+			debug(
+				'file correctly fixed: ' +
+					getRelativeToProjectRootPath(xcodeProjectConfigPath, projectRoot)
+			);
+		} else {
+			debug(
+				'file is correct: ' +
+					getRelativeToProjectRootPath(xcodeProjectConfigPath, projectRoot)
+			);
+		}
 	});
 };
-
-
-function debug(msg) {
-	console.log('iosrtc-swift-support.js [INFO] ' + msg);
-}
-
-
-function debugerror(msg) {
-	console.error('iosrtc-swift-support.js [ERROR] ' + msg);
-}
diff --git a/extra/ios-websocket-hack.js b/extra/ios-websocket-hack.js
index 5fb3d576..553cb784 100644
--- a/extra/ios-websocket-hack.js
+++ b/extra/ios-websocket-hack.js
@@ -6,7 +6,7 @@
  * buggy WebSocket context.
  *
  * The issue is described here:
- *   https://github.com/eface2face/cordova-plugin-iosrtc/issues/12
+ *   https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/12
  *
  * USAGE:
  *
@@ -15,7 +15,6 @@
  * deteriorates the WebSocket performance.
  */
 
-
 (function () {
 	// run on iOS Cordova only
 	if (!(window.cordova && window.cordova.platformId === 'ios')) {
@@ -29,16 +28,13 @@
 	window.WebSocket = FakeWebSocket;
 	window.NativeWebSocket = NativeWebSocket;
 
-
 	// Fake WebSocket class that ill override the native one.
 	function FakeWebSocket() {
-		var
-			self = this,
+		var self = this,
 			url = arguments[0],
 			protocols = arguments[1],
 			listeners = {};
 
-
 		// WebSocket is an EventTarget as per W3C spec.
 
 		this.addEventListener = function (type, newListener) {
@@ -87,13 +83,13 @@
 		};
 
 		this.dispatchEvent = function (event) {
-			var
-				self = this,
+			var self = this,
 				type,
 				listenersType,
 				dummyListener,
 				stopImmediatePropagation = false,
-				i, listener;
+				i,
+				listener;
 
 			if (!(event instanceof Event)) {
 				throw new Error('first argument must be an instance of Event');
@@ -101,7 +97,7 @@
 
 			type = event.type;
 
-			listenersType = (listeners[type] || []);
+			listenersType = listeners[type] || [];
 
 			dummyListener = this['on' + type];
 			if (typeof dummyListener === 'function') {
@@ -164,7 +160,6 @@
 		});
 	}
 
-
 	// Expose W3C WebSocket attributes and setters.
 
 	Object.defineProperties(FakeWebSocket.prototype, {
@@ -251,7 +246,6 @@
 		}
 	});
 
-
 	// Expose W3C WebSocket methods.
 
 	FakeWebSocket.prototype.send = function (data) {
diff --git a/extra/ios_arch.js b/extra/ios_arch.js
new file mode 100644
index 00000000..4b4b53a1
--- /dev/null
+++ b/extra/ios_arch.js
@@ -0,0 +1,111 @@
+#!/usr/bin/env node
+
+/*jshint esversion: 6 */
+'use strict';
+
+const os = require('os');
+const fs = require('fs');
+const path = require('path');
+const exec = require('child_process').execSync;
+
+const LIB_PATH = path.join(__dirname, '../lib');
+const WEBRTC_LIB_PATH = path.join(LIB_PATH, 'WebRTC.framework');
+const WEBRTC_XC_LIB_PATH = path.join(LIB_PATH, 'WebRTC.xcframework');
+
+const TMP_PATH = os.tmpdir();
+const WEBRTC_SIM_LIB_PATH = path.join(TMP_PATH, '/sim/WebRTC.framework');
+const WEBRTC_DEVICE_LIB_PATH = path.join(TMP_PATH, '/device/WebRTC.framework');
+
+const ARCH_SIM_TYPES = ['i386', 'x86_64'];
+const ARCH_DEVICE_TYPES = ['armv7', 'arm64'];
+const ARCH_TYPES = ARCH_SIM_TYPES.concat(ARCH_DEVICE_TYPES);
+
+/* === Example to strip simulator archs for Apple Store Submission ===
+ *
+ * Step 1. extract all archs first
+ *    `node ios_arch.js --extract`
+ * Step 2. re-package binary without simulator archs
+ *    `node ios_arch.js --device`
+ *
+ * That's it!
+ * Remember to delete generated/backed up files (WebRTC-*) from step 1 if needed
+ *
+ * === How to Verify ===
+ *
+ * You can check current archs use this command:
+ *     `file $YOUR_PATH/WebRTC.framework/WebRTC`
+ */
+
+function extractArchitecture(archs, cwd) {
+	archs.forEach((arch) => {
+		//console.log(`lipo -extract ${arch} WebRTC -o WebRTC-${arch}`, cwd);
+		exec(`lipo -extract ${arch} WebRTC -o WebRTC-${arch}`, { cwd: cwd });
+	});
+}
+
+function exportArchitecture(archs, cwd) {
+	exec(
+		`lipo -o WebRTC -create ${archs
+			.map(function (arch) {
+				return `WebRTC-${arch}`;
+			})
+			.join(' ')}`,
+		{ cwd: cwd }
+	);
+}
+
+if (process.argv[2] === '--extract' || process.argv[2] === '-e') {
+	console.log('Extracting...');
+	// extract all archs, you might want to delete it later.
+	extractArchitecture(ARCH_TYPES, WEBRTC_LIB_PATH);
+	exec('cp WebRTC WebRTC-all', { cwd: WEBRTC_LIB_PATH }); // make a backup
+} else if (process.argv[2] === '--simulator' || process.argv[2] === '-s') {
+	// re-package simulator related archs only. ( i386, x86_64 )
+	console.log('Compiling simulator...');
+	exportArchitecture(ARCH_SIM_TYPES, WEBRTC_LIB_PATH);
+} else if (process.argv[2] === '--device' || process.argv[2] === '-d') {
+	// re-package device related archs only. ( armv7, arm64 )
+	console.log('Compiling device...');
+	exportArchitecture(ARCH_DEVICE_TYPES, WEBRTC_LIB_PATH);
+} else if (process.argv[2] === '--xcframework' || process.argv[2] === '-x') {
+	// extract all archs, you might want to delete it later.
+	extractArchitecture(ARCH_TYPES, WEBRTC_LIB_PATH);
+
+	// xcframework
+	console.log('Generating WebRTC.framework for device...');
+	fs.mkdirSync(WEBRTC_DEVICE_LIB_PATH, { recursive: true });
+	exec(`cp -r 'WebRTC.framework/'* '${WEBRTC_DEVICE_LIB_PATH}'`, { cwd: LIB_PATH });
+	exportArchitecture(ARCH_DEVICE_TYPES, WEBRTC_DEVICE_LIB_PATH);
+	exec(`rm -f WebRTC-*`, { cwd: WEBRTC_DEVICE_LIB_PATH });
+
+	console.log('Generating WebRTC.framework for simulator...');
+	fs.mkdirSync(WEBRTC_SIM_LIB_PATH, { recursive: true });
+	exec(`cp -r 'WebRTC.framework/'* '${WEBRTC_SIM_LIB_PATH}'`, { cwd: LIB_PATH });
+	exportArchitecture(ARCH_SIM_TYPES, WEBRTC_SIM_LIB_PATH);
+	exec(`rm -f WebRTC-*`, { cwd: WEBRTC_SIM_LIB_PATH });
+
+	console.log('Generating WebRTC.xcframework...');
+	exec(`rm -rf ${WEBRTC_XC_LIB_PATH}`, { cwd: LIB_PATH });
+
+	exec(
+		`xcodebuild -create-xcframework -framework ${WEBRTC_DEVICE_LIB_PATH} -framework ${WEBRTC_SIM_LIB_PATH} -output '${WEBRTC_XC_LIB_PATH}'`,
+		{ cwd: LIB_PATH }
+	);
+
+	exec(`rm -f WebRTC-*`, { cwd: WEBRTC_LIB_PATH });
+	exec(`rm -fr ${WEBRTC_DEVICE_LIB_PATH}`, { cwd: TMP_PATH });
+	exec(`rm -fr ${WEBRTC_SIM_LIB_PATH}`, { cwd: TMP_PATH });
+} else if (process.argv[2] === '--list' || process.argv[2] === '-l') {
+	// List WebRTC architectures
+	console.log('List WebRTC architectures...');
+	console.log(exec(`file WebRTC`, { cwd: WEBRTC_LIB_PATH }).toString().trim());
+} else if (process.argv[2] === '--clean' || process.argv[2] === '-l') {
+	// Delete WebRTC-* architectures
+	console.log('Clean WebRTC architectures...');
+	exec(`rm -f WebRTC-*`, { cwd: WEBRTC_LIB_PATH });
+} else {
+	console.log('Unknow options');
+}
+
+console.log('List WebRTC files...');
+console.log(exec('ls -ahl | grep WebRTC', { cwd: LIB_PATH }).toString().trim());
diff --git a/extra/libwebrtc-objc-iosrtc-dtmf.patch b/extra/libwebrtc-objc-iosrtc-dtmf.patch
deleted file mode 100644
index 79ea693f..00000000
--- a/extra/libwebrtc-objc-iosrtc-dtmf.patch
+++ /dev/null
@@ -1,336 +0,0 @@
---- a/talk/app/webrtc/legacy_objc_api.gyp
-+++ b/talk/app/webrtc/legacy_objc_api.gyp
-@@ -41,6 +41,8 @@
-             'objc/RTCAudioTrack.mm',
-             'objc/RTCDataChannel+Internal.h',
-             'objc/RTCDataChannel.mm',
-+            'objc/RTCDTMFSender+Internal.h',
-+            'objc/RTCDTMFSender.mm',
-             'objc/RTCEnumConverter.h',
-             'objc/RTCEnumConverter.mm',
-             'objc/RTCI420Frame+Internal.h',
-@@ -84,6 +86,7 @@
-             'objc/public/RTCAudioSource.h',
-             'objc/public/RTCAudioTrack.h',
-             'objc/public/RTCDataChannel.h',
-+            'objc/public/RTCDTMFSender.h',
-             'objc/public/RTCFileLogger.h',
-             'objc/public/RTCI420Frame.h',
-             'objc/public/RTCICECandidate.h',
---- /dev/null
-+++ b/talk/app/webrtc/objc/RTCDTMFSender+Internal.h
-@@ -0,0 +1,42 @@
-+/*
-+ * libjingle
-+ * Copyright 2014 Google Inc.
-+ *
-+ * Redistribution and use in source and binary forms, with or without
-+ * modification, are permitted provided that the following conditions are met:
-+ *
-+ *  1. Redistributions of source code must retain the above copyright notice,
-+ *     this list of conditions and the following disclaimer.
-+ *  2. Redistributions in binary form must reproduce the above copyright notice,
-+ *     this list of conditions and the following disclaimer in the documentation
-+ *     and/or other materials provided with the distribution.
-+ *  3. The name of the author may not be used to endorse or promote products
-+ *     derived from this software without specific prior written permission.
-+ *
-+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
-+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
-+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
-+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
-+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
-+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
-+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
-+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-+ */
-+
-+#import "RTCDTMFSender.h"
-+
-+#include "webrtc/api/dtmfsenderinterface.h"
-+#include "webrtc/base/scoped_ref_ptr.h"
-+
-+
-+@interface RTCDTMFSender (Internal)
-+
-+@property(nonatomic, readonly)
-+    rtc::scoped_refptr<webrtc::DtmfSenderInterface> dtmfSender;
-+
-+- (instancetype)initWithDtmfSender:
-+        (rtc::scoped_refptr<webrtc::DtmfSenderInterface>)dtmfSender;
-+
-+@end
---- /dev/null
-+++ b/talk/app/webrtc/objc/RTCDTMFSender.mm
-@@ -0,0 +1,150 @@
-+/*
-+ * libjingle
-+ * Copyright 2014 Google Inc.
-+ *
-+ * Redistribution and use in source and binary forms, with or without
-+ * modification, are permitted provided that the following conditions are met:
-+ *
-+ *  1. Redistributions of source code must retain the above copyright notice,
-+ *     this list of conditions and the following disclaimer.
-+ *  2. Redistributions in binary form must reproduce the above copyright notice,
-+ *     this list of conditions and the following disclaimer in the documentation
-+ *     and/or other materials provided with the distribution.
-+ *  3. The name of the author may not be used to endorse or promote products
-+ *     derived from this software without specific prior written permission.
-+ *
-+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
-+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
-+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
-+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
-+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
-+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
-+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
-+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-+ */
-+
-+#if !defined(__has_feature) || !__has_feature(objc_arc)
-+#error "This file requires ARC support."
-+#endif
-+
-+#include "RTCDTMFSender.h"
-+#include "RTCAudioTrack+Internal.h"
-+#include "RTCMediaStreamTrack+Internal.h"
-+
-+#include <memory>
-+
-+#include "webrtc/api/dtmfsenderinterface.h"
-+
-+
-+// utilities
-+static NSString* NSStringFromStdString(const std::string& stdString) {
-+  // std::string may contain null termination character so we construct
-+  // using length.
-+  return [[NSString alloc] initWithBytes:stdString.data()
-+                                  length:stdString.length()
-+                                encoding:NSUTF8StringEncoding];
-+}
-+
-+static std::string StdStringFromNSString(NSString* nsString) {
-+  NSData* charData = [nsString dataUsingEncoding:NSUTF8StringEncoding];
-+  return std::string(reinterpret_cast<const char*>([charData bytes]),
-+                     [charData length]);
-+}
-+
-+
-+namespace webrtc {
-+
-+class RTCDTMFSenderObserver : public DtmfSenderObserverInterface {
-+ public:
-+  RTCDTMFSenderObserver(RTCDTMFSender* sender) { _sender = sender; }
-+
-+  void OnToneChange(const std::string& tone) override {
-+    if (!_sender.delegate) {
-+        return;
-+    }
-+    [_sender.delegate toneChange:NSStringFromStdString(tone)];
-+  }
-+
-+ private:
-+  __weak RTCDTMFSender* _sender;
-+};
-+}
-+
-+
-+@implementation RTCDTMFSender {
-+  rtc::scoped_refptr<webrtc::DtmfSenderInterface> _dtmfSender;
-+  std::unique_ptr<webrtc::RTCDTMFSenderObserver> _observer;
-+  BOOL _isObserverRegistered;
-+}
-+
-+- (void)dealloc {
-+  // Handles unregistering the observer properly.
-+  self.delegate = nil;
-+}
-+
-+- (NSString*)toneBuffer {
-+  return NSStringFromStdString(_dtmfSender->tones());
-+}
-+
-+- (BOOL)canInsertDTMF {
-+  return _dtmfSender->CanInsertDtmf();
-+}
-+
-+- (NSInteger)duration {
-+  return _dtmfSender->duration();
-+}
-+
-+- (NSInteger)interToneGap {
-+  return _dtmfSender->inter_tone_gap();
-+}
-+
-+- (RTCAudioTrack*)track {
-+  rtc::scoped_refptr<webrtc::AudioTrackInterface> audioTrack(
-+    const_cast<webrtc::AudioTrackInterface*>(_dtmfSender->track()));
-+  if (audioTrack) {
-+    return [[RTCAudioTrack alloc] initWithMediaTrack:audioTrack];
-+  }
-+  return nil;
-+}
-+
-+- (void)setDelegate:(id<RTCDTMFSenderDelegate>)delegate {
-+  if (_delegate == delegate) {
-+    return;
-+  }
-+  if (_isObserverRegistered) {
-+    _dtmfSender->UnregisterObserver();
-+    _isObserverRegistered = NO;
-+  }
-+  _delegate = delegate;
-+  if (_delegate) {
-+    _dtmfSender->RegisterObserver(_observer.get());
-+    _isObserverRegistered = YES;
-+  }
-+}
-+
-+- (BOOL)insertDTMF:(NSString*)tones withDuration:(NSInteger)duration andInterToneGap:(NSInteger)interToneGap {
-+  return _dtmfSender->InsertDtmf(StdStringFromNSString(tones), duration, interToneGap);
-+}
-+
-+@end
-+
-+@implementation RTCDTMFSender (Internal)
-+
-+- (instancetype)initWithDtmfSender:
-+                    (rtc::scoped_refptr<webrtc::DtmfSenderInterface>)
-+                dtmfSender {
-+  NSAssert(dtmfSender != NULL, @"dtmfSender cannot be NULL");
-+  if (self = [super init]) {
-+    _dtmfSender = dtmfSender;
-+    _observer.reset(new webrtc::RTCDTMFSenderObserver(self));
-+  }
-+  return self;
-+}
-+
-+- (rtc::scoped_refptr<webrtc::DtmfSenderInterface>)dtmfSender {
-+  return _dtmfSender;
-+}
-+
-+@end
---- a/talk/app/webrtc/objc/RTCPeerConnection.mm
-+++ b/talk/app/webrtc/objc/RTCPeerConnection.mm
-@@ -31,7 +31,9 @@
- 
- #import "RTCPeerConnection+Internal.h"
- 
-+#import "RTCAudioTrack+Internal.h"
- #import "RTCDataChannel+Internal.h"
-+#import "RTCDTMFSender+Internal.h"
- #import "RTCEnumConverter.h"
- #import "RTCICECandidate+Internal.h"
- #import "RTCICEServer+Internal.h"
-@@ -171,6 +173,13 @@ - (RTCDataChannel*)createDataChannelWithLabel:(NSString*)label
-   return [[RTCDataChannel alloc] initWithDataChannel:dataChannel];
- }
- 
-+- (RTCDTMFSender*)createDTMFSenderForTrack:(RTCAudioTrack*)track {
-+  rtc::scoped_refptr<webrtc::DtmfSenderInterface> dtmfSender =
-+      self.peerConnection->CreateDtmfSender(track.audioTrack);
-+  return dtmfSender ? [[RTCDTMFSender alloc] initWithDtmfSender:dtmfSender]
-+                    : nil;
-+}
-+
- - (void)createAnswerWithDelegate:(id<RTCSessionDescriptionDelegate>)delegate
-                      constraints:(RTCMediaConstraints*)constraints {
-   rtc::scoped_refptr<webrtc::RTCCreateSessionDescriptionObserver>
---- /dev/null
-+++ b/talk/app/webrtc/objc/public/RTCDTMFSender.h
-@@ -0,0 +1,66 @@
-+/*
-+ * libjingle
-+ * Copyright 2014 Google Inc.
-+ *
-+ * Redistribution and use in source and binary forms, with or without
-+ * modification, are permitted provided that the following conditions are met:
-+ *
-+ *  1. Redistributions of source code must retain the above copyright notice,
-+ *     this list of conditions and the following disclaimer.
-+ *  2. Redistributions in binary form must reproduce the above copyright notice,
-+ *     this list of conditions and the following disclaimer in the documentation
-+ *     and/or other materials provided with the distribution.
-+ *  3. The name of the author may not be used to endorse or promote products
-+ *     derived from this software without specific prior written permission.
-+ *
-+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
-+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
-+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
-+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
-+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
-+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
-+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
-+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-+ */
-+
-+#import <Foundation/Foundation.h>
-+
-+#import "RTCAudioTrack.h"
-+
-+NS_ASSUME_NONNULL_BEGIN
-+
-+@class RTCDTMFSender;
-+// Protocol for receving tone change events.
-+@protocol RTCDTMFSenderDelegate<NSObject>
-+
-+// Called when a DTMF tone is played out.
-+- (void)toneChange:(NSString*)tone;
-+
-+@end
-+
-+// ObjectiveC wrapper for a DtmfSender object.
-+// See webrtc/api/dtmfsenderinterface.h
-+@interface RTCDTMFSender : NSObject
-+
-+@property(nonatomic, readonly) BOOL canInsertDTMF;
-+@property(nonatomic, readonly) NSString* toneBuffer;
-+@property(nonatomic, readonly) NSInteger duration;
-+@property(nonatomic, readonly) NSInteger interToneGap;
-+@property(nonatomic, weak) id<RTCDTMFSenderDelegate> delegate;
-+// The track associated with this DTMF sender. This property
-+// returns a copy of the RTCMediaStreamTrack
-+@property(nonatomic, copy, nullable) RTCAudioTrack *track;
-+
-+- (BOOL)insertDTMF:(NSString*)tones withDuration:(NSInteger)duration andInterToneGap:(NSInteger)interToneGap;
-+
-+#ifndef DOXYGEN_SHOULD_SKIP_THIS
-+// Disallow init and don't add to documentation
-+- (id)init __attribute__((
-+    unavailable("init is not a supported initializer for this class.")));
-+#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-+
-+@end
-+
-+NS_ASSUME_NONNULL_END
---- a/talk/app/webrtc/objc/public/RTCPeerConnection.h
-+++ b/talk/app/webrtc/objc/public/RTCPeerConnection.h
-@@ -27,9 +27,11 @@
- 
- #import "RTCPeerConnectionDelegate.h"
- 
-+@class RTCAudioTrack;
- @class RTCConfiguration;
- @class RTCDataChannel;
- @class RTCDataChannelInit;
-+@class RTCDTMFSender;
- @class RTCICECandidate;
- @class RTCICEServers;
- @class RTCMediaConstraints;
-@@ -76,6 +78,9 @@
- - (RTCDataChannel*)createDataChannelWithLabel:(NSString*)label
-                                        config:(RTCDataChannelInit*)config;
- 
-+// Create a DTMF sender.
-+- (RTCDTMFSender*)createDTMFSenderForTrack:(RTCAudioTrack*)audioTrack;
-+
- // Create a new offer.
- // Success or failure will be reported via RTCSessionDescriptionDelegate.
- - (void)createOfferWithDelegate:(id<RTCSessionDescriptionDelegate>)delegate
diff --git a/extra/libwebrtc-objc-iosrtc-misc.patch b/extra/libwebrtc-objc-iosrtc-misc.patch
deleted file mode 100644
index 31446e6b..00000000
--- a/extra/libwebrtc-objc-iosrtc-misc.patch
+++ /dev/null
@@ -1,22 +0,0 @@
---- a/webrtc/build/common.gypi
-+++ b/webrtc/build/common.gypi
-@@ -155,7 +155,7 @@
- 
-     # Enable this to use HW H.264 encoder/decoder on iOS/Mac PeerConnections.
-     # Enabling this may break interop with Android clients that support H264.
--    'use_objc_h264%': 0,
-+    'use_objc_h264%': 1,
- 
-     # Enable this to prevent extern symbols from being hidden on iOS builds.
-     # The chromium settings we inherit hide symbols by default on Release
---- a/webrtc/system_wrappers/system_wrappers.gyp
-+++ b/webrtc/system_wrappers/system_wrappers.gyp
-@@ -76,6 +76,8 @@
-         'source/trace_posix.h',
-         'source/trace_win.cc',
-         'source/trace_win.h',
-+        'include/metrics_default.h',
-+        'source/metrics_default.cc',
-       ],
-       'conditions': [
-         ['enable_data_logging==1', {
diff --git a/extra/libwebrtc-objc-iosrtc.patch b/extra/libwebrtc-objc-iosrtc.patch
deleted file mode 100644
index 57953817..00000000
--- a/extra/libwebrtc-objc-iosrtc.patch
+++ /dev/null
@@ -1,284 +0,0 @@
---- a/talk/app/webrtc/objc/RTCMediaStream+Internal.h
-+++ b/talk/app/webrtc/objc/RTCMediaStream+Internal.h
-@@ -37,4 +37,12 @@
- - (id)initWithMediaStream:
-         (rtc::scoped_refptr<webrtc::MediaStreamInterface>)mediaStream;
- 
-+- (void)update;
-+- (BOOL)hasNativeAudioTrack:(RTCAudioTrack *)objcTrack;
-+- (BOOL)hasNativeVideoTrack:(RTCVideoTrack *)objcTrack;
-+- (BOOL)hasObjcAudioTrack:
-+    (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeTrack;
-+- (BOOL)hasObjcVideoTrack:
-+    (rtc::scoped_refptr<webrtc::VideoTrackInterface>)nativeTrack;
-+
- @end
---- a/talk/app/webrtc/objc/RTCMediaStream.mm
-+++ b/talk/app/webrtc/objc/RTCMediaStream.mm
-@@ -31,16 +31,31 @@
- 
- #import "RTCMediaStream+Internal.h"
- 
--#import "RTCAudioTrack+Internal.h"
- #import "RTCMediaStreamTrack+Internal.h"
-+#import "RTCAudioTrack+Internal.h"
- #import "RTCVideoTrack+Internal.h"
- 
- #include "webrtc/api/mediastreaminterface.h"
- 
-+namespace webrtc {
-+  class RTCMediaStreamObserver : public ObserverInterface {
-+    public:
-+    RTCMediaStreamObserver(RTCMediaStream* stream) { _stream = stream; }
-+
-+    void OnChanged() override {
-+      [_stream update];
-+    }
-+
-+   private:
-+    __weak RTCMediaStream* _stream;
-+  };
-+}
-+
- @implementation RTCMediaStream {
-   NSMutableArray* _audioTracks;
-   NSMutableArray* _videoTracks;
-   rtc::scoped_refptr<webrtc::MediaStreamInterface> _mediaStream;
-+  std::unique_ptr<webrtc::RTCMediaStreamObserver> _observer;
- }
- 
- - (NSString*)description {
-@@ -64,7 +79,6 @@ - (NSString*)label {
- 
- - (BOOL)addAudioTrack:(RTCAudioTrack*)track {
-   if (self.mediaStream->AddTrack(track.audioTrack)) {
--    [_audioTracks addObject:track];
-     return YES;
-   }
-   return NO;
-@@ -72,7 +86,6 @@ - (BOOL)addAudioTrack:(RTCAudioTrack*)track {
- 
- - (BOOL)addVideoTrack:(RTCVideoTrack*)track {
-   if (self.mediaStream->AddTrack(track.nativeVideoTrack)) {
--    [_videoTracks addObject:track];
-     return YES;
-   }
-   return NO;
-@@ -83,7 +96,6 @@ - (BOOL)removeAudioTrack:(RTCAudioTrack*)track {
-   NSAssert(index != NSNotFound,
-            @"|removeAudioTrack| called on unexpected RTCAudioTrack");
-   if (index != NSNotFound && self.mediaStream->RemoveTrack(track.audioTrack)) {
--    [_audioTracks removeObjectAtIndex:index];
-     return YES;
-   }
-   return NO;
-@@ -93,9 +105,7 @@ - (BOOL)removeVideoTrack:(RTCVideoTrack*)track {
-   NSUInteger index = [_videoTracks indexOfObjectIdenticalTo:track];
-   NSAssert(index != NSNotFound,
-            @"|removeAudioTrack| called on unexpected RTCVideoTrack");
--  if (index != NSNotFound &&
--      self.mediaStream->RemoveTrack(track.nativeVideoTrack)) {
--    [_videoTracks removeObjectAtIndex:index];
-+  if (index != NSNotFound && self.mediaStream->RemoveTrack(track.nativeVideoTrack)) {
-     return YES;
-   }
-   return NO;
-@@ -119,6 +129,8 @@ - (id)initWithMediaStream:
-     _audioTracks = [NSMutableArray arrayWithCapacity:audio_tracks.size()];
-     _videoTracks = [NSMutableArray arrayWithCapacity:video_tracks.size()];
-     _mediaStream = mediaStream;
-+    _observer.reset(new webrtc::RTCMediaStreamObserver(self));
-+    _mediaStream->RegisterObserver(_observer.get());
- 
-     for (size_t i = 0; i < audio_tracks.size(); ++i) {
-       rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
-@@ -139,8 +151,157 @@ - (id)initWithMediaStream:
-   return self;
- }
- 
-+- (void)dealloc {
-+  _mediaStream->UnregisterObserver(_observer.get());
-+}
-+
- - (rtc::scoped_refptr<webrtc::MediaStreamInterface>)mediaStream {
-   return _mediaStream;
- }
- 
-+- (void)update {
-+  size_t i;
-+  webrtc::AudioTrackVector native_audio_tracks = _mediaStream->GetAudioTracks();
-+  webrtc::VideoTrackVector native_video_tracks = _mediaStream->GetVideoTracks();
-+  std::vector<size_t> removedAudioTrackIndexes;
-+  std::vector<size_t> removedVideoTrackIndexes;
-+
-+  // Detect audio tracks removal.
-+  for (i = 0; i < [_audioTracks count]; i++) {
-+    RTCAudioTrack* objcTrack = [_audioTracks objectAtIndex:i];
-+    NSNumber* index;
-+
-+    if (![self hasNativeAudioTrack:objcTrack]) {
-+      removedAudioTrackIndexes.push_back(i);
-+    }
-+  }
-+
-+  // Detect video tracks removal.
-+  for (i = 0; i < [_videoTracks count]; i++) {
-+    RTCVideoTrack* objcTrack = [_videoTracks objectAtIndex:i];
-+    NSNumber* index;
-+
-+    if (![self hasNativeVideoTrack:objcTrack]) {
-+      removedVideoTrackIndexes.push_back(i);
-+    }
-+  }
-+
-+  // Remove old audio tracks and notify the delegate.
-+  for (std::vector<size_t>::iterator it = removedAudioTrackIndexes.begin();
-+    it != removedAudioTrackIndexes.end(); ++it) {
-+    RTCAudioTrack* objcTrack = [_audioTracks objectAtIndex:*it];
-+
-+    // Remove the track from the ObjC container.
-+    [_audioTracks removeObjectAtIndex:*it];
-+
-+    // Notify the delegate.
-+    [_delegate OnRemoveAudioTrack:self track:objcTrack];
-+  }
-+
-+  // Remove old video tracks and notify the delegate.
-+  for (std::vector<size_t>::iterator it = removedVideoTrackIndexes.begin();
-+    it != removedVideoTrackIndexes.end(); ++it) {
-+    RTCVideoTrack* objcTrack = [_videoTracks objectAtIndex:*it];
-+
-+    // Remove the track from the ObjC container.
-+    [_videoTracks removeObjectAtIndex:*it];
-+
-+    // Notify the delegate.
-+    [_delegate OnRemoveVideoTrack:self track:objcTrack];
-+  }
-+
-+  // Detect audio tracks addition and notify the delegate.
-+  for (i = 0; i < native_audio_tracks.size(); ++i) {
-+    rtc::scoped_refptr<webrtc::AudioTrackInterface> nativeTrack =
-+      native_audio_tracks[i];
-+
-+    if (![self hasObjcAudioTrack:nativeTrack]) {
-+      // Create the RTCAudioTrack instance and add it to the ObjC container.
-+      RTCAudioTrack* objcTrack =
-+        [[RTCAudioTrack alloc] initWithMediaTrack:nativeTrack];
-+      [_audioTracks addObject:objcTrack];
-+
-+      // Notify the delegate.
-+      [_delegate OnAddAudioTrack:self track:objcTrack];
-+    }
-+  }
-+
-+  // Detect video tracks addition and notify the delegate.
-+  for (i = 0; i < native_video_tracks.size(); ++i) {
-+    rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeTrack =
-+      native_video_tracks[i];
-+
-+    if (![self hasObjcVideoTrack:nativeTrack]) {
-+      // Create the RTCVideoTrack instance and add it to the ObjC container.
-+      RTCVideoTrack* objcTrack =
-+        [[RTCVideoTrack alloc] initWithMediaTrack:nativeTrack];
-+      [_videoTracks addObject:objcTrack];
-+
-+      // Notify the delegate.
-+      [_delegate OnAddVideoTrack:self track:objcTrack];
-+    }
-+  }
-+
-+  unsigned long num_native_audios = native_audio_tracks.size();
-+  unsigned long num_native_videos = native_video_tracks.size();
-+  unsigned long num_objc_audios = (unsigned long)[_audioTracks count];
-+  unsigned long num_objc_videos = (unsigned long)[_videoTracks count];
-+
-+  NSAssert(num_native_audios == num_objc_audios,
-+    @"ObjC audio tracks does not match number of native audio tracks");
-+
-+  NSAssert(num_native_videos == num_objc_videos,
-+    @"ObjC video tracks does not match number of native video tracks");
-+}
-+
-+- (BOOL)hasNativeAudioTrack:(RTCAudioTrack *)objcTrack {
-+  webrtc::AudioTrackVector audio_tracks = _mediaStream->GetAudioTracks();
-+
-+  for (size_t i = 0; i < audio_tracks.size(); ++i) {
-+    rtc::scoped_refptr<webrtc::AudioTrackInterface> track = audio_tracks[i];
-+
-+    if (track->id().compare(objcTrack.audioTrack->id()) == 0) {
-+      return YES;
-+    }
-+  }
-+  return NO;
-+}
-+
-+- (BOOL)hasNativeVideoTrack:(RTCVideoTrack *)objcTrack {
-+  webrtc::VideoTrackVector video_tracks = _mediaStream->GetVideoTracks();
-+
-+  for (size_t i = 0; i < video_tracks.size(); ++i) {
-+    rtc::scoped_refptr<webrtc::VideoTrackInterface> track = video_tracks[i];
-+
-+    if (track->id().compare(objcTrack.nativeVideoTrack->id()) == 0) {
-+      return YES;
-+    }
-+  }
-+  return NO;
-+}
-+
-+- (BOOL)hasObjcAudioTrack:
-+    (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeTrack {
-+  for (size_t i = 0; i < [_audioTracks count]; i++) {
-+    RTCAudioTrack* track = [_audioTracks objectAtIndex:i];
-+
-+    if (track.audioTrack->id().compare(nativeTrack->id()) == 0) {
-+      return YES;
-+    }
-+  }
-+  return NO;
-+}
-+
-+- (BOOL)hasObjcVideoTrack:
-+    (rtc::scoped_refptr<webrtc::VideoTrackInterface>)nativeTrack {
-+  for (size_t i = 0; i < [_videoTracks count]; i++) {
-+    RTCVideoTrack* track = [_videoTracks objectAtIndex:i];
-+
-+    if (track.nativeVideoTrack->id().compare(nativeTrack->id()) == 0) {
-+      return YES;
-+    }
-+  }
-+  return NO;
-+}
-+
- @end
---- a/talk/app/webrtc/objc/public/RTCMediaStream.h
-+++ b/talk/app/webrtc/objc/public/RTCMediaStream.h
-@@ -29,6 +29,21 @@
- 
- @class RTCAudioTrack;
- @class RTCVideoTrack;
-+@class RTCMediaStream;
-+@class RTCMediaStreamTrack;
-+
-+@protocol RTCMediaStreamDelegate<NSObject>
-+
-+- (void)OnAddAudioTrack:
-+    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
-+- (void)OnAddVideoTrack:
-+    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
-+- (void)OnRemoveAudioTrack:
-+    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
-+- (void)OnRemoveVideoTrack:
-+    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
-+
-+@end
- 
- // RTCMediaStream is an ObjectiveC wrapper for MediaStreamInterface.
- @interface RTCMediaStream : NSObject
-@@ -36,6 +51,7 @@
- @property(nonatomic, strong, readonly) NSArray *audioTracks;
- @property(nonatomic, strong, readonly) NSArray *videoTracks;
- @property(nonatomic, strong, readonly) NSString *label;
-+@property(nonatomic, weak) id<RTCMediaStreamDelegate> delegate;
- 
- - (BOOL)addAudioTrack:(RTCAudioTrack *)track;
- - (BOOL)addVideoTrack:(RTCVideoTrack *)track;
diff --git a/extra/renderer-and-libwebrtc-tests.js b/extra/renderer-and-libwebrtc-tests.js
new file mode 100644
index 00000000..533538fd
--- /dev/null
+++ b/extra/renderer-and-libwebrtc-tests.js
@@ -0,0 +1,494 @@
+/* global RTCPeerConnection, MediaStream */
+// jshint unused:false
+
+var cordova = window.cordova;
+
+// Expose WebRTC Globals
+if (cordova && cordova.plugins && cordova.plugins.iosrtc) {
+	cordova.plugins.iosrtc.registerGlobals();
+	//cordova.plugins.iosrtc.debug.disable('*', true);
+	//cordova.plugins.iosrtc.debug.enable('*', true);
+	cordova.plugins.iosrtc.initAudioDevices();
+	cordova.plugins.iosrtc.turnOnSpeaker(true);
+	cordova.plugins.iosrtc.requestPermission(true, true, function (result) {
+		console.log('requestPermission.result', result);
+	});
+}
+
+//
+// Container
+//
+
+document.body.innerHTML = '';
+var appContainer = document.body;
+
+//
+// getUserMedia
+//
+
+var localStream;
+var localVideoEl;
+function TestGetUserMedia() {
+	// Note: Support for Multiple TestRTCPeerConnection calls
+	if (localStream) {
+		// Close local Stream if already connected
+		localStream.stop();
+
+		// Clear local video element
+		if (localVideoEl.srcObject) {
+			localVideoEl.srcObject = null;
+			if (localVideoEl.parentNode === appContainer) {
+				appContainer.removeChild(localVideoEl);
+			}
+		}
+	}
+
+	localVideoEl = document.createElement('video');
+	localVideoEl.setAttribute('autoplay', 'autoplay');
+	localVideoEl.setAttribute('playsinline', 'playsinline');
+
+	// Note: Test CSS positioning
+	localVideoEl.style.backgroundColor = 'purple'; // Cause zIndex - 1 failure
+	localVideoEl.style.position = 'absolute';
+	localVideoEl.style.top = 0;
+	localVideoEl.style.left = 0;
+	localVideoEl.style.width = '100px';
+	localVideoEl.style.height = '100px';
+	localVideoEl.style.transform = 'scaleX(-1)';
+	appContainer.appendChild(localVideoEl);
+
+	navigator.mediaDevices.enumerateDevices().then(
+		function (devices) {
+			console.log('getMediaDevices.ok', devices);
+			devices.forEach(function (device, idx) {
+				console.log(
+					'getMediaDevices.devices',
+					idx,
+					device.label,
+					device.kind,
+					device.deviceId
+				);
+			});
+		},
+		function (err) {
+			console.log('getMediaDevices.err', err);
+		}
+	);
+
+	return navigator.mediaDevices
+		.getUserMedia({
+			video: true,
+			audio: true
+			/*
+    video: {
+      // Test Back Camera
+      //deviceId: 'com.apple.avfoundation.avcapturedevice.built-in_video:0'
+      //sourceId: 'com.apple.avfoundation.avcapturedevice.built-in_video:0'
+      deviceId: {
+        exact: 'com.apple.avfoundation.avcapturedevice.built-in_video:0'
+      }
+    },
+    audio: {
+      exact: 'Built-In Microphone'
+    }*/
+		})
+		.then(function (stream) {
+			console.log('getUserMedia.stream', stream);
+			console.log('getUserMedia.stream.getTracks', stream.getTracks());
+
+			// Note: Expose for debug
+			localStream = stream;
+
+			// Attach local stream to video element
+			localVideoEl.srcObject = localStream;
+
+			// Test HTML over Video element
+			TestPluginMediaStreamRenderer(localVideoEl);
+
+			return localStream;
+		})
+		.catch(function (err) {
+			console.log('getUserMediaError', err, err.stack);
+		});
+}
+
+var useAnimateVideo = false;
+function TestPluginMediaStreamRenderer(localVideoEl) {
+	// Animate video position
+	var currentPosition = {
+		x: 0,
+		y: 0
+	};
+
+	var animateTimer;
+	function animateVideo() {
+		currentPosition.x =
+			currentPosition.x < window.innerWidth - parseInt(localVideoEl.style.width, 10)
+				? currentPosition.x + 1
+				: 0;
+		currentPosition.y =
+			currentPosition.y < window.innerHeight - parseInt(localVideoEl.style.height, 10)
+				? currentPosition.y + 1
+				: 0;
+
+		localVideoEl.style.top = currentPosition.y + 'px';
+		localVideoEl.style.left = currentPosition.x + 'px';
+
+		if (cordova && cordova.plugins && cordova.plugins.iosrtc) {
+			//cordova.plugins.iosrtc.refreshVideos();
+		}
+
+		return (animateTimer = requestAnimationFrame(animateVideo));
+	}
+
+	if (useAnimateVideo) {
+		animateTimer = animateVideo();
+	}
+
+	//
+	// Test Video behind Element
+	//
+
+	// Note: Test HTML over video requiring Z-Index -1 and transparent <html> and <body>
+	document.body.style.background = 'transparent';
+	document.documentElement.style.background = 'transparent';
+	localVideoEl.style.backgroundColor = '';
+	localVideoEl.style.zIndex = -1;
+
+	var overEl = document.createElement('button');
+	overEl.style.backgroundColor = 'red';
+	overEl.style.position = 'absolute';
+	overEl.style.left = 0;
+	overEl.style.top = 0;
+	overEl.style.width = '100px';
+	overEl.style.height = '100px';
+
+	overEl.addEventListener('click', function () {
+		overEl.style.backgroundColor = overEl.style.backgroundColor === 'red' ? 'green' : 'red';
+
+		if (overEl.style.backgroundColor === 'red') {
+			animateTimer = animateVideo();
+		} else {
+			cancelAnimationFrame(animateTimer);
+		}
+	});
+
+	appContainer.appendChild(overEl);
+
+	overEl.style.left = window.innerWidth / 2 - parseInt(overEl.style.width, 10) + 'px';
+	overEl.style.top = window.innerHeight / 2 - parseInt(overEl.style.height, 10) + 'px';
+}
+
+//
+// Test RTCPeerConnection
+//
+
+var peerConnectionConfig = {
+	offerToReceiveVideo: true,
+	//iceTransportPolicy: 'relay',
+	sdpSemantics: 'unified-plan',
+	offerToReceiveAudio: true,
+	//sdpSemantics: 'plan-b',
+	//bundlePolicy: 'max-compat',
+	//rtcpMuxPolicy: 'negotiate',
+	iceServers: [
+		{
+			url: 'stun:stun.stunprotocol.org'
+		}
+	]
+};
+
+// This plugin handle 'addstream' and 'track' event for MediaStream creation.
+var useTrackEvent = Object.getOwnPropertyDescriptors(RTCPeerConnection.prototype).ontrack;
+
+var pc1 = new RTCPeerConnection(peerConnectionConfig),
+	pc2 = new RTCPeerConnection(peerConnectionConfig);
+
+var sendChannel, receiveChannel;
+function TestPeerDataChannel() {
+	sendChannel = pc1.createDataChannel('sendChannel');
+
+	sendChannel.addEventListener('message', function (event) {
+		console.log('sendChannel.message', event, event.data);
+	});
+
+	sendChannel.addEventListener('open', function (event) {
+		console.log('sendChannel.open', event);
+		sendChannel.send('data message Date: ' + new Date());
+	});
+
+	sendChannel.addEventListener('close', function (event) {
+		console.log('sendChannel.close', event);
+	});
+
+	pc2.addEventListener('datachannel', function (event) {
+		receiveChannel = event.channel;
+		receiveChannel.addEventListener('message', function (event) {
+			console.log('receiveChannel.message', event, event.data);
+		});
+		receiveChannel.addEventListener('open', function (event) {
+			console.log('receiveChannel.open', event);
+		});
+		receiveChannel.addEventListener('close', function (event) {
+			console.log('receiveChannel.close', event);
+		});
+	});
+}
+
+var peerVideoEl;
+var peerStream;
+function TestRTCPeerConnection(localStream) {
+	// Note: Support for Multiple TestRTCPeerConnection calls
+	if (peerStream) {
+		// Close peer Stream if already connected
+		peerStream.stop();
+
+		// Clear peer video element
+		if (peerVideoEl.srcObject) {
+			peerVideoEl.srcObject = null;
+			appContainer.removeChild(peerVideoEl);
+		}
+
+		// Disconnect peer
+		if (pc1.iceConnectionState === 'completed') {
+			pc1.close();
+
+			// Current you cannot reuse previous RTCPeerConnection
+			pc1 = new RTCPeerConnection();
+			pc2 = new RTCPeerConnection();
+		}
+	}
+
+	if (useTrackEvent) {
+		// Add local stream tracks to RTCPeerConnection
+		var localPeerStream = new MediaStream();
+		localStream.getTracks().forEach(function (track) {
+			console.log('pc1.addTrack', track, localPeerStream);
+			pc1.addTrack(track, localPeerStream);
+		});
+	} else {
+		// Note: Deprecated but supported
+		pc1.addStream(localStream);
+
+		// Note: Deprecated Test removeStream
+		// pc1.removeStream(pc1.getLocalStreams()[0]);<
+	}
+
+	function onAddIceCandidate(pc, can) {
+		console.log('addIceCandidate', pc, can);
+		return (
+			can &&
+			pc.addIceCandidate(can).catch(function (err) {
+				console.log('addIceCandidateError', err);
+			})
+		);
+	}
+
+	pc1.addEventListener('icecandidate', function (e) {
+		onAddIceCandidate(pc2, e.candidate);
+	});
+
+	pc2.addEventListener('icecandidate', function (e) {
+		onAddIceCandidate(pc1, e.candidate);
+	});
+
+	pc2.addEventListener('track', function (e) {
+		console.log('pc2.track', e);
+	});
+
+	pc2.addEventListener('removestream', function (e) {
+		console.log('pc2.removeStream', e);
+	});
+
+	function setPeerVideoStream(stream) {
+		peerVideoEl = document.createElement('video');
+		peerVideoEl.setAttribute('autoplay', 'autoplay');
+		peerVideoEl.setAttribute('playsinline', 'playsinline');
+
+		// Note: Test Object-fix
+		peerVideoEl.style.objectFit = 'cover';
+
+		// Note: Test CSS positioning
+		peerVideoEl.style.backgroundColor = 'blue';
+		peerVideoEl.style.position = 'fixed';
+		peerVideoEl.style.width = '100px';
+		peerVideoEl.style.height = '100px';
+		peerVideoEl.style.top = 0;
+		peerVideoEl.style.left = window.innerWidth - parseInt(peerVideoEl.style.width, 10) + 'px';
+		appContainer.appendChild(peerVideoEl);
+
+		// Note: Expose for debug
+		peerStream = stream;
+
+		// Attach peer stream to video element
+		peerVideoEl.srcObject = peerStream;
+	}
+
+	if (useTrackEvent) {
+		var newPeerStream;
+		pc2.addEventListener('track', function (e) {
+			console.log('pc2.track', e);
+			newPeerStream = e.streams[0] || newPeerStream || new MediaStream();
+			setPeerVideoStream(newPeerStream);
+			newPeerStream.addTrack(e.track);
+		});
+	} else {
+		// Note: Deprecated but supported
+		pc2.addEventListener('addstream', function (e) {
+			console.log('pc2.addStream', e);
+			setPeerVideoStream(e.stream);
+		});
+	}
+
+	pc1.addEventListener('iceconnectionstatechange', function (e) {
+		console.log('pc1.iceConnectionState', e, pc1.iceConnectionState);
+
+		if (pc1.iceConnectionState === 'completed') {
+			console.log('pc1.getSenders', pc1.getSenders());
+			console.log('pc2.getReceivers', pc2.getReceivers());
+		}
+	});
+
+	pc1.addEventListener('icegatheringstatechange', function (e) {
+		console.log('pc1.iceGatheringStateChange', e);
+	});
+
+	// https://stackoverflow.com/questions/48963787/failed-to-set-local-answer-sdp-called-in-wrong-state-kstable
+	// https://bugs.chromium.org/p/chromium/issues/detail?id=740501
+	var isNegotiating = false;
+	pc1.addEventListener('signalingstatechange', function (e) {
+		console.log('pc1.signalingstatechange', e);
+		isNegotiating = pc1.signalingState !== 'stable';
+	});
+
+	pc1.addEventListener('negotiationneeded', function (e) {
+		if (isNegotiating) {
+			// Should not trigger on iosrtc cause of PluginRTCPeerConnection.swift fix
+			console.log('pc1.negotiatioNeeded', 'SKIP nested negotiations');
+			return;
+		}
+		isNegotiating = true;
+
+		console.log('pc1.negotiatioNeeded', e);
+
+		return pc1
+			.createOffer()
+			.then(function (d) {
+				var desc = {
+					type: d.type,
+					sdp: d.sdp
+				};
+				console.log('pc1.setLocalDescription', desc);
+				return pc1.setLocalDescription(desc);
+			})
+			.then(function () {
+				var desc = {
+					type: pc1.localDescription.type,
+					sdp: pc1.localDescription.sdp
+				};
+				console.log('pc2.setLocalDescription', desc);
+				return pc2.setRemoteDescription(desc);
+			})
+			.then(function () {
+				console.log('pc2.createAnswer');
+				return pc2.createAnswer();
+			})
+			.then(function (d) {
+				var desc = {
+					type: d.type,
+					sdp: d.sdp
+				};
+				console.log('pc2.setLocalDescription', desc);
+				return pc2.setLocalDescription(d);
+			})
+			.then(function () {
+				var desc = {
+					type: pc2.localDescription.type,
+					sdp: pc2.localDescription.sdp
+				};
+				console.log('pc1.setRemoteDescription', desc);
+				return pc1.setRemoteDescription(desc);
+			})
+			.then(function () {
+				TestMediaRenderCatpure(peerVideoEl);
+				TestMediaRenderCatpure(localVideoEl);
+				TestPeerDataChannel();
+			})
+			.catch(function (err) {
+				console.log('pc1.createOfferError', err);
+			});
+	});
+}
+
+var canvasEl;
+function TestMediaRenderCatpure(videoEl) {
+	if (!canvasEl) {
+		canvasEl = document.createElement('canvas');
+	}
+
+	var ctx = canvasEl.getContext('2d');
+	canvasEl.style.position = 'absolute';
+	canvasEl.style.left = 0;
+	canvasEl.style.bottom = 0;
+	appContainer.appendChild(canvasEl);
+	var image = new Image();
+	image.addEventListener('load', function () {
+		canvasEl.width = image.width;
+		canvasEl.height = image.height;
+		ctx.drawImage(image, 0, 0);
+	});
+	localVideoEl.render.save(function (data) {
+		image.src = 'data:image/jpg;base64,' + data;
+	});
+}
+
+var testMediaRenderCatpureAnimateFrame;
+function TestMediaRenderCatpureAnimate() {
+	TestMediaRenderCatpure(peerVideoEl);
+	testMediaRenderCatpureAnimateFrame = requestAnimationFrame(TestMediaRenderCatpureAnimate);
+}
+
+// Disabled to avoid confusion with remoteStream
+var cloneStream;
+var cloneVideoEl;
+function TestPluginMediaStreamClone(mediaStream) {
+	cloneVideoEl = document.createElement('video');
+	cloneVideoEl.setAttribute('autoplay', 'autoplay');
+	cloneVideoEl.setAttribute('playsinline', 'playsinline');
+	cloneVideoEl.style.backgroundColor = 'purple';
+	cloneVideoEl.style.position = 'absolute';
+	cloneVideoEl.style.bottom = 0;
+	cloneVideoEl.style.left = 0;
+	cloneVideoEl.style.width = '100px';
+	cloneVideoEl.style.height = '100px';
+	cloneVideoEl.style.transform = 'scaleX(-1)';
+
+	cloneStream = mediaStream.clone();
+	cloneVideoEl.srcObject = cloneStream;
+
+	appContainer.appendChild(cloneVideoEl);
+}
+
+var useWebRTCAdapter = false;
+
+// Expose webrtc-adapter
+if (useWebRTCAdapter && typeof window.adapter === 'undefined') {
+	// load adapter.js
+	var version = 'latest';
+	var script = document.createElement('script');
+	script.type = 'text/javascript';
+	//script.src = "adapter-latest.js";
+	script.src = 'https://webrtc.github.io/adapter/adapter-' + version + '.js';
+	script.async = false;
+	document.getElementsByTagName('head')[0].appendChild(script);
+	script.addEventListener('load', function () {
+		console.log('useWebRTCAdapter.loaded', script.src);
+		TestGetUserMedia().then(function (localStream) {
+			TestRTCPeerConnection(localStream);
+		});
+	});
+} else {
+	TestGetUserMedia().then(function (localStream) {
+		TestRTCPeerConnection(localStream);
+	});
+}
diff --git a/extra/webrtc/build-webrtc.py b/extra/webrtc/build-webrtc.py
new file mode 100644
index 00000000..4e23b603
--- /dev/null
+++ b/extra/webrtc/build-webrtc.py
@@ -0,0 +1,242 @@
+from __future__ import print_function
+
+import argparse
+import errno
+import os
+import shutil
+import subprocess
+import sys
+
+
+# Constants
+
+ANDROID_CPU_ABI_MAP = {
+    'arm'   : 'armeabi-v7a',
+    'arm64' : 'arm64-v8a',
+    'x86'   : 'x86',
+    'x64'   : 'x86_64'
+}
+ANDROID_BUILD_CPUS = ['arm', 'arm64', 'x86', 'x64']
+IOS_BUILD_ARCHS = ['arm64', 'arm','x64','x86']
+
+def build_gn_args(platform_args):
+    return "--args='" + ' '.join(GN_COMMON_ARGS + platform_args) + "'"
+
+GN_COMMON_ARGS = [
+    'treat_warnings_as_errors=false',
+    'is_component_build=false',
+    'rtc_libvpx_build_vp9=true',
+    'is_debug=%s',
+    'target_cpu="%s"'
+]
+
+_GN_IOS_ARGS = [
+    'enable_dsyms=true',
+    'enable_ios_bitcode=%s',
+    'ios_deployment_target="10.0"',
+    'ios_enable_code_signing=false',
+    'target_os="ios"',
+    'use_xcode_clang=true'
+]
+GN_IOS_ARGS = build_gn_args(_GN_IOS_ARGS)
+
+_GN_ANDROID_ARGS = [
+    'target_os="android"'
+]
+GN_ANDROID_ARGS = build_gn_args(_GN_ANDROID_ARGS)
+
+
+# Utilities
+
+def sh(cmd, env=None):
+    print('Running cmd: %s' % cmd)
+    try:
+        subprocess.check_call(cmd, env=env, shell=True, stdin=sys.stdin, stdout=sys.stdout, stderr=subprocess.STDOUT)
+    except subprocess.CalledProcessError as e:
+        sys.exit(e.returncode)
+    except KeyboardInterrupt:
+        pass
+
+def mkdirp(path):
+    try:
+        os.makedirs(path)
+    except OSError as e:
+        if e.errno != errno.EEXIST:
+            raise
+
+def rmr(path):
+    try:
+        shutil.rmtree(path)
+    except OSError as e:
+        if e.errno != errno.ENOENT:
+            raise
+
+
+# The Real Deal
+
+def setup(target_dir, platform):
+    mkdirp(target_dir)
+    os.chdir(target_dir)
+
+    # Maybe fetch depot_tools
+    depot_tools_dir = os.path.join(target_dir, 'depot_tools')
+    if not os.path.isdir(depot_tools_dir):
+        print('Fetching Chromium depot_tools...')
+        sh('git clone https://chromium.googlesource.com/chromium/tools/depot_tools.git')
+
+    # Prepare environment
+    env = os.environ.copy()
+    env['PATH'] = '%s:%s' % (env['PATH'], depot_tools_dir)
+
+    # Maybe fetch WebRTC
+    webrtc_dir = os.path.join(target_dir, 'webrtc', platform)
+    if not os.path.isdir(webrtc_dir):
+        mkdirp(webrtc_dir)
+        os.chdir(webrtc_dir)
+        print('Fetching WebRTC for %s...' % platform)
+        sh('fetch --nohooks webrtc_%s' % platform, env)
+
+    # Run glient
+    sh('gclient sync', env)
+
+    # Install dependencies
+    if platform == 'android':
+        webrtc_dir = os.path.join(target_dir, 'webrtc', platform, 'src')
+        os.chdir(webrtc_dir)
+        sh('./build/install-build-deps.sh')
+
+
+def build(target_dir, platform, debug, bitcode):
+    build_dir = os.path.join(target_dir, 'build', platform)
+    build_type = 'Debug' if debug else 'Release'
+    depot_tools_dir = os.path.join(target_dir, 'depot_tools')
+    webrtc_dir = os.path.join(target_dir, 'webrtc', platform, 'src')
+
+    if not os.path.isdir(webrtc_dir):
+        print('WebRTC source not found, did you forget to run --setup?')
+        sys.exit(1)
+
+    # Prepare environment
+    env = os.environ.copy()
+    path_parts = [env['PATH'], depot_tools_dir]
+    if platform == 'android':
+        # Same as . build/android/envsetup.sh
+        android_sdk_root = os.path.join(webrtc_dir, 'third_party/android_tools/sdk')
+        path_parts.append(os.path.join(android_sdk_root, 'platform-tools'))
+        path_parts.append(os.path.join(android_sdk_root, 'tools'))
+        path_parts.append(os.path.join(webrtc_dir, 'build/android'))
+    env['PATH'] = ':'.join(path_parts)
+
+    os.chdir(webrtc_dir)
+
+    # Run glient
+    sh('gclient sync', env)
+
+    # Cleanup old build
+    rmr('out')
+
+    # Run GN
+    if platform == 'ios':
+        for arch in IOS_BUILD_ARCHS:
+            gn_out_dir = 'out/%s-%s' % (build_type, arch)
+            gn_args = GN_IOS_ARGS % (str(debug).lower(), arch, str(bitcode).lower())
+            gn_cmd = 'gn gen %s %s' % (gn_out_dir, gn_args)
+            sh(gn_cmd, env)
+    else:
+        for cpu in ANDROID_BUILD_CPUS:
+            gn_out_dir = 'out/%s-%s' % (build_type, cpu)
+            gn_args = GN_ANDROID_ARGS % (str(debug).lower(), cpu)
+            gn_cmd = 'gn gen %s %s' % (gn_out_dir, gn_args)
+            sh(gn_cmd, env)
+
+    # Build with Ninja
+    if platform == 'ios':
+        for arch in IOS_BUILD_ARCHS:
+            gn_out_dir = 'out/%s-%s' % (build_type, arch)
+            ninja_cmd = 'ninja -C %s framework_objc' % gn_out_dir
+            sh(ninja_cmd, env)
+    else:
+        for cpu in ANDROID_BUILD_CPUS:
+            gn_out_dir = 'out/%s-%s' % (build_type, cpu)
+            ninja_cmd = 'ninja -C %s libwebrtc libjingle_peerconnection_so' % gn_out_dir
+            sh(ninja_cmd, env)
+
+    # Cleanup build dir
+    rmr(build_dir)
+    mkdirp(build_dir)
+
+    # Copy build artifacts to build directory
+    if platform == 'ios':
+        # Framework
+        gn_out_dir = 'out/%s-%s' % (build_type, IOS_BUILD_ARCHS[0])
+        shutil.copytree(os.path.join(gn_out_dir, 'WebRTC.framework'), os.path.join(build_dir, 'WebRTC.framework'))
+        out_lib_path = os.path.join(build_dir, 'WebRTC.framework', 'WebRTC')
+        os.unlink(out_lib_path)
+        slice_paths = [os.path.join('out/%s-%s' % (build_type, arch), 'WebRTC.framework', 'WebRTC') for arch in IOS_BUILD_ARCHS]
+        sh('lipo %s -create -output %s' % (' '.join(slice_paths), out_lib_path))
+        # dSYM
+        shutil.copytree(os.path.join(gn_out_dir, 'WebRTC.dSYM'), os.path.join(build_dir, 'WebRTC.dSYM'))
+        out_dsym_path = os.path.join(build_dir, 'WebRTC.dSYM', 'Contents', 'Resources', 'DWARF', 'WebRTC')
+        os.unlink(out_dsym_path)
+        dsym_slice_paths = [os.path.join('out/%s-%s' % (build_type, arch), 'WebRTC.dSYM', 'Contents', 'Resources', 'DWARF', 'WebRTC') for arch in IOS_BUILD_ARCHS]
+        sh('lipo %s -create -output %s' % (' '.join(dsym_slice_paths), out_dsym_path))
+    else:
+        gn_out_dir = 'out/%s-%s' % (build_type, ANDROID_BUILD_CPUS[0])
+        shutil.copy(os.path.join(gn_out_dir, 'lib.java/sdk/android/libwebrtc.jar'), build_dir)
+
+        for cpu in ANDROID_BUILD_CPUS:
+            lib_dir = os.path.join(build_dir, 'lib', ANDROID_CPU_ABI_MAP[cpu])
+            mkdirp(lib_dir)
+            gn_out_dir = 'out/%s-%s' % (build_type, cpu)
+            shutil.copy(os.path.join(gn_out_dir, 'libjingle_peerconnection_so.so'), lib_dir)
+
+        os.chdir(build_dir)
+        sh('jar cvfM libjingle_peerconnection.so.jar lib')
+        rmr('lib')
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument('dir', help='Target directory')
+    parser.add_argument('--setup', help='Prepare the target directory for building', action='store_true')
+    parser.add_argument('--build', help='Build WebRTC in the target directory', action='store_true')
+    parser.add_argument('--ios', help='Use iOS as the target platform', action='store_true')
+    parser.add_argument('--android', help='Use Android as the target platform', action='store_true')
+    parser.add_argument('--debug', help='Make a Debug build (defaults to false)', action='store_true')
+    parser.add_argument('--bitcode', help='Enable bitcode (defaults to false)', action='store_true')
+
+    args = parser.parse_args()
+
+    if not (args.setup or args.build):
+        print('--setup or --build must be specified!')
+        sys.exit(1)
+
+    if args.setup and args.build:
+        print('--setup and --build cannot be specified at the same time!')
+        sys.exit(1)
+
+    if not (args.ios or args.android):
+        print('--ios or --android must be specified!')
+        sys.exit(1)
+
+    if args.ios and args.android:
+        print('--ios and --android cannot be specified at the same time!')
+        sys.exit(1)
+
+    if not os.path.isdir(args.dir):
+        print('The specified directory does not exist!')
+        sys.exit(1)
+
+    target_dir = os.path.abspath(os.path.join(args.dir, 'build_webrtc'))
+    platform = 'ios' if args.ios else 'android'
+
+    if args.setup:
+        setup(target_dir, platform)
+        print('WebRTC setup for %s completed in %s' % (platform, target_dir))
+        sys.exit(0)
+
+    if args.build:
+        build(target_dir, platform, args.debug, args.bitcode)
+        print('WebRTC build for %s completed in %s' % (platform, target_dir))
+        sys.exit(0)
+
diff --git a/extra/webrtc/m69.patch b/extra/webrtc/m69.patch
new file mode 100644
index 00000000..e09622be
--- /dev/null
+++ b/extra/webrtc/m69.patch
@@ -0,0 +1,22 @@
+diff --git i/sdk/objc/Framework/Classes/PeerConnection/RTCCameraVideoCapturer.m w/sdk/objc/Framework/Classes/PeerConnection/RTCCameraVideoCapturer.m
+index 8f93937c75..e21cc76280 100644
+--- i/sdk/objc/Framework/Classes/PeerConnection/RTCCameraVideoCapturer.m
++++ w/sdk/objc/Framework/Classes/PeerConnection/RTCCameraVideoCapturer.m
+@@ -148,7 +148,7 @@ const int64_t kNanosecondsPerSecond = 1000000000;
+              completionHandler:(nullable void (^)(NSError *))completionHandler {
+   _willBeRunning = YES;
+   [RTCDispatcher
+-      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
++      dispatchAsyncOnType:RTCDispatcherTypeMain
+                     block:^{
+                       RTCLogInfo("startCaptureWithDevice %@ @ %ld fps", format, (long)fps);
+
+@@ -184,7 +184,7 @@ const int64_t kNanosecondsPerSecond = 1000000000;
+ - (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler {
+   _willBeRunning = NO;
+   [RTCDispatcher
+-      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
++      dispatchAsyncOnType:RTCDispatcherTypeMain
+                     block:^{
+                       RTCLogInfo("Stop");
+                       _currentDevice = nil;
diff --git a/gulpfile.js b/gulpfile.js
index 5cf0e00a..3e95ff74 100644
--- a/gulpfile.js
+++ b/gulpfile.js
@@ -2,8 +2,6 @@
  * Dependencies.
  */
 var gulp = require('gulp'),
-	jscs = require('gulp-jscs'),
-	stylish = require('gulp-jscs-stylish'),
 	browserify = require('browserify'),
 	vinyl_source_stream = require('vinyl-source-stream'),
 	vinyl_buffer = require('vinyl-buffer'),
@@ -11,45 +9,41 @@ var gulp = require('gulp'),
 	header = require('gulp-header'),
 	path = require('path'),
 	fs = require('fs'),
-
-/**
- * Constants.
- */
+	derequire = require('gulp-derequire'),
+	/**
+	 * Constants.
+	 */
 	PKG = require('./package.json'),
-
-/**
- * Banner.
- */
+	/**
+	 * Banner.
+	 */
 	banner = fs.readFileSync('banner.txt').toString(),
 	banner_options = {
 		pkg: PKG,
-		currentYear: (new Date()).getFullYear()
+		currentYear: new Date().getFullYear()
 	};
 
-
 gulp.task('lint', function () {
 	var src = ['gulpfile.js', 'js/**/*.js', 'hooks/**/*.js', 'extra/**/*.js'];
 
-	return gulp.src(src)
-		.pipe(jshint('.jshintrc'))  // Enforce good practics.
-		.pipe(jscs('.jscsrc'))  // Enforce style guide.
-		.pipe(stylish.combineWithHintResults())
-		.pipe(jshint.reporter('jshint-stylish', {verbose: true}))
+	return gulp
+		.src(src)
+		.pipe(jshint('.jshintrc')) // Enforce good practics.
+		.pipe(jshint.reporter('jshint-stylish', { verbose: true }))
 		.pipe(jshint.reporter('fail'));
 });
 
-
 gulp.task('browserify', function () {
 	return browserify([path.join(__dirname, PKG.main)], {
 		standalone: 'iosrtc'
 	})
-		.exclude('cordova/exec')  // Exclude require('cordova/exec').
+		.exclude('cordova/exec') // Exclude require('cordova/exec').
 		.bundle()
 		.pipe(vinyl_source_stream(PKG.name + '.js'))
 		.pipe(vinyl_buffer())
 		.pipe(header(banner, banner_options))
-		.pipe(gulp.dest('dist/'));
+		.pipe(derequire())
+		.pipe(gulp.dest('www/'));
 });
 
-
 gulp.task('default', gulp.series('lint', 'browserify'));
diff --git a/js/Errors.js b/js/Errors.js
index b2eb67c3..223eb5b2 100644
--- a/js/Errors.js
+++ b/js/Errors.js
@@ -1,18 +1,14 @@
 /**
  * Expose an object with WebRTC Errors.
  */
-var Errors = module.exports = {},
-
-
-/**
- * Local variables.
- */
+var Errors = (module.exports = {}),
+	/**
+	 * Local variables.
+	 */
 	IntermediateInheritor = function () {};
 
-
 IntermediateInheritor.prototype = Error.prototype;
 
-
 /**
  * Create error classes.
  */
@@ -21,7 +17,6 @@ addError('InvalidSessionDescriptionError');
 addError('InternalError');
 addError('MediaStreamError');
 
-
 function addError(name) {
 	Errors[name] = function () {
 		var tmp = Error.apply(this, arguments);
@@ -40,3 +35,13 @@ function addError(name) {
 
 	Errors[name].prototype = new IntermediateInheritor();
 }
+
+// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+// TODO remove on 6.0.0
+Errors.detectDeprecatedCallbaksUsage = function detectDeprecatedCallbaksUsage(funcName, arg) {
+	if (typeof arg[1] === 'function' || typeof arg[2] === 'function') {
+		throw new Error(
+			'Callbacks are not supported by "' + funcName + '" anymore, use Promise instead.'
+		);
+	}
+};
diff --git a/js/EventTarget.js b/js/EventTarget.js
new file mode 100644
index 00000000..089adbe6
--- /dev/null
+++ b/js/EventTarget.js
@@ -0,0 +1,30 @@
+/**
+ * Dependencies.
+ */
+var YaetiEventTarget = require('yaeti').EventTarget;
+
+var EventTarget = function () {
+	YaetiEventTarget.call(this);
+};
+
+EventTarget.prototype = Object.create(YaetiEventTarget.prototype);
+EventTarget.prototype.constructor = EventTarget;
+
+Object.defineProperties(
+	EventTarget.prototype,
+	Object.getOwnPropertyDescriptors(YaetiEventTarget.prototype)
+);
+
+EventTarget.prototype.dispatchEvent = function (event) {
+	Object.defineProperty(event, 'target', {
+		value: this,
+		writable: false
+	});
+
+	YaetiEventTarget.prototype.dispatchEvent.call(this, event);
+};
+
+/**
+ * Expose the EventTarget class.
+ */
+module.exports = EventTarget;
diff --git a/js/MediaDeviceInfo.js b/js/MediaDeviceInfo.js
index ac3544f2..398470a0 100644
--- a/js/MediaDeviceInfo.js
+++ b/js/MediaDeviceInfo.js
@@ -3,7 +3,6 @@
  */
 module.exports = MediaDeviceInfo;
 
-
 function MediaDeviceInfo(data) {
 	data = data || {};
 
diff --git a/js/MediaDevices.js b/js/MediaDevices.js
new file mode 100644
index 00000000..1af4e3d1
--- /dev/null
+++ b/js/MediaDevices.js
@@ -0,0 +1,78 @@
+/**
+ * Expose the MediaDevices class.
+ */
+module.exports = MediaDevices;
+
+/**
+ * Spec: https://w3c.github.io/mediacapture-main/#dom-mediadevices
+ */
+
+/**
+ * Dependencies.
+ */
+var EventTarget = require('./EventTarget'),
+	getUserMedia = require('./getUserMedia'),
+	enumerateDevices = require('./enumerateDevices');
+
+function MediaDevices(data) {
+	//ondevicechange
+	//enumerateDevices
+	//getDisplayMedia
+	//getSupportedConstraints
+	//getUserMedia
+
+	var self = this;
+
+	// Make this an EventTarget.
+	EventTarget.call(self);
+
+	data = data || {};
+}
+
+MediaDevices.prototype = Object.create(EventTarget.prototype);
+MediaDevices.prototype.constructor = MediaDevices;
+
+MediaDevices.prototype.getUserMedia = function (constraints) {
+	return getUserMedia(constraints);
+};
+
+MediaDevices.prototype.enumerateDevices = function () {
+	return enumerateDevices();
+};
+
+MediaDevices.prototype.getSupportedConstraints = function () {
+	return {
+		// Supported
+		height: true,
+		width: true,
+		deviceId: true,
+		frameRate: true,
+		sampleRate: true,
+		aspectRatio: true,
+		// Not Supported
+		autoGainControl: false,
+		brightness: false,
+		channelCount: false,
+		colorTemperature: false,
+		contrast: false,
+		echoCancellation: false,
+		exposureCompensation: false,
+		exposureMode: false,
+		exposureTime: false,
+		facingMode: true,
+		focusDistance: false,
+		focusMode: false,
+		groupId: false,
+		iso: false,
+		latency: false,
+		noiseSuppression: false,
+		pointsOfInterest: false,
+		resizeMode: false,
+		sampleSize: false,
+		saturation: false,
+		sharpness: false,
+		torch: false,
+		whiteBalanceMode: false,
+		zoom: false
+	};
+};
diff --git a/js/MediaStream.js b/js/MediaStream.js
index 402807b1..cb47b1f1 100644
--- a/js/MediaStream.js
+++ b/js/MediaStream.js
@@ -1,76 +1,187 @@
 /**
  * Expose the MediaStream class.
- * Make MediaStream be a Blob so it can be consumed by URL.createObjectURL().
  */
-var MediaStream = module.exports = window.Blob,
-
+module.exports = MediaStream;
 
 /**
  * Spec: http://w3c.github.io/mediacapture-main/#mediastream
  */
 
-
 /**
  * Dependencies.
  */
-	debug = require('debug')('iosrtc:MediaStream'),
+var debug = require('debug')('iosrtc:MediaStream'),
 	exec = require('cordova/exec'),
-	EventTarget = require('yaeti').EventTarget,
-	MediaStreamTrack = require('./MediaStreamTrack'),
+	EventTarget = require('./EventTarget'),
+	{ MediaStreamTrack } = require('./MediaStreamTrack'),
+	/**
+	 * Local variables.
+	 */
 
-
-/**
- * Local variables.
- */
-
-	// Dictionary of MediaStreams (provided via setMediaStreams() class method).
+	// Dictionary of MediaStreams (provided via getMediaStreams() class method).
 	// - key: MediaStream blobId.
 	// - value: MediaStream.
 	mediaStreams;
 
+// TODO longer UUID like native call
+// - "4021904575-2849079001-3048689102-1644344044-4021904575-2849079001-3048689102-1644344044"
+function newMediaStreamId() {
+	return window.crypto.getRandomValues(new Uint32Array(4)).join('-');
+}
+
+// Save original MediaStream
+var originalMediaStream = window.MediaStream || window.Blob;
+//var originalMediaStream = window.Blob;
+var originalMediaStreamTrack = MediaStreamTrack.originalMediaStreamTrack;
 
 /**
- * Class methods.
+ * Expose the MediaStream class.
  */
+function MediaStream(arg, id) {
+	debug('new MediaStream(arg) | [arg:%o]', arg);
+
+	// Detect native MediaStream usage
+	// new MediaStream(originalMediaStream) // stream
+	// new MediaStream(originalMediaStreamTrack[]) // tracks
+	if (
+		(!(arg instanceof window.Blob) &&
+			arg instanceof originalMediaStream &&
+			typeof arg.getBlobId === 'undefined') ||
+		(Array.isArray(arg) && arg[0] instanceof originalMediaStreamTrack)
+	) {
+		return new originalMediaStream(arg);
+	}
 
+	// new MediaStream(MediaStream) // stream
+	// new MediaStream(MediaStreamTrack[]) // tracks
+	// new MediaStream() // empty
 
-MediaStream.setMediaStreams = function (_mediaStreams) {
-	mediaStreams = _mediaStreams;
-};
-
-
-MediaStream.create = function (dataFromEvent) {
-	debug('create() | [dataFromEvent:%o]', dataFromEvent);
-
-	var stream,
-		blobId = 'MediaStream_' + dataFromEvent.id,
-		trackId,
-		track;
+	id = id || newMediaStreamId();
+	var blobId = 'MediaStream_' + id;
 
-	// Note that this is the Blob constructor.
-	stream = new MediaStream([blobId], {
-		type: 'stream'
-	});
+	// Extend returned MediaTream with custom MediaStream
+	var stream;
+	if (originalMediaStream !== window.Blob) {
+		stream = new (Function.prototype.bind.apply(originalMediaStream.bind(this), []))();
+	} else {
+		// Fallback on Blob if originalMediaStream is not a MediaStream and Emulate EventTarget
+		stream = new Blob([blobId], {
+			type: 'stream'
+		});
+
+		var target = document.createTextNode(null);
+		stream.addEventListener = target.addEventListener.bind(target);
+		stream.removeEventListener = target.removeEventListener.bind(target);
+		stream.dispatchEvent = target.dispatchEvent.bind(target);
+	}
 
-	// Store the stream into the dictionary.
-	mediaStreams[blobId] = stream;
+	Object.defineProperties(stream, Object.getOwnPropertyDescriptors(MediaStream.prototype));
 
 	// Make it an EventTarget.
 	EventTarget.call(stream);
 
 	// Public atributes.
-	stream.id = dataFromEvent.id;
-	stream.label = dataFromEvent.id;  // Backwards compatibility.
-	stream.active = true;
+	stream._id = id || newMediaStreamId();
+	stream._active = true;
+
+	// Init Stream by Id
+	exec(null, null, 'iosrtcPlugin', 'MediaStream_init', [stream.id]);
 
 	// Public but internal attributes.
 	stream.connected = false;
 
+	stream._addedToConnection = false;
+
 	// Private attributes.
-	stream._blobId = blobId;
 	stream._audioTracks = {};
 	stream._videoTracks = {};
 
+	// Store the stream into the dictionary.
+	stream._blobId = blobId;
+	mediaStreams[stream._blobId] = stream;
+
+	// Convert arg to array of tracks if possible
+	if (arg instanceof MediaStream || arg instanceof MediaStream.originalMediaStream) {
+		arg = arg.getTracks();
+	}
+
+	if (Array.isArray(arg)) {
+		arg.forEach(function (track) {
+			stream.addTrack(track);
+		});
+	} else if (typeof arg !== 'undefined') {
+		throw new TypeError(
+			"Failed to construct 'MediaStream': No matching constructor signature."
+		);
+	}
+
+	function onResultOK(data) {
+		onEvent.call(stream, data);
+	}
+	exec(onResultOK, null, 'iosrtcPlugin', 'MediaStream_setListener', [stream.id]);
+
+	return stream;
+}
+
+MediaStream.prototype = Object.create(originalMediaStream.prototype, {
+	id: {
+		get: function () {
+			return this._id;
+		}
+	},
+	active: {
+		get: function () {
+			return this._active;
+		}
+	},
+	// Backwards compatibility.
+	label: {
+		get: function () {
+			return this._id;
+		}
+	},
+	addedToConnection: {
+		get: function () {
+			return this._addedToConnection;
+		},
+		set: function (value) {
+			this._addedToConnection = value;
+		}
+	}
+});
+
+Object.defineProperties(
+	MediaStream.prototype,
+	Object.getOwnPropertyDescriptors(EventTarget.prototype)
+);
+
+MediaStream.prototype.constructor = MediaStream;
+
+// Static reference to original MediaStream
+MediaStream.originalMediaStream = originalMediaStream;
+
+/**
+ * Class methods.
+ */
+
+MediaStream.setMediaStreams = function (_mediaStreams) {
+	mediaStreams = _mediaStreams;
+};
+
+MediaStream.getMediaStreams = function () {
+	return mediaStreams;
+};
+
+MediaStream.create = function (dataFromEvent) {
+	debug('create() | [dataFromEvent:%o]', dataFromEvent);
+
+	var trackId,
+		track,
+		stream = new MediaStream([], dataFromEvent.id);
+
+	// We do not use addTrack to prevent false positive "ERROR: video track not added" and "ERROR: audio track not added"
+	// cause the rtcMediaStream already has them internaly.
+
 	for (trackId in dataFromEvent.audioTracks) {
 		if (dataFromEvent.audioTracks.hasOwnProperty(trackId)) {
 			track = new MediaStreamTrack(dataFromEvent.audioTracks[trackId]);
@@ -91,15 +202,12 @@ MediaStream.create = function (dataFromEvent) {
 		}
 	}
 
-	function onResultOK(data) {
-		onEvent.call(stream, data);
-	}
-
-	exec(onResultOK, null, 'iosrtcPlugin', 'MediaStream_setListener', [stream.id]);
-
 	return stream;
 };
 
+MediaStream.prototype.getBlobId = function () {
+	return this._blobId;
+};
 
 MediaStream.prototype.getAudioTracks = function () {
 	debug('getAudioTracks()');
@@ -116,7 +224,6 @@ MediaStream.prototype.getAudioTracks = function () {
 	return tracks;
 };
 
-
 MediaStream.prototype.getVideoTracks = function () {
 	debug('getVideoTracks()');
 
@@ -132,7 +239,6 @@ MediaStream.prototype.getVideoTracks = function () {
 	return tracks;
 };
 
-
 MediaStream.prototype.getTracks = function () {
 	debug('getTracks()');
 
@@ -154,14 +260,12 @@ MediaStream.prototype.getTracks = function () {
 	return tracks;
 };
 
-
 MediaStream.prototype.getTrackById = function (id) {
 	debug('getTrackById()');
 
 	return this._audioTracks[id] || this._videoTracks[id] || null;
 };
 
-
 MediaStream.prototype.addTrack = function (track) {
 	debug('addTrack() [track:%o]', track);
 
@@ -184,8 +288,11 @@ MediaStream.prototype.addTrack = function (track) {
 	addListenerForTrackEnded.call(this, track);
 
 	exec(null, null, 'iosrtcPlugin', 'MediaStream_addTrack', [this.id, track.id]);
-};
 
+	this.dispatchEvent(new Event('update'));
+
+	this.emitConnected();
+};
 
 MediaStream.prototype.removeTrack = function (track) {
 	debug('removeTrack() [track:%o]', track);
@@ -208,9 +315,19 @@ MediaStream.prototype.removeTrack = function (track) {
 
 	exec(null, null, 'iosrtcPlugin', 'MediaStream_removeTrack', [this.id, track.id]);
 
+	this.dispatchEvent(new Event('update'));
+
 	checkActive.call(this);
 };
 
+MediaStream.prototype.clone = function () {
+	var newStream = MediaStream();
+	this.getTracks().forEach(function (track) {
+		newStream.addTrack(track.clone());
+	});
+
+	return newStream;
+};
 
 // Backwards compatible API.
 MediaStream.prototype.stop = function () {
@@ -231,15 +348,12 @@ MediaStream.prototype.stop = function () {
 	}
 };
 
-
 // TODO: API methods and events.
 
-
 /**
  * Private API.
  */
 
-
 MediaStream.prototype.emitConnected = function () {
 	debug('emitConnected()');
 
@@ -250,12 +364,17 @@ MediaStream.prototype.emitConnected = function () {
 	}
 	this.connected = true;
 
-	setTimeout(function () {
-		self.dispatchEvent(new Event('connected'));
-	});
+	setTimeout(
+		function (self) {
+			var event = new Event('connected');
+			Object.defineProperty(event, 'target', { value: self, enumerable: true });
+			self.dispatchEvent(event);
+		},
+		0,
+		self
+	);
 };
 
-
 function addListenerForTrackEnded(track) {
 	var self = this;
 
@@ -270,7 +389,6 @@ function addListenerForTrackEnded(track) {
 	});
 }
 
-
 function checkActive() {
 	// A MediaStream object is said to be active when it has at least one MediaStreamTrack
 	// that has not ended. A MediaStream that does not have any tracks or only has tracks
@@ -282,8 +400,15 @@ function checkActive() {
 	if (!this.active) {
 		return;
 	}
+	// Fixes Twilio fails to read a local video if the stream is released.
+	if (this._addedToConnection) {
+		return;
+	}
 
-	if (Object.keys(this._audioTracks).length === 0 && Object.keys(this._videoTracks).length === 0) {
+	if (
+		Object.keys(this._audioTracks).length === 0 &&
+		Object.keys(this._videoTracks).length === 0
+	) {
 		debug('no tracks, releasing MediaStream');
 
 		release();
@@ -306,11 +431,11 @@ function checkActive() {
 		}
 	}
 
-	debug('all tracks are ended, releasing MediaStream');
 	release();
 
 	function release() {
-		self.active = false;
+		debug('all tracks are ended, releasing MediaStream %s', self.id);
+		self._active = false;
 		self.dispatchEvent(new Event('inactive'));
 
 		// Remove the stream from the dictionary.
@@ -320,7 +445,6 @@ function checkActive() {
 	}
 }
 
-
 function onEvent(data) {
 	var type = data.type,
 		event,
@@ -330,14 +454,22 @@ function onEvent(data) {
 
 	switch (type) {
 		case 'addtrack':
-			track = new MediaStreamTrack(data.track);
-
-			if (track.kind === 'audio') {
-				this._audioTracks[track.id] = track;
-			} else if (track.kind === 'video') {
-				this._videoTracks[track.id] = track;
+			// check if a track already exists before initializing a new
+			// track and calling setListener again.
+			if (data.track.kind === 'audio') {
+				track = this._audioTracks[data.track.id];
+			} else if (data.track.kind === 'video') {
+				track = this._videoTracks[data.track.id];
+			}
+			if (!track) {
+				track = new MediaStreamTrack(data.track);
+				if (track.kind === 'audio') {
+					this._audioTracks[track.id] = track;
+				} else if (track.kind === 'video') {
+					this._videoTracks[track.id] = track;
+				}
+				addListenerForTrackEnded.call(this, track);
 			}
-			addListenerForTrackEnded.call(this, track);
 
 			event = new Event('addtrack');
 			event.track = track;
@@ -358,13 +490,16 @@ function onEvent(data) {
 			}
 
 			if (!track) {
-				throw new Error('"removetrack" event fired on MediaStream for a non existing MediaStreamTrack');
+				throw new Error(
+					'"removetrack" event fired on MediaStream for a non existing MediaStreamTrack'
+				);
 			}
 
 			event = new Event('removetrack');
 			event.track = track;
 
 			this.dispatchEvent(event);
+
 			// Also emit 'update' for the MediaStreamRenderer.
 			this.dispatchEvent(new Event('update'));
 
diff --git a/js/MediaStreamRenderer.js b/js/MediaStreamRenderer.js
index 25646e51..39c30c5e 100644
--- a/js/MediaStreamRenderer.js
+++ b/js/MediaStreamRenderer.js
@@ -3,18 +3,15 @@
  */
 module.exports = MediaStreamRenderer;
 
-
 /**
  * Dependencies.
  */
-var
-	debug = require('debug')('iosrtc:MediaStreamRenderer'),
+var debug = require('debug')('iosrtc:MediaStreamRenderer'),
 	exec = require('cordova/exec'),
-	randomNumber = require('random-number').generator({min: 10000, max: 99999, integer: true}),
-	EventTarget = require('yaeti').EventTarget,
+	randomNumber = require('random-number').generator({ min: 10000, max: 99999, integer: true }),
+	EventTarget = require('./EventTarget'),
 	MediaStream = require('./MediaStream');
 
-
 function MediaStreamRenderer(element) {
 	debug('new() | [element:"%s"]', element);
 
@@ -42,22 +39,31 @@ function MediaStreamRenderer(element) {
 
 	exec(onResultOK, null, 'iosrtcPlugin', 'new_MediaStreamRenderer', [this.id]);
 
-	this.refresh(this);
+	this.refresh();
+
+	// TODO cause video resizing jiggling add semaphore
+	//this.refreshInterval = setInterval(function () {
+	//	self.refresh(self);
+	//}, 500);
+
+	element.render = this;
 }
 
+MediaStreamRenderer.prototype = Object.create(EventTarget.prototype);
+MediaStreamRenderer.prototype.constructor = MediaStreamRenderer;
 
 MediaStreamRenderer.prototype.render = function (stream) {
 	debug('render() [stream:%o]', stream);
 
 	var self = this;
 
-	if (!(stream instanceof MediaStream)) {
+	if (!(stream instanceof MediaStream.originalMediaStream)) {
 		throw new Error('render() requires a MediaStream instance as argument');
 	}
 
-	this.stream = stream;
+	self.stream = stream;
 
-	exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_render', [this.id, stream.id]);
+	exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_render', [self.id, stream.id]);
 
 	// Subscribe to 'update' event so we call native mediaStreamChanged() on it.
 	stream.addEventListener('update', function () {
@@ -84,8 +90,8 @@ MediaStreamRenderer.prototype.render = function (stream) {
 
 	if (stream.connected) {
 		connected();
-	// Otherwise subscribe to 'connected' event to emulate video elements events.
 	} else {
+		// Otherwise subscribe to 'connected' event to emulate video elements events.
 		stream.addEventListener('connected', function () {
 			if (self.stream !== stream) {
 				return;
@@ -104,6 +110,24 @@ MediaStreamRenderer.prototype.render = function (stream) {
 	}
 };
 
+MediaStreamRenderer.prototype.save = function (callback) {
+	debug('save()');
+
+	if (!this.stream) {
+		callback(null);
+		return;
+	}
+
+	function onResultOK(data) {
+		callback(data);
+	}
+
+	function onResultError() {
+		callback(null);
+	}
+
+	exec(onResultOK, onResultError, 'iosrtcPlugin', 'MediaStreamRenderer_save', [this.id]);
+};
 
 MediaStreamRenderer.prototype.refresh = function () {
 	debug('refresh()');
@@ -128,10 +152,23 @@ MediaStreamRenderer.prototype.refresh = function () {
 		paddingTop,
 		paddingBottom,
 		paddingLeft,
-		paddingRight;
+		paddingRight,
+		backgroundColorRgba,
+		self = this;
 
 	computedStyle = window.getComputedStyle(this.element);
 
+	// get background color
+	backgroundColorRgba = computedStyle.backgroundColor
+		.replace(/rgba?\((.*)\)/, '$1')
+		.split(',')
+		.map(function (x) {
+			return x.trim();
+		});
+	backgroundColorRgba[3] = '0';
+	this.element.style.backgroundColor = 'rgba(' + backgroundColorRgba.join(',') + ')';
+	backgroundColorRgba.length = 3;
+
 	// get padding values
 	paddingTop = parseInt(computedStyle.paddingTop) | 0;
 	paddingBottom = parseInt(computedStyle.paddingBottom) | 0;
@@ -143,8 +180,8 @@ MediaStreamRenderer.prototype.refresh = function () {
 	elementTop += paddingTop;
 
 	// fix width and height according to padding
-	elementWidth -= (paddingLeft + paddingRight);
-	elementHeight -= (paddingTop + paddingBottom);
+	elementWidth -= paddingLeft + paddingRight;
+	elementHeight -= paddingTop + paddingBottom;
 
 	videoViewWidth = elementWidth;
 	videoViewHeight = elementHeight;
@@ -153,7 +190,7 @@ MediaStreamRenderer.prototype.refresh = function () {
 	if (computedStyle.visibility === 'hidden') {
 		visible = false;
 	} else {
-		visible = !!this.element.offsetHeight;  // Returns 0 if element or any parent is hidden.
+		visible = !!this.element.offsetHeight; // Returns 0 if element or any parent is hidden.
 	}
 
 	// opacity
@@ -163,8 +200,10 @@ MediaStreamRenderer.prototype.refresh = function () {
 	zIndex = parseFloat(computedStyle.zIndex) || parseFloat(this.element.style.zIndex) || 0;
 
 	// mirrored (detect "-webkit-transform: scaleX(-1);" or equivalent)
-	if (computedStyle.transform === 'matrix(-1, 0, 0, 1, 0, 0)' ||
-		computedStyle['-webkit-transform'] === 'matrix(-1, 0, 0, 1, 0, 0)') {
+	if (
+		computedStyle.transform === 'matrix(-1, 0, 0, 1, 0, 0)' ||
+		computedStyle['-webkit-transform'] === 'matrix(-1, 0, 0, 1, 0, 0)'
+	) {
 		mirrored = true;
 	} else {
 		mirrored = false;
@@ -222,8 +261,8 @@ MediaStreamRenderer.prototype.refresh = function () {
 			if (elementRatio >= videoRatio) {
 				videoViewWidth = elementWidth;
 				videoViewHeight = videoViewWidth / videoRatio;
-			// The element has lower width/height ratio than the video.
 			} else if (elementRatio < videoRatio) {
+				// The element has lower width/height ratio than the video.
 				videoViewHeight = elementHeight;
 				videoViewWidth = videoViewHeight * videoRatio;
 			}
@@ -244,29 +283,29 @@ MediaStreamRenderer.prototype.refresh = function () {
 			if (this.videoWidth <= elementWidth && this.videoHeight <= elementHeight) {
 				videoViewHeight = this.videoHeight;
 				videoViewWidth = this.videoWidth;
-			// Same as 'contain'.
 			} else {
-				// The element has higher or equal width/height ratio than the video.
+				// Same as 'contain'.
 				if (elementRatio >= videoRatio) {
+					// The element has higher or equal width/height ratio than the video.
 					videoViewHeight = elementHeight;
 					videoViewWidth = videoViewHeight * videoRatio;
-				// The element has lower width/height ratio than the video.
 				} else if (elementRatio < videoRatio) {
+					// The element has lower width/height ratio than the video.
 					videoViewWidth = elementWidth;
 					videoViewHeight = videoViewWidth / videoRatio;
 				}
 			}
 			break;
 
-		// 'contain'.
 		default:
+			// 'contain'.
 			objectFit = 'contain';
-			// The element has higher or equal width/height ratio than the video.
 			if (elementRatio >= videoRatio) {
+				// The element has higher or equal width/height ratio than the video.
 				videoViewHeight = elementHeight;
 				videoViewWidth = videoViewHeight * videoRatio;
-			// The element has lower width/height ratio than the video.
 			} else if (elementRatio < videoRatio) {
+				// The element has lower width/height ratio than the video.
 				videoViewWidth = elementWidth;
 				videoViewHeight = videoViewWidth / videoRatio;
 			}
@@ -275,22 +314,41 @@ MediaStreamRenderer.prototype.refresh = function () {
 
 	nativeRefresh.call(this);
 
+	function hash(str) {
+		var hash = 5381,
+			i = str.length;
+
+		while (i) {
+			hash = (hash * 33) ^ str.charCodeAt(--i);
+		}
+
+		return hash >>> 0;
+	}
+
 	function nativeRefresh() {
 		var data = {
-			elementLeft: elementLeft,
-			elementTop: elementTop,
-			elementWidth: elementWidth,
-			elementHeight: elementHeight,
-			videoViewWidth: videoViewWidth,
-			videoViewHeight: videoViewHeight,
-			visible: visible,
-			opacity: opacity,
-			zIndex: zIndex,
-			mirrored: mirrored,
-			objectFit: objectFit,
-			clip: clip,
-			borderRadius: borderRadius
-		};
+				elementLeft: Math.round(elementLeft),
+				elementTop: Math.round(elementTop),
+				elementWidth: Math.round(elementWidth),
+				elementHeight: Math.round(elementHeight),
+				videoViewWidth: Math.round(videoViewWidth),
+				videoViewHeight: Math.round(videoViewHeight),
+				visible: visible,
+				backgroundColor: backgroundColorRgba.join(','),
+				opacity: opacity,
+				zIndex: zIndex,
+				mirrored: mirrored,
+				objectFit: objectFit,
+				clip: clip,
+				borderRadius: borderRadius
+			},
+			newRefreshCached = hash(JSON.stringify(data));
+
+		if (newRefreshCached === self.refreshCached) {
+			return;
+		}
+
+		self.refreshCached = newRefreshCached;
 
 		debug('refresh() | [data:%o]', data);
 
@@ -298,7 +356,6 @@ MediaStreamRenderer.prototype.refresh = function () {
 	}
 };
 
-
 MediaStreamRenderer.prototype.close = function () {
 	debug('close()');
 
@@ -308,14 +365,16 @@ MediaStreamRenderer.prototype.close = function () {
 	this.stream = undefined;
 
 	exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_close', [this.id]);
+	if (this.refreshInterval) {
+		clearInterval(this.refreshInterval);
+		delete this.refreshInterval;
+	}
 };
 
-
 /**
  * Private API.
  */
 
-
 function onEvent(data) {
 	var type = data.type,
 		event;
@@ -326,7 +385,7 @@ function onEvent(data) {
 		case 'videoresize':
 			this.videoWidth = data.size.width;
 			this.videoHeight = data.size.height;
-			this.refresh(this);
+			this.refresh();
 
 			event = new Event(type);
 			event.videoWidth = data.size.width;
@@ -337,14 +396,13 @@ function onEvent(data) {
 	}
 }
 
-
 function getElementPositionAndSize() {
 	var rect = this.element.getBoundingClientRect();
 
 	return {
-		left:   rect.left + this.element.clientLeft,
-		top:    rect.top + this.element.clientTop,
-		width:  this.element.clientWidth,
+		left: rect.left + this.element.clientLeft,
+		top: rect.top + this.element.clientTop,
+		width: this.element.clientWidth,
 		height: this.element.clientHeight
 	};
 }
diff --git a/js/MediaStreamTrack.js b/js/MediaStreamTrack.js
index 6f34752a..8adecaab 100644
--- a/js/MediaStreamTrack.js
+++ b/js/MediaStreamTrack.js
@@ -1,25 +1,38 @@
 /**
  * Expose the MediaStreamTrack class.
  */
-module.exports = MediaStreamTrack;
-
+module.exports.MediaStreamTrack = MediaStreamTrack;
+module.exports.newMediaStreamTrackId = newMediaStreamTrackId;
 
 /**
  * Spec: http://w3c.github.io/mediacapture-main/#mediastreamtrack
  */
 
-
 /**
  * Dependencies.
  */
-var
-	debug = require('debug')('iosrtc:MediaStreamTrack'),
+var debug = require('debug')('iosrtc:MediaStreamTrack'),
 	exec = require('cordova/exec'),
 	enumerateDevices = require('./enumerateDevices'),
-	EventTarget = require('yaeti').EventTarget;
+	MediaTrackCapabilities = require('./MediaTrackCapabilities'),
+	MediaTrackSettings = require('./MediaTrackSettings'),
+	EventTarget = require('./EventTarget');
 
+// Save original MediaStreamTrack
+var originalMediaStreamTrack = window.MediaStreamTrack || function dummyMediaStreamTrack() {};
+
+// All "active" tracks, to ensure RTCRtpSender gets the same object
+var activeMediaStreamTracks = new Map();
+
+function newMediaStreamTrackId() {
+	return window.crypto.getRandomValues(new Uint32Array(4)).join('-');
+}
 
 function MediaStreamTrack(dataFromEvent) {
+	if (!dataFromEvent) {
+		throw new Error('Illegal constructor');
+	}
+
 	debug('new() | [dataFromEvent:%o]', dataFromEvent);
 
 	var self = this;
@@ -28,16 +41,21 @@ function MediaStreamTrack(dataFromEvent) {
 	EventTarget.call(this);
 
 	// Public atributes.
-	this.id = dataFromEvent.id;  // NOTE: It's a string.
+	this.id = dataFromEvent.id; // NOTE: It's a string.
 	this.kind = dataFromEvent.kind;
 	this.label = dataFromEvent.label;
-	this.muted = false;  // TODO: No "muted" property in ObjC API.
+	this.muted = false; // TODO: No "muted" property in ObjC API.
+	this.capabilities = dataFromEvent.capabilities;
 	this.readyState = dataFromEvent.readyState;
 
 	// Private attributes.
 	this._enabled = dataFromEvent.enabled;
 	this._ended = false;
 
+	this.dataFromEvent = dataFromEvent;
+
+	activeMediaStreamTracks.set(this.id, this);
+
 	function onResultOK(data) {
 		onEvent.call(self, data);
 	}
@@ -45,6 +63,11 @@ function MediaStreamTrack(dataFromEvent) {
 	exec(onResultOK, null, 'iosrtcPlugin', 'MediaStreamTrack_setListener', [this.id]);
 }
 
+MediaStreamTrack.prototype = Object.create(EventTarget.prototype);
+MediaStreamTrack.prototype.constructor = MediaStreamTrack;
+
+// Static reference to original MediaStreamTrack
+MediaStreamTrack.originalMediaStreamTrack = originalMediaStreamTrack;
 
 // Setters.
 Object.defineProperty(MediaStreamTrack.prototype, 'enabled', {
@@ -59,6 +82,40 @@ Object.defineProperty(MediaStreamTrack.prototype, 'enabled', {
 	}
 });
 
+MediaStreamTrack.prototype.getConstraints = function () {
+	debug('MediaStreamTrack.prototype.getConstraints  is not implemented.');
+	return {};
+};
+
+MediaStreamTrack.prototype.applyConstraints = function (constraints) {
+	debug('MediaStreamTrack.prototype.applyConstraints  is not implemented.', constraints);
+	return Promise.reject(new Error('applyConstraints is not implemented.'));
+};
+
+MediaStreamTrack.prototype.clone = function () {
+	var newTrackId = newMediaStreamTrackId();
+
+	exec(null, null, 'iosrtcPlugin', 'MediaStreamTrack_clone', [this.id, newTrackId]);
+
+	return new MediaStreamTrack({
+		id: newTrackId,
+		kind: this.kind,
+		label: this.label,
+		readyState: this.readyState,
+		enabled: this.enabled,
+		trackId: this.dataFromEvent.trackId
+	});
+};
+
+MediaStreamTrack.prototype.getCapabilities = function () {
+	return new MediaTrackCapabilities(this.capabilities);
+};
+
+MediaStreamTrack.prototype.getSettings = function () {
+	//throw new Error('Not implemented.');
+	// SHAM
+	return new MediaTrackSettings();
+};
 
 MediaStreamTrack.prototype.stop = function () {
 	debug('stop()');
@@ -70,27 +127,29 @@ MediaStreamTrack.prototype.stop = function () {
 	exec(null, null, 'iosrtcPlugin', 'MediaStreamTrack_stop', [this.id]);
 };
 
-
 // TODO: API methods and events.
 
-
 /**
  * Class methods.
  */
 
-
 MediaStreamTrack.getSources = function () {
 	debug('getSources()');
 
 	return enumerateDevices.apply(this, arguments);
 };
 
+MediaStreamTrack.findOrCreate = function (trackData) {
+	var id = trackData.id;
+	debug('findOrCreate()', id);
+
+	return activeMediaStreamTracks.get(id) || new MediaStreamTrack(trackData);
+};
 
 /**
  * Private API.
  */
 
-
 function onEvent(data) {
 	var type = data.type;
 
@@ -107,6 +166,7 @@ function onEvent(data) {
 				case 'live':
 					break;
 				case 'ended':
+					activeMediaStreamTracks.delete(this.id);
 					this._ended = true;
 					this.dispatchEvent(new Event('ended'));
 					break;
diff --git a/js/MediaTrackCapabilities.js b/js/MediaTrackCapabilities.js
new file mode 100644
index 00000000..555093bd
--- /dev/null
+++ b/js/MediaTrackCapabilities.js
@@ -0,0 +1,11 @@
+/**
+ * Expose the MediaTrackSettings class.
+ */
+module.exports = MediaTrackCapabilities;
+
+// Ref https://www.w3.org/TR/mediacapture-streams/#dom-mediatrackcapabilities
+function MediaTrackCapabilities(data) {
+	data = data || {};
+
+	this.deviceId = data.deviceId;
+}
diff --git a/js/MediaTrackSettings.js b/js/MediaTrackSettings.js
new file mode 100644
index 00000000..931c8488
--- /dev/null
+++ b/js/MediaTrackSettings.js
@@ -0,0 +1,9 @@
+/**
+ * Expose the MediaTrackSettings class.
+ */
+module.exports = MediaTrackSettings;
+
+// Ref https://www.w3.org/TR/mediacapture-streams/#dom-mediatracksettings
+function MediaTrackSettings(data) {
+	data = data || {};
+}
diff --git a/js/RTCDTMFSender.js b/js/RTCDTMFSender.js
index 5810e35e..1bbd6fb0 100644
--- a/js/RTCDTMFSender.js
+++ b/js/RTCDTMFSender.js
@@ -3,21 +3,17 @@
  */
 module.exports = RTCDTMFSender;
 
-
 /**
  * Dependencies.
  */
-var
-	debug = require('debug')('iosrtc:RTCDTMFSender'),
+var debug = require('debug')('iosrtc:RTCDTMFSender'),
 	debugerror = require('debug')('iosrtc:ERROR:RTCDTMFSender'),
 	exec = require('cordova/exec'),
-	randomNumber = require('random-number').generator({min: 10000, max: 99999, integer: true}),
-	EventTarget = require('yaeti').EventTarget;
-
+	randomNumber = require('random-number').generator({ min: 10000, max: 99999, integer: true }),
+	EventTarget = require('./EventTarget');
 
 debugerror.log = console.warn.bind(console);
 
-
 function RTCDTMFSender(peerConnection, track) {
 	var self = this;
 
@@ -41,10 +37,15 @@ function RTCDTMFSender(peerConnection, track) {
 		onEvent.call(self, data);
 	}
 
-	exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_createDTMFSender', [this.peerConnection.pcId, this.dsId, this._track.id]);
-
+	exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_createDTMFSender', [
+		this.peerConnection.pcId,
+		this.dsId,
+		this._track.id
+	]);
 }
 
+RTCDTMFSender.prototype = Object.create(EventTarget.prototype);
+RTCDTMFSender.prototype.constructor = RTCDTMFSender;
 
 Object.defineProperty(RTCDTMFSender.prototype, 'canInsertDTMF', {
 	get: function () {
@@ -53,35 +54,30 @@ Object.defineProperty(RTCDTMFSender.prototype, 'canInsertDTMF', {
 	}
 });
 
-
 Object.defineProperty(RTCDTMFSender.prototype, 'track', {
 	get: function () {
 		return this._track;
 	}
 });
 
-
 Object.defineProperty(RTCDTMFSender.prototype, 'duration', {
 	get: function () {
 		return this._duration;
 	}
 });
 
-
 Object.defineProperty(RTCDTMFSender.prototype, 'interToneGap', {
 	get: function () {
 		return this._interToneGap;
 	}
 });
 
-
 Object.defineProperty(RTCDTMFSender.prototype, 'toneBuffer', {
 	get: function () {
 		return this._toneBuffer;
 	}
 });
 
-
 RTCDTMFSender.prototype.insertDTMF = function (tones, duration, interToneGap) {
 	if (isClosed.call(this)) {
 		return;
@@ -102,20 +98,23 @@ RTCDTMFSender.prototype.insertDTMF = function (tones, duration, interToneGap) {
 		onEvent.call(self, data);
 	}
 
-	exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDTMFSender_insertDTMF', [this.peerConnection.pcId, this.dsId, tones, this._duration, this._interToneGap]);
+	exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDTMFSender_insertDTMF', [
+		this.peerConnection.pcId,
+		this.dsId,
+		tones,
+		this._duration,
+		this._interToneGap
+	]);
 };
 
-
 /**
  * Private API.
  */
 
-
 function isClosed() {
 	return this.peerConnection.signalingState === 'closed';
 }
 
-
 function onEvent(data) {
 	var type = data.type,
 		event;
diff --git a/js/RTCDataChannel.js b/js/RTCDataChannel.js
index cbefc5b3..d1febc01 100644
--- a/js/RTCDataChannel.js
+++ b/js/RTCDataChannel.js
@@ -3,21 +3,17 @@
  */
 module.exports = RTCDataChannel;
 
-
 /**
  * Dependencies.
  */
-var
-	debug = require('debug')('iosrtc:RTCDataChannel'),
+var debug = require('debug')('iosrtc:RTCDataChannel'),
 	debugerror = require('debug')('iosrtc:ERROR:RTCDataChannel'),
 	exec = require('cordova/exec'),
-	randomNumber = require('random-number').generator({min: 10000, max: 99999, integer: true}),
-	EventTarget = require('yaeti').EventTarget;
-
+	randomNumber = require('random-number').generator({ min: 10000, max: 99999, integer: true }),
+	EventTarget = require('./EventTarget');
 
 debugerror.log = console.warn.bind(console);
 
-
 function RTCDataChannel(peerConnection, label, options, dataFromEvent) {
 	var self = this;
 
@@ -34,7 +30,10 @@ function RTCDataChannel(peerConnection, label, options, dataFromEvent) {
 
 		options = options || {};
 
-		if (options.hasOwnProperty('maxPacketLifeTime') && options.hasOwnProperty('maxRetransmits')) {
+		if (
+			options.hasOwnProperty('maxPacketLifeTime') &&
+			options.hasOwnProperty('maxRetransmits')
+		) {
 			throw new SyntaxError('both maxPacketLifeTime and maxRetransmits can not be present');
 		}
 
@@ -45,15 +44,21 @@ function RTCDataChannel(peerConnection, label, options, dataFromEvent) {
 			// TODO:
 			//   https://code.google.com/p/webrtc/issues/detail?id=4618
 			if (options.id > 1023) {
-				throw new SyntaxError('id cannot be greater than 1023 (https://code.google.com/p/webrtc/issues/detail?id=4614)');
+				throw new SyntaxError(
+					'id cannot be greater than 1023 (https://code.google.com/p/webrtc/issues/detail?id=4614)'
+				);
 			}
 		}
 
 		// Public atributes.
 		this.label = label;
 		this.ordered = options.hasOwnProperty('ordered') ? !!options.ordered : true;
-		this.maxPacketLifeTime = options.hasOwnProperty('maxPacketLifeTime') ? Number(options.maxPacketLifeTime) : null;
-		this.maxRetransmits = options.hasOwnProperty('maxRetransmits') ? Number(options.maxRetransmits) : null;
+		this.maxPacketLifeTime = options.hasOwnProperty('maxPacketLifeTime')
+			? Number(options.maxPacketLifeTime)
+			: null;
+		this.maxRetransmits = options.hasOwnProperty('maxRetransmits')
+			? Number(options.maxRetransmits)
+			: null;
 		this.protocol = options.hasOwnProperty('protocol') ? String(options.protocol) : '';
 		this.negotiated = options.hasOwnProperty('negotiated') ? !!options.negotiated : false;
 		this.id = options.hasOwnProperty('id') ? Number(options.id) : undefined;
@@ -65,9 +70,14 @@ function RTCDataChannel(peerConnection, label, options, dataFromEvent) {
 		this.peerConnection = peerConnection;
 		this.dcId = randomNumber();
 
-		exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_createDataChannel', [this.peerConnection.pcId, this.dcId, label, options]);
-	// Created via pc.ondatachannel.
+		exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_createDataChannel', [
+			this.peerConnection.pcId,
+			this.dcId,
+			label,
+			options
+		]);
 	} else {
+		// Created via pc.ondatachannel.
 		debug('new() | [dataFromEvent:%o]', dataFromEvent);
 
 		// Public atributes.
@@ -86,14 +96,17 @@ function RTCDataChannel(peerConnection, label, options, dataFromEvent) {
 		this.peerConnection = peerConnection;
 		this.dcId = dataFromEvent.dcId;
 
-		exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_setListener', [this.peerConnection.pcId, this.dcId]);
+		exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_setListener', [
+			this.peerConnection.pcId,
+			this.dcId
+		]);
 	}
 
 	function onResultOK(data) {
 		if (data.type) {
 			onEvent.call(self, data);
-		// Special handler for received binary mesage.
 		} else {
+			// Special handler for received binary message.
 			onEvent.call(self, {
 				type: 'message',
 				message: data
@@ -102,6 +115,8 @@ function RTCDataChannel(peerConnection, label, options, dataFromEvent) {
 	}
 }
 
+RTCDataChannel.prototype = Object.create(EventTarget.prototype);
+RTCDataChannel.prototype.constructor = RTCDataChannel;
 
 // Just 'arraybuffer' binaryType is implemented in Chromium.
 Object.defineProperty(RTCDataChannel.prototype, 'binaryType', {
@@ -115,7 +130,6 @@ Object.defineProperty(RTCDataChannel.prototype, 'binaryType', {
 	}
 });
 
-
 RTCDataChannel.prototype.send = function (data) {
 	if (isClosed.call(this) || this.readyState !== 'open') {
 		return;
@@ -127,29 +141,42 @@ RTCDataChannel.prototype.send = function (data) {
 		return;
 	}
 
-	if (typeof data === 'string' || data instanceof String) {
-		exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendString', [this.peerConnection.pcId, this.dcId, data]);
-	} else if (window.ArrayBuffer && data instanceof window.ArrayBuffer) {
-		exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendBinary', [this.peerConnection.pcId, this.dcId, data]);
-	} else if (
-		(window.Int8Array && data instanceof window.Int8Array) ||
-		(window.Uint8Array && data instanceof window.Uint8Array) ||
-		(window.Uint8ClampedArray && data instanceof window.Uint8ClampedArray) ||
-		(window.Int16Array && data instanceof window.Int16Array) ||
-		(window.Uint16Array && data instanceof window.Uint16Array) ||
-		(window.Int32Array && data instanceof window.Int32Array) ||
-		(window.Uint32Array && data instanceof window.Uint32Array) ||
-		(window.Float32Array && data instanceof window.Float32Array) ||
-		(window.Float64Array && data instanceof window.Float64Array) ||
-		(window.DataView && data instanceof window.DataView)
-	) {
-		exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendBinary', [this.peerConnection.pcId, this.dcId, data.buffer]);
-	} else {
-		throw new Error('invalid data type');
-	}
+	setImmediate(() => {
+		if (typeof data === 'string' || data instanceof String) {
+			exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendString', [
+				this.peerConnection.pcId,
+				this.dcId,
+				data
+			]);
+		} else if (window.ArrayBuffer && data instanceof window.ArrayBuffer) {
+			exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendBinary', [
+				this.peerConnection.pcId,
+				this.dcId,
+				data
+			]);
+		} else if (
+			(window.Int8Array && data instanceof window.Int8Array) ||
+			(window.Uint8Array && data instanceof window.Uint8Array) ||
+			(window.Uint8ClampedArray && data instanceof window.Uint8ClampedArray) ||
+			(window.Int16Array && data instanceof window.Int16Array) ||
+			(window.Uint16Array && data instanceof window.Uint16Array) ||
+			(window.Int32Array && data instanceof window.Int32Array) ||
+			(window.Uint32Array && data instanceof window.Uint32Array) ||
+			(window.Float32Array && data instanceof window.Float32Array) ||
+			(window.Float64Array && data instanceof window.Float64Array) ||
+			(window.DataView && data instanceof window.DataView)
+		) {
+			exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendBinary', [
+				this.peerConnection.pcId,
+				this.dcId,
+				data.buffer
+			]);
+		} else {
+			throw new Error('invalid data type');
+		}
+	});
 };
 
-
 RTCDataChannel.prototype.close = function () {
 	if (isClosed.call(this)) {
 		return;
@@ -159,20 +186,24 @@ RTCDataChannel.prototype.close = function () {
 
 	this.readyState = 'closing';
 
-	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_close', [this.peerConnection.pcId, this.dcId]);
+	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_close', [
+		this.peerConnection.pcId,
+		this.dcId
+	]);
 };
 
-
 /**
  * Private API.
  */
 
-
 function isClosed() {
-	return this.readyState === 'closed' || this.readyState === 'closing' || this.peerConnection.signalingState === 'closed';
+	return (
+		this.readyState === 'closed' ||
+		this.readyState === 'closing' ||
+		this.peerConnection.signalingState === 'closed'
+	);
 }
 
-
 function onEvent(data) {
 	var type = data.type,
 		event;
@@ -218,7 +249,10 @@ function onEvent(data) {
 		case 'bufferedamount':
 			this.bufferedAmount = data.bufferedAmount;
 
-			if (this.bufferedAmountLowThreshold > 0 && this.bufferedAmountLowThreshold > this.bufferedAmount) {
+			if (
+				this.bufferedAmountLowThreshold > 0 &&
+				this.bufferedAmountLowThreshold > this.bufferedAmount
+			) {
 				event = new Event('bufferedamountlow');
 				event.bufferedAmount = this.bufferedAmount;
 				this.dispatchEvent(event);
diff --git a/js/RTCIceCandidate.js b/js/RTCIceCandidate.js
index 4c47eba6..15e48e35 100644
--- a/js/RTCIceCandidate.js
+++ b/js/RTCIceCandidate.js
@@ -3,7 +3,161 @@
  */
 module.exports = RTCIceCandidate;
 
+/**
+* RFC-5245: http://tools.ietf.org/html/rfc5245#section-15.1
+*
+* candidate-attribute   = "candidate" ":" foundation SP component-id SP
+                           transport SP
+                           priority SP
+                           connection-address SP     ;from RFC 4566
+                           port         ;port from RFC 4566
+                           SP cand-type
+                           [SP rel-addr]
+                           [SP rel-port]
+                           *(SP extension-att-name SP
+                                extension-att-value)
+*
+* foundation            = 1*32ice-char
+* component-id          = 1*5DIGIT
+* transport             = "UDP" / transport-extension
+* transport-extension   = token              ; from RFC 3261
+* priority              = 1*10DIGIT
+* cand-type             = "typ" SP candidate-types
+* candidate-types       = "host" / "srflx" / "prflx" / "relay" / token
+* rel-addr              = "raddr" SP connection-address
+* rel-port              = "rport" SP port
+* extension-att-name    = byte-string    ;from RFC 4566
+* extension-att-value   = byte-string
+* ice-char              = ALPHA / DIGIT / "+" / "/"
+*/
+
+/**
+* RFC-3261: https://tools.ietf.org/html/rfc3261#section-25.1
+*
+* token          =  1*(alphanum / "-" / "." / "!" / "%" / "*"
+                     / "_" / "+" / "`" / "'" / "~" )
+*/
+
+/*
+* RFC-4566: https://tools.ietf.org/html/rfc4566#section-9
+*
+* port =                1*DIGIT
+* IP4-address =         b1 3("." decimal-uchar)
+* b1 =                  decimal-uchar
+                         ; less than "224"
+* ; The following is consistent with RFC 2373 [30], Appendix B.
+* IP6-address =         hexpart [ ":" IP4-address ]
+* hexpart =             hexseq / hexseq "::" [ hexseq ] /
+                         "::" [ hexseq ]
+* hexseq  =             hex4 *( ":" hex4)
+* hex4    =             1*4HEXDIG
+* decimal-uchar =       DIGIT
+                         / POS-DIGIT DIGIT
+                         / ("1" 2*(DIGIT))
+                         / ("2" ("0"/"1"/"2"/"3"/"4") DIGIT)
+                         / ("2" "5" ("0"/"1"/"2"/"3"/"4"/"5"))
+*/
+
+var candidateToJson = (function () {
+	var candidateFieldName = {
+		FOUNDATION: 'foundation',
+		COMPONENT_ID: 'componentId',
+		TRANSPORT: 'transport',
+		PRIORITY: 'priority',
+		CONNECTION_ADDRESS: 'connectionAddress',
+		PORT: 'port',
+		CANDIDATE_TYPE: 'candidateType',
+		REMOTE_CANDIDATE_ADDRESS: 'remoteConnectionAddress',
+		REMOTE_CANDIDATE_PORT: 'remotePort'
+	};
+
+	var candidateType = {
+		HOST: 'host',
+		SRFLX: 'srflx',
+		PRFLX: 'prflx',
+		RELAY: 'relay'
+	};
+
+	var transport = {
+		TCP: 'TCP',
+		UDP: 'UDP'
+	};
+
+	var IPV4SEG = '(?:25[0-5]|(?:2[0-4]|1{0,1}[0-9]){0,1}[0-9])';
+	var IPV4ADDR = `(?:${IPV4SEG}\\.){3}${IPV4SEG}`;
+	var IPV6SEG = '[0-9a-fA-F]{1,4}';
+	var IPV6ADDR =
+		`(?:${IPV6SEG}:){7,7}${IPV6SEG}|` + // 1:2:3:4:5:6:7:8
+		`(?:${IPV6SEG}:){1,7}:|` + // 1::                              1:2:3:4:5:6:7::
+		`(?:${IPV6SEG}:){1,6}:${IPV6SEG}|` + // 1::8             1:2:3:4:5:6::8  1:2:3:4:5:6::8
+		`(?:${IPV6SEG}:){1,5}(?::${IPV6SEG}){1,2}|` + // 1::7:8           1:2:3:4:5::7:8  1:2:3:4:5::8
+		`(?:${IPV6SEG}:){1,4}(?::${IPV6SEG}){1,3}|` + // 1::6:7:8         1:2:3:4::6:7:8  1:2:3:4::8
+		`(?:${IPV6SEG}:){1,3}(?::${IPV6SEG}){1,4}|` + // 1::5:6:7:8       1:2:3::5:6:7:8  1:2:3::8
+		`(?:${IPV6SEG}:){1,2}(?::${IPV6SEG}){1,5}|` + // 1::4:5:6:7:8     1:2::4:5:6:7:8  1:2::8
+		`${IPV6SEG}:(?:(?::${IPV6SEG}){1,6})|` + // 1::3:4:5:6:7:8   1::3:4:5:6:7:8  1::8
+		`:(?:(?::${IPV6SEG}){1,7}|:)|` + // ::2:3:4:5:6:7:8  ::2:3:4:5:6:7:8 ::8       ::
+		`fe80:(?::${IPV6SEG}){0,4}%[0-9a-zA-Z]{1,}|` + // fe80::7:8%eth0   fe80::7:8%1     (link-local IPv6 addresses with zone index)
+		`::(?:ffff(?::0{1,4}){0,1}:){0,1}${IPV4ADDR}|` + // ::255.255.255.255   ::ffff:255.255.255.255  ::ffff:0:255.255.255.255 (IPv4-mapped IPv6 addresses and IPv4-translated addresses)
+		`(?:${IPV6SEG}:){1,4}:${IPV4ADDR}`; // 2001:db8:3:4::192.0.2.33  64:ff9b::192.0.2.33 (IPv4-Embedded IPv6 Address)
+
+	var TOKEN = "[0-9a-zA-Z\\-\\.!\\%\\*_\\+\\`\\'\\~]+";
 
+	var CANDIDATE_TYPE = '';
+	Object.keys(candidateType).forEach(function (key) {
+		CANDIDATE_TYPE += candidateType[key] + '|';
+	});
+	CANDIDATE_TYPE += TOKEN;
+
+	var pattern = {
+		COMPONENT_ID: '[0-9]{1,5}',
+		FOUNDATION: '[a-zA-Z0-9\\+\\/\\-]+',
+		PRIORITY: '[0-9]{1,10}',
+		TRANSPORT: transport.UDP + '|' + TOKEN,
+		CONNECTION_ADDRESS: IPV4ADDR + '|' + IPV6ADDR,
+		PORT: '[0-9]{1,5}',
+		CANDIDATE_TYPE: CANDIDATE_TYPE
+	};
+
+	return function candidateToJson(iceCandidate) {
+		var iceCandidateJson = null;
+
+		if (iceCandidate && typeof iceCandidate === 'string') {
+			var ICE_CANDIDATE_PATTERN = new RegExp(
+				`candidate:(${pattern.FOUNDATION})` + // 10
+					`\\s(${pattern.COMPONENT_ID})` + // 1
+					`\\s(${pattern.TRANSPORT})` + // UDP
+					`\\s(${pattern.PRIORITY})` + // 1845494271
+					`\\s(${pattern.CONNECTION_ADDRESS})` + // 13.93.107.159
+					`\\s(${pattern.PORT})` + // 53705
+					'\\s' +
+					'typ' +
+					`\\s(${pattern.CANDIDATE_TYPE})` + // typ prflx
+					'(?:\\s' +
+					'raddr' +
+					`\\s(${pattern.CONNECTION_ADDRESS})` + // raddr 10.1.221.7
+					'\\s' +
+					'rport' +
+					`\\s(${pattern.PORT}))?` // rport 54805
+			);
+
+			var iceCandidateFields = iceCandidate.match(ICE_CANDIDATE_PATTERN);
+			if (iceCandidateFields) {
+				iceCandidateJson = {};
+				Object.keys(candidateFieldName).forEach(function (key, i) {
+					// i+1 because match returns the entire match result
+					// and the parentheses-captured matched results.
+					if (iceCandidateFields.length > i + 1 && iceCandidateFields[i + 1]) {
+						iceCandidateJson[candidateFieldName[key]] = iceCandidateFields[i + 1];
+					}
+				});
+			}
+		}
+
+		return iceCandidateJson;
+	};
+})();
+
+// See https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidate/RTCIceCandidate
 function RTCIceCandidate(data) {
 	data = data || {};
 
@@ -11,4 +165,22 @@ function RTCIceCandidate(data) {
 	this.sdpMid = data.sdpMid;
 	this.sdpMLineIndex = data.sdpMLineIndex;
 	this.candidate = data.candidate;
+
+	// Parse candidate SDP:
+	// Example: candidate:1829696681 1 udp 2122262783 2a01:cb05:8d3e:a300:e1ad:79c1:7096:8ba0 49778 typ host generation 0 ufrag c9L6 network-id 2 network-cost 10
+	var iceCandidateFields = candidateToJson(this.candidate);
+	if (iceCandidateFields) {
+		this.foundation = iceCandidateFields.foundation;
+		this.component = iceCandidateFields.componentId;
+		this.priority = iceCandidateFields.priority;
+		this.type = iceCandidateFields.candidateType;
+
+		this.address = iceCandidateFields.connectionAddress;
+		this.ip = iceCandidateFields.connectionAddress;
+		this.protocol = iceCandidateFields.transport;
+		this.port = iceCandidateFields.port;
+
+		this.relatedAddress = iceCandidateFields.remoteConnectionAddress || null;
+		this.relatedPort = iceCandidateFields.remotePort || null;
+	}
 }
diff --git a/js/RTCPeerConnection.js b/js/RTCPeerConnection.js
index 3e6f62c5..d8cebef1 100644
--- a/js/RTCPeerConnection.js
+++ b/js/RTCPeerConnection.js
@@ -3,29 +3,35 @@
  */
 module.exports = RTCPeerConnection;
 
-
 /**
  * Dependencies.
  */
-var
-	debug = require('debug')('iosrtc:RTCPeerConnection'),
+var debug = require('debug')('iosrtc:RTCPeerConnection'),
 	debugerror = require('debug')('iosrtc:ERROR:RTCPeerConnection'),
 	exec = require('cordova/exec'),
-	randomNumber = require('random-number').generator({min: 10000, max: 99999, integer: true}),
-	EventTarget = require('yaeti').EventTarget,
+	randomNumber = require('random-number').generator({ min: 10000, max: 99999, integer: true }),
+	EventTarget = require('./EventTarget'),
 	RTCSessionDescription = require('./RTCSessionDescription'),
 	RTCIceCandidate = require('./RTCIceCandidate'),
 	RTCDataChannel = require('./RTCDataChannel'),
 	RTCDTMFSender = require('./RTCDTMFSender'),
-	RTCStatsResponse = require('./RTCStatsResponse'),
+	RTCRtpReceiver = require('./RTCRtpReceiver'),
+	RTCRtpSender = require('./RTCRtpSender'),
+	{ RTCRtpTransceiver, addTransceiverToPeerConnection } = require('./RTCRtpTransceiver'),
 	RTCStatsReport = require('./RTCStatsReport'),
 	MediaStream = require('./MediaStream'),
-	MediaStreamTrack = require('./MediaStreamTrack'),
+	{ MediaStreamTrack, newMediaStreamTrackId } = require('./MediaStreamTrack'),
 	Errors = require('./Errors');
 
-
 debugerror.log = console.warn.bind(console);
 
+function deprecateWarning(method, newMethod) {
+	if (!newMethod) {
+		console.warn(method + ' is deprecated.');
+	} else {
+		console.warn(method + ' method is deprecated, use ' + newMethod + ' instead.');
+	}
+}
 
 function RTCPeerConnection(pcConfig, pcConstraints) {
 	debug('new() | [pcConfig:%o, pcConstraints:%o]', pcConfig, pcConstraints);
@@ -35,8 +41,32 @@ function RTCPeerConnection(pcConfig, pcConstraints) {
 	// Make this an EventTarget.
 	EventTarget.call(this);
 
+	// Restore corrupted RTCPeerConnection.prototype
+	// TODO find why webrtc-adapter prevent events onnegotiationneeded to be trigger.
+	// Object.defineProperties(this, RTCPeerConnection.prototype_descriptor);
+
+	// Fix webrtc-adapter bad SHIM on addTrack causing error when original does support multiple streams.
+	// NotSupportedError: The adapter.js addTrack, addStream polyfill only supports a single stream which is associated with the specified track.
+	Object.defineProperty(this, 'addTrack', RTCPeerConnection.prototype_descriptor.addTrack);
+	Object.defineProperty(this, 'addStream', RTCPeerConnection.prototype_descriptor.addStream);
+	Object.defineProperty(
+		this,
+		'getLocalStreams',
+		RTCPeerConnection.prototype_descriptor.getLocalStreams
+	);
+	Object.defineProperty(
+		this,
+		'addTransceiver',
+		RTCPeerConnection.prototype_descriptor.addTransceiver
+	);
+	Object.defineProperty(
+		this,
+		'getOrCreateTrack',
+		RTCPeerConnection.prototype_descriptor.getOrCreateTrack
+	);
+
 	// Public atributes.
-	this.localDescription = null;
+	this._localDescription = null;
 	this.remoteDescription = null;
 	this.signalingState = 'stable';
 	this.iceGatheringState = 'new';
@@ -48,29 +78,79 @@ function RTCPeerConnection(pcConfig, pcConstraints) {
 	this.localStreams = {};
 	this.remoteStreams = {};
 
+	this.tracks = {};
+	this.transceivers = [];
+
 	function onResultOK(data) {
 		onEvent.call(self, data);
 	}
 
-	exec(onResultOK, null, 'iosrtcPlugin', 'new_RTCPeerConnection', [this.pcId, this.pcConfig, pcConstraints]);
+	exec(onResultOK, null, 'iosrtcPlugin', 'new_RTCPeerConnection', [
+		this.pcId,
+		this.pcConfig,
+		pcConstraints
+	]);
 }
 
+RTCPeerConnection.prototype = Object.create(EventTarget.prototype);
+RTCPeerConnection.prototype.constructor = RTCPeerConnection;
 
-RTCPeerConnection.prototype.createOffer = function () {
-	var self = this,
-		isPromise,
-		options,
-		callback, errback;
-
-	if (typeof arguments[0] !== 'function') {
-		isPromise = true;
-		options = arguments[0];
-	} else {
-		isPromise = false;
-		callback = arguments[0];
-		errback = arguments[1];
-		options = arguments[2];
+Object.defineProperties(RTCPeerConnection.prototype, {
+	localDescription: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		get: function () {
+			return this._localDescription;
+		}
+	},
+	connectionState: {
+		get: function () {
+			return this.iceConnectionState;
+		}
+	},
+	onicecandidate: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		set: function (callback) {
+			return this.addEventListener('icecandidate', callback);
+		}
+	},
+	onaddstream: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		set: function (callback) {
+			return this.addEventListener('addstream', callback);
+		}
+	},
+	ontrack: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		set: function (callback) {
+			return this.addEventListener('track', callback);
+		}
+	},
+	oniceconnectionstatechange: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		set: function (callback) {
+			return this.addEventListener('iceconnectionstatechange', callback);
+		}
+	},
+	onnegotiationneeded: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		set: function (callback) {
+			return this.addEventListener('negotiationneeded', callback);
+		}
 	}
+});
+
+RTCPeerConnection.prototype.createOffer = function (options) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('RTCPeerConnection.prototype.createOffer', arguments);
+
+	var self = this;
 
 	if (isClosed.call(this)) {
 		return;
@@ -78,73 +158,44 @@ RTCPeerConnection.prototype.createOffer = function () {
 
 	debug('createOffer() [options:%o]', options);
 
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				var desc = new RTCSessionDescription(data);
-
-				debug('createOffer() | success [desc:%o]', desc);
-				resolve(desc);
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			if (isClosed.call(self)) {
+				return;
 			}
 
-			function onResultError(error) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('createOffer() | failure: %s', error);
-				reject(new global.DOMError(error));
+			if (data.transceivers) {
+				self.updateTransceiversState(data.transceivers);
 			}
 
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createOffer', [self.pcId, options]);
-		});
-	}
+			var desc = new RTCSessionDescription(data);
 
-	function onResultOK(data) {
-		if (isClosed.call(self)) {
-			return;
+			debug('createOffer() | success [desc:%o]', desc);
+			resolve(desc);
 		}
 
-		var desc = new RTCSessionDescription(data);
-
-		debug('createOffer() | success [desc:%o]', desc);
-		callback(desc);
-	}
-
-	function onResultError(error) {
-		if (isClosed.call(self)) {
-			return;
-		}
+		function onResultError(error) {
+			if (isClosed.call(self)) {
+				return;
+			}
 
-		debugerror('createOffer() | failure: %s', error);
-		if (typeof errback === 'function') {
-			errback(new global.DOMError(error));
+			debugerror('createOffer() | failure: %s', error);
+			reject(new global.DOMException(error));
 		}
-	}
 
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createOffer', [this.pcId, options]);
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createOffer', [
+			self.pcId,
+			options
+		]);
+	});
 };
 
+RTCPeerConnection.prototype.createAnswer = function (options) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('RTCPeerConnection.prototype.createAnswer', arguments);
 
-RTCPeerConnection.prototype.createAnswer = function () {
-	var self = this,
-		isPromise,
-		options,
-		callback, errback;
-
-	if (typeof arguments[0] !== 'function') {
-		isPromise = true;
-		options = arguments[0];
-	} else {
-		isPromise = false;
-		callback = arguments[0];
-		errback = arguments[1];
-		options = arguments[2];
-	}
+	var self = this;
 
 	if (isClosed.call(this)) {
 		return;
@@ -152,351 +203,240 @@ RTCPeerConnection.prototype.createAnswer = function () {
 
 	debug('createAnswer() [options:%o]', options);
 
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				var desc = new RTCSessionDescription(data);
-
-				debug('createAnswer() | success [desc:%o]', desc);
-				resolve(desc);
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			if (isClosed.call(self)) {
+				return;
 			}
 
-			function onResultError(error) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('createAnswer() | failure: %s', error);
-				reject(new global.DOMError(error));
+			if (data.transceivers) {
+				self.updateTransceiversState(data.transceivers);
 			}
 
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createAnswer', [self.pcId, options]);
-		});
-	}
+			var desc = new RTCSessionDescription(data);
 
-	function onResultOK(data) {
-		if (isClosed.call(self)) {
-			return;
+			debug('createAnswer() | success [desc:%o]', desc);
+			resolve(desc);
 		}
 
-		var desc = new RTCSessionDescription(data);
-
-		debug('createAnswer() | success [desc:%o]', desc);
-		callback(desc);
-	}
-
-	function onResultError(error) {
-		if (isClosed.call(self)) {
-			return;
-		}
+		function onResultError(error) {
+			if (isClosed.call(self)) {
+				return;
+			}
 
-		debugerror('createAnswer() | failure: %s', error);
-		if (typeof errback === 'function') {
-			errback(new global.DOMError(error));
+			debugerror('createAnswer() | failure: %s', error);
+			reject(new global.DOMException(error));
 		}
-	}
 
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createAnswer', [this.pcId, options]);
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createAnswer', [
+			self.pcId,
+			options
+		]);
+	});
 };
 
-
-
 RTCPeerConnection.prototype.setLocalDescription = function (desc) {
-	var self = this,
-		isPromise,
-		callback, errback;
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage(
+		'RTCPeerConnection.prototype.setLocalDescription',
+		arguments
+	);
 
-	if (typeof arguments[1] !== 'function') {
-		isPromise = true;
-	} else {
-		isPromise = false;
-		callback = arguments[1];
-		errback = arguments[2];
-	}
+	var self = this;
 
 	if (isClosed.call(this)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.InvalidStateError('peerconnection is closed'));
-			});
-		} else {
-			throw new Errors.InvalidStateError('peerconnection is closed');
-		}
+		return new Promise(function (resolve, reject) {
+			reject(new Errors.InvalidStateError('peerconnection is closed'));
+		});
 	}
 
-	debug('setLocalDescription() [desc:%o]', desc);
+	// "This is no longer necessary, however; RTCPeerConnection.setLocalDescription() and other
+	// methods which take SDP as input now directly accept an object conforming to the RTCSessionDescriptionInit dictionary,
+	// so you don't have to instantiate an RTCSessionDescription yourself.""
+	// Source: https://developer.mozilla.org/en-US/docs/Web/API/RTCSessionDescription/RTCSessionDescription#Example
+	// Still we do instnanciate RTCSessionDescription, so internal object is used properly.
 
 	if (!(desc instanceof RTCSessionDescription)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.InvalidSessionDescriptionError('setLocalDescription() must be called with a RTCSessionDescription instance as first argument'));
-			});
-		} else {
-			if (typeof errback === 'function') {
-				errback(new Errors.InvalidSessionDescriptionError('setLocalDescription() must be called with a RTCSessionDescription instance as first argument'));
-			}
-			return;
-		}
+		desc = new RTCSessionDescription(desc);
 	}
 
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debug('setLocalDescription() | success');
-				// Update localDescription.
-				self.localDescription = new RTCSessionDescription(data);
-				resolve();
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			if (isClosed.call(self)) {
+				return;
 			}
 
-			function onResultError(error) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('setLocalDescription() | failure: %s', error);
-				reject(new Errors.InvalidSessionDescriptionError('setLocalDescription() failed: ' + error));
+			if (data.transceivers) {
+				self.updateTransceiversState(data.transceivers);
 			}
 
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setLocalDescription', [self.pcId, desc]);
-		});
-	}
-
-	function onResultOK(data) {
-		if (isClosed.call(self)) {
-			return;
-		}
-
-		debug('setLocalDescription() | success');
-		// Update localDescription.
-		self.localDescription = new RTCSessionDescription(data);
-		callback();
-	}
-
-	function onResultError(error) {
-		if (isClosed.call(self)) {
-			return;
+			debug('setLocalDescription() | success');
+			// Update localDescription.
+			self._localDescription = data.type === '' ? null : new RTCSessionDescription(data);
+			resolve();
 		}
 
-		debugerror('setLocalDescription() | failure: %s', error);
+		function onResultError(error) {
+			if (isClosed.call(self)) {
+				return;
+			}
 
-		if (typeof errback === 'function') {
-			errback(new Errors.InvalidSessionDescriptionError('setLocalDescription() failed: ' + error));
+			debugerror('setLocalDescription() | failure: %s', error);
+			reject(
+				new Errors.InvalidSessionDescriptionError('setLocalDescription() failed: ' + error)
+			);
 		}
-	}
 
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setLocalDescription', [this.pcId, desc]);
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setLocalDescription', [
+			self.pcId,
+			desc
+		]);
+	});
 };
 
-
 RTCPeerConnection.prototype.setRemoteDescription = function (desc) {
-	var self = this,
-		isPromise,
-		callback, errback;
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage(
+		'RTCPeerConnection.prototype.setRemoteDescription',
+		arguments
+	);
 
-	if (typeof arguments[1] !== 'function') {
-		isPromise = true;
-	} else {
-		isPromise = false;
-		callback = arguments[1];
-		errback = arguments[2];
-	}
+	var self = this;
 
 	if (isClosed.call(this)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.InvalidStateError('peerconnection is closed'));
-			});
-		} else {
-			throw new Errors.InvalidStateError('peerconnection is closed');
-		}
+		return new Promise(function (resolve, reject) {
+			reject(new Errors.InvalidStateError('peerconnection is closed'));
+		});
 	}
 
 	debug('setRemoteDescription() [desc:%o]', desc);
 
-	if (!(desc instanceof RTCSessionDescription)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.InvalidSessionDescriptionError('setRemoteDescription() must be called with a RTCSessionDescription instance as first argument'));
-			});
-		} else {
-			if (typeof errback === 'function') {
-				errback(new Errors.InvalidSessionDescriptionError('setRemoteDescription() must be called with a RTCSessionDescription instance as first argument'));
-			}
-			return;
-		}
+	// Remove extmap-allow-mixed sdp header
+	if (desc && desc.sdp && desc.sdp.indexOf('\na=extmap-allow-mixed') !== -1) {
+		desc = new RTCSessionDescription({
+			type: desc.type,
+			sdp: desc.sdp
+				.split('\n')
+				.filter((line) => {
+					return line.trim() !== 'a=extmap-allow-mixed';
+				})
+				.join('\n')
+		});
 	}
 
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debug('setRemoteDescription() | success');
-				// Update remoteDescription.
-				self.remoteDescription = new RTCSessionDescription(data);
-				resolve();
-			}
-
-			function onResultError(error) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('setRemoteDescription() | failure: %s', error);
-				reject(new Errors.InvalidSessionDescriptionError('setRemoteDescription() failed: ' + error));
-			}
+	// "This is no longer necessary, however; RTCPeerConnection.setLocalDescription() and other
+	// methods which take SDP as input now directly accept an object conforming to the RTCSessionDescriptionInit dictionary,
+	// so you don't have to instantiate an RTCSessionDescription yourself.""
+	// Source: https://developer.mozilla.org/en-US/docs/Web/API/RTCSessionDescription/RTCSessionDescription#Example
+	// Still we do instnanciate RTCSessionDescription so internal object is used properly.
 
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setRemoteDescription', [self.pcId, desc]);
-		});
+	if (!(desc instanceof RTCSessionDescription)) {
+		desc = new RTCSessionDescription(desc);
 	}
 
-	function onResultOK(data) {
-		if (isClosed.call(self)) {
-			return;
-		}
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			if (isClosed.call(self)) {
+				return;
+			}
 
-		debug('setRemoteDescription() | success');
-		// Update remoteDescription.
-		self.remoteDescription = new RTCSessionDescription(data);
-		callback();
-	}
+			if (data.transceivers) {
+				self.updateTransceiversState(data.transceivers);
+			}
 
-	function onResultError(error) {
-		if (isClosed.call(self)) {
-			return;
+			debug('setRemoteDescription() | success');
+			// Update remoteDescription.
+			self.remoteDescription = new RTCSessionDescription(data);
+			resolve();
 		}
 
-		debugerror('setRemoteDescription() | failure: %s', error);
+		function onResultError(error) {
+			if (isClosed.call(self)) {
+				return;
+			}
 
-		if (typeof errback === 'function') {
-			errback(new Errors.InvalidSessionDescriptionError('setRemoteDescription() failed: ' + error));
+			debugerror('setRemoteDescription() | failure: %s', error);
+			reject(
+				new Errors.InvalidSessionDescriptionError('setRemoteDescription() failed: ' + error)
+			);
 		}
-	}
 
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setRemoteDescription', [this.pcId, desc]);
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setRemoteDescription', [
+			self.pcId,
+			desc
+		]);
+	});
 };
 
-
-
 RTCPeerConnection.prototype.addIceCandidate = function (candidate) {
-	var self = this,
-		isPromise,
-		callback, errback;
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('RTCPeerConnection.prototype.addIceCandidate', arguments);
 
-	if (typeof arguments[1] !== 'function') {
-		isPromise = true;
-	} else {
-		isPromise = false;
-		callback = arguments[1];
-		errback = arguments[2];
-	}
+	var self = this;
 
 	if (isClosed.call(this)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.InvalidStateError('peerconnection is closed'));
-			});
-		} else {
-			throw new Errors.InvalidStateError('peerconnection is closed');
-		}
+		return new Promise(function (resolve, reject) {
+			reject(new Errors.InvalidStateError('peerconnection is closed'));
+		});
 	}
 
 	debug('addIceCandidate() | [candidate:%o]', candidate);
 
-	if (!(candidate instanceof RTCIceCandidate)) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new global.DOMError('addIceCandidate() must be called with a RTCIceCandidate instance as first argument'));
-			});
-		} else {
-			if (typeof errback === 'function') {
-				errback(new global.DOMError('addIceCandidate() must be called with a RTCIceCandidate instance as first argument'));
-			}
-			return;
-		}
-	}
-
-	if (isPromise) {
+	if (typeof candidate !== 'object') {
 		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debug('addIceCandidate() | success');
-				// Update remoteDescription.
-				if (self.remoteDescription && data.remoteDescription) {
-					self.remoteDescription.type = data.remoteDescription.type;
-					self.remoteDescription.sdp = data.remoteDescription.sdp;
-				} else if (data.remoteDescription) {
-					self.remoteDescription = new RTCSessionDescription(data.remoteDescription);
-				}
-				resolve();
-			}
-
-			function onResultError() {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('addIceCandidate() | failure');
-				reject(new global.DOMError('addIceCandidate() failed'));
-			}
-
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_addIceCandidate', [self.pcId, candidate]);
+			reject(
+				new global.DOMException(
+					'addIceCandidate() must be called with a RTCIceCandidate instance or RTCIceCandidateInit object as argument'
+				)
+			);
 		});
 	}
 
-	function onResultOK(data) {
-		if (isClosed.call(self)) {
-			return;
-		}
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			if (isClosed.call(self)) {
+				return;
+			}
 
-		debug('addIceCandidate() | success');
-		// Update remoteDescription.
-		if (self.remoteDescription && data.remoteDescription) {
-			self.remoteDescription.type = data.remoteDescription.type;
-			self.remoteDescription.sdp = data.remoteDescription.sdp;
-		} else if (data.remoteDescription) {
-			self.remoteDescription = new RTCSessionDescription(data.remoteDescription);
+			debug('addIceCandidate() | success');
+			// Update remoteDescription.
+			if (self.remoteDescription && data.remoteDescription) {
+				self.remoteDescription.type = data.remoteDescription.type;
+				self.remoteDescription.sdp = data.remoteDescription.sdp;
+			} else if (data.remoteDescription) {
+				self.remoteDescription = new RTCSessionDescription(data.remoteDescription);
+			}
+			resolve();
 		}
-		callback();
-	}
 
-	function onResultError() {
-		if (isClosed.call(self)) {
-			return;
-		}
+		function onResultError() {
+			if (isClosed.call(self)) {
+				return;
+			}
 
-		debugerror('addIceCandidate() | failure');
-		if (typeof errback === 'function') {
-			errback(new global.DOMError('addIceCandidate() failed'));
+			debugerror('addIceCandidate() | failure');
+			reject(new global.DOMException('addIceCandidate() failed'));
 		}
-	}
 
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_addIceCandidate', [this.pcId, candidate]);
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_addIceCandidate', [
+			self.pcId,
+			candidate
+		]);
+	});
 };
 
-
 RTCPeerConnection.prototype.getConfiguration = function () {
 	debug('getConfiguration()');
 
 	return this.pcConfig;
 };
 
-
 RTCPeerConnection.prototype.getLocalStreams = function () {
 	debug('getLocalStreams()');
+	deprecateWarning('getLocalStreams', 'getSenders');
 
 	var streams = [],
 		id;
@@ -510,9 +450,9 @@ RTCPeerConnection.prototype.getLocalStreams = function () {
 	return streams;
 };
 
-
 RTCPeerConnection.prototype.getRemoteStreams = function () {
 	debug('getRemoteStreams()');
+	deprecateWarning('getRemoteStreams', 'getReceivers');
 
 	var streams = [],
 		id;
@@ -526,6 +466,258 @@ RTCPeerConnection.prototype.getRemoteStreams = function () {
 	return streams;
 };
 
+RTCPeerConnection.prototype.getReceivers = function () {
+	return this.getTransceivers()
+		.filter((transceiver) => !transceiver.stopped)
+		.map((transceiver) => transceiver.receiver);
+};
+
+RTCPeerConnection.prototype.getSenders = function () {
+	return this.getTransceivers()
+		.filter((transceiver) => !transceiver.stopped)
+		.map((transceiver) => transceiver.sender);
+};
+
+RTCPeerConnection.prototype.getTransceivers = function () {
+	return this.transceivers;
+};
+
+RTCPeerConnection.prototype.addTrack = function (track, ...streams) {
+	var stream = streams[0];
+
+	if (isClosed.call(this)) {
+		throw new Errors.InvalidStateError('peerconnection is closed');
+	}
+
+	// Add localStreams if missing
+	// Disable to match browser behavior
+	//stream = stream || Object.values(this.localStreams)[0] || new MediaStream();
+
+	// Fix webrtc-adapter bad SHIM on addStream
+	if (stream) {
+		this.addStream(stream);
+	}
+
+	var transceiver = new RTCRtpTransceiver(this, {
+		sender: new RTCRtpSender(this, {
+			track: track
+		}),
+		receiver: new RTCRtpReceiver(this),
+		direction: 'sendrecv',
+		currentDirection: null,
+		mid: null
+	});
+
+	for (var streamId in this.localStreams) {
+		if (this.localStreams.hasOwnProperty(streamId)) {
+			// Target provided stream argument or first added stream to group track
+			if (!stream || (stream && stream.id === streamId)) {
+				stream = this.localStreams[streamId];
+				stream.addTrack(track);
+				exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_addTrack', [
+					this.pcId,
+					track.id,
+					transceiver._id,
+					transceiver.receiver._id,
+					transceiver.sender._id,
+					streamId
+				]);
+				break;
+			}
+		}
+	}
+
+	// No Stream matched add track without stream
+	if (!stream) {
+		exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_addTrack', [
+			this.pcId,
+			track.id,
+			transceiver._id,
+			transceiver.receiver._id,
+			transceiver.sender._id,
+			null
+		]);
+	}
+
+	this.getOrCreateTrack(track);
+
+	this.transceivers.push(transceiver);
+
+	return transceiver.sender;
+};
+
+RTCPeerConnection.prototype.removeTrack = function (sender) {
+	var id, track, stream, hasTrack;
+
+	if (!(sender instanceof RTCRtpSender)) {
+		throw new Error('removeTrack() must be called with a RTCRtpSender instance as argument');
+	}
+
+	track = sender.track;
+
+	// No sender track found
+	if (!track) {
+		return;
+	}
+
+	function matchLocalTrack(localTrack) {
+		return localTrack.id === track.id;
+	}
+
+	for (id in this.localStreams) {
+		if (this.localStreams.hasOwnProperty(id)) {
+			// Check if track is belong to stream
+			hasTrack = this.localStreams[id].getTracks().filter(matchLocalTrack).length > 0;
+
+			if (hasTrack) {
+				stream = this.localStreams[id];
+				stream.removeTrack(track);
+				exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_removeTrack', [
+					this.pcId,
+					sender._id
+				]);
+				delete this.tracks[track.id];
+				break;
+			}
+		}
+	}
+
+	// No Stream matched remove track without stream
+	if (!stream) {
+		for (id in this.tracks) {
+			if (this.tracks.hasOwnProperty(id)) {
+				if (track.id === id) {
+					exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_removeTrack', [
+						this.pcId,
+						sender._id
+					]);
+					delete this.tracks[track.id];
+				}
+			}
+		}
+	}
+};
+
+RTCPeerConnection.prototype.getOrCreateTrack = function (trackInput) {
+	var { id } = trackInput,
+		existingTrack = this.tracks[id];
+
+	if (existingTrack) {
+		return existingTrack;
+	}
+
+	var track;
+	if (trackInput instanceof MediaStreamTrack) {
+		track = trackInput;
+	} else {
+		track = MediaStreamTrack.findOrCreate(trackInput);
+	}
+
+	this.tracks[id] = track;
+	track.addEventListener('ended', () => {
+		delete this.tracks[id];
+	});
+
+	return track;
+};
+
+RTCPeerConnection.prototype.addTransceiver = function (trackOrKind, init) {
+	if (isClosed.call(this)) {
+		throw new Errors.InvalidStateError('peerconnection is closed');
+	}
+
+	var kind,
+		track = null,
+		self = this,
+		trackIdOrKind;
+	if (trackOrKind instanceof MediaStreamTrack) {
+		kind = trackOrKind.kind;
+		track = trackOrKind;
+		trackIdOrKind = trackOrKind.id;
+	} else {
+		if (!(trackOrKind === 'audio' || trackOrKind === 'video')) {
+			throw new TypeError(
+				'An invalid string was specified as trackOrKind. The string must be either "audio" or "video".'
+			);
+		}
+		kind = trackOrKind;
+		trackIdOrKind = trackOrKind;
+	}
+
+	var receiverTrackID = newMediaStreamTrackId();
+
+	var receiver = new RTCRtpReceiver(this, {
+		track: new MediaStreamTrack({
+			id: receiverTrackID,
+			kind,
+			enabled: true,
+			readyState: 'live',
+			trackId: 'unknown'
+		})
+	});
+
+	var sender = new RTCRtpSender(this, {
+		track
+	});
+
+	var data = init || {};
+	data.sender = sender;
+	data.receiver = receiver;
+
+	var transceiver = new RTCRtpTransceiver(this, data);
+
+	this.transceivers.push(transceiver);
+
+	var initJson = {};
+	if (init && init.direction) {
+		initJson.direction = init.direction;
+	} else {
+		initJson.direction = 'sendrecv';
+	}
+
+	if (init && init.sendEncodings) {
+		initJson.sendEncodings = init.sendEncodings;
+	}
+
+	if (init && init.streams) {
+		initJson.streams = init.streams.map((stream) => stream.id);
+	} else {
+		initJson.streams = [];
+	}
+
+	debug(
+		'addTransceiver() [trackIdOrKind:%s, init:%o, initJson: %o]',
+		trackIdOrKind,
+		init,
+		initJson
+	);
+
+	addTransceiverToPeerConnection(this, trackIdOrKind, initJson, transceiver)
+		.then((update) => {
+			self.updateTransceiversState(update.transceivers);
+		})
+		.catch((error) => {
+			debugerror('addTransceiver() | failure: %s', error);
+		});
+
+	return transceiver;
+};
+
+RTCPeerConnection.prototype.updateTransceiversState = function (transceivers) {
+	debug('updateTransceiversState()');
+	this.transceivers = transceivers.map((transceiver) => {
+		const existingTransceiver = this.transceivers.find(
+			(localTransceiver) => localTransceiver._id === transceiver.id
+		);
+
+		if (existingTransceiver) {
+			existingTransceiver.update(transceiver);
+			return existingTransceiver;
+		}
+
+		return new RTCRtpTransceiver(this, transceiver);
+	});
+};
 
 RTCPeerConnection.prototype.getStreamById = function (id) {
 	debug('getStreamById()');
@@ -533,15 +725,16 @@ RTCPeerConnection.prototype.getStreamById = function (id) {
 	return this.localStreams[id] || this.remoteStreams[id] || null;
 };
 
-
 RTCPeerConnection.prototype.addStream = function (stream) {
+	var self = this;
+
 	if (isClosed.call(this)) {
 		throw new Errors.InvalidStateError('peerconnection is closed');
 	}
 
 	debug('addStream()');
 
-	if (!(stream instanceof MediaStream)) {
+	if (!(stream instanceof MediaStream.originalMediaStream)) {
 		throw new Error('addStream() must be called with a MediaStream instance as argument');
 	}
 
@@ -552,18 +745,25 @@ RTCPeerConnection.prototype.addStream = function (stream) {
 
 	this.localStreams[stream.id] = stream;
 
+	stream.addedToConnection = true;
+
+	stream.getTracks().forEach(function (track) {
+		self.getOrCreateTrack(track);
+	});
+
 	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_addStream', [this.pcId, stream.id]);
 };
 
-
 RTCPeerConnection.prototype.removeStream = function (stream) {
+	var self = this;
+
 	if (isClosed.call(this)) {
 		throw new Errors.InvalidStateError('peerconnection is closed');
 	}
 
 	debug('removeStream()');
 
-	if (!(stream instanceof MediaStream)) {
+	if (!(stream instanceof MediaStream.originalMediaStream)) {
 		throw new Error('removeStream() must be called with a MediaStream instance as argument');
 	}
 
@@ -574,10 +774,13 @@ RTCPeerConnection.prototype.removeStream = function (stream) {
 
 	delete this.localStreams[stream.id];
 
+	stream.getTracks().forEach(function (track) {
+		delete self.tracks[track.id];
+	});
+
 	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_removeStream', [this.pcId, stream.id]);
 };
 
-
 RTCPeerConnection.prototype.createDataChannel = function (label, options) {
 	if (isClosed.call(this)) {
 		throw new Errors.InvalidStateError('peerconnection is closed');
@@ -588,7 +791,6 @@ RTCPeerConnection.prototype.createDataChannel = function (label, options) {
 	return new RTCDataChannel(this, label, options);
 };
 
-
 RTCPeerConnection.prototype.createDTMFSender = function (track) {
 	if (isClosed.call(this)) {
 		throw new Errors.InvalidStateError('peerconnection is closed');
@@ -599,83 +801,48 @@ RTCPeerConnection.prototype.createDTMFSender = function (track) {
 	return new RTCDTMFSender(this, track);
 };
 
-RTCPeerConnection.prototype.getStats = function () {
-	var self = this,
-		isPromise,
-		selector,
-		callback, errback;
+RTCPeerConnection.prototype.getStats = function (selector) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('RTCPeerConnection.prototype.getStats', arguments);
 
-	if (typeof arguments[0] !== 'function') {
-		isPromise = true;
-		selector = arguments[0];
-	} else {
-		isPromise = false;
-		callback = arguments[0];
-		selector = arguments[1];
-		errback = arguments[2];
-	}
+	var self = this;
 
 	if (selector && !(selector instanceof MediaStreamTrack)) {
-		throw new Error('getStats() must be called with null or a valid MediaStreamTrack instance as argument');
+		throw new Error(
+			'getStats() must be called with null or a valid MediaStreamTrack instance as argument'
+		);
 	}
 
 	if (isClosed.call(this)) {
 		throw new Errors.InvalidStateError('peerconnection is closed');
 	}
 
-	debug('getStats() [selector:%o]', selector);
-
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(array) {
-				if (isClosed.call(self)) {
-					return;
-				}
+	// debug('getStats() [selector:%o]', selector);
 
-				var res = [];
-				array.forEach(function (stat) {
-					res.push(new RTCStatsReport(stat));
-				});
-				resolve(new RTCStatsResponse(res));
+	return new Promise(function (resolve, reject) {
+		function onResultOK(array) {
+			if (isClosed.call(self)) {
+				return;
 			}
 
-			function onResultError(error) {
-				if (isClosed.call(self)) {
-					return;
-				}
-
-				debugerror('getStats() | failure: %s', error);
-				reject(new global.DOMError(error));
-			}
-
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_getStats', [self.pcId, selector ? selector.id : null]);
-		});
-	}
-
-	function onResultOK(array) {
-		if (isClosed.call(self)) {
-			return;
+			resolve(new RTCStatsReport(array));
 		}
 
-		var res = [];
-		array.forEach(function (stat) {
-			res.push(new RTCStatsReport(stat));
-		});
-		callback(new RTCStatsResponse(res));
-	}
-
-	function onResultError(error) {
-		if (isClosed.call(self)) {
-			return;
-		}
+		function onResultError(error) {
+			if (isClosed.call(self)) {
+				return;
+			}
 
-		debugerror('getStats() | failure: %s', error);
-		if (typeof errback === 'function') {
-			errback(new global.DOMError(error));
+			debugerror('getStats() | failure: %s', error);
+			reject(new global.DOMException(error));
 		}
-	}
 
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_getStats', [this.pcId, selector ? selector.id : null]);
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_getStats', [
+			self.pcId,
+			selector ? selector.id : null
+		]);
+	});
 };
 
 RTCPeerConnection.prototype.close = function () {
@@ -688,12 +855,14 @@ RTCPeerConnection.prototype.close = function () {
 	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_close', [this.pcId]);
 };
 
+// Save current RTCPeerConnection.prototype
+RTCPeerConnection.prototype_descriptor = Object.getOwnPropertyDescriptors(
+	RTCPeerConnection.prototype
+);
 
 /**
  * Private API.
  */
-
-
 function fixPcConfig(pcConfig) {
 	if (!pcConfig) {
 		return {
@@ -702,7 +871,9 @@ function fixPcConfig(pcConfig) {
 	}
 
 	var iceServers = pcConfig.iceServers,
-		i, len, iceServer;
+		i,
+		len,
+		iceServer;
 
 	if (!Array.isArray(iceServers)) {
 		pcConfig.iceServers = [];
@@ -725,21 +896,26 @@ function fixPcConfig(pcConfig) {
 	return pcConfig;
 }
 
-
 function isClosed() {
 	return this.signalingState === 'closed';
 }
 
-
 function onEvent(data) {
 	var type = data.type,
+		self = this,
 		event = new Event(type),
-		stream,
 		dataChannel,
+		transceiver,
 		id;
 
+	Object.defineProperty(event, 'target', { value: self, enumerable: true });
+
 	debug('onEvent() | [type:%s, data:%o]', type, data);
 
+	if (data.transceivers) {
+		this.updateTransceiversState(data.transceivers);
+	}
+
 	switch (type) {
 		case 'signalingstatechange':
 			this.signalingState = data.signalingState;
@@ -768,37 +944,58 @@ function onEvent(data) {
 			} else {
 				event.candidate = null;
 			}
-			// Update localDescription.
-			if (this.localDescription) {
-				this.localDescription.type = data.localDescription.type;
-				this.localDescription.sdp = data.localDescription.sdp;
-			} else {
-				this.localDescription = new RTCSessionDescription(data);
+			// Update _localDescription.
+			if (this._localDescription && data.localDescription) {
+				this._localDescription.type = data.localDescription.type;
+				this._localDescription.sdp = data.localDescription.sdp;
+			} else if (data.localDescription) {
+				this._localDescription = new RTCSessionDescription(data.localDescription);
 			}
 			break;
 
 		case 'negotiationneeded':
 			break;
 
-		case 'addstream':
-			stream = MediaStream.create(data.stream);
-			event.stream = stream;
+		case 'track':
+			var track = (event.track = new MediaStreamTrack(data.track));
+			event.receiver = new RTCRtpReceiver(self, { track: track });
 
+			transceiver = this.transceivers.find((t) => t.receiver.track.id === track.id);
+			event.transceiver = transceiver;
+			event.streams = [];
+
+			// Add stream only if available in case of Unified-Plan of track event without stream
+			if (data.stream && data.streamId) {
+				var stream = this.remoteStreams[data.streamId] || MediaStream.create(data.stream);
+				event.streams.push(stream);
+			}
+
+			// Store remote track
+			this.getOrCreateTrack(track);
+
+			break;
+
+		case 'addstream':
 			// Append to the remote streams.
-			this.remoteStreams[stream.id] = stream;
+			this.remoteStreams[data.streamId] =
+				this.remoteStreams[data.streamId] || MediaStream.create(data.stream);
+
+			event.stream = this.remoteStreams[data.streamId];
 
 			// Emit "connected" on the stream if ICE connected.
-			if (this.iceConnectionState === 'connected' || this.iceConnectionState === 'completed') {
-				stream.emitConnected();
+			if (
+				this.iceConnectionState === 'connected' ||
+				this.iceConnectionState === 'completed'
+			) {
+				event.stream.emitConnected();
 			}
 			break;
 
 		case 'removestream':
-			stream = this.remoteStreams[data.streamId];
-			event.stream = stream;
+			event.stream = this.remoteStreams[data.streamId];
 
 			// Remove from the remote streams.
-			delete this.remoteStreams[stream.id];
+			delete this.remoteStreams[data.streamId];
 			break;
 
 		case 'datachannel':
diff --git a/js/RTCRtpEncodingParameters.js b/js/RTCRtpEncodingParameters.js
new file mode 100644
index 00000000..0c6a9626
--- /dev/null
+++ b/js/RTCRtpEncodingParameters.js
@@ -0,0 +1,16 @@
+class RTCRtpEncodingParameters {
+	constructor(data) {
+		if (data) {
+			Object.assign(this, data);
+		}
+		this.codecs = this.codecs || [];
+		this.encodings = this.encodings || [];
+		this.headerExtensions = this.headerExtensions || [];
+		this.rtcp = this.rtcp || {};
+		this.transactionId = this.transactionId || null;
+	}
+}
+/**
+ * Expose the RTCRtpEncodingParameters class.
+ */
+module.exports = RTCRtpEncodingParameters;
diff --git a/js/RTCRtpReceiver.js b/js/RTCRtpReceiver.js
new file mode 100644
index 00000000..a604eed3
--- /dev/null
+++ b/js/RTCRtpReceiver.js
@@ -0,0 +1,33 @@
+/**
+ * Expose the RTCRtpReceiver class.
+ */
+module.exports = RTCRtpReceiver;
+
+var randomNumber = require('random-number').generator({ min: 10000, max: 99999, integer: true });
+
+function RTCRtpReceiver(pc, data) {
+	data = data || {};
+	this._id = data.id || randomNumber();
+
+	this._pc = pc;
+	this.track = data.track ? pc.getOrCreateTrack(data.track) : null;
+	this.params = data.params || {};
+}
+
+RTCRtpReceiver.prototype.getParameters = function () {
+	return this.params;
+};
+
+RTCRtpReceiver.prototype.getStats = function () {
+	return this._pc.getStats();
+};
+
+RTCRtpReceiver.prototype.update = function ({ track, params }) {
+	if (track) {
+		this.track = this._pc.getOrCreateTrack(track);
+	} else {
+		this.track = null;
+	}
+
+	this.params = params;
+};
diff --git a/js/RTCRtpSendParameters.js b/js/RTCRtpSendParameters.js
new file mode 100644
index 00000000..d1b71f65
--- /dev/null
+++ b/js/RTCRtpSendParameters.js
@@ -0,0 +1,11 @@
+const RTCRtpEncodingParameters = require('./RTCRtpEncodingParameters');
+
+class RTCRtpSendParameters {
+	constructor(params) {
+		this.encodings = params || new RTCRtpEncodingParameters();
+	}
+}
+/**
+ * Expose the RTCRtpSendParameters class.
+ */
+module.exports = RTCRtpSendParameters;
diff --git a/js/RTCRtpSender.js b/js/RTCRtpSender.js
new file mode 100644
index 00000000..0f55c6d2
--- /dev/null
+++ b/js/RTCRtpSender.js
@@ -0,0 +1,102 @@
+/**
+ * Expose the RTCRtpSender class.
+ */
+module.exports = RTCRtpSender;
+
+/**
+ * Dependencies.
+ */
+var exec = require('cordova/exec'),
+	{ MediaStreamTrack } = require('./MediaStreamTrack'),
+	randomNumber = require('random-number').generator({ min: 10000, max: 99999, integer: true });
+
+function RTCRtpSender(pc, data) {
+	data = data || {};
+	this._id = data.id || randomNumber();
+
+	this._pc = pc;
+	this.track = data.track ? pc.getOrCreateTrack(data.track) : null;
+	this.params = data.params || {};
+	if (this.track) {
+		this.dtmf = pc.createDTMFSender(this.track);
+	}
+}
+
+RTCRtpSender.prototype.getParameters = function () {
+	return this.params;
+};
+
+RTCRtpSender.prototype.getStats = function () {
+	return this._pc.getStats();
+};
+
+RTCRtpSender.prototype.setParameters = function (params) {
+	Object.assign(this.params, params);
+	return new Promise((resolve, reject) => {
+		function onResultOK(result) {
+			resolve(result);
+		}
+
+		function onResultError(error) {
+			reject(error);
+		}
+
+		exec(
+			onResultOK,
+			onResultError,
+			'iosrtcPlugin',
+			'RTCPeerConnection_RTCRtpSender_setParameters',
+			[this._pc.pcId, this._id, params]
+		);
+	});
+};
+
+RTCRtpSender.prototype.replaceTrack = function (withTrack) {
+	var self = this,
+		pc = self._pc;
+
+	return new Promise((resolve, reject) => {
+		function onResultOK(result) {
+			self.track = result.track ? MediaStreamTrack.findOrCreate(result.track) : null;
+			resolve();
+		}
+
+		function onResultError(error) {
+			reject(error);
+		}
+
+		var trackId = withTrack ? withTrack.id : null;
+
+		exec(
+			onResultOK,
+			onResultError,
+			'iosrtcPlugin',
+			'RTCPeerConnection_RTCRtpSender_replaceTrack',
+			[this._pc.pcId, this._id, trackId]
+		);
+
+		// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/negotiationneeded_event
+		var event = new Event('negotiationneeded');
+		pc.dispatchEvent(event);
+
+		pc.addEventListener('signalingstatechange', function listener() {
+			if (pc.signalingState === 'closed') {
+				pc.removeEventListener('signalingstatechange', listener);
+				reject();
+			} else if (pc.signalingState === 'stable') {
+				pc.removeEventListener('signalingstatechange', listener);
+				resolve();
+			}
+		});
+	});
+};
+
+RTCRtpSender.prototype.update = function ({ track, params }) {
+	if (track) {
+		this.track = this._pc.getOrCreateTrack(track);
+	} else {
+		this.track = null;
+	}
+
+	this.params = params;
+};
diff --git a/js/RTCRtpTransceiver.js b/js/RTCRtpTransceiver.js
new file mode 100644
index 00000000..b2f2178b
--- /dev/null
+++ b/js/RTCRtpTransceiver.js
@@ -0,0 +1,155 @@
+const RTCRtpSender = require('./RTCRtpSender');
+const RTCRtpReceiver = require('./RTCRtpReceiver');
+
+/**
+ * Expose the RTCRtpTransceiver class.
+ */
+module.exports = { RTCRtpTransceiver, addTransceiverToPeerConnection };
+
+/**
+ * Dependencies.
+ */
+var debugerror = require('debug')('iosrtc:ERROR:RTCRtpTransceiver'),
+	exec = require('cordova/exec'),
+	randomNumber = require('random-number').generator({ min: 10000, max: 99999, integer: true }),
+	EventTarget = require('./EventTarget');
+
+debugerror.log = console.warn.bind(console);
+
+function addTransceiverToPeerConnection(peerConnection, trackIdOrKind, init, transceiver) {
+	return new Promise((resolve, reject) => {
+		exec(onResultOK, reject, 'iosrtcPlugin', 'RTCPeerConnection_addTransceiver', [
+			peerConnection.pcId,
+			trackIdOrKind,
+			init,
+			transceiver._id,
+			transceiver.sender ? transceiver.sender._id : 0,
+			transceiver.receiver ? transceiver.receiver._id : 0,
+			transceiver.receiver && transceiver.receiver.track
+				? transceiver.receiver.track.id
+				: null
+		]);
+
+		function onResultOK(data) {
+			resolve(data);
+		}
+	});
+}
+
+function RTCRtpTransceiver(peerConnection, data) {
+	data = data || {};
+
+	this.peerConnection = peerConnection;
+
+	this._id = data.id || randomNumber();
+	this._receiver =
+		data.receiver instanceof RTCRtpReceiver
+			? data.receiver
+			: new RTCRtpReceiver(peerConnection, data.receiver);
+	this._sender =
+		data.sender instanceof RTCRtpSender
+			? data.sender
+			: new RTCRtpSender(peerConnection, data.sender);
+	this._stopped = false;
+
+	if (data.mid) {
+		this._mid = data.mid;
+	} else {
+		this._mid = null;
+	}
+
+	if (data.direction) {
+		this._direction = data.direction;
+		this._currentDirection = data.direction;
+	} else {
+		this._direction = 'sendrecv';
+		this._currentDirection = 'sendrecv';
+	}
+}
+
+RTCRtpTransceiver.prototype = Object.create(EventTarget.prototype);
+RTCRtpTransceiver.prototype.constructor = RTCRtpTransceiver;
+
+Object.defineProperties(RTCRtpTransceiver.prototype, {
+	currentDirection: {
+		get: function () {
+			return this._currentDirection;
+		}
+	},
+	direction: {
+		get: function () {
+			return this._direction;
+		},
+		set: function (direction) {
+			var self = this;
+			this._direction = direction;
+
+			exec(
+				onResultOK,
+				null,
+				'iosrtcPlugin',
+				'RTCPeerConnection_RTCRtpTransceiver_setDirection',
+				[this.peerConnection.pcId, this._id, direction]
+			);
+
+			function onResultOK(data) {
+				self.peerConnection.updateTransceiversState(data.transceivers);
+			}
+		}
+	},
+	mid: {
+		get: function () {
+			return this._mid;
+		}
+	},
+	receiver: {
+		get: function () {
+			return this._receiver;
+		}
+	},
+	sender: {
+		get: function () {
+			return this._sender;
+		}
+	},
+	stopped: {
+		get: function () {
+			return this._stopped;
+		}
+	},
+	receiverId: {
+		get: function () {
+			return this._receiver.track.id;
+		}
+	}
+});
+
+RTCRtpTransceiver.prototype.stop = function () {
+	var self = this;
+
+	exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCRtpTransceiver_stop', [
+		this.peerConnection.pcId,
+		this._id
+	]);
+
+	function onResultOK(data) {
+		self.peerConnection.updateTransceiversState(data.transceivers);
+	}
+};
+
+RTCRtpTransceiver.prototype.update = function (data) {
+	if (data.direction) {
+		this._direction = data.direction;
+	}
+	if (data.currentDirection) {
+		this._currentDirection = data.currentDirection;
+	}
+	if (data.mid) {
+		this._mid = data.mid;
+	}
+
+	this._stopped = data.stopped;
+
+	this.receiver.update(data.receiver);
+	this.sender.update(data.sender);
+};
diff --git a/js/RTCSessionDescription.js b/js/RTCSessionDescription.js
index 58a277bd..43549fac 100644
--- a/js/RTCSessionDescription.js
+++ b/js/RTCSessionDescription.js
@@ -3,7 +3,6 @@
  */
 module.exports = RTCSessionDescription;
 
-
 function RTCSessionDescription(data) {
 	data = data || {};
 
diff --git a/js/RTCStatsReport.js b/js/RTCStatsReport.js
index 895fa4ba..15b26e34 100644
--- a/js/RTCStatsReport.js
+++ b/js/RTCStatsReport.js
@@ -1,20 +1,39 @@
-/**
- * Expose the RTCStatsReport class.
- */
-module.exports = RTCStatsReport;
-
-function RTCStatsReport(data) {
-	data = data || [];
+class RTCStatsReport {
+	constructor(data) {
+		this.data = new Map((data || []).map((el) => [el.reportId, el]));
+	}
 
-	this.id = data.reportId;
-	this.timestamp = data.timestamp;
-	this.type = data.type;
+	get size() {
+		return this.data.size;
+	}
 
-	this.names = function () {
-		return Object.keys(data.values);
-	};
+	has(key) {
+		return this.data.has(key);
+	}
+	get(key) {
+		return this.data.get(key);
+	}
+	forEach(callbackfn, thisArg) {
+		return this.data.forEach(callbackfn, thisArg);
+	}
 
-	this.stat = function (key) {
-		return data.values[key] || '';
-	};
+	keys() {
+		return this.data.keys();
+	}
+	values() {
+		return this.data.values();
+	}
+	entries() {
+		return this.data.entries();
+	}
+	*[Symbol.iterator]() {
+		for (const value of this.data) {
+			yield value;
+		}
+	}
 }
+
+/**
+ * Expose the RTCStatsReport class.
+ */
+module.exports = RTCStatsReport;
diff --git a/js/RTCStatsResponse.js b/js/RTCStatsResponse.js
index a8331282..fd85fed9 100644
--- a/js/RTCStatsResponse.js
+++ b/js/RTCStatsResponse.js
@@ -10,7 +10,15 @@ function RTCStatsResponse(data) {
 		return data;
 	};
 
+	this.forEach = function (callback, thisArg) {
+		return data.forEach(callback, thisArg);
+	};
+
 	this.namedItem = function () {
 		return null;
 	};
+
+	this.values = function () {
+		return data;
+	};
 }
diff --git a/js/enumerateDevices.js b/js/enumerateDevices.js
index 7f8710bc..cfda0052 100644
--- a/js/enumerateDevices.js
+++ b/js/enumerateDevices.js
@@ -3,54 +3,33 @@
  */
 module.exports = enumerateDevices;
 
-
 /**
  * Dependencies.
  */
-var
-	debug = require('debug')('iosrtc:enumerateDevices'),
+var debug = require('debug')('iosrtc:enumerateDevices'),
 	exec = require('cordova/exec'),
-	MediaDeviceInfo = require('./MediaDeviceInfo');
-
+	MediaDeviceInfo = require('./MediaDeviceInfo'),
+	Errors = require('./Errors');
 
 function enumerateDevices() {
-	debug('');
-
-	var isPromise,
-		callback;
-
-	if (typeof arguments[0] !== 'function') {
-		isPromise = true;
-	} else {
-		isPromise = false;
-		callback = arguments[0];
-	}
-
-	if (isPromise) {
-		return new Promise(function (resolve) {
-			function onResultOK(data) {
-				debug('enumerateDevices() | success');
-				resolve(getMediaDeviceInfos(data.devices));
-			}
-
-			exec(onResultOK, null, 'iosrtcPlugin', 'enumerateDevices', []);
-		});
-	}
-
-	function onResultOK(data) {
-		debug('enumerateDevices() | success');
-		callback(getMediaDeviceInfos(data.devices));
-	}
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('cordova.plugins.iosrtc.enumerateDevices', arguments);
+
+	return new Promise(function (resolve) {
+		function onResultOK(data) {
+			debug('enumerateDevices() | success');
+			resolve(getMediaDeviceInfos(data.devices));
+		}
 
-	exec(onResultOK, null, 'iosrtcPlugin', 'enumerateDevices', []);
+		exec(onResultOK, null, 'iosrtcPlugin', 'enumerateDevices', []);
+	});
 }
 
-
 /**
  * Private API.
  */
 
-
 function getMediaDeviceInfos(devices) {
 	debug('getMediaDeviceInfos() | [devices:%o]', devices);
 
diff --git a/js/getUserMedia.js b/js/getUserMedia.js
index 75e86a11..f6e6fc85 100644
--- a/js/getUserMedia.js
+++ b/js/getUserMedia.js
@@ -3,20 +3,15 @@
  */
 module.exports = getUserMedia;
 
-
 /**
  * Dependencies.
  */
-var
-	debug = require('debug')('iosrtc:getUserMedia'),
+var debug = require('debug')('iosrtc:getUserMedia'),
 	debugerror = require('debug')('iosrtc:ERROR:getUserMedia'),
 	exec = require('cordova/exec'),
 	MediaStream = require('./MediaStream'),
 	Errors = require('./Errors');
 
-debugerror.log = console.warn.bind(console);
-
-
 function isPositiveInteger(number) {
 	return typeof number === 'number' && number >= 0 && number % 1 === 0;
 }
@@ -25,148 +20,440 @@ function isPositiveFloat(number) {
 	return typeof number === 'number' && number >= 0;
 }
 
-
 function getUserMedia(constraints) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('cordova.plugins.iosrtc.getUserMedia', arguments);
+
 	debug('[original constraints:%o]', constraints);
 
-	var
-		isPromise,
-		callback, errback,
-		audioRequested = false,
+	var audioRequested = false,
 		videoRequested = false,
-		newConstraints = {
-			audio: false,
-			video: false
-		};
-
-	if (typeof arguments[1] !== 'function') {
-		isPromise = true;
-	} else {
-		isPromise = false;
-		callback = arguments[1];
-		errback = arguments[2];
-	}
+		newConstraints = {};
 
 	if (
 		typeof constraints !== 'object' ||
 		(!constraints.hasOwnProperty('audio') && !constraints.hasOwnProperty('video'))
 	) {
-		if (isPromise) {
-			return new Promise(function (resolve, reject) {
-				reject(new Errors.MediaStreamError('constraints must be an object with at least "audio" or "video" keys'));
-			});
-		} else {
-			if (typeof errback === 'function') {
-				errback(new Errors.MediaStreamError('constraints must be an object with at least "audio" or "video" keys'));
-			}
-			return;
-		}
+		return new Promise(function (resolve, reject) {
+			reject(
+				new Errors.MediaStreamError(
+					'constraints must be an object with at least "audio" or "video" keys'
+				)
+			);
+		});
 	}
 
 	if (constraints.audio) {
 		audioRequested = true;
-		newConstraints.audio = true;
 	}
+
 	if (constraints.video) {
 		videoRequested = true;
-		newConstraints.video = true;
 	}
 
+	// https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints
 	// Example:
 	//
 	// getUserMedia({
-	//  audio: true,
+	//  audio: {
+	//      deviceId: 'azer-asdf-zxcv',
+	//  },
 	//  video: {
-	//  	deviceId: 'qwer-asdf-zxcv',
-	//  	width: {
-	//  		min: 400,
-	//  		max: 600
-	//  	},
-	//  	frameRate: {
-	//  		min: 1.0,
-	//  		max: 60.0
-	//  	}
+	//      deviceId: 'qwer-asdf-zxcv',
+	//      aspectRatio: 1.777.
+	//      facingMode: 'user',
+	//      width: {
+	//          min: 400,
+	//          max: 600
+	//      },
+	//      frameRate: {
+	//          min: 1.0,
+	//          max: 60.0
+	//      }
 	//  }
 	// });
 
+	/*
+	// See: https://www.w3.org/TR/mediacapture-streams/#media-track-constraints
+	dictionary MediaTrackConstraintSet {
+	 ConstrainULong     width;
+	 ConstrainULong     height;
+	 ConstrainDouble    aspectRatio;
+	 ConstrainDouble    frameRate;
+	 ConstrainDOMString facingMode;
+	 ConstrainDOMString resizeMode;
+	 ConstrainULong     sampleRate;
+	 ConstrainULong     sampleSize;
+	 ConstrainBoolean   echoCancellation;
+	 ConstrainBoolean   autoGainControl;
+	 ConstrainBoolean   noiseSuppression;
+	 ConstrainDouble    latency;
+	 ConstrainULong     channelCount;
+	 ConstrainDOMString deviceId;
+	 ConstrainDOMString groupId;
+	};
+
+	 // typedef ([Clamp] unsigned long or ConstrainULongRange) ConstrainULong;
+	 // We convert unsigned long to ConstrainULongRange.exact
+
+	 dictionary ULongRange {
+		[Clamp] unsigned long max;
+		[Clamp] unsigned long min;
+	 };
+
+	 dictionary ConstrainULongRange : ULongRange {
+		  [Clamp] unsigned long exact;
+		  [Clamp] unsigned long ideal;
+	 };
+
+	 // See: https://www.w3.org/TR/mediacapture-streams/#dom-doublerange
+	 // typedef (double or ConstrainDoubleRange) ConstrainDouble;
+	 // We convert double to ConstrainDoubleRange.exact
+	 dictionary ConstrainDouble {
+		double max;
+		double min;
+	 };
+
+	 dictionary ConstrainDoubleRange : DoubleRange {
+		double exact;
+		double ideal;
+	 };
+
+	 // typedef (boolean or ConstrainBooleanParameters) ConstrainBoolean;
+	 dictionary ConstrainBooleanParameters {
+		boolean exact;
+		boolean ideal;
+	 };
+
+	 // typedef (DOMString or sequence<DOMString> or ConstrainDOMStringParameters) ConstrainDOMString;
+	 // We convert DOMString to ConstrainDOMStringParameters.exact
+	 dictionary ConstrainDOMStringParameters {
+		(DOMString or sequence<DOMString>) exact;
+		(DOMString or sequence<DOMString>) ideal;
+	 };
+	*/
+
 	// Get video constraints
 	if (videoRequested) {
+		// Handle object video constraints
+		newConstraints.video = {};
+
+		// Handle Stupid not up-to-date webrtc-adapter
+		// Note: Firefox [38+] does support a subset of constraints with getUserMedia(), but not the outdated syntax that Chrome and Opera are using.
+		// The mandatory / optional syntax was deprecated a in 2014, and minWidth and minHeight the year before that.
+		if (
+			typeof constraints.video === 'object' &&
+			(typeof constraints.video.optional === 'object' ||
+				typeof constraints.video.mandatory === 'object')
+		) {
+			var videoConstraints = {};
+
+			if (Array.isArray(constraints.video.optional)) {
+				/*
+				Example of constraints.video.optional:
+					{
+						"optional": [
+							{
+								"minWidth": 640
+							},
+							{
+								"maxWidth": 640
+							},
+							{
+								"minHeight": 480
+							},
+							{
+								"maxHeight": 480
+							},
+							{
+								"sourceId": "com.apple.avfoundation.avcapturedevice.built-in_video:0"
+							}
+						]
+					}
+				*/
+
+				// Convert optional array to object
+				Object.values(constraints.video.optional).forEach(function (optional) {
+					var optionalConstraintName = Object.keys(optional)[0],
+						optionalConstraintValue = optional[optionalConstraintName];
+					videoConstraints[optionalConstraintName] = Array.isArray(
+						optionalConstraintValue
+					)
+						? optionalConstraintValue[0]
+						: optionalConstraintValue;
+				});
+			} else if (typeof constraints.video.mandatory === 'object') {
+				videoConstraints = constraints.video.mandatory;
+			}
+
+			if (typeof videoConstraints.sourceId === 'string') {
+				newConstraints.video.deviceId = {
+					ideal: videoConstraints.sourceId
+				};
+			}
+
+			if (isPositiveInteger(videoConstraints.minWidth)) {
+				newConstraints.video.width = {
+					min: videoConstraints.minWidth
+				};
+			}
+
+			if (isPositiveInteger(videoConstraints.maxWidth)) {
+				newConstraints.video.width = newConstraints.video.width || {};
+				newConstraints.video.width.max = videoConstraints.maxWidth;
+			}
+
+			if (isPositiveInteger(videoConstraints.minHeight)) {
+				newConstraints.video.height = {
+					min: videoConstraints.minHeight
+				};
+			}
+
+			if (isPositiveInteger(videoConstraints.maxHeight)) {
+				newConstraints.video.height = newConstraints.video.height || {};
+				newConstraints.video.height.max = videoConstraints.maxHeight;
+			}
+
+			if (isPositiveFloat(videoConstraints.minFrameRate)) {
+				newConstraints.video.frameRate = {
+					min: parseFloat(videoConstraints.minFrameRate, 10)
+				};
+			}
+
+			if (isPositiveFloat(videoConstraints.maxFrameRate)) {
+				newConstraints.video.frameRate = newConstraints.video.frameRate || {};
+				newConstraints.video.frameRate.max = parseFloat(videoConstraints.maxFrameRate, 10);
+			}
+		}
+
 		// Get requested video deviceId.
 		if (typeof constraints.video.deviceId === 'string') {
-			newConstraints.videoDeviceId = constraints.video.deviceId;
+			newConstraints.video.deviceId = {
+				exact: constraints.video.deviceId
+			};
+		} else if (typeof constraints.video.sourceId === 'string') {
+			// Also check video sourceId (mangled by adapter.js).
+			newConstraints.video.deviceId = {
+				exact: constraints.video.sourceId
+			};
+		} else if (typeof constraints.video.deviceId === 'object') {
+			// Also check deviceId.(exact|ideal)
+			if (!!constraints.video.deviceId.exact) {
+				newConstraints.video.deviceId = {
+					exact: Array.isArray(constraints.video.deviceId.exact)
+						? constraints.video.deviceId.exact[0]
+						: constraints.video.deviceId.exact
+				};
+			} else if (!!constraints.video.deviceId.ideal) {
+				newConstraints.video.deviceId = {
+					ideal: Array.isArray(constraints.video.deviceId.ideal)
+						? constraints.video.deviceId.ideal[0]
+						: constraints.video.deviceId.ideal
+				};
+			}
 		}
 
-		// Get requested min/max width.
+		// Get requested width min/max, exact.
 		if (typeof constraints.video.width === 'object') {
+			newConstraints.video.width = {};
 			if (isPositiveInteger(constraints.video.width.min)) {
-				newConstraints.videoMinWidth = constraints.video.width.min;
+				newConstraints.video.width.min = constraints.video.width.min;
 			}
 			if (isPositiveInteger(constraints.video.width.max)) {
-				newConstraints.videoMaxWidth = constraints.video.width.max;
+				newConstraints.video.width.max = constraints.video.width.max;
 			}
+			if (isPositiveInteger(constraints.video.width.exact)) {
+				newConstraints.video.width.exact = constraints.video.width.exact;
+			}
+			if (isPositiveInteger(constraints.video.width.ideal)) {
+				newConstraints.video.width.ideal = constraints.video.width.ideal;
+			}
+		} else if (isPositiveInteger(constraints.video.width)) {
+			// Get requested width long as exact
+			newConstraints.video.width = {
+				exact: constraints.video.width
+			};
 		}
-		// Get requested min/max height.
+
+		// Get requested height min/max, exact.
 		if (typeof constraints.video.height === 'object') {
+			newConstraints.video.height = {};
 			if (isPositiveInteger(constraints.video.height.min)) {
-				newConstraints.videoMinHeight = constraints.video.height.min;
+				newConstraints.video.height.min = constraints.video.height.min;
 			}
 			if (isPositiveInteger(constraints.video.height.max)) {
-				newConstraints.videoMaxHeight = constraints.video.height.max;
+				newConstraints.video.height.max = constraints.video.height.max;
+			}
+			if (isPositiveInteger(constraints.video.height.exact)) {
+				newConstraints.video.height.exact = constraints.video.height.exact;
 			}
+			if (isPositiveInteger(constraints.video.height.ideal)) {
+				newConstraints.video.height.ideal = constraints.video.height.ideal;
+			}
+		} else if (isPositiveInteger(constraints.video.height)) {
+			// Get requested height long as exact
+			newConstraints.video.height = {
+				exact: constraints.video.height
+			};
 		}
-		// Get requested min/max frame rate.
+
+		// Get requested frameRate min/max.
 		if (typeof constraints.video.frameRate === 'object') {
+			newConstraints.video.frameRate = {};
 			if (isPositiveFloat(constraints.video.frameRate.min)) {
-				newConstraints.videoMinFrameRate = constraints.video.frameRate.min;
+				newConstraints.video.frameRate.min = parseFloat(
+					constraints.video.frameRate.min,
+					10
+				);
 			}
 			if (isPositiveFloat(constraints.video.frameRate.max)) {
-				newConstraints.videoMaxFrameRate = constraints.video.frameRate.max;
+				newConstraints.video.frameRate.max = parseFloat(
+					constraints.video.frameRate.max,
+					10
+				);
+			}
+			if (isPositiveInteger(constraints.video.frameRate.exact)) {
+				newConstraints.video.frameRate.exact = constraints.video.frameRate.exact;
+			}
+			if (isPositiveInteger(constraints.video.frameRate.ideal)) {
+				newConstraints.video.frameRate.ideal = constraints.video.frameRate.ideal;
 			}
 		} else if (isPositiveFloat(constraints.video.frameRate)) {
-			newConstraints.videoMinFrameRate = constraints.video.frameRate;
-			newConstraints.videoMaxFrameRate = constraints.video.frameRate;
+			// Get requested frameRate double as exact
+			newConstraints.video.frameRate = {
+				exact: parseFloat(constraints.video.frameRate, 10)
+			};
 		}
-	}
-
-	debug('[computed constraints:%o]', newConstraints);
 
-	if (isPromise) {
-		return new Promise(function (resolve, reject) {
-			function onResultOK(data) {
-				debug('getUserMedia() | success');
-				var stream = MediaStream.create(data.stream);
-				resolve(stream);
-				// Emit "connected" on the stream.
-				stream.emitConnected();
+		// get aspectRatio (e.g 1.7777777777777777)
+		// TODO ConstrainDouble min, max
+		if (typeof constraints.video.aspectRatio === 'object') {
+			newConstraints.video.aspectRatio = {};
+			if (isPositiveFloat(constraints.video.aspectRatio.min)) {
+				newConstraints.video.aspectRatio.min = parseFloat(
+					constraints.video.aspectRatio.min,
+					10
+				);
 			}
-
-			function onResultError(error) {
-				debugerror('getUserMedia() | failure: %s', error);
-				reject(new Errors.MediaStreamError('getUserMedia() failed: ' + error));
+			if (isPositiveFloat(constraints.video.aspectRatio.max)) {
+				newConstraints.video.aspectRatio.max = parseFloat(
+					constraints.video.aspectRatio.max,
+					10
+				);
+			}
+			if (isPositiveInteger(constraints.video.aspectRatio.exact)) {
+				newConstraints.video.aspectRatio.exact = constraints.video.aspectRatio.exact;
 			}
+			if (isPositiveInteger(constraints.video.aspectRatio.ideal)) {
+				newConstraints.video.aspectRatio.ideal = constraints.video.aspectRatio.ideal;
+			}
+		} else if (isPositiveFloat(constraints.video.aspectRatio)) {
+			newConstraints.video.aspectRatio = {
+				exact: parseFloat(constraints.video.aspectRatio, 10)
+			};
+		}
 
-			exec(onResultOK, onResultError, 'iosrtcPlugin', 'getUserMedia', [newConstraints]);
-		});
+		// get facingMode (e.g environment, user)
+		// TODO ConstrainDOMStringParameters ideal, exact
+		if (typeof constraints.video.facingMode === 'string') {
+			newConstraints.video.facingMode = {
+				exact: constraints.video.facingMode
+			};
+		} else if (typeof constraints.video.facingMode === 'object') {
+			if (typeof constraints.video.facingMode.exact === 'string') {
+				newConstraints.video.facingMode = {
+					exact: constraints.video.facingMode.exact
+				};
+			} else if (typeof constraints.video.facingMode.ideal === 'string') {
+				newConstraints.video.facingMode = {
+					ideal: constraints.video.facingMode.ideal
+				};
+			}
+		}
 	}
 
-	function onResultOK(data) {
-		debug('getUserMedia() | success');
-
-		var stream = MediaStream.create(data.stream);
+	// Get audio constraints
+	if (audioRequested) {
+		// Handle object audio constraints
+		newConstraints.audio = {};
 
-		callback(stream);
+		// Handle Stupid not up-to-date webrtc-adapter
+		// Note: Firefox [38+] does support a subset of constraints with getUserMedia(), but not the outdated syntax that Chrome and Opera are using.
+		// The mandatory / optional syntax was deprecated a in 2014, and minWidth and minHeight the year before that.
+		if (
+			typeof constraints.audio === 'object' &&
+			(typeof constraints.audio.optional === 'object' ||
+				typeof constraints.audio.mandatory === 'object')
+		) {
+			if (typeof constraints.audio.optional === 'object') {
+				if (typeof constraints.audio.optional.sourceId === 'string') {
+					newConstraints.audio.deviceId = {
+						ideal: constraints.audio.optional.sourceId
+					};
+				} else if (
+					Array.isArray(constraints.audio.optional) &&
+					typeof constraints.audio.optional[0] === 'object' &&
+					typeof constraints.audio.optional[0].sourceId === 'string'
+				) {
+					newConstraints.audio.deviceId = {
+						ideal: constraints.audio.optional[0].sourceId
+					};
+				}
+			} else if (
+				constraints.audio.mandatory &&
+				typeof constraints.audio.mandatory.sourceId === 'string'
+			) {
+				newConstraints.audio.deviceId = {
+					exact: constraints.audio.mandatory.sourceId
+				};
+			}
+		}
 
-		// Emit "connected" on the stream.
-		stream.emitConnected();
+		// Get requested audio deviceId.
+		if (typeof constraints.audio.deviceId === 'string') {
+			newConstraints.audio.deviceId = {
+				exact: constraints.audio.deviceId
+			};
+		} else if (typeof constraints.audio.sourceId === 'string') {
+			// Also check audio sourceId (mangled by adapter.js).
+			newConstraints.audio.deviceId = {
+				exact: constraints.audio.sourceId
+			};
+		} else if (typeof constraints.audio.deviceId === 'object') {
+			// Also check deviceId.(exact|ideal)
+			if (!!constraints.audio.deviceId.exact) {
+				newConstraints.audio.deviceId = {
+					exact: Array.isArray(constraints.audio.deviceId.exact)
+						? constraints.audio.deviceId.exact[0]
+						: constraints.audio.deviceId.exact
+				};
+			} else if (!!constraints.audio.deviceId.ideal) {
+				newConstraints.audio.deviceId = {
+					ideal: Array.isArray(constraints.audio.deviceId.ideal)
+						? constraints.audio.deviceId.ideal[0]
+						: constraints.audio.deviceId.ideal
+				};
+			}
+		}
 	}
 
-	function onResultError(error) {
-		debugerror('getUserMedia() | failure: %s', error);
+	debug('[computed constraints:%o]', newConstraints);
 
-		if (typeof errback === 'function') {
-			errback(new Errors.MediaStreamError('getUserMedia() failed: ' + error));
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			debug('getUserMedia() | success');
+			var stream = MediaStream.create(data.stream);
+			resolve(stream);
+			// Emit "connected" on the stream.
+			stream.emitConnected();
+		}
+
+		function onResultError(error) {
+			debugerror('getUserMedia() | failure: %s', error);
+			reject(new Errors.MediaStreamError('getUserMedia() failed: ' + error));
 		}
-	}
 
-	exec(onResultOK, onResultError, 'iosrtcPlugin', 'getUserMedia', [newConstraints]);
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'getUserMedia', [newConstraints]);
+	});
 }
diff --git a/js/iosrtc.js b/js/iosrtc.js
index 38b3d2ec..2e69cbf6 100644
--- a/js/iosrtc.js
+++ b/js/iosrtc.js
@@ -2,92 +2,92 @@
  * Variables.
  */
 
-var
-	// Dictionary of MediaStreamRenderers.
+var // Dictionary of MediaStreamRenderers.
 	// - key: MediaStreamRenderer id.
 	// - value: MediaStreamRenderer.
 	mediaStreamRenderers = {},
-
 	// Dictionary of MediaStreams.
 	// - key: MediaStream blobId.
 	// - value: MediaStream.
 	mediaStreams = {},
-
-
-/**
- * Dependencies.
- */
-	debug                  = require('debug')('iosrtc'),
-	exec                   = require('cordova/exec'),
-	domready               = require('domready'),
-
-	getUserMedia           = require('./getUserMedia'),
-	enumerateDevices       = require('./enumerateDevices'),
-	RTCPeerConnection      = require('./RTCPeerConnection'),
-	RTCSessionDescription  = require('./RTCSessionDescription'),
-	RTCIceCandidate        = require('./RTCIceCandidate'),
-	MediaStream            = require('./MediaStream'),
-	MediaStreamTrack       = require('./MediaStreamTrack'),
-	videoElementsHandler   = require('./videoElementsHandler'),
-	rtcninjaPlugin         = require('./rtcninjaPlugin');
-
+	/**
+	 * Dependencies.
+	 */
+	debug = require('debug')('iosrtc'),
+	exec = require('cordova/exec'),
+	domready = require('domready'),
+	getUserMedia = require('./getUserMedia'),
+	enumerateDevices = require('./enumerateDevices'),
+	RTCPeerConnection = require('./RTCPeerConnection'),
+	RTCSessionDescription = require('./RTCSessionDescription'),
+	RTCIceCandidate = require('./RTCIceCandidate'),
+	MediaDevices = require('./MediaDevices'),
+	MediaStream = require('./MediaStream'),
+	{ MediaStreamTrack } = require('./MediaStreamTrack'),
+	videoElementsHandler = require('./videoElementsHandler'),
+	{ RTCRtpTransceiver } = require('./RTCRtpTransceiver');
 
 /**
  * Expose the iosrtc object.
  */
 module.exports = {
 	// Expose WebRTC classes and functions.
-	getUserMedia:          getUserMedia,
-	enumerateDevices:      enumerateDevices,
-	getMediaDevices:       enumerateDevices,  // TMP
-	RTCPeerConnection:     RTCPeerConnection,
+	getUserMedia: getUserMedia,
+	enumerateDevices: enumerateDevices,
+	getMediaDevices: enumerateDevices, // TMP
+	RTCPeerConnection: RTCPeerConnection,
 	RTCSessionDescription: RTCSessionDescription,
-	RTCIceCandidate:       RTCIceCandidate,
-	MediaStream:           MediaStream,
-	MediaStreamTrack:      MediaStreamTrack,
+	RTCIceCandidate: RTCIceCandidate,
+	MediaDevices: MediaDevices,
+	MediaStream: MediaStream,
+	MediaStreamTrack: MediaStreamTrack,
 
 	// Expose a function to refresh current videos rendering a MediaStream.
-	refreshVideos:         refreshVideos,
+	refreshVideos: videoElementsHandler.refreshVideos,
 
 	// Expose a function to handle a video not yet inserted in the DOM.
-	observeVideo:          videoElementsHandler.observeVideo,
+	observeVideo: videoElementsHandler.observeVideo,
 
 	// Select audio output (earpiece or speaker).
-	selectAudioOutput:     selectAudioOutput,
+	selectAudioOutput: selectAudioOutput,
 
-	// Expose a function to pollute window and naigator namespaces.
-	registerGlobals:       registerGlobals,
+	// Set default audio output (earpiece or speaker).
+	setDefaultAudioOutput: setDefaultAudioOutput,
+
+	// turnOnSpeaker with options
+	turnOnSpeaker: turnOnSpeaker,
 
-	// Expose the rtcninjaPlugin module.
-	rtcninjaPlugin:        rtcninjaPlugin,
+	// Checking permission (audio and camera)
+	requestPermission: requestPermission,
+
+	// Expose a function to initAudioDevices if needed, sets the audio session active
+	initAudioDevices: initAudioDevices,
+
+	// Expose a function to pollute window and naigator namespaces.
+	registerGlobals: registerGlobals,
 
 	// Expose the debug module.
-	debug:                 require('debug'),
+	debug: require('debug'),
 
 	// Debug function to see what happens internally.
-	dump:                  dump
-};
+	dump: dump,
 
+	// Debug Stores to see what happens internally.
+	mediaStreamRenderers: mediaStreamRenderers,
+	mediaStreams: mediaStreams
+};
 
 domready(function () {
 	// Let the MediaStream class and the videoElementsHandler share same MediaStreams container.
 	MediaStream.setMediaStreams(mediaStreams);
 	videoElementsHandler(mediaStreams, mediaStreamRenderers);
-});
-
-
-function refreshVideos() {
-	debug('refreshVideos()');
-
-	var id;
-
-	for (id in mediaStreamRenderers) {
-		if (mediaStreamRenderers.hasOwnProperty(id)) {
-			mediaStreamRenderers[id].refresh();
-		}
-	}
-}
 
+	// refreshVideos on device orientation change to resize peers video
+	// while local video will resize du orientation change
+	window.addEventListener('resize', function () {
+		videoElementsHandler.refreshVideos();
+	});
+});
 
 function selectAudioOutput(output) {
 	debug('selectAudioOutput() | [output:"%s"]', output);
@@ -104,29 +104,196 @@ function selectAudioOutput(output) {
 	}
 }
 
+function setDefaultAudioOutput(output) {
+	debug('setDefaultAudioOutput() | [output:"%s"]', output);
+	if (!['earpiece', 'speaker'].includes(output)) {
+		throw new Error('output must be "earpiece" or "speaker"');
+	}
+	exec(null, null, 'iosrtcPlugin', 'setDefaultAudioOutput', [output === 'speaker']);
+}
+
+function turnOnSpeaker(isTurnOn) {
+	debug('turnOnSpeaker() | [isTurnOn:"%s"]', isTurnOn);
+
+	exec(null, null, 'iosrtcPlugin', 'RTCTurnOnSpeaker', [isTurnOn]);
+}
+
+function requestPermission(needMic, needCamera, callback) {
+	debug('requestPermission() | [needMic:"%s", needCamera:"%s"]', needMic, needCamera);
+
+	function ok() {
+		callback(true);
+	}
+
+	function error() {
+		callback(false);
+	}
+	exec(ok, error, 'iosrtcPlugin', 'RTCRequestPermission', [needMic, needCamera]);
+}
+
+function initAudioDevices() {
+	debug('initAudioDevices()');
+
+	exec(null, null, 'iosrtcPlugin', 'initAudioDevices', []);
+}
+
+function callbackifyMethod(originalMethod) {
+	return function () {
+		var success,
+			failure,
+			originalArgs = Array.prototype.slice.call(arguments);
+
+		var callbackArgs = [];
+		originalArgs.forEach(function (arg) {
+			if (typeof arg === 'function') {
+				if (!success) {
+					success = arg;
+				} else {
+					failure = arg;
+				}
+			} else {
+				callbackArgs.push(arg);
+			}
+		});
+
+		var promiseResult = originalMethod.apply(this, callbackArgs);
+
+		// Only apply then if callback success available
+		if (typeof success === 'function') {
+			promiseResult = promiseResult.then(success);
+		}
+
+		// Only apply catch if callback failure available
+		if (typeof failure === 'function') {
+			promiseResult = promiseResult.catch(failure);
+		}
+
+		return promiseResult;
+	};
+}
+
+function callbackifyPrototype(proto, method) {
+	var originalMethod = proto[method];
+	proto[method] = callbackifyMethod(originalMethod);
+}
+
+function restoreCallbacksSupport() {
+	debug('restoreCallbacksSupport()');
+	getUserMedia = callbackifyMethod(getUserMedia);
+	enumerateDevices = callbackifyMethod(enumerateDevices);
+	callbackifyPrototype(RTCPeerConnection.prototype, 'createAnswer');
+	callbackifyPrototype(RTCPeerConnection.prototype, 'createOffer');
+	callbackifyPrototype(RTCPeerConnection.prototype, 'setRemoteDescription');
+	callbackifyPrototype(RTCPeerConnection.prototype, 'setLocalDescription');
+	callbackifyPrototype(RTCPeerConnection.prototype, 'addIceCandidate');
+	callbackifyPrototype(RTCPeerConnection.prototype, 'getStats');
+}
+
+function registerGlobals(doNotRestoreCallbacksSupport) {
+	debug('registerGlobals()');
 
-function registerGlobals() {
 	if (!global.navigator) {
 		global.navigator = {};
 	}
 
-	if (!navigator.mediaDevices) {
-		navigator.mediaDevices = {};
+	// Restore Callback support
+	if (!doNotRestoreCallbacksSupport) {
+		restoreCallbacksSupport();
 	}
 
-	navigator.getUserMedia                  = getUserMedia;
-	navigator.webkitGetUserMedia            = getUserMedia;
-	navigator.mediaDevices.getUserMedia     = getUserMedia;
-	navigator.mediaDevices.enumerateDevices = enumerateDevices;
-	window.RTCPeerConnection                = RTCPeerConnection;
-	window.webkitRTCPeerConnection          = RTCPeerConnection;
-	window.RTCSessionDescription            = RTCSessionDescription;
-	window.RTCIceCandidate                  = RTCIceCandidate;
-	window.MediaStream                      = MediaStream;
-	window.webkitMediaStream                = MediaStream;
-	window.MediaStreamTrack                 = MediaStreamTrack;
-}
+	navigator.getUserMedia = getUserMedia;
+	navigator.webkitGetUserMedia = getUserMedia;
+
+	// Prevent WebRTC-adapter to overide navigator.mediaDevices after shim is applied since ios 14.3
+	if (!(navigator.mediaDevices instanceof MediaDevices)) {
+		Object.defineProperty(
+			navigator,
+			'mediaDevices',
+			{
+				value: new MediaDevices(),
+				writable: false
+			},
+			{
+				enumerable: false,
+				configurable: false,
+				writable: false,
+				value: 'static'
+			}
+		);
+	}
+
+	window.RTCPeerConnection = RTCPeerConnection;
+	window.webkitRTCPeerConnection = RTCPeerConnection;
+	window.RTCSessionDescription = RTCSessionDescription;
+	window.RTCIceCandidate = RTCIceCandidate;
+	window.MediaStream = MediaStream;
+	window.webkitMediaStream = MediaStream;
+	window.MediaStreamTrack = MediaStreamTrack;
+	window.RTCRtpTransceiver = RTCRtpTransceiver;
+	window.MediaStreamTrackEvent = window.Event;
+
+	// Apply CanvasRenderingContext2D.drawImage monkey patch
+	var drawImage = CanvasRenderingContext2D.prototype.drawImage;
+	CanvasRenderingContext2D.prototype.drawImage = (function () {
+		// Methods to address the memory leaks problems in Safari
+		let temporaryImage, imageElement;
+		const BASE64_MARKER = ';base64,';
+		const objectURL = window.URL || window.webkitURL;
+
+		function convertDataURIToBlob(dataURI) {
+			// Validate input data
+			if (!dataURI) {
+				return;
+			}
+
+			// Convert image (in base64) to binary data
+			const base64Index = dataURI.indexOf(BASE64_MARKER) + BASE64_MARKER.length;
+			const base64 = dataURI.substring(base64Index);
+			const raw = window.atob(base64);
+			const rawLength = raw.length;
+			let array = new Uint8Array(new ArrayBuffer(rawLength));
 
+			for (let i = 0; i < rawLength; i++) {
+				array[i] = raw.charCodeAt(i);
+			}
+
+			// Create and return a new blob object using binary data
+			return new Blob([array], { type: 'image/jpeg' });
+		}
+
+		return function (arg) {
+			const context = this;
+			let args = Array.prototype.slice.call(arguments);
+			if (arg instanceof HTMLVideoElement && arg.render) {
+				arg.render.save(function (base64Image) {
+					// Destroy old image
+					if (temporaryImage) {
+						objectURL.revokeObjectURL(temporaryImage);
+					}
+
+					// Create a new image from binary data
+					const imageDataBlob = convertDataURIToBlob(
+						'data:image/jpg;base64,' + base64Image
+					);
+
+					// Create a new object URL object
+					imageElement = imageElement || new Image();
+					temporaryImage = objectURL.createObjectURL(imageDataBlob);
+
+					imageElement.addEventListener('load', function () {
+						args.splice(0, 1, imageElement);
+						drawImage.apply(context, args);
+					});
+
+					// Set the new image
+					imageElement.src = temporaryImage;
+				});
+			} else {
+				return drawImage.apply(context, args);
+			}
+		};
+	})();
+}
 
 function dump() {
 	exec(null, null, 'iosrtcPlugin', 'dump', []);
diff --git a/js/rtcninjaPlugin.js b/js/rtcninjaPlugin.js
deleted file mode 100644
index 396b0199..00000000
--- a/js/rtcninjaPlugin.js
+++ /dev/null
@@ -1,30 +0,0 @@
-/**
- * Expose the rtcninjaPlugin object.
- */
-module.exports = {
-	isRequired: function () {
-		return true;
-	},
-
-	isInstalled: function () {
-		return true;
-	},
-
-	interface: {
-		getUserMedia:          require('./getUserMedia'),
-		enumerateDevices:      require('./enumerateDevices'),
-		getMediaDevices:       require('./enumerateDevices'),  // TMP
-		RTCPeerConnection:     require('./RTCPeerConnection'),
-		RTCSessionDescription: require('./RTCSessionDescription'),
-		RTCIceCandidate:       require('./RTCIceCandidate'),
-		MediaStreamTrack:      require('./MediaStreamTrack'),
-		attachMediaStream:     attachMediaStream,
-		canRenegotiate:        true
-	}
-};
-
-
-function attachMediaStream(element, stream) {
-	element.src = URL.createObjectURL(stream);
-	return element;
-}
diff --git a/js/videoElementsHandler.js b/js/videoElementsHandler.js
index 9b9a5c13..8ec57d9a 100644
--- a/js/videoElementsHandler.js
+++ b/js/videoElementsHandler.js
@@ -4,39 +4,30 @@
  */
 module.exports = videoElementsHandler;
 module.exports.observeVideo = observeVideo;
-
+module.exports.refreshVideos = refreshVideos;
 
 /**
  * Dependencies.
  */
 var debug = require('debug')('iosrtc:videoElementsHandler'),
 	MediaStreamRenderer = require('./MediaStreamRenderer'),
-
-
-/**
- * Local variables.
- */
-
-	// RegExp for MediaStream blobId.
-	MEDIASTREAM_ID_REGEXP = new RegExp(/^MediaStream_/),
+	/**
+	 * Local variables.
+	 */
 
 	// RegExp for Blob URI.
-	BLOB_URI_REGEX = new RegExp(/^blob:/),
-
+	BLOB_INTERNAL_URI_REGEX = new RegExp(/^blob:/),
 	// Dictionary of MediaStreamRenderers (provided via module argument).
 	// - key: MediaStreamRenderer id.
 	// - value: MediaStreamRenderer.
 	mediaStreamRenderers,
-
 	// Dictionary of MediaStreams (provided via module argument).
 	// - key: MediaStream blobId.
 	// - value: MediaStream.
 	mediaStreams,
-
 	// Video element mutation observer.
 	videoObserver = new MutationObserver(function (mutations) {
-		var i, numMutations, mutation,
-			video;
+		var i, numMutations, mutation, video;
 
 		for (i = 0, numMutations = mutations.length; i < numMutations; i++) {
 			mutation = mutations[i];
@@ -44,8 +35,8 @@ var debug = require('debug')('iosrtc:videoElementsHandler'),
 			// HTML video element.
 			video = mutation.target;
 
-			// .src removed.
-			if (!video.src) {
+			// .srcObject removed.
+			if (!video.srcObject && !video.src) {
 				// If this video element was previously handling a MediaStreamRenderer, release it.
 				releaseMediaStreamRenderer(video);
 				continue;
@@ -54,11 +45,9 @@ var debug = require('debug')('iosrtc:videoElementsHandler'),
 			handleVideo(video);
 		}
 	}),
-
 	// DOM mutation observer.
 	domObserver = new MutationObserver(function (mutations) {
-		var i, numMutations, mutation,
-			j, numNodes, node;
+		var i, numMutations, mutation, j, numNodes, node;
 
 		for (i = 0, numMutations = mutations.length; i < numMutations; i++) {
 			mutation = mutations[i];
@@ -125,10 +114,22 @@ var debug = require('debug')('iosrtc:videoElementsHandler'),
 		}
 	});
 
+function refreshVideos() {
+	debug('refreshVideos()');
+
+	var id;
+
+	for (id in mediaStreamRenderers) {
+		if (mediaStreamRenderers.hasOwnProperty(id)) {
+			mediaStreamRenderers[id].refresh();
+		}
+	}
+}
 
 function videoElementsHandler(_mediaStreams, _mediaStreamRenderers) {
 	var existingVideos = document.querySelectorAll('video'),
-		i, len,
+		i,
+		len,
 		video;
 
 	mediaStreams = _mediaStreams;
@@ -165,17 +166,17 @@ function videoElementsHandler(_mediaStreams, _mediaStreamRenderers) {
 	});
 }
 
-
 function observeVideo(video) {
 	debug('observeVideo()');
 
-	// If the video already has a src property but is not yet handled by the plugin
+	// If the video already has a srcObject property but is not yet handled by the plugin
 	// then handle it now.
-	if (video.src && !video._iosrtcMediaStreamRendererId) {
+	var hasStream = video.srcObject || video.src;
+	if (hasStream && !video._iosrtcMediaStreamRendererId) {
 		handleVideo(video);
 	}
 
-	// Add .src observer to the video element.
+	// Add .srcObject observer to the video element.
 	videoObserver.observe(video, {
 		// Set to true if additions and removals of the target node's child elements (including text
 		// nodes) are to be observed.
@@ -194,13 +195,46 @@ function observeVideo(video) {
 		characterDataOldValue: false,
 		// Set to an array of attribute local names (without namespace) if not all attribute mutations
 		// need to be observed.
-		// TODO: Add srcObject, mozSrcObject
-		attributeFilter: ['src']
+		// srcObject DO not trigger MutationObserver
+		attributeFilter: ['srcObject', 'src']
+	});
+
+	// MutationObserver fail to trigger when using srcObject on ony tested browser.
+	// But video.srcObject = new MediaStream() will trigger onloadstart and
+	// video.srcObject = null will trigger onemptied events.
+
+	video.addEventListener('loadstart', function () {
+		var hasStream = video.srcObject || video.src;
+
+		if (hasStream && !video._iosrtcMediaStreamRendererId) {
+			// If this video element was NOT previously handling a MediaStreamRenderer, release it.
+			handleVideo(video);
+		} else if (hasStream && video._iosrtcMediaStreamRendererId) {
+			// The video element has received a new srcObject.
+			var stream = video.srcObject;
+			if (stream && typeof stream.getBlobId === 'function') {
+				// Release previous renderer
+				releaseMediaStreamRenderer(video);
+				// Install new renderer
+				provideMediaStreamRenderer(video, stream.getBlobId());
+			}
+		}
+	});
+
+	video.addEventListener('emptied', function () {
+		var hasStream = video.srcObject || video.src;
+		if (!hasStream && video._iosrtcMediaStreamRendererId) {
+			// If this video element was previously handling a MediaStreamRenderer, release it.
+			releaseMediaStreamRenderer(video);
+		}
 	});
 
 	// Intercept video 'error' events if it's due to the attached MediaStream.
 	video.addEventListener('error', function (event) {
-		if (video.error.code === global.MediaError.MEDIA_ERR_SRC_NOT_SUPPORTED && BLOB_URI_REGEX.test(video.src)) {
+		if (
+			video.error.code === window.MediaError.MEDIA_ERR_SRC_NOT_SUPPORTED &&
+			BLOB_INTERNAL_URI_REGEX.test(video.src)
+		) {
 			debug('stopping "error" event propagation for video element');
 
 			event.stopImmediatePropagation();
@@ -208,52 +242,66 @@ function observeVideo(video) {
 	});
 }
 
-
 /**
  * Private API.
  */
 
 function handleVideo(video) {
-	var xhr = new XMLHttpRequest();
+	var stream;
 
-	xhr.open('GET', video.src, true);
-	xhr.responseType = 'blob';
-	xhr.onload = function () {
-		if (xhr.status !== 200) {
-			// If this video element was previously handling a MediaStreamRenderer, release it.
-			releaseMediaStreamRenderer(video);
-
-			return;
-		}
+	// The app has set video.srcObject.
+	if (video.srcObject) {
+		stream = video.srcObject;
+		if (stream && typeof stream.getBlobId === 'function') {
+			if (!stream.getBlobId()) {
+				// If this video element was previously handling a MediaStreamRenderer, release it.
+				releaseMediaStreamRenderer(video);
 
-		var reader = new FileReader();
+				return;
+			}
 
-		// Some versions of Safari fail to set onloadend property, some others do not react
-		// on 'loadend' event. Try everything here.
-		try {
-			reader.onloadend = onloadend;
-		} catch (error) {
-			reader.addEventListener('loadend', onloadend);
+			provideMediaStreamRenderer(video, stream.getBlobId());
 		}
-		reader.readAsText(xhr.response);
-
-		function onloadend() {
-			var mediaStreamBlobId = reader.result;
-
-			// The retrieved URL does not point to a MediaStream.
-			if (!mediaStreamBlobId || typeof mediaStreamBlobId !== 'string' || !MEDIASTREAM_ID_REGEXP.test(mediaStreamBlobId)) {
+	} else if (video.src) {
+		var xhr = new XMLHttpRequest();
+		xhr.open('GET', video.src, true);
+		xhr.responseType = 'blob';
+		xhr.onload = function () {
+			if (xhr.status !== 200) {
 				// If this video element was previously handling a MediaStreamRenderer, release it.
 				releaseMediaStreamRenderer(video);
 
 				return;
 			}
 
-			provideMediaStreamRenderer(video, mediaStreamBlobId);
-		}
-	};
-	xhr.send();
-}
+			var reader = new FileReader();
+
+			// Some versions of Safari fail to set onloadend property, some others do not react
+			// on 'loadend' event. Try everything here.
+			try {
+				reader.onloadend = onloadend;
+			} catch (error) {
+				reader.addEventListener('loadend', onloadend);
+			}
+			reader.readAsText(xhr.response);
 
+			function onloadend() {
+				var mediaStreamBlobId = reader.result;
+
+				// The retrieved URL does not point to a MediaStream.
+				if (!mediaStreamBlobId || typeof mediaStreamBlobId !== 'string') {
+					// If this video element was previously handling a MediaStreamRenderer, release it.
+					releaseMediaStreamRenderer(video);
+
+					return;
+				}
+
+				provideMediaStreamRenderer(video, mediaStreamBlobId);
+			}
+		};
+		xhr.send();
+	}
+}
 
 function provideMediaStreamRenderer(video, mediaStreamBlobId) {
 	var mediaStream = mediaStreams[mediaStreamBlobId],
@@ -302,7 +350,11 @@ function provideMediaStreamRenderer(video, mediaStreamBlobId) {
 		readyState: {
 			configurable: true,
 			get: function () {
-				if (mediaStreamRenderer && mediaStreamRenderer.stream && mediaStreamRenderer.stream.connected) {
+				if (
+					mediaStreamRenderer &&
+					mediaStreamRenderer.stream &&
+					mediaStreamRenderer.stream.connected
+				) {
 					return video.HAVE_ENOUGH_DATA;
 				} else {
 					return video.HAVE_NOTHING;
@@ -312,7 +364,6 @@ function provideMediaStreamRenderer(video, mediaStreamBlobId) {
 	});
 }
 
-
 function releaseMediaStreamRenderer(video) {
 	if (!video._iosrtcMediaStreamRendererId) {
 		return;
diff --git a/lib/WebRTC.xcframework/Info.plist b/lib/WebRTC.xcframework/Info.plist
new file mode 100644
index 00000000..3258f5c7
--- /dev/null
+++ b/lib/WebRTC.xcframework/Info.plist
@@ -0,0 +1,41 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>AvailableLibraries</key>
+	<array>
+		<dict>
+			<key>LibraryIdentifier</key>
+			<string>ios-arm64_armv7</string>
+			<key>LibraryPath</key>
+			<string>WebRTC.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>arm64</string>
+				<string>armv7</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>ios</string>
+		</dict>
+		<dict>
+			<key>LibraryIdentifier</key>
+			<string>ios-i386_x86_64-simulator</string>
+			<key>LibraryPath</key>
+			<string>WebRTC.framework</string>
+			<key>SupportedArchitectures</key>
+			<array>
+				<string>i386</string>
+				<string>x86_64</string>
+			</array>
+			<key>SupportedPlatform</key>
+			<string>ios</string>
+			<key>SupportedPlatformVariant</key>
+			<string>simulator</string>
+		</dict>
+	</array>
+	<key>CFBundlePackageType</key>
+	<string>XFWK</string>
+	<key>XCFrameworkFormatVersion</key>
+	<string>1.0</string>
+</dict>
+</plist>
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioSession.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioSession.h
new file mode 100644
index 00000000..f917e327
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioSession.h
@@ -0,0 +1,264 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const kRTCAudioSessionErrorDomain;
+/** Method that requires lock was called without lock. */
+extern NSInteger const kRTCAudioSessionErrorLockRequired;
+/** Unknown configuration error occurred. */
+extern NSInteger const kRTCAudioSessionErrorConfiguration;
+
+@class RTC_OBJC_TYPE(RTCAudioSession);
+@class RTC_OBJC_TYPE(RTCAudioSessionConfiguration);
+
+// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
+// from AVAudioSession and handle them before calling these delegate methods,
+// at which point applications can perform additional processing if required.
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCAudioSessionDelegate)<NSObject>
+
+    @optional
+/** Called on a system notification thread when AVAudioSession starts an
+ *  interruption event.
+ */
+- (void)audioSessionDidBeginInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a system notification thread when AVAudioSession ends an
+ *  interruption event.
+ */
+- (void)audioSessionDidEndInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                   shouldResumeSession:(BOOL)shouldResumeSession;
+
+/** Called on a system notification thread when AVAudioSession changes the
+ *  route.
+ */
+- (void)audioSessionDidChangeRoute:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                            reason:(AVAudioSessionRouteChangeReason)reason
+                     previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
+
+/** Called on a system notification thread when AVAudioSession media server
+ *  terminates.
+ */
+- (void)audioSessionMediaServerTerminated:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a system notification thread when AVAudioSession media server
+ *  restarts.
+ */
+- (void)audioSessionMediaServerReset:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+    didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+
+/** Called on a WebRTC thread when the audio device is notified to begin
+ *  playback or recording.
+ */
+- (void)audioSessionDidStartPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a WebRTC thread when the audio device is notified to stop
+ *  playback or recording.
+ */
+- (void)audioSessionDidStopPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called when the AVAudioSession output volume value changes. */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didChangeOutputVolume:(float)outputVolume;
+
+/** Called when the audio device detects a playout glitch. The argument is the
+ *  number of glitches detected so far in the current audio playout session.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
+
+/** Called when the audio session is about to change the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession willSetActive:(BOOL)active;
+
+/** Called after the audio session sucessfully changed the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession didSetActive:(BOOL)active;
+
+/** Called after the audio session failed to change the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    failedToSetActive:(BOOL)active
+                error:(NSError *)error;
+
+@end
+
+/** This is a protocol used to inform RTCAudioSession when the audio session
+ *  activation state has changed outside of RTCAudioSession. The current known use
+ *  case of this is when CallKit activates the audio session for the application
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCAudioSessionActivationDelegate)<NSObject>
+
+    /** Called when the audio session is activated outside of the app by iOS. */
+    - (void)audioSessionDidActivate : (AVAudioSession *)session;
+
+/** Called when the audio session is deactivated outside of the app by iOS. */
+- (void)audioSessionDidDeactivate:(AVAudioSession *)session;
+
+@end
+
+/** Proxy class for AVAudioSession that adds a locking mechanism similar to
+ *  AVCaptureDevice. This is used to that interleaving configurations between
+ *  WebRTC and the application layer are avoided.
+ *
+ *  RTCAudioSession also coordinates activation so that the audio session is
+ *  activated only once. See |setActive:error:|.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSession) : NSObject <RTC_OBJC_TYPE(RTCAudioSessionActivationDelegate)>
+
+/** Convenience property to access the AVAudioSession singleton. Callers should
+ *  not call setters on AVAudioSession directly, but other method invocations
+ *  are fine.
+ */
+@property(nonatomic, readonly) AVAudioSession *session;
+
+/** Our best guess at whether the session is active based on results of calls to
+ *  AVAudioSession.
+ */
+@property(nonatomic, readonly) BOOL isActive;
+/** Whether RTCAudioSession is currently locked for configuration. */
+@property(nonatomic, readonly) BOOL isLocked;
+
+/** If YES, WebRTC will not initialize the audio unit automatically when an
+ *  audio track is ready for playout or recording. Instead, applications should
+ *  call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
+ *  as soon as an audio track is ready for playout or recording.
+ */
+@property(nonatomic, assign) BOOL useManualAudio;
+
+/** This property is only effective if useManualAudio is YES.
+ *  Represents permission for WebRTC to initialize the VoIP audio unit.
+ *  When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
+ *  stopped and uninitialized. This will stop incoming and outgoing audio.
+ *  When set to YES, WebRTC will initialize and start the audio unit when it is
+ *  needed (e.g. due to establishing an audio connection).
+ *  This property was introduced to work around an issue where if an AVPlayer is
+ *  playing audio while the VoIP audio unit is initialized, its audio would be
+ *  either cut off completely or played at a reduced volume. By preventing
+ *  the audio unit from being initialized until after the audio has completed,
+ *  we are able to prevent the abrupt cutoff.
+ */
+@property(nonatomic, assign) BOOL isAudioEnabled;
+
+// Proxy properties.
+@property(readonly) NSString *category;
+@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
+@property(readonly) NSString *mode;
+@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
+@property(readonly) AVAudioSessionRouteDescription *currentRoute;
+@property(readonly) NSInteger maximumInputNumberOfChannels;
+@property(readonly) NSInteger maximumOutputNumberOfChannels;
+@property(readonly) float inputGain;
+@property(readonly) BOOL inputGainSettable;
+@property(readonly) BOOL inputAvailable;
+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *inputDataSources;
+@property(readonly, nullable) AVAudioSessionDataSourceDescription *inputDataSource;
+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *outputDataSources;
+@property(readonly, nullable) AVAudioSessionDataSourceDescription *outputDataSource;
+@property(readonly) double sampleRate;
+@property(readonly) double preferredSampleRate;
+@property(readonly) NSInteger inputNumberOfChannels;
+@property(readonly) NSInteger outputNumberOfChannels;
+@property(readonly) float outputVolume;
+@property(readonly) NSTimeInterval inputLatency;
+@property(readonly) NSTimeInterval outputLatency;
+@property(readonly) NSTimeInterval IOBufferDuration;
+@property(readonly) NSTimeInterval preferredIOBufferDuration;
+
+/**
+ When YES, calls to -setConfiguration:error: and -setConfiguration:active:error: ignore errors in
+ configuring the audio session's "preferred" attributes (e.g. preferredInputNumberOfChannels).
+ Typically, configurations to preferred attributes are optimizations, and ignoring this type of
+ configuration error allows code flow to continue along the happy path when these optimization are
+ not available. The default value of this property is NO.
+ */
+@property(nonatomic) BOOL ignoresPreferredAttributeConfigurationErrors;
+
+/** Default constructor. */
++ (instancetype)sharedInstance;
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Adds a delegate, which is held weakly. */
+- (void)addDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+/** Removes an added delegate. */
+- (void)removeDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+
+/** Request exclusive access to the audio session for configuration. This call
+ *  will block if the lock is held by another object.
+ */
+- (void)lockForConfiguration;
+/** Relinquishes exclusive access to the audio session. */
+- (void)unlockForConfiguration;
+
+/** If |active|, activates the audio session if it isn't already active.
+ *  Successful calls must be balanced with a setActive:NO when activation is no
+ *  longer required. If not |active|, deactivates the audio session if one is
+ *  active and this is the last balanced call. When deactivating, the
+ *  AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
+ *  AVAudioSession.
+ */
+- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
+
+// The following methods are proxies for the associated methods on
+// AVAudioSession. |lockForConfiguration| must be called before using them
+// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
+
+- (BOOL)setCategory:(NSString *)category
+        withOptions:(AVAudioSessionCategoryOptions)options
+              error:(NSError **)outError;
+- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
+- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
+- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
+- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration error:(NSError **)outError;
+- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
+- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
+- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride error:(NSError **)outError;
+- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort error:(NSError **)outError;
+- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                     error:(NSError **)outError;
+- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                      error:(NSError **)outError;
+@end
+
+@interface RTC_OBJC_TYPE (RTCAudioSession)
+(Configuration)
+
+    /** Applies the configuration to the current session. Attempts to set all
+     *  properties even if previous ones fail. Only the last error will be
+     *  returned.
+     *  |lockForConfiguration| must be called first.
+     */
+    - (BOOL)setConfiguration : (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration error
+    : (NSError **)outError;
+
+/** Convenience method that calls both setConfiguration and setActive.
+ *  |lockForConfiguration| must be called first.
+ */
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioSessionConfiguration.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioSessionConfiguration.h
new file mode 100644
index 00000000..4582b805
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioSessionConfiguration.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN const int kRTCAudioSessionPreferredNumberOfChannels;
+RTC_EXTERN const double kRTCAudioSessionHighPerformanceSampleRate;
+RTC_EXTERN const double kRTCAudioSessionLowComplexitySampleRate;
+RTC_EXTERN const double kRTCAudioSessionHighPerformanceIOBufferDuration;
+RTC_EXTERN const double kRTCAudioSessionLowComplexityIOBufferDuration;
+
+// Struct to hold configuration values.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSessionConfiguration) : NSObject
+
+@property(nonatomic, strong) NSString *category;
+@property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
+@property(nonatomic, strong) NSString *mode;
+@property(nonatomic, assign) double sampleRate;
+@property(nonatomic, assign) NSTimeInterval ioBufferDuration;
+@property(nonatomic, assign) NSInteger inputNumberOfChannels;
+@property(nonatomic, assign) NSInteger outputNumberOfChannels;
+
+/** Initializes configuration to defaults. */
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+/** Returns the current configuration of the audio session. */
++ (instancetype)currentConfiguration;
+/** Returns the configuration that WebRTC needs. */
++ (instancetype)webRTCConfiguration;
+/** Provide a way to override the default configuration. */
++ (void)setWebRTCConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioSource.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioSource.h
new file mode 100644
index 00000000..9f78dcd9
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioSource.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMediaSource.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSource) : RTC_OBJC_TYPE(RTCMediaSource)
+
+- (instancetype)init NS_UNAVAILABLE;
+
+// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
+// [0, 10].
+// Temporary fix to be able to modify volume of remote audio tracks.
+// TODO(kthelgason): Property stays here temporarily until a proper volume-api
+// is available on the surface exposed by webrtc.
+@property(nonatomic, assign) double volume;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioTrack.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioTrack.h
new file mode 100644
index 00000000..95eb5d3d
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCAudioTrack.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMacros.h"
+#import "RTCMediaStreamTrack.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioSource);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioTrack) : RTC_OBJC_TYPE(RTCMediaStreamTrack)
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** The audio source for this audio track. */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCAudioSource) * source;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCVPixelBuffer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCVPixelBuffer.h
new file mode 100644
index 00000000..17eebd0a
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCVPixelBuffer.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrameBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCVideoFrameBuffer containing a CVPixelBufferRef */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCVPixelBuffer) : NSObject <RTC_OBJC_TYPE(RTCVideoFrameBuffer)>
+
+@property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
+@property(nonatomic, readonly) int cropX;
+@property(nonatomic, readonly) int cropY;
+@property(nonatomic, readonly) int cropWidth;
+@property(nonatomic, readonly) int cropHeight;
+
++ (NSSet<NSNumber *> *)supportedPixelFormats;
+
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer;
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                       adaptedWidth:(int)adaptedWidth
+                      adaptedHeight:(int)adaptedHeight
+                          cropWidth:(int)cropWidth
+                         cropHeight:(int)cropHeight
+                              cropX:(int)cropX
+                              cropY:(int)cropY;
+
+- (BOOL)requiresCropping;
+- (BOOL)requiresScalingToWidth:(int)width height:(int)height;
+- (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
+
+/** The minimum size of the |tmpBuffer| must be the number of bytes returned from the
+ * bufferSizeForCroppingAndScalingToWidth:height: method.
+ * If that size is 0, the |tmpBuffer| may be nil.
+ */
+- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
+        withTempBuffer:(nullable uint8_t *)tmpBuffer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCallbackLogger.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCallbackLogger.h
new file mode 100644
index 00000000..c1aeb825
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCallbackLogger.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCLogging.h"
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef void (^RTCCallbackLoggerMessageHandler)(NSString *message);
+typedef void (^RTCCallbackLoggerMessageAndSeverityHandler)(NSString *message,
+                                                           RTCLoggingSeverity severity);
+
+// This class intercepts WebRTC logs and forwards them to a registered block.
+// This class is not threadsafe.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCallbackLogger) : NSObject
+
+// The severity level to capture. The default is kRTCLoggingSeverityInfo.
+@property(nonatomic, assign) RTCLoggingSeverity severity;
+
+// The callback handler will be called on the same thread that does the
+// logging, so if the logging callback can be slow it may be a good idea
+// to implement dispatching to some other queue.
+- (void)start:(nullable RTCCallbackLoggerMessageHandler)handler;
+- (void)startWithMessageAndSeverityHandler:
+        (nullable RTCCallbackLoggerMessageAndSeverityHandler)handler;
+
+- (void)stop;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCameraPreviewView.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCameraPreviewView.h
new file mode 100644
index 00000000..db9b15a4
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCameraPreviewView.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCMacros.h"
+
+@class AVCaptureSession;
+
+/** RTCCameraPreviewView is a view that renders local video from an
+ *  AVCaptureSession.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCameraPreviewView) : UIView
+
+/** The capture session being rendered in the view. Capture session
+ *  is assigned to AVCaptureVideoPreviewLayer async in the same
+ *  queue that the AVCaptureSession is started/stopped.
+ */
+@property(nonatomic, strong) AVCaptureSession* captureSession;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCameraVideoCapturer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCameraVideoCapturer.h
new file mode 100644
index 00000000..fed5a378
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCameraVideoCapturer.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+// Camera capture that implements RTCVideoCapturer. Delivers frames to a
+// RTCVideoCapturerDelegate (usually RTCVideoSource).
+NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
+@interface RTC_OBJC_TYPE (RTCCameraVideoCapturer) : RTC_OBJC_TYPE(RTCVideoCapturer)
+
+// Capture session that is used for capturing. Valid from initialization to dealloc.
+@property(readonly, nonatomic) AVCaptureSession *captureSession;
+
+// Returns list of available capture devices that support video capture.
++ (NSArray<AVCaptureDevice *> *)captureDevices;
+// Returns list of formats that are supported by this class for this device.
++ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device;
+
+// Returns the most efficient supported output pixel format for this capturer.
+- (FourCharCode)preferredOutputPixelFormat;
+
+// Starts the capture session asynchronously and notifies callback on completion.
+// The device will capture video in the format given in the `format` parameter. If the pixel format
+// in `format` is supported by the WebRTC pipeline, the same pixel format will be used for the
+// output. Otherwise, the format returned by `preferredOutputPixelFormat` will be used.
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps
+             completionHandler:(nullable void (^)(NSError *))completionHandler;
+// Stops the capture session asynchronously and notifies callback on completion.
+- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler;
+
+// Starts the capture session asynchronously.
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps;
+// Stops the capture session asynchronously.
+- (void)stopCapture;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCertificate.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCertificate.h
new file mode 100644
index 00000000..5ac8984d
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCertificate.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCertificate) : NSObject <NSCopying>
+
+/** Private key in PEM. */
+@property(nonatomic, readonly, copy) NSString *private_key;
+
+/** Public key in an x509 cert encoded in PEM. */
+@property(nonatomic, readonly, copy) NSString *certificate;
+
+/**
+ * Initialize an RTCCertificate with PEM strings for private_key and certificate.
+ */
+- (instancetype)initWithPrivateKey:(NSString *)private_key
+                       certificate:(NSString *)certificate NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Generate a new certificate for 're' use.
+ *
+ *  Optional dictionary of parameters. Defaults to KeyType ECDSA if none are
+ *  provided.
+ *  - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
+ */
++ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:(NSDictionary *)params;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCodecSpecificInfo.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCodecSpecificInfo.h
new file mode 100644
index 00000000..5e7800e5
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCodecSpecificInfo.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Implement this protocol to pass codec specific info from the encoder.
+ *  Corresponds to webrtc::CodecSpecificInfo.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCCodecSpecificInfo)<NSObject> @end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCodecSpecificInfoH264.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCodecSpecificInfoH264.h
new file mode 100644
index 00000000..ae3003a1
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCodecSpecificInfoH264.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCodecSpecificInfo.h"
+#import "RTCMacros.h"
+
+/** Class for H264 specific config. */
+typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
+  RTCH264PacketizationModeNonInterleaved = 0,  // Mode 1 - STAP-A, FU-A is allowed
+  RTCH264PacketizationModeSingleNalUnit        // Mode 0 - only single NALU allowed
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCodecSpecificInfoH264) : NSObject <RTC_OBJC_TYPE(RTCCodecSpecificInfo)>
+
+@property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCConfiguration.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCConfiguration.h
new file mode 100644
index 00000000..86eaa6ce
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCConfiguration.h
@@ -0,0 +1,221 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCertificate.h"
+#import "RTCCryptoOptions.h"
+#import "RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCIceServer);
+
+/**
+ * Represents the ice transport policy. This exposes the same states in C++,
+ * which include one more state than what exists in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
+  RTCIceTransportPolicyNone,
+  RTCIceTransportPolicyRelay,
+  RTCIceTransportPolicyNoHost,
+  RTCIceTransportPolicyAll
+};
+
+/** Represents the bundle policy. */
+typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
+  RTCBundlePolicyBalanced,
+  RTCBundlePolicyMaxCompat,
+  RTCBundlePolicyMaxBundle
+};
+
+/** Represents the rtcp mux policy. */
+typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
+
+/** Represents the tcp candidate policy. */
+typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
+  RTCTcpCandidatePolicyEnabled,
+  RTCTcpCandidatePolicyDisabled
+};
+
+/** Represents the candidate network policy. */
+typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) {
+  RTCCandidateNetworkPolicyAll,
+  RTCCandidateNetworkPolicyLowCost
+};
+
+/** Represents the continual gathering policy. */
+typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
+  RTCContinualGatheringPolicyGatherOnce,
+  RTCContinualGatheringPolicyGatherContinually
+};
+
+/** Represents the encryption key type. */
+typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
+  RTCEncryptionKeyTypeRSA,
+  RTCEncryptionKeyTypeECDSA,
+};
+
+/** Represents the chosen SDP semantics for the RTCPeerConnection. */
+typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
+  RTCSdpSemanticsPlanB,
+  RTCSdpSemanticsUnifiedPlan,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCConfiguration) : NSObject
+
+/** If true, allows DSCP codes to be set on outgoing packets, configured using
+ *  networkPriority field of RTCRtpEncodingParameters. Defaults to false.
+ */
+@property(nonatomic, assign) BOOL enableDscp;
+
+/** An array of Ice Servers available to be used by ICE. */
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCIceServer) *> *iceServers;
+
+/** An RTCCertificate for 're' use. */
+@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCertificate) * certificate;
+
+/** Which candidates the ICE agent is allowed to use. The W3C calls it
+ * |iceTransportPolicy|, while in C++ it is called |type|. */
+@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
+
+/** The media-bundling policy to use when gathering ICE candidates. */
+@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
+
+/** The rtcp-mux policy to use when gathering ICE candidates. */
+@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
+@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
+@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
+@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
+
+/** If set to YES, don't gather IPv6 ICE candidates.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL disableIPV6;
+
+/** If set to YES, don't gather IPv6 ICE candidates on Wi-Fi.
+ *  Only intended to be used on specific devices. Certain phones disable IPv6
+ *  when the screen is turned off and it would be better to just disable the
+ *  IPv6 ICE candidates on Wi-Fi in those cases.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL disableIPV6OnWiFi;
+
+/** By default, the PeerConnection will use a limited number of IPv6 network
+ *  interfaces, in order to avoid too many ICE candidate pairs being created
+ *  and delaying ICE completion.
+ *
+ *  Can be set to INT_MAX to effectively disable the limit.
+ */
+@property(nonatomic, assign) int maxIPv6Networks;
+
+/** Exclude link-local network interfaces
+ *  from considertaion for gathering ICE candidates.
+ *  Defaults to NO.
+ */
+@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
+
+@property(nonatomic, assign) int audioJitterBufferMaxPackets;
+@property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
+@property(nonatomic, assign) int iceConnectionReceivingTimeout;
+@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
+
+/** Key type used to generate SSL identity. Default is ECDSA. */
+@property(nonatomic, assign) RTCEncryptionKeyType keyType;
+
+/** ICE candidate pool size as defined in JSEP. Default is 0. */
+@property(nonatomic, assign) int iceCandidatePoolSize;
+
+/** Prune turn ports on the same network to the same turn server.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL shouldPruneTurnPorts;
+
+/** If set to YES, this means the ICE transport should presume TURN-to-TURN
+ *  candidate pairs will succeed, even before a binding response is received.
+ */
+@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
+
+/* This flag is only effective when |continualGatheringPolicy| is
+ * RTCContinualGatheringPolicyGatherContinually.
+ *
+ * If YES, after the ICE transport type is changed such that new types of
+ * ICE candidates are allowed by the new transport type, e.g. from
+ * RTCIceTransportPolicyRelay to RTCIceTransportPolicyAll, candidates that
+ * have been gathered by the ICE transport but not matching the previous
+ * transport type and as a result not observed by PeerConnectionDelegateAdapter,
+ * will be surfaced to the delegate.
+ */
+@property(nonatomic, assign) BOOL shouldSurfaceIceCandidatesOnIceTransportTypeChanged;
+
+/** If set to non-nil, controls the minimal interval between consecutive ICE
+ *  check packets.
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
+
+/** Configure the SDP semantics used by this PeerConnection. Note that the
+ *  WebRTC 1.0 specification requires UnifiedPlan semantics. The
+ *  RTCRtpTransceiver API is only available with UnifiedPlan semantics.
+ *
+ *  PlanB will cause RTCPeerConnection to create offers and answers with at
+ *  most one audio and one video m= section with multiple RTCRtpSenders and
+ *  RTCRtpReceivers specified as multiple a=ssrc lines within the section. This
+ *  will also cause RTCPeerConnection to ignore all but the first m= section of
+ *  the same media type.
+ *
+ *  UnifiedPlan will cause RTCPeerConnection to create offers and answers with
+ *  multiple m= sections where each m= section maps to one RTCRtpSender and one
+ *  RTCRtpReceiver (an RTCRtpTransceiver), either both audio or both
+ *  video. This will also cause RTCPeerConnection) to ignore all but the first a=ssrc
+ *  lines that form a Plan B stream.
+ *
+ *  For users who wish to send multiple audio/video streams and need to stay
+ *  interoperable with legacy WebRTC implementations or use legacy APIs,
+ *  specify PlanB.
+ *
+ *  For all other users, specify UnifiedPlan.
+ */
+@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
+
+/** Actively reset the SRTP parameters when the DTLS transports underneath are
+ *  changed after offer/answer negotiation. This is only intended to be a
+ *  workaround for crbug.com/835958
+ */
+@property(nonatomic, assign) BOOL activeResetSrtpParams;
+
+/** If the remote side support mid-stream codec switches then allow encoder
+ *  switching to be performed.
+ */
+
+@property(nonatomic, assign) BOOL allowCodecSwitching;
+
+/**
+ * Defines advanced optional cryptographic settings related to SRTP and
+ * frame encryption for native WebRTC. Setting this will overwrite any
+ * options set through the PeerConnectionFactory (which is deprecated).
+ */
+@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCryptoOptions) * cryptoOptions;
+
+/**
+ * Time interval between audio RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
+
+/**
+ * Time interval between video RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCryptoOptions.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCryptoOptions.h
new file mode 100644
index 00000000..759a45e0
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCCryptoOptions.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * Objective-C bindings for webrtc::CryptoOptions. This API had to be flattened
+ * as Objective-C doesn't support nested structures.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCryptoOptions) : NSObject
+
+/**
+ * Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
+ * if both sides enable it
+ */
+@property(nonatomic, assign) BOOL srtpEnableGcmCryptoSuites;
+/**
+ * If set to true, the (potentially insecure) crypto cipher
+ * SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
+ * during negotiation. It will only be used if both peers support it and no
+ * other ciphers get preferred.
+ */
+@property(nonatomic, assign) BOOL srtpEnableAes128Sha1_32CryptoCipher;
+/**
+ * If set to true, encrypted RTP header extensions as defined in RFC 6904
+ * will be negotiated. They will only be used if both peers support them.
+ */
+@property(nonatomic, assign) BOOL srtpEnableEncryptedRtpHeaderExtensions;
+
+/**
+ * If set all RtpSenders must have an FrameEncryptor attached to them before
+ * they are allowed to send packets. All RtpReceivers must have a
+ * FrameDecryptor attached to them before they are able to receive packets.
+ */
+@property(nonatomic, assign) BOOL sframeRequireFrameEncryption;
+
+/**
+ * Initializes CryptoOptions with all possible options set explicitly. This
+ * is done when converting from a native RTCConfiguration.crypto_options.
+ */
+- (instancetype)initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
+              srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
+           srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
+                     sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDataChannel.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDataChannel.h
new file mode 100644
index 00000000..2d0661f1
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDataChannel.h
@@ -0,0 +1,132 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AvailabilityMacros.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataBuffer) : NSObject
+
+/** NSData representation of the underlying buffer. */
+@property(nonatomic, readonly) NSData *data;
+
+/** Indicates whether |data| contains UTF-8 or binary data. */
+@property(nonatomic, readonly) BOOL isBinary;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Initialize an RTCDataBuffer from NSData. |isBinary| indicates whether |data|
+ * contains UTF-8 or binary data.
+ */
+- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary;
+
+@end
+
+@class RTC_OBJC_TYPE(RTCDataChannel);
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCDataChannelDelegate)<NSObject>
+
+    /** The data channel state changed. */
+    - (void)dataChannelDidChangeState : (RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
+
+/** The data channel successfully received a data buffer. */
+- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
+    didReceiveMessageWithBuffer:(RTC_OBJC_TYPE(RTCDataBuffer) *)buffer;
+
+@optional
+/** The data channel's |bufferedAmount| changed. */
+- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
+    didChangeBufferedAmount:(uint64_t)amount;
+
+@end
+
+/** Represents the state of the data channel. */
+typedef NS_ENUM(NSInteger, RTCDataChannelState) {
+  RTCDataChannelStateConnecting,
+  RTCDataChannelStateOpen,
+  RTCDataChannelStateClosing,
+  RTCDataChannelStateClosed,
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataChannel) : NSObject
+
+/**
+ * A label that can be used to distinguish this data channel from other data
+ * channel objects.
+ */
+@property(nonatomic, readonly) NSString *label;
+
+/** Whether the data channel can send messages in unreliable mode. */
+@property(nonatomic, readonly) BOOL isReliable DEPRECATED_ATTRIBUTE;
+
+/** Returns whether this data channel is ordered or not. */
+@property(nonatomic, readonly) BOOL isOrdered;
+
+/** Deprecated. Use maxPacketLifeTime. */
+@property(nonatomic, readonly) NSUInteger maxRetransmitTime DEPRECATED_ATTRIBUTE;
+
+/**
+ * The length of the time window (in milliseconds) during which transmissions
+ * and retransmissions may occur in unreliable mode.
+ */
+@property(nonatomic, readonly) uint16_t maxPacketLifeTime;
+
+/**
+ * The maximum number of retransmissions that are attempted in unreliable mode.
+ */
+@property(nonatomic, readonly) uint16_t maxRetransmits;
+
+/**
+ * The name of the sub-protocol used with this data channel, if any. Otherwise
+ * this returns an empty string.
+ */
+@property(nonatomic, readonly) NSString *protocol;
+
+/**
+ * Returns whether this data channel was negotiated by the application or not.
+ */
+@property(nonatomic, readonly) BOOL isNegotiated;
+
+/** Deprecated. Use channelId. */
+@property(nonatomic, readonly) NSInteger streamId DEPRECATED_ATTRIBUTE;
+
+/** The identifier for this data channel. */
+@property(nonatomic, readonly) int channelId;
+
+/** The state of the data channel. */
+@property(nonatomic, readonly) RTCDataChannelState readyState;
+
+/**
+ * The number of bytes of application data that have been queued using
+ * |sendData:| but that have not yet been transmitted to the network.
+ */
+@property(nonatomic, readonly) uint64_t bufferedAmount;
+
+/** The delegate for this data channel. */
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCDataChannelDelegate)> delegate;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Closes the data channel. */
+- (void)close;
+
+/** Attempt to send |data| on this data channel's underlying data transport. */
+- (BOOL)sendData:(RTC_OBJC_TYPE(RTCDataBuffer) *)data;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDataChannelConfiguration.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDataChannelConfiguration.h
new file mode 100644
index 00000000..9459ae0a
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDataChannelConfiguration.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AvailabilityMacros.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataChannelConfiguration) : NSObject
+
+/** Set to YES if ordered delivery is required. */
+@property(nonatomic, assign) BOOL isOrdered;
+
+/** Deprecated. Use maxPacketLifeTime. */
+@property(nonatomic, assign) NSInteger maxRetransmitTimeMs DEPRECATED_ATTRIBUTE;
+
+/**
+ * Max period in milliseconds in which retransmissions will be sent. After this
+ * time, no more retransmissions will be sent. -1 if unset.
+ */
+@property(nonatomic, assign) int maxPacketLifeTime;
+
+/** The max number of retransmissions. -1 if unset. */
+@property(nonatomic, assign) int maxRetransmits;
+
+/** Set to YES if the channel has been externally negotiated and we do not send
+ * an in-band signalling in the form of an "open" message.
+ */
+@property(nonatomic, assign) BOOL isNegotiated;
+
+/** Deprecated. Use channelId. */
+@property(nonatomic, assign) int streamId DEPRECATED_ATTRIBUTE;
+
+/** The id of the data channel. */
+@property(nonatomic, assign) int channelId;
+
+/** Set by the application and opaque to the WebRTC implementation. */
+@property(nonatomic) NSString* protocol;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDefaultVideoDecoderFactory.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDefaultVideoDecoderFactory.h
new file mode 100644
index 00000000..de5a9c46
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDefaultVideoDecoderFactory.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoderFactory.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This decoder factory include support for all codecs bundled with WebRTC. If using custom
+ *  codecs, create custom implementations of RTCVideoEncoderFactory and
+ *  RTCVideoDecoderFactory.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDefaultVideoDecoderFactory) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoderFactory)>
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDefaultVideoEncoderFactory.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDefaultVideoEncoderFactory.h
new file mode 100644
index 00000000..92ab40c9
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDefaultVideoEncoderFactory.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoderFactory.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This encoder factory include support for all codecs bundled with WebRTC. If using custom
+ *  codecs, create custom implementations of RTCVideoEncoderFactory and
+ *  RTCVideoDecoderFactory.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDefaultVideoEncoderFactory) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoderFactory)>
+
+@property(nonatomic, retain) RTC_OBJC_TYPE(RTCVideoCodecInfo) *preferredCodec;
+
++ (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDispatcher.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDispatcher.h
new file mode 100644
index 00000000..e148af6d
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDispatcher.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
+  // Main dispatcher queue.
+  RTCDispatcherTypeMain,
+  // Used for starting/stopping AVCaptureSession, and assigning
+  // capture session to AVCaptureVideoPreviewLayer.
+  RTCDispatcherTypeCaptureSession,
+  // Used for operations on AVAudioSession.
+  RTCDispatcherTypeAudioSession,
+  // Used for operations on NWPathMonitor.
+  RTCDispatcherTypeNetworkMonitor,
+};
+
+/** Dispatcher that asynchronously dispatches blocks to a specific
+ *  shared dispatch queue.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDispatcher) : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Dispatch the block asynchronously on the queue for dispatchType.
+ *  @param dispatchType The queue type to dispatch on.
+ *  @param block The block to dispatch asynchronously.
+ */
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType block:(dispatch_block_t)block;
+
+/** Returns YES if run on queue for the dispatchType otherwise NO.
+ *  Useful for asserting that a method is run on a correct queue.
+ */
++ (BOOL)isOnQueueForType:(RTCDispatcherQueueType)dispatchType;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDtmfSender.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDtmfSender.h
new file mode 100644
index 00000000..0f1b6ba4
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCDtmfSender.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCDtmfSender)<NSObject>
+
+    /**
+     * Returns true if this RTCDtmfSender is capable of sending DTMF. Otherwise
+     * returns false. To be able to send DTMF, the associated RTCRtpSender must be
+     * able to send packets, and a "telephone-event" codec must be negotiated.
+     */
+    @property(nonatomic, readonly) BOOL canInsertDtmf;
+
+/**
+ * Queues a task that sends the DTMF tones. The tones parameter is treated
+ * as a series of characters. The characters 0 through 9, A through D, #, and *
+ * generate the associated DTMF tones. The characters a to d are equivalent
+ * to A to D. The character ',' indicates a delay of 2 seconds before
+ * processing the next character in the tones parameter.
+ *
+ * Unrecognized characters are ignored.
+ *
+ * @param duration The parameter indicates the duration to use for each
+ * character passed in the tones parameter. The duration cannot be more
+ * than 6000 or less than 70 ms.
+ *
+ * @param interToneGap The parameter indicates the gap between tones.
+ * This parameter must be at least 50 ms but should be as short as
+ * possible.
+ *
+ * If InsertDtmf is called on the same object while an existing task for this
+ * object to generate DTMF is still running, the previous task is canceled.
+ * Returns true on success and false on failure.
+ */
+- (BOOL)insertDtmf:(nonnull NSString *)tones
+          duration:(NSTimeInterval)duration
+      interToneGap:(NSTimeInterval)interToneGap;
+
+/** The tones remaining to be played out */
+- (nonnull NSString *)remainingTones;
+
+/**
+ * The current tone duration value. This value will be the value last set via the
+ * insertDtmf method, or the default value of 100 ms if insertDtmf was never called.
+ */
+- (NSTimeInterval)duration;
+
+/**
+ * The current value of the between-tone gap. This value will be the value last set
+ * via the insertDtmf() method, or the default value of 50 ms if insertDtmf() was never
+ * called.
+ */
+- (NSTimeInterval)interToneGap;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCEAGLVideoView.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCEAGLVideoView.h
new file mode 100644
index 00000000..24b26cd6
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCEAGLVideoView.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoRenderer.h"
+#import "RTCVideoViewShading.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCEAGLVideoView);
+
+/**
+ * RTCEAGLVideoView is an RTCVideoRenderer which renders video frames
+ * in its bounds using OpenGLES 2.0 or OpenGLES 3.0.
+ */
+RTC_OBJC_EXPORT
+NS_EXTENSION_UNAVAILABLE_IOS("Rendering not available in app extensions.")
+@interface RTC_OBJC_TYPE (RTCEAGLVideoView) : UIView <RTC_OBJC_TYPE(RTCVideoRenderer)>
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoViewDelegate)> delegate;
+
+- (instancetype)initWithFrame:(CGRect)frame
+                       shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)initWithCoder:(NSCoder *)aDecoder
+                       shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader
+    NS_DESIGNATED_INITIALIZER;
+
+/** @abstract Wrapped RTCVideoRotation, or nil.
+ */
+@property(nonatomic, nullable) NSValue *rotationOverride;
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCEncodedImage.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCEncodedImage.h
new file mode 100644
index 00000000..5fec8a22
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCEncodedImage.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Represents an encoded frame's type. */
+typedef NS_ENUM(NSUInteger, RTCFrameType) {
+  RTCFrameTypeEmptyFrame = 0,
+  RTCFrameTypeAudioFrameSpeech = 1,
+  RTCFrameTypeAudioFrameCN = 2,
+  RTCFrameTypeVideoFrameKey = 3,
+  RTCFrameTypeVideoFrameDelta = 4,
+};
+
+typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
+  RTCVideoContentTypeUnspecified,
+  RTCVideoContentTypeScreenshare,
+};
+
+/** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCEncodedImage) : NSObject
+
+@property(nonatomic, strong) NSData *buffer;
+@property(nonatomic, assign) int32_t encodedWidth;
+@property(nonatomic, assign) int32_t encodedHeight;
+@property(nonatomic, assign) uint32_t timeStamp;
+@property(nonatomic, assign) int64_t captureTimeMs;
+@property(nonatomic, assign) int64_t ntpTimeMs;
+@property(nonatomic, assign) uint8_t flags;
+@property(nonatomic, assign) int64_t encodeStartMs;
+@property(nonatomic, assign) int64_t encodeFinishMs;
+@property(nonatomic, assign) RTCFrameType frameType;
+@property(nonatomic, assign) RTCVideoRotation rotation;
+@property(nonatomic, assign) BOOL completeFrame;
+@property(nonatomic, strong) NSNumber *qp;
+@property(nonatomic, assign) RTCVideoContentType contentType;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCFieldTrials.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCFieldTrials.h
new file mode 100644
index 00000000..7477ad02
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCFieldTrials.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */
+RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweKey;
+RTC_EXTERN NSString * const kRTCFieldTrialAudioForceNoTWCCKey;
+RTC_EXTERN NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey;
+RTC_EXTERN NSString * const kRTCFieldTrialSendSideBweWithOverheadKey;
+RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03AdvertisedKey;
+RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03Key;
+RTC_EXTERN NSString * const kRTCFieldTrialH264HighProfileKey;
+RTC_EXTERN NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey;
+RTC_EXTERN NSString *const kRTCFieldTrialUseNWPathMonitor;
+
+/** The valid value for field trials above. */
+RTC_EXTERN NSString * const kRTCFieldTrialEnabledValue;
+
+/** Initialize field trials using a dictionary mapping field trial keys to their
+ * values. See above for valid keys and values. Must be called before any other
+ * call into WebRTC. See: webrtc/system_wrappers/include/field_trial.h
+ */
+RTC_EXTERN void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials);
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCFileLogger.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCFileLogger.h
new file mode 100644
index 00000000..853e673a
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCFileLogger.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSUInteger, RTCFileLoggerSeverity) {
+  RTCFileLoggerSeverityVerbose,
+  RTCFileLoggerSeverityInfo,
+  RTCFileLoggerSeverityWarning,
+  RTCFileLoggerSeverityError
+};
+
+typedef NS_ENUM(NSUInteger, RTCFileLoggerRotationType) {
+  RTCFileLoggerTypeCall,
+  RTCFileLoggerTypeApp,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+// This class intercepts WebRTC logs and saves them to a file. The file size
+// will not exceed the given maximum bytesize. When the maximum bytesize is
+// reached, logs are rotated according to the rotationType specified.
+// For kRTCFileLoggerTypeCall, logs from the beginning and the end
+// are preserved while the middle section is overwritten instead.
+// For kRTCFileLoggerTypeApp, the oldest log is overwritten.
+// This class is not threadsafe.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCFileLogger) : NSObject
+
+// The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
+@property(nonatomic, assign) RTCFileLoggerSeverity severity;
+
+// The rotation type for this file logger. The default is
+// kRTCFileLoggerTypeCall.
+@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
+
+// Disables buffering disk writes. Should be set before |start|. Buffering
+// is enabled by default for performance.
+@property(nonatomic, assign) BOOL shouldDisableBuffering;
+
+// Default constructor provides default settings for dir path, file size and
+// rotation type.
+- (instancetype)init;
+
+// Create file logger with default rotation type.
+- (instancetype)initWithDirPath:(NSString *)dirPath maxFileSize:(NSUInteger)maxFileSize;
+
+- (instancetype)initWithDirPath:(NSString *)dirPath
+                    maxFileSize:(NSUInteger)maxFileSize
+                   rotationType:(RTCFileLoggerRotationType)rotationType NS_DESIGNATED_INITIALIZER;
+
+// Starts writing WebRTC logs to disk if not already started. Overwrites any
+// existing file(s).
+- (void)start;
+
+// Stops writing WebRTC logs to disk. This method is also called on dealloc.
+- (void)stop;
+
+// Returns the current contents of the logs, or nil if start has been called
+// without a stop.
+- (nullable NSData *)logData;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCFileVideoCapturer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCFileVideoCapturer.h
new file mode 100644
index 00000000..19262c64
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCFileVideoCapturer.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCVideoCapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * Error passing block.
+ */
+typedef void (^RTCFileVideoCapturerErrorBlock)(NSError *error);
+
+/**
+ * Captures buffers from bundled video file.
+ *
+ * See @c RTCVideoCapturer for more info on capturers.
+ */
+RTC_OBJC_EXPORT
+
+NS_CLASS_AVAILABLE_IOS(10)
+@interface RTC_OBJC_TYPE (RTCFileVideoCapturer) : RTC_OBJC_TYPE(RTCVideoCapturer)
+
+/**
+ * Starts asynchronous capture of frames from video file.
+ *
+ * Capturing is not started if error occurs. Underlying error will be
+ * relayed in the errorBlock if one is provided.
+ * Successfully captured video frames will be passed to the delegate.
+ *
+ * @param nameOfFile The name of the bundled video file to be read.
+ * @errorBlock block to be executed upon error.
+ */
+- (void)startCapturingFromFileNamed:(NSString *)nameOfFile
+                            onError:(__nullable RTCFileVideoCapturerErrorBlock)errorBlock;
+
+/**
+ * Immediately stops capture.
+ */
+- (void)stopCapture;
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCH264ProfileLevelId.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCH264ProfileLevelId.h
new file mode 100644
index 00000000..dac7bb56
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCH264ProfileLevelId.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+RTC_OBJC_EXPORT extern NSString *const kRTCVideoCodecH264Name;
+RTC_OBJC_EXPORT extern NSString *const kRTCLevel31ConstrainedHigh;
+RTC_OBJC_EXPORT extern NSString *const kRTCLevel31ConstrainedBaseline;
+RTC_OBJC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh;
+RTC_OBJC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline;
+
+/** H264 Profiles and levels. */
+typedef NS_ENUM(NSUInteger, RTCH264Profile) {
+  RTCH264ProfileConstrainedBaseline,
+  RTCH264ProfileBaseline,
+  RTCH264ProfileMain,
+  RTCH264ProfileConstrainedHigh,
+  RTCH264ProfileHigh,
+};
+
+typedef NS_ENUM(NSUInteger, RTCH264Level) {
+  RTCH264Level1_b = 0,
+  RTCH264Level1 = 10,
+  RTCH264Level1_1 = 11,
+  RTCH264Level1_2 = 12,
+  RTCH264Level1_3 = 13,
+  RTCH264Level2 = 20,
+  RTCH264Level2_1 = 21,
+  RTCH264Level2_2 = 22,
+  RTCH264Level3 = 30,
+  RTCH264Level3_1 = 31,
+  RTCH264Level3_2 = 32,
+  RTCH264Level4 = 40,
+  RTCH264Level4_1 = 41,
+  RTCH264Level4_2 = 42,
+  RTCH264Level5 = 50,
+  RTCH264Level5_1 = 51,
+  RTCH264Level5_2 = 52
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCH264ProfileLevelId) : NSObject
+
+@property(nonatomic, readonly) RTCH264Profile profile;
+@property(nonatomic, readonly) RTCH264Level level;
+@property(nonatomic, readonly) NSString *hexString;
+
+- (instancetype)initWithHexString:(NSString *)hexString;
+- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCI420Buffer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCI420Buffer.h
new file mode 100644
index 00000000..b97f05a5
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCI420Buffer.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCYUVPlanarBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Protocol for RTCYUVPlanarBuffers containing I420 data */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCI420Buffer)<RTC_OBJC_TYPE(RTCYUVPlanarBuffer)> @end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCIceCandidate.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCIceCandidate.h
new file mode 100644
index 00000000..f84843af
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCIceCandidate.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCIceCandidate) : NSObject
+
+/**
+ * If present, the identifier of the "media stream identification" for the media
+ * component this candidate is associated with.
+ */
+@property(nonatomic, readonly, nullable) NSString *sdpMid;
+
+/**
+ * The index (starting at zero) of the media description this candidate is
+ * associated with in the SDP.
+ */
+@property(nonatomic, readonly) int sdpMLineIndex;
+
+/** The SDP string for this candidate. */
+@property(nonatomic, readonly) NSString *sdp;
+
+/** The URL of the ICE server which this candidate is gathered from. */
+@property(nonatomic, readonly, nullable) NSString *serverUrl;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Initialize an RTCIceCandidate from SDP.
+ */
+- (instancetype)initWithSdp:(NSString *)sdp
+              sdpMLineIndex:(int)sdpMLineIndex
+                     sdpMid:(nullable NSString *)sdpMid NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCIceServer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCIceServer.h
new file mode 100644
index 00000000..dd66c61a
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCIceServer.h
@@ -0,0 +1,114 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSUInteger, RTCTlsCertPolicy) {
+  RTCTlsCertPolicySecure,
+  RTCTlsCertPolicyInsecureNoCheck
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCIceServer) : NSObject
+
+/** URI(s) for this server represented as NSStrings. */
+@property(nonatomic, readonly) NSArray<NSString *> *urlStrings;
+
+/** Username to use if this RTCIceServer object is a TURN server. */
+@property(nonatomic, readonly, nullable) NSString *username;
+
+/** Credential to use if this RTCIceServer object is a TURN server. */
+@property(nonatomic, readonly, nullable) NSString *credential;
+
+/**
+ * TLS certificate policy to use if this RTCIceServer object is a TURN server.
+ */
+@property(nonatomic, readonly) RTCTlsCertPolicy tlsCertPolicy;
+
+/**
+  If the URIs in |urls| only contain IP addresses, this field can be used
+  to indicate the hostname, which may be necessary for TLS (using the SNI
+  extension). If |urls| itself contains the hostname, this isn't necessary.
+ */
+@property(nonatomic, readonly, nullable) NSString *hostname;
+
+/** List of protocols to be used in the TLS ALPN extension. */
+@property(nonatomic, readonly) NSArray<NSString *> *tlsAlpnProtocols;
+
+/**
+  List elliptic curves to be used in the TLS elliptic curves extension.
+  Only curve names supported by OpenSSL should be used (eg. "P-256","X25519").
+  */
+@property(nonatomic, readonly) NSArray<NSString *> *tlsEllipticCurves;
+
+- (nonnull instancetype)init NS_UNAVAILABLE;
+
+/** Convenience initializer for a server with no authentication (e.g. STUN). */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, and credentialType.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, and TLS cert policy.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, TLS cert policy and hostname.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(nullable NSString *)hostname;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, TLS cert policy, hostname and ALPN protocols.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(nullable NSString *)hostname
+                  tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, TLS cert policy, hostname, ALPN protocols and
+ * elliptic curves.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(nullable NSString *)hostname
+                  tlsAlpnProtocols:(nullable NSArray<NSString *> *)tlsAlpnProtocols
+                 tlsEllipticCurves:(nullable NSArray<NSString *> *)tlsEllipticCurves
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCLegacyStatsReport.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCLegacyStatsReport.h
new file mode 100644
index 00000000..b3bd12c5
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCLegacyStatsReport.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This does not currently conform to the spec. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCLegacyStatsReport) : NSObject
+
+/** Time since 1970-01-01T00:00:00Z in milliseconds. */
+@property(nonatomic, readonly) CFTimeInterval timestamp;
+
+/** The type of stats held by this object. */
+@property(nonatomic, readonly) NSString *type;
+
+/** The identifier for this object. */
+@property(nonatomic, readonly) NSString *reportId;
+
+/** A dictionary holding the actual stats. */
+@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *values;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCLogging.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCLogging.h
new file mode 100644
index 00000000..754945c8
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCLogging.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+// Subset of rtc::LoggingSeverity.
+typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
+  RTCLoggingSeverityVerbose,
+  RTCLoggingSeverityInfo,
+  RTCLoggingSeverityWarning,
+  RTCLoggingSeverityError,
+  RTCLoggingSeverityNone,
+};
+
+// Wrapper for C++ RTC_LOG(sev) macros.
+// Logs the log string to the webrtc logstream for the given severity.
+RTC_EXTERN void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
+
+// Wrapper for rtc::LogMessage::LogToDebug.
+// Sets the minimum severity to be logged to console.
+RTC_EXTERN void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
+
+// Returns the filename with the path prefix removed.
+RTC_EXTERN NSString* RTCFileName(const char* filePath);
+
+// Some convenience macros.
+
+#define RTCLogString(format, ...)                                           \
+  [NSString stringWithFormat:@"(%@:%d %s): " format, RTCFileName(__FILE__), \
+                             __LINE__, __FUNCTION__, ##__VA_ARGS__]
+
+#define RTCLogFormat(severity, format, ...)                     \
+  do {                                                          \
+    NSString* log_string = RTCLogString(format, ##__VA_ARGS__); \
+    RTCLogEx(severity, log_string);                             \
+  } while (false)
+
+#define RTCLogVerbose(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__)
+
+#define RTCLogInfo(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__)
+
+#define RTCLogWarning(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__)
+
+#define RTCLogError(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__)
+
+#if !defined(NDEBUG)
+#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
+#else
+#define RTCLogDebug(format, ...) \
+  do {                           \
+  } while (false)
+#endif
+
+#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMTLVideoView.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMTLVideoView.h
new file mode 100644
index 00000000..5678112a
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMTLVideoView.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrame.h"
+#import "RTCVideoRenderer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * RTCMTLVideoView is thin wrapper around MTKView.
+ *
+ * It has id<RTCVideoRenderer> property that renders video frames in the view's
+ * bounds using Metal.
+ * NOTE: always check if metal is available on the running device via
+ * RTC_SUPPORTS_METAL macro before initializing this class.
+ */
+NS_CLASS_AVAILABLE_IOS(9)
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMTLVideoView) : UIView<RTC_OBJC_TYPE(RTCVideoRenderer)>
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoViewDelegate)> delegate;
+
+@property(nonatomic) UIViewContentMode videoContentMode;
+
+/** @abstract Enables/disables rendering.
+ */
+@property(nonatomic, getter=isEnabled) BOOL enabled;
+
+/** @abstract Wrapped RTCVideoRotation, or nil.
+ */
+@property(nonatomic, nullable) NSValue* rotationOverride;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMacros.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMacros.h
new file mode 100644
index 00000000..e527ff6b
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMacros.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_BASE_RTCMACROS_H_
+#define SDK_OBJC_BASE_RTCMACROS_H_
+
+// Internal macros used to correctly concatenate symbols.
+#define RTC_SYMBOL_CONCAT_HELPER(a, b) a##b
+#define RTC_SYMBOL_CONCAT(a, b) RTC_SYMBOL_CONCAT_HELPER(a, b)
+
+// RTC_OBJC_TYPE_PREFIX
+//
+// Macro used to prepend a prefix to the API types that are exported with
+// RTC_OBJC_EXPORT.
+//
+// Clients can patch the definition of this macro locally and build
+// WebRTC.framework with their own prefix in case symbol clashing is a
+// problem.
+//
+// This macro must only be defined here and not on via compiler flag to
+// ensure it has a unique value.
+#define RTC_OBJC_TYPE_PREFIX
+
+// RCT_OBJC_TYPE
+//
+// Macro used internally to declare API types. Declaring an API type without
+// using this macro will not include the declared type in the set of types
+// that will be affected by the configurable RTC_OBJC_TYPE_PREFIX.
+#define RTC_OBJC_TYPE(type_name) RTC_SYMBOL_CONCAT(RTC_OBJC_TYPE_PREFIX, type_name)
+
+#define RTC_OBJC_EXPORT __attribute__((visibility("default")))
+
+#if defined(__cplusplus)
+#define RTC_EXTERN extern "C" RTC_OBJC_EXPORT
+#else
+#define RTC_EXTERN extern RTC_OBJC_EXPORT
+#endif
+
+#ifdef __OBJC__
+#define RTC_FWD_DECL_OBJC_CLASS(classname) @class classname
+#else
+#define RTC_FWD_DECL_OBJC_CLASS(classname) typedef struct objc_object classname
+#endif
+
+#endif  // SDK_OBJC_BASE_RTCMACROS_H_
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaConstraints.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaConstraints.h
new file mode 100644
index 00000000..c5baf20c
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaConstraints.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Constraint keys for media sources. */
+/** The value for this key should be a base64 encoded string containing
+ *  the data from the serialized configuration proto.
+ */
+RTC_EXTERN NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
+
+/** Constraint keys for generating offers and answers. */
+RTC_EXTERN NSString *const kRTCMediaConstraintsIceRestart;
+RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveAudio;
+RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveVideo;
+RTC_EXTERN NSString *const kRTCMediaConstraintsVoiceActivityDetection;
+
+/** Constraint values for Boolean parameters. */
+RTC_EXTERN NSString *const kRTCMediaConstraintsValueTrue;
+RTC_EXTERN NSString *const kRTCMediaConstraintsValueFalse;
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaConstraints) : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize with mandatory and/or optional constraints. */
+- (instancetype)
+    initWithMandatoryConstraints:(nullable NSDictionary<NSString *, NSString *> *)mandatory
+             optionalConstraints:(nullable NSDictionary<NSString *, NSString *> *)optional
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaSource.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaSource.h
new file mode 100644
index 00000000..ba19c2a3
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaSource.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSInteger, RTCSourceState) {
+  RTCSourceStateInitializing,
+  RTCSourceStateLive,
+  RTCSourceStateEnded,
+  RTCSourceStateMuted,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaSource) : NSObject
+
+/** The current state of the RTCMediaSource. */
+@property(nonatomic, readonly) RTCSourceState state;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaStream.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaStream.h
new file mode 100644
index 00000000..2d56f15c
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaStream.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@class RTC_OBJC_TYPE(RTCVideoTrack);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaStream) : NSObject
+
+/** The audio tracks in this stream. */
+@property(nonatomic, strong, readonly) NSArray<RTC_OBJC_TYPE(RTCAudioTrack) *> *audioTracks;
+
+/** The video tracks in this stream. */
+@property(nonatomic, strong, readonly) NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *videoTracks;
+
+/** An identifier for this media stream. */
+@property(nonatomic, readonly) NSString *streamId;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Adds the given audio track to this media stream. */
+- (void)addAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack;
+
+/** Adds the given video track to this media stream. */
+- (void)addVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack;
+
+/** Removes the given audio track to this media stream. */
+- (void)removeAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack;
+
+/** Removes the given video track to this media stream. */
+- (void)removeVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaStreamTrack.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaStreamTrack.h
new file mode 100644
index 00000000..2200122c
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMediaStreamTrack.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/**
+ * Represents the state of the track. This exposes the same states in C++.
+ */
+typedef NS_ENUM(NSInteger, RTCMediaStreamTrackState) {
+  RTCMediaStreamTrackStateLive,
+  RTCMediaStreamTrackStateEnded
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN NSString *const kRTCMediaStreamTrackKindAudio;
+RTC_EXTERN NSString *const kRTCMediaStreamTrackKindVideo;
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaStreamTrack) : NSObject
+
+/**
+ * The kind of track. For example, "audio" if this track represents an audio
+ * track and "video" if this track represents a video track.
+ */
+@property(nonatomic, readonly) NSString *kind;
+
+/** An identifier string. */
+@property(nonatomic, readonly) NSString *trackId;
+
+/** The enabled state of the track. */
+@property(nonatomic, assign) BOOL isEnabled;
+
+/** The state of the track. */
+@property(nonatomic, readonly) RTCMediaStreamTrackState readyState;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMetrics.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMetrics.h
new file mode 100644
index 00000000..fddbb27c
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMetrics.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMetricsSampleInfo.h"
+
+/**
+ * Enables gathering of metrics (which can be fetched with
+ * RTCGetAndResetMetrics). Must be called before any other call into WebRTC.
+ */
+RTC_EXTERN void RTCEnableMetrics(void);
+
+/** Gets and clears native histograms. */
+RTC_EXTERN NSArray<RTC_OBJC_TYPE(RTCMetricsSampleInfo) *>* RTCGetAndResetMetrics(void);
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMetricsSampleInfo.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMetricsSampleInfo.h
new file mode 100644
index 00000000..47a877b6
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMetricsSampleInfo.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMetricsSampleInfo) : NSObject
+
+/**
+ * Example of RTCMetricsSampleInfo:
+ * name: "WebRTC.Video.InputFramesPerSecond"
+ * min: 1
+ * max: 100
+ * bucketCount: 50
+ * samples: [29]:2 [30]:1
+ */
+
+/** The name of the histogram. */
+@property(nonatomic, readonly) NSString *name;
+
+/** The minimum bucket value. */
+@property(nonatomic, readonly) int min;
+
+/** The maximum bucket value. */
+@property(nonatomic, readonly) int max;
+
+/** The number of buckets. */
+@property(nonatomic, readonly) int bucketCount;
+
+/** A dictionary holding the samples <value, # of events>. */
+@property(nonatomic, readonly) NSDictionary<NSNumber *, NSNumber *> *samples;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMutableI420Buffer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMutableI420Buffer.h
new file mode 100644
index 00000000..cde72198
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMutableI420Buffer.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCI420Buffer.h"
+#import "RTCMutableYUVPlanarBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Extension of the I420 buffer with mutable data access */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCMutableI420Buffer)<RTC_OBJC_TYPE(RTCI420Buffer), RTC_OBJC_TYPE(RTCMutableYUVPlanarBuffer)> @end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMutableYUVPlanarBuffer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMutableYUVPlanarBuffer.h
new file mode 100644
index 00000000..bd14e3bc
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCMutableYUVPlanarBuffer.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCYUVPlanarBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Extension of the YUV planar data buffer with mutable data access */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCMutableYUVPlanarBuffer)<RTC_OBJC_TYPE(RTCYUVPlanarBuffer)>
+
+    @property(nonatomic, readonly) uint8_t *mutableDataY;
+@property(nonatomic, readonly) uint8_t *mutableDataU;
+@property(nonatomic, readonly) uint8_t *mutableDataV;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCNativeI420Buffer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCNativeI420Buffer.h
new file mode 100644
index 00000000..3afe2090
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCNativeI420Buffer.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCI420Buffer.h"
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCI420Buffer implements the RTCI420Buffer protocol */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCI420Buffer) : NSObject<RTC_OBJC_TYPE(RTCI420Buffer)>
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCNativeMutableI420Buffer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCNativeMutableI420Buffer.h
new file mode 100644
index 00000000..053a10a3
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCNativeMutableI420Buffer.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMutableI420Buffer.h"
+#import "RTCNativeI420Buffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Mutable version of RTCI420Buffer */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMutableI420Buffer) : RTC_OBJC_TYPE(RTCI420Buffer)<RTC_OBJC_TYPE(RTCMutableI420Buffer)>
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCNetworkMonitor.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCNetworkMonitor.h
new file mode 100644
index 00000000..21d22f54
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCNetworkMonitor.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2020 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Listens for NWPathMonitor updates and forwards the results to a C++
+ *  observer.
+ */
+@interface RTCNetworkMonitor : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCPeerConnection.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCPeerConnection.h
new file mode 100644
index 00000000..5401666f
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCPeerConnection.h
@@ -0,0 +1,374 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCConfiguration);
+@class RTC_OBJC_TYPE(RTCDataChannel);
+@class RTC_OBJC_TYPE(RTCDataChannelConfiguration);
+@class RTC_OBJC_TYPE(RTCIceCandidate);
+@class RTC_OBJC_TYPE(RTCMediaConstraints);
+@class RTC_OBJC_TYPE(RTCMediaStream);
+@class RTC_OBJC_TYPE(RTCMediaStreamTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@class RTC_OBJC_TYPE(RTCRtpReceiver);
+@class RTC_OBJC_TYPE(RTCRtpSender);
+@class RTC_OBJC_TYPE(RTCRtpTransceiver);
+@class RTC_OBJC_TYPE(RTCRtpTransceiverInit);
+@class RTC_OBJC_TYPE(RTCSessionDescription);
+@class RTC_OBJC_TYPE(RTCStatisticsReport);
+@class RTC_OBJC_TYPE(RTCLegacyStatsReport);
+
+typedef NS_ENUM(NSInteger, RTCRtpMediaType);
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const kRTCPeerConnectionErrorDomain;
+extern int const kRTCSessionDescriptionErrorCode;
+
+/** Represents the signaling state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCSignalingState) {
+  RTCSignalingStateStable,
+  RTCSignalingStateHaveLocalOffer,
+  RTCSignalingStateHaveLocalPrAnswer,
+  RTCSignalingStateHaveRemoteOffer,
+  RTCSignalingStateHaveRemotePrAnswer,
+  // Not an actual state, represents the total number of states.
+  RTCSignalingStateClosed,
+};
+
+/** Represents the ice connection state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCIceConnectionState) {
+  RTCIceConnectionStateNew,
+  RTCIceConnectionStateChecking,
+  RTCIceConnectionStateConnected,
+  RTCIceConnectionStateCompleted,
+  RTCIceConnectionStateFailed,
+  RTCIceConnectionStateDisconnected,
+  RTCIceConnectionStateClosed,
+  RTCIceConnectionStateCount,
+};
+
+/** Represents the combined ice+dtls connection state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCPeerConnectionState) {
+  RTCPeerConnectionStateNew,
+  RTCPeerConnectionStateConnecting,
+  RTCPeerConnectionStateConnected,
+  RTCPeerConnectionStateDisconnected,
+  RTCPeerConnectionStateFailed,
+  RTCPeerConnectionStateClosed,
+};
+
+/** Represents the ice gathering state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCIceGatheringState) {
+  RTCIceGatheringStateNew,
+  RTCIceGatheringStateGathering,
+  RTCIceGatheringStateComplete,
+};
+
+/** Represents the stats output level. */
+typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
+  RTCStatsOutputLevelStandard,
+  RTCStatsOutputLevelDebug,
+};
+
+@class RTC_OBJC_TYPE(RTCPeerConnection);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCPeerConnectionDelegate)<NSObject>
+
+    /** Called when the SignalingState changed. */
+    - (void)peerConnection
+    : (RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection didChangeSignalingState
+    : (RTCSignalingState)stateChanged;
+
+/** Called when media is received on a new stream from remote peer. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+          didAddStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Called when a remote peer closes a stream.
+ *  This is not called when RTCSdpSemanticsUnifiedPlan is specified.
+ */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+       didRemoveStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Called when negotiation is needed, for example ICE has restarted. */
+- (void)peerConnectionShouldNegotiate:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection;
+
+/** Called any time the IceConnectionState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeIceConnectionState:(RTCIceConnectionState)newState;
+
+/** Called any time the IceGatheringState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeIceGatheringState:(RTCIceGatheringState)newState;
+
+/** New ice candidate has been found. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didGenerateIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate;
+
+/** Called when a group of local Ice candidates have been removed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didRemoveIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+
+/** New data channel has been opened. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didOpenDataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
+
+/** Called when signaling indicates a transceiver will be receiving media from
+ *  the remote endpoint.
+ *  This is only called with RTCSdpSemanticsUnifiedPlan specified.
+ */
+@optional
+/** Called any time the IceConnectionState changes following standardized
+ * transition. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeStandardizedIceConnectionState:(RTCIceConnectionState)newState;
+
+/** Called any time the PeerConnectionState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeConnectionState:(RTCPeerConnectionState)newState;
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didStartReceivingOnTransceiver:(RTC_OBJC_TYPE(RTCRtpTransceiver) *)transceiver;
+
+/** Called when a receiver and its track are created. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+        didAddReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver
+               streams:(NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *)mediaStreams;
+
+/** Called when the receiver and its track are removed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+     didRemoveReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver;
+
+/** Called when the selected ICE candidate pair is changed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeLocalCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)local
+            remoteCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)remote
+             lastReceivedMs:(int)lastDataReceivedMs
+               changeReason:(NSString *)reason;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnection) : NSObject
+
+/** The object that will be notifed about events such as state changes and
+ *  streams being added or removed.
+ */
+@property(nonatomic, weak, nullable) id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)> delegate;
+/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
+ *  |senders| instead.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *localStreams;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * localDescription;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * remoteDescription;
+@property(nonatomic, readonly) RTCSignalingState signalingState;
+@property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
+@property(nonatomic, readonly) RTCPeerConnectionState connectionState;
+@property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
+@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCConfiguration) * configuration;
+
+/** Gets all RTCRtpSenders associated with this peer connection.
+ *  Note: reading this property returns different instances of RTCRtpSender.
+ *  Use isEqual: instead of == to compare RTCRtpSender instances.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpSender) *> *senders;
+
+/** Gets all RTCRtpReceivers associated with this peer connection.
+ *  Note: reading this property returns different instances of RTCRtpReceiver.
+ *  Use isEqual: instead of == to compare RTCRtpReceiver instances.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpReceiver) *> *receivers;
+
+/** Gets all RTCRtpTransceivers associated with this peer connection.
+ *  Note: reading this property returns different instances of
+ *  RTCRtpTransceiver. Use isEqual: instead of == to compare
+ *  RTCRtpTransceiver instances. This is only available with
+ * RTCSdpSemanticsUnifiedPlan specified.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpTransceiver) *> *transceivers;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Sets the PeerConnection's global configuration to |configuration|.
+ *  Any changes to STUN/TURN servers or ICE candidate policy will affect the
+ *  next gathering phase, and cause the next call to createOffer to generate
+ *  new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
+ *  cannot be changed with this method.
+ */
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration;
+
+/** Terminate all media and close the transport. */
+- (void)close;
+
+/** Provide a remote candidate to the ICE Agent. */
+- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate;
+
+/** Remove a group of remote candidates from the ICE Agent. */
+- (void)removeIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+
+/** Add a new media stream to be sent on this peer connection.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  addTrack instead.
+ */
+- (void)addStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Remove the given media stream from this peer connection.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  removeTrack instead.
+ */
+- (void)removeStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Add a new media stream track to be sent on this peer connection, and return
+ *  the newly created RTCRtpSender. The RTCRtpSender will be
+ * associated with the streams specified in the |streamIds| list.
+ *
+ *  Errors: If an error occurs, returns nil. An error can occur if:
+ *  - A sender already exists for the track.
+ *  - The peer connection is closed.
+ */
+- (RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+                                streamIds:(NSArray<NSString *> *)streamIds;
+
+/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
+ *
+ *  With UnifiedPlan semantics, sets sender's track to null and removes the
+ *  send component from the associated RTCRtpTransceiver's direction.
+ *
+ *  Returns YES on success.
+ */
+- (BOOL)removeTrack:(RTC_OBJC_TYPE(RTCRtpSender) *)sender;
+
+/** addTransceiver creates a new RTCRtpTransceiver and adds it to the set of
+ *  transceivers. Adding a transceiver will cause future calls to CreateOffer
+ *  to add a media description for the corresponding transceiver.
+ *
+ *  The initial value of |mid| in the returned transceiver is nil. Setting a
+ *  new session description may change it to a non-nil value.
+ *
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
+ *
+ *  Optionally, an RtpTransceiverInit structure can be specified to configure
+ *  the transceiver from construction. If not specified, the transceiver will
+ *  default to having a direction of kSendRecv and not be part of any streams.
+ *
+ *  These methods are only available when Unified Plan is enabled (see
+ *  RTCConfiguration).
+ */
+
+/** Adds a transceiver with a sender set to transmit the given track. The kind
+ *  of the transceiver (and sender/receiver) will be derived from the kind of
+ *  the track.
+ */
+- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
+    (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track;
+- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)
+    addTransceiverWithTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+                       init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
+
+/** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
+ *  or RTCRtpMediaTypeVideo.
+ */
+- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
+- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType
+                                                      init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)
+                                                               init;
+
+/** Generate an SDP offer. */
+- (void)offerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+          completionHandler:(nullable void (^)(RTC_OBJC_TYPE(RTCSessionDescription) * _Nullable sdp,
+                                               NSError *_Nullable error))completionHandler;
+
+/** Generate an SDP answer. */
+- (void)answerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+           completionHandler:
+               (nullable void (^)(RTC_OBJC_TYPE(RTCSessionDescription) * _Nullable sdp,
+                                  NSError *_Nullable error))completionHandler;
+
+/** Apply the supplied RTCSessionDescription as the local description. */
+- (void)setLocalDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+          completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
+
+/** Apply the supplied RTCSessionDescription as the remote description. */
+- (void)setRemoteDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+           completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
+
+/** Limits the bandwidth allocated for all RTP streams sent by this
+ *  PeerConnection. Nil parameters will be unchanged. Setting
+ * |currentBitrateBps| will force the available bitrate estimate to the given
+ *  value. Returns YES if the parameters were successfully updated.
+ */
+- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
+          currentBitrateBps:(nullable NSNumber *)currentBitrateBps
+              maxBitrateBps:(nullable NSNumber *)maxBitrateBps;
+
+/** Start or stop recording an Rtc EventLog. */
+- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
+- (void)stopRtcEventLog;
+
+@end
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(Media)
+
+    /** Create an RTCRtpSender with the specified kind and media stream ID.
+     *  See RTCMediaStreamTrack.h for available kinds.
+     *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+     *  addTransceiver instead.
+     */
+    - (RTC_OBJC_TYPE(RTCRtpSender) *)senderWithKind : (NSString *)kind streamId
+    : (NSString *)streamId;
+
+@end
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(DataChannel)
+
+    /** Create a new data channel with the given label and configuration. */
+    - (nullable RTC_OBJC_TYPE(RTCDataChannel) *)dataChannelForLabel
+    : (NSString *)label configuration : (RTC_OBJC_TYPE(RTCDataChannelConfiguration) *)configuration;
+
+@end
+
+typedef void (^RTCStatisticsCompletionHandler)(RTC_OBJC_TYPE(RTCStatisticsReport) *);
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(Stats)
+
+    /** Gather stats for the given RTCMediaStreamTrack. If |mediaStreamTrack| is nil
+     *  statistics are gathered for all tracks.
+     */
+    - (void)statsForTrack
+    : (nullable RTC_OBJC_TYPE(RTCMediaStreamTrack) *)mediaStreamTrack statsOutputLevel
+    : (RTCStatsOutputLevel)statsOutputLevel completionHandler
+    : (nullable void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))completionHandler;
+
+/** Gather statistic through the v2 statistics API. */
+- (void)statisticsWithCompletionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+/** Spec-compliant getStats() performing the stats selection algorithm with the
+ *  sender.
+ */
+- (void)statisticsForSender:(RTC_OBJC_TYPE(RTCRtpSender) *)sender
+          completionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+/** Spec-compliant getStats() performing the stats selection algorithm with the
+ *  receiver.
+ */
+- (void)statisticsForReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)receiver
+            completionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+@end
+
+NS_ASSUME_NONNULL_END
\ No newline at end of file
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCPeerConnectionFactory.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCPeerConnectionFactory.h
new file mode 100644
index 00000000..3dcd3b64
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCPeerConnectionFactory.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioSource);
+@class RTC_OBJC_TYPE(RTCAudioTrack);
+@class RTC_OBJC_TYPE(RTCConfiguration);
+@class RTC_OBJC_TYPE(RTCMediaConstraints);
+@class RTC_OBJC_TYPE(RTCMediaStream);
+@class RTC_OBJC_TYPE(RTCPeerConnection);
+@class RTC_OBJC_TYPE(RTCVideoSource);
+@class RTC_OBJC_TYPE(RTCVideoTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions);
+@protocol RTC_OBJC_TYPE
+(RTCPeerConnectionDelegate);
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoderFactory);
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderFactory);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactory) : NSObject
+
+/* Initialize object with default H264 video encoder/decoder factories */
+- (instancetype)init;
+
+/* Initialize object with injectable video encoder/decoder factories */
+- (instancetype)
+    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
+
+/** Initialize an RTCAudioSource with constraints. */
+- (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
+    (nullable RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints;
+
+/** Initialize an RTCAudioTrack with an id. Convenience ctor to use an audio source
+ * with no constraints.
+ */
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithTrackId:(NSString *)trackId;
+
+/** Initialize an RTCAudioTrack with a source and an id. */
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:(RTC_OBJC_TYPE(RTCAudioSource) *)source
+                                               trackId:(NSString *)trackId;
+
+/** Initialize a generic RTCVideoSource. The RTCVideoSource should be
+ * passed to a RTCVideoCapturer implementation, e.g.
+ * RTCCameraVideoCapturer, in order to produce frames.
+ */
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSource;
+
+/** Initialize an RTCVideoTrack with a source and an id. */
+- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
+                                               trackId:(NSString *)trackId;
+
+/** Initialize an RTCMediaStream with an id. */
+- (RTC_OBJC_TYPE(RTCMediaStream) *)mediaStreamWithStreamId:(NSString *)streamId;
+
+/** Initialize an RTCPeerConnection with a configuration, constraints, and
+ *  delegate.
+ */
+- (RTC_OBJC_TYPE(RTCPeerConnection) *)
+    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                           delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+
+/** Set the options to be used for subsequently created RTCPeerConnections */
+- (void)setOptions:(nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options;
+
+/** Start an AecDump recording. This API call will likely change in the future. */
+- (BOOL)startAecDumpWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
+
+/* Stop an active AecDump recording */
+- (void)stopAecDump;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCPeerConnectionFactoryOptions.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCPeerConnectionFactoryOptions.h
new file mode 100644
index 00000000..bfc54a5d
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCPeerConnectionFactoryOptions.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactoryOptions) : NSObject
+
+@property(nonatomic, assign) BOOL disableEncryption;
+
+@property(nonatomic, assign) BOOL disableNetworkMonitor;
+
+@property(nonatomic, assign) BOOL ignoreLoopbackNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreVPNNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreCellularNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreWiFiNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreEthernetNetworkAdapter;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtcpParameters.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtcpParameters.h
new file mode 100644
index 00000000..1bbaedcf
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtcpParameters.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtcpParameters) : NSObject
+
+/** The Canonical Name used by RTCP. */
+@property(nonatomic, readonly, copy) NSString *cname;
+
+/** Whether reduced size RTCP is configured or compound RTCP. */
+@property(nonatomic, assign) BOOL isReducedSize;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpCodecParameters.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpCodecParameters.h
new file mode 100644
index 00000000..a68d9eb8
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpCodecParameters.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN const NSString *const kRTCRtxCodecName;
+RTC_EXTERN const NSString *const kRTCRedCodecName;
+RTC_EXTERN const NSString *const kRTCUlpfecCodecName;
+RTC_EXTERN const NSString *const kRTCFlexfecCodecName;
+RTC_EXTERN const NSString *const kRTCOpusCodecName;
+RTC_EXTERN const NSString *const kRTCIsacCodecName;
+RTC_EXTERN const NSString *const kRTCL16CodecName;
+RTC_EXTERN const NSString *const kRTCG722CodecName;
+RTC_EXTERN const NSString *const kRTCIlbcCodecName;
+RTC_EXTERN const NSString *const kRTCPcmuCodecName;
+RTC_EXTERN const NSString *const kRTCPcmaCodecName;
+RTC_EXTERN const NSString *const kRTCDtmfCodecName;
+RTC_EXTERN const NSString *const kRTCComfortNoiseCodecName;
+RTC_EXTERN const NSString *const kRTCVp8CodecName;
+RTC_EXTERN const NSString *const kRTCVp9CodecName;
+RTC_EXTERN const NSString *const kRTCH264CodecName;
+
+/** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTC_OBJC_TYPE(RTCRtpCodecParameters) */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpCodecParameters) : NSObject
+
+/** The RTP payload type. */
+@property(nonatomic, assign) int payloadType;
+
+/**
+ * The codec MIME subtype. Valid types are listed in:
+ * http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2
+ *
+ * Several supported types are represented by the constants above.
+ */
+@property(nonatomic, readonly, nonnull) NSString *name;
+
+/**
+ * The media type of this codec. Equivalent to MIME top-level type.
+ *
+ * Valid values are kRTCMediaStreamTrackKindAudio and
+ * kRTCMediaStreamTrackKindVideo.
+ */
+@property(nonatomic, readonly, nonnull) NSString *kind;
+
+/** The codec clock rate expressed in Hertz. */
+@property(nonatomic, readonly, nullable) NSNumber *clockRate;
+
+/**
+ * The number of channels (mono=1, stereo=2).
+ * Set to null for video codecs.
+ **/
+@property(nonatomic, readonly, nullable) NSNumber *numChannels;
+
+/** The "format specific parameters" field from the "a=fmtp" line in the SDP */
+@property(nonatomic, readonly, nonnull) NSDictionary *parameters;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpEncodingParameters.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpEncodingParameters.h
new file mode 100644
index 00000000..88b8e49e
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpEncodingParameters.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Corresponds to webrtc::Priority. */
+typedef NS_ENUM(NSInteger, RTCPriority) {
+  RTCPriorityVeryLow,
+  RTCPriorityLow,
+  RTCPriorityMedium,
+  RTCPriorityHigh
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpEncodingParameters) : NSObject
+
+/** The idenfifier for the encoding layer. This is used in simulcast. */
+@property(nonatomic, copy, nullable) NSString *rid;
+
+/** Controls whether the encoding is currently transmitted. */
+@property(nonatomic, assign) BOOL isActive;
+
+/** The maximum bitrate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *maxBitrateBps;
+
+/** The minimum bitrate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *minBitrateBps;
+
+/** The maximum framerate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *maxFramerate;
+
+/** The requested number of temporal layers to use for the encoding, or nil
+ * if the default should be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *numTemporalLayers;
+
+/** Scale the width and height down by this factor for video. If nil,
+ * implementation default scaling factor will be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *scaleResolutionDownBy;
+
+/** The SSRC being used by this encoding. */
+@property(nonatomic, readonly, nullable) NSNumber *ssrc;
+
+/** The relative bitrate priority. */
+@property(nonatomic, assign) double bitratePriority;
+
+/** The relative DiffServ Code Point priority. */
+@property(nonatomic, assign) RTCPriority networkPriority;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
\ No newline at end of file
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpFragmentationHeader.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpFragmentationHeader.h
new file mode 100644
index 00000000..001b4e9d
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpFragmentationHeader.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Information for header. Corresponds to webrtc::RTPFragmentationHeader. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpFragmentationHeader) : NSObject
+
+@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationOffset;
+@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationLength;
+@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationTimeDiff;
+@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationPlType;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpHeaderExtension.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpHeaderExtension.h
new file mode 100644
index 00000000..15be5af5
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpHeaderExtension.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpHeaderExtension) : NSObject
+
+/** The URI of the RTP header extension, as defined in RFC5285. */
+@property(nonatomic, readonly, copy) NSString *uri;
+
+/** The value put in the RTP packet to identify the header extension. */
+@property(nonatomic, readonly) int id;
+
+/** Whether the header extension is encrypted or not. */
+@property(nonatomic, readonly, getter=isEncrypted) BOOL encrypted;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpParameters.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpParameters.h
new file mode 100644
index 00000000..fff6a858
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpParameters.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCRtcpParameters.h"
+#import "RTCRtpCodecParameters.h"
+#import "RTCRtpEncodingParameters.h"
+#import "RTCRtpHeaderExtension.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Corresponds to webrtc::DegradationPreference. */
+typedef NS_ENUM(NSInteger, RTCDegradationPreference) {
+  RTCDegradationPreferenceDisabled,
+  RTCDegradationPreferenceMaintainFramerate,
+  RTCDegradationPreferenceMaintainResolution,
+  RTCDegradationPreferenceBalanced
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpParameters) : NSObject
+
+/** A unique identifier for the last set of parameters applied. */
+@property(nonatomic, copy) NSString *transactionId;
+
+/** Parameters used for RTCP. */
+@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCRtcpParameters) * rtcp;
+
+/** An array containing parameters for RTP header extensions. */
+@property(nonatomic, readonly, copy)
+    NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtension) *> *headerExtensions;
+
+/** The currently active encodings in the order of preference. */
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *encodings;
+
+/** The negotiated set of send codecs in order of preference. */
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpCodecParameters) *> *codecs;
+
+/**
+ * Degradation preference in case of CPU adaptation or constrained bandwidth.
+ * If nil, implementation default degradation preference will be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *degradationPreference;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpReceiver.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpReceiver.h
new file mode 100644
index 00000000..7ab2cfae
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpReceiver.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMediaStreamTrack.h"
+#import "RTCRtpParameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Represents the media type of the RtpReceiver. */
+typedef NS_ENUM(NSInteger, RTCRtpMediaType) {
+  RTCRtpMediaTypeAudio,
+  RTCRtpMediaTypeVideo,
+  RTCRtpMediaTypeData,
+};
+
+@class RTC_OBJC_TYPE(RTCRtpReceiver);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpReceiverDelegate)<NSObject>
+
+    /** Called when the first RTP packet is received.
+     *
+     *  Note: Currently if there are multiple RtpReceivers of the same media type,
+     *  they will all call OnFirstPacketReceived at once.
+     *
+     *  For example, if we create three audio receivers, A/B/C, they will listen to
+     *  the same signal from the underneath network layer. Whenever the first audio packet
+     *  is received, the underneath signal will be fired. All the receivers A/B/C will be
+     *  notified and the callback of the receiver's delegate will be called.
+     *
+     *  The process is the same for video receivers.
+     */
+    - (void)rtpReceiver
+    : (RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver didReceiveFirstPacketForMediaType
+    : (RTCRtpMediaType)mediaType;
+
+@end
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpReceiver)<NSObject>
+
+    /** A unique identifier for this receiver. */
+    @property(nonatomic, readonly) NSString *receiverId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ *
+ *  The WebRTC specification only defines RTCRtpParameters in terms of senders,
+ *  but this API also applies them to receivers, similar to ORTC:
+ *  http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
+ */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpParameters) * parameters;
+
+/** The RTCMediaStreamTrack associated with the receiver.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track;
+
+/** The delegate for this RtpReceiver. */
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCRtpReceiverDelegate)> delegate;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpReceiver) : NSObject <RTC_OBJC_TYPE(RTCRtpReceiver)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpSender.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpSender.h
new file mode 100644
index 00000000..fcdf1998
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpSender.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCDtmfSender.h"
+#import "RTCMacros.h"
+#import "RTCMediaStreamTrack.h"
+#import "RTCRtpParameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpSender)<NSObject>
+
+/** A unique identifier for this sender. */
+@property(nonatomic, readonly) NSString *senderId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ */
+@property(nonatomic, copy) RTC_OBJC_TYPE(RTCRtpParameters) * parameters;
+
+/** The RTCMediaStreamTrack associated with the sender.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, copy, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track;
+
+/** IDs of streams associated with the RTP sender */
+@property(nonatomic, copy) NSArray<NSString *> *streamIds;
+
+/** The RTCDtmfSender accociated with the RTP sender. */
+@property(nonatomic, readonly, nullable) id<RTC_OBJC_TYPE(RTCDtmfSender)> dtmfSender;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpSender) : NSObject <RTC_OBJC_TYPE(RTCRtpSender)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpTransceiver.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpTransceiver.h
new file mode 100644
index 00000000..fd590136
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCRtpTransceiver.h
@@ -0,0 +1,137 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCRtpReceiver.h"
+#import "RTCRtpSender.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const kRTCRtpTransceiverErrorDomain;
+
+/** https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverdirection */
+typedef NS_ENUM(NSInteger, RTCRtpTransceiverDirection) {
+  RTCRtpTransceiverDirectionSendRecv,
+  RTCRtpTransceiverDirectionSendOnly,
+  RTCRtpTransceiverDirectionRecvOnly,
+  RTCRtpTransceiverDirectionInactive,
+  RTCRtpTransceiverDirectionStopped
+};
+
+/** Structure for initializing an RTCRtpTransceiver in a call to
+ *  RTCPeerConnection.addTransceiver.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpTransceiverInit) : NSObject
+
+/** Direction of the RTCRtpTransceiver. See RTCRtpTransceiver.direction. */
+@property(nonatomic) RTCRtpTransceiverDirection direction;
+
+/** The added RTCRtpTransceiver will be added to these streams. */
+@property(nonatomic) NSArray<NSString *> *streamIds;
+
+/** TODO(bugs.webrtc.org/7600): Not implemented. */
+@property(nonatomic) NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *sendEncodings;
+
+@end
+
+@class RTC_OBJC_TYPE(RTCRtpTransceiver);
+
+/** The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the
+ *  WebRTC specification. A transceiver represents a combination of an RTCRtpSender
+ *  and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
+ *  RTCRtpTransceiver is said to be associated with a media description if its
+ *  mid property is non-nil; otherwise, it is said to be disassociated.
+ *  JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
+ *
+ *  Note that RTCRtpTransceivers are only supported when using
+ *  RTCPeerConnection with Unified Plan SDP.
+ *
+ *  WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpTransceiver)<NSObject>
+
+    /** Media type of the transceiver. The sender and receiver will also have this
+     *  type.
+     */
+    @property(nonatomic, readonly) RTCRtpMediaType mediaType;
+
+/** The mid attribute is the mid negotiated and present in the local and
+ *  remote descriptions. Before negotiation is complete, the mid value may be
+ *  nil. After rollbacks, the value may change from a non-nil value to nil.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
+ */
+@property(nonatomic, readonly) NSString *mid;
+
+/** The sender attribute exposes the RTCRtpSender corresponding to the RTP
+ *  media that may be sent with the transceiver's mid. The sender is always
+ *  present, regardless of the direction of media.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
+ */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpSender) * sender;
+
+/** The receiver attribute exposes the RTCRtpReceiver corresponding to the RTP
+ *  media that may be received with the transceiver's mid. The receiver is
+ *  always present, regardless of the direction of media.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
+ */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpReceiver) * receiver;
+
+/** The isStopped attribute indicates that the sender of this transceiver will
+ *  no longer send, and that the receiver will no longer receive. It is true if
+ *  either stop has been called or if setting the local or remote description
+ *  has caused the RTCRtpTransceiver to be stopped.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
+ */
+@property(nonatomic, readonly) BOOL isStopped;
+
+/** The direction attribute indicates the preferred direction of this
+ *  transceiver, which will be used in calls to createOffer and createAnswer.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
+ */
+@property(nonatomic, readonly) RTCRtpTransceiverDirection direction;
+
+/** The currentDirection attribute indicates the current direction negotiated
+ *  for this transceiver. If this transceiver has never been represented in an
+ *  offer/answer exchange, or if the transceiver is stopped, the value is not
+ *  present and this method returns NO.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
+ */
+- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut;
+
+/** The stop method irreversibly stops the RTCRtpTransceiver. The sender of
+ *  this transceiver will no longer send, the receiver will no longer receive.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
+ */
+- (void)stopInternal;
+
+/** An update of directionality does not take effect immediately. Instead,
+ *  future calls to createOffer and createAnswer mark the corresponding media
+ *  descriptions as sendrecv, sendonly, recvonly, or inactive.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
+ */
+- (void)setDirection:(RTCRtpTransceiverDirection)direction error:(NSError **)error;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpTransceiver) : NSObject <RTC_OBJC_TYPE(RTCRtpTransceiver)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCSSLAdapter.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCSSLAdapter.h
new file mode 100644
index 00000000..f68bc5e9
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCSSLAdapter.h
@@ -0,0 +1,20 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/**
+ * Initialize and clean up the SSL library. Failure is fatal. These call the
+ * corresponding functions in webrtc/rtc_base/ssladapter.h.
+ */
+RTC_EXTERN BOOL RTCInitializeSSL(void);
+RTC_EXTERN BOOL RTCCleanupSSL(void);
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCSessionDescription.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCSessionDescription.h
new file mode 100644
index 00000000..8a9479d5
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCSessionDescription.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/**
+ * Represents the session description type. This exposes the same types that are
+ * in C++, which doesn't include the rollback type that is in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCSdpType) {
+  RTCSdpTypeOffer,
+  RTCSdpTypePrAnswer,
+  RTCSdpTypeAnswer,
+  RTCSdpTypeRollback,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCSessionDescription) : NSObject
+
+/** The type of session description. */
+@property(nonatomic, readonly) RTCSdpType type;
+
+/** The SDP string representation of this session description. */
+@property(nonatomic, readonly) NSString *sdp;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize a session description with a type and SDP string. */
+- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp NS_DESIGNATED_INITIALIZER;
+
++ (NSString *)stringForType:(RTCSdpType)type;
+
++ (RTCSdpType)typeForString:(NSString *)string;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCStatisticsReport.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCStatisticsReport.h
new file mode 100644
index 00000000..38d93e87
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCStatisticsReport.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCStatistics);
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** A statistics report. Encapsulates a number of RTCStatistics objects. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCStatisticsReport) : NSObject
+
+/** The timestamp of the report in microseconds since 1970-01-01T00:00:00Z. */
+@property(nonatomic, readonly) CFTimeInterval timestamp_us;
+
+/** RTCStatistics objects by id. */
+@property(nonatomic, readonly) NSDictionary<NSString *, RTC_OBJC_TYPE(RTCStatistics) *> *statistics;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+/** A part of a report (a subreport) covering a certain area. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCStatistics) : NSObject
+
+/** The id of this subreport, e.g. "RTCMediaStreamTrack_receiver_2". */
+@property(nonatomic, readonly) NSString *id;
+
+/** The timestamp of the subreport in microseconds since 1970-01-01T00:00:00Z. */
+@property(nonatomic, readonly) CFTimeInterval timestamp_us;
+
+/** The type of the subreport, e.g. "track", "codec". */
+@property(nonatomic, readonly) NSString *type;
+
+/** The keys and values of the subreport, e.g. "totalFramesDuration = 5.551".
+    The values are either NSNumbers or NSStrings, or NSArrays encapsulating NSNumbers
+    or NSStrings. */
+@property(nonatomic, readonly) NSDictionary<NSString *, NSObject *> *values;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCTracing.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCTracing.h
new file mode 100644
index 00000000..5c66e5a6
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCTracing.h
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+RTC_EXTERN void RTCSetupInternalTracer(void);
+/** Starts capture to specified file. Must be a valid writable path.
+ *  Returns YES if capture starts.
+ */
+RTC_EXTERN BOOL RTCStartInternalCapture(NSString* filePath);
+RTC_EXTERN void RTCStopInternalCapture(void);
+RTC_EXTERN void RTCShutdownInternalTracer(void);
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoCapturer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoCapturer.h
new file mode 100644
index 00000000..a1ffdcf3
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoCapturer.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoFrame.h"
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCVideoCapturer);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoCapturerDelegate)<NSObject> -
+    (void)capturer : (RTC_OBJC_TYPE(RTCVideoCapturer) *)capturer didCaptureVideoFrame
+    : (RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoCapturer) : NSObject
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)> delegate;
+
+- (instancetype)initWithDelegate:(id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoCodecConstants.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoCodecConstants.h
new file mode 100644
index 00000000..03f36e22
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoCodecConstants.h
@@ -0,0 +1,16 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+RTC_OBJC_EXPORT extern NSString* const kRTCVideoCodecVp8Name;
+RTC_OBJC_EXPORT extern NSString* const kRTCVideoCodecVp9Name;
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoCodecInfo.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoCodecInfo.h
new file mode 100644
index 00000000..fa28958f
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoCodecInfo.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Holds information to identify a codec. Corresponds to webrtc::SdpVideoFormat. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoCodecInfo) : NSObject <NSCoding>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+- (instancetype)initWithName:(NSString *)name;
+
+- (instancetype)initWithName:(NSString *)name
+                  parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters
+    NS_DESIGNATED_INITIALIZER;
+
+- (BOOL)isEqualToCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+
+@property(nonatomic, readonly) NSString *name;
+@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *parameters;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoder.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoder.h
new file mode 100644
index 00000000..0fc88c91
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoder.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCodecSpecificInfo.h"
+#import "RTCEncodedImage.h"
+#import "RTCMacros.h"
+#import "RTCVideoEncoderSettings.h"
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Callback block for decoder. */
+typedef void (^RTCVideoDecoderCallback)(RTC_OBJC_TYPE(RTCVideoFrame) * frame);
+
+/** Protocol for decoder implementations. */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoder)<NSObject>
+
+    - (void)setCallback : (RTCVideoDecoderCallback)callback;
+- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores;
+- (NSInteger)releaseDecoder;
+- (NSInteger)decode:(RTC_OBJC_TYPE(RTCEncodedImage) *)encodedImage
+        missingFrames:(BOOL)missingFrames
+    codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
+         renderTimeMs:(int64_t)renderTimeMs;
+- (NSString *)implementationName;
+
+@end
+
+NS_ASSUME_NONNULL_END
\ No newline at end of file
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderFactory.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderFactory.h
new file mode 100644
index 00000000..8d901385
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderFactory.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCodecInfo.h"
+#import "RTCVideoDecoder.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoderFactory)<NSObject>
+
+    - (nullable id<RTC_OBJC_TYPE(RTCVideoDecoder)>)createDecoder
+    : (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)
+    supportedCodecs;  // TODO(andersc): "supportedFormats" instead?
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderFactoryH264.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderFactoryH264.h
new file mode 100644
index 00000000..88bacbbd
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderFactoryH264.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoderFactory.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderFactoryH264) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoderFactory)>
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderH264.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderH264.h
new file mode 100644
index 00000000..a12e4212
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderH264.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderH264) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoder)>
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderVP8.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderVP8.h
new file mode 100644
index 00000000..a118b25e
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderVP8.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderVP8) : NSObject
+
+/* This returns a VP8 decoder that can be returned from a RTCVideoDecoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)vp8Decoder;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderVP9.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderVP9.h
new file mode 100644
index 00000000..b3a17430
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoDecoderVP9.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderVP9) : NSObject
+
+/* This returns a VP9 decoder that can be returned from a RTCVideoDecoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)vp9Decoder;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoder.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoder.h
new file mode 100644
index 00000000..7d1a7afd
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoder.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCodecSpecificInfo.h"
+#import "RTCEncodedImage.h"
+#import "RTCMacros.h"
+#import "RTCRtpFragmentationHeader.h"
+#import "RTCVideoEncoderQpThresholds.h"
+#import "RTCVideoEncoderSettings.h"
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Callback block for encoder. */
+typedef BOOL (^RTCVideoEncoderCallback)(RTC_OBJC_TYPE(RTCEncodedImage) * frame,
+                                        id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)> info,
+                                        RTC_OBJC_TYPE(RTCRtpFragmentationHeader) * header);
+
+/** Protocol for encoder implementations. */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoder)<NSObject>
+
+    - (void)setCallback : (RTCVideoEncoderCallback)callback;
+- (NSInteger)startEncodeWithSettings:(RTC_OBJC_TYPE(RTCVideoEncoderSettings) *)settings
+                       numberOfCores:(int)numberOfCores;
+- (NSInteger)releaseEncoder;
+- (NSInteger)encode:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame
+    codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
+           frameTypes:(NSArray<NSNumber *> *)frameTypes;
+- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate;
+- (NSString *)implementationName;
+
+/** Returns QP scaling settings for encoder. The quality scaler adjusts the resolution in order to
+ *  keep the QP from the encoded images within the given range. Returning nil from this function
+ *  disables quality scaling. */
+- (nullable RTC_OBJC_TYPE(RTCVideoEncoderQpThresholds) *)scalingSettings;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderFactory.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderFactory.h
new file mode 100644
index 00000000..733cb277
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderFactory.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCodecInfo.h"
+#import "RTCVideoEncoder.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCVideoEncoderFactory is an Objective-C version of
+ webrtc::VideoEncoderFactory::VideoEncoderSelector.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderSelector)<NSObject>
+
+    - (void)registerCurrentEncoderInfo : (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+- (nullable RTC_OBJC_TYPE(RTCVideoCodecInfo) *)encoderForBitrate:(NSInteger)bitrate;
+- (nullable RTC_OBJC_TYPE(RTCVideoCodecInfo) *)encoderForBrokenEncoder;
+
+@end
+
+/** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderFactory)<NSObject>
+
+    - (nullable id<RTC_OBJC_TYPE(RTCVideoEncoder)>)createEncoder
+    : (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)
+    supportedCodecs;  // TODO(andersc): "supportedFormats" instead?
+
+@optional
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)implementations;
+- (nullable id<RTC_OBJC_TYPE(RTCVideoEncoderSelector)>)encoderSelector;
+
+@optional
+- (NSArray<RTCVideoCodecInfo *> *)implementations;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderFactoryH264.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderFactoryH264.h
new file mode 100644
index 00000000..45fc4be2
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderFactoryH264.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoderFactory.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderFactoryH264) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoderFactory)>
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderH264.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderH264.h
new file mode 100644
index 00000000..9f4f4c7c
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderH264.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCodecInfo.h"
+#import "RTCVideoEncoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderH264) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoder)>
+
+- (instancetype)initWithCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)codecInfo;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderQpThresholds.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderQpThresholds.h
new file mode 100644
index 00000000..1a6e9e88
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderQpThresholds.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderQpThresholds) : NSObject
+
+- (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high;
+
+@property(nonatomic, readonly) NSInteger low;
+@property(nonatomic, readonly) NSInteger high;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderSettings.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderSettings.h
new file mode 100644
index 00000000..ae792eab
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderSettings.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
+  RTCVideoCodecModeRealtimeVideo,
+  RTCVideoCodecModeScreensharing,
+};
+
+/** Settings for encoder. Corresponds to webrtc::VideoCodec. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderSettings) : NSObject
+
+@property(nonatomic, strong) NSString *name;
+
+@property(nonatomic, assign) unsigned short width;
+@property(nonatomic, assign) unsigned short height;
+
+@property(nonatomic, assign) unsigned int startBitrate;  // kilobits/sec.
+@property(nonatomic, assign) unsigned int maxBitrate;
+@property(nonatomic, assign) unsigned int minBitrate;
+
+@property(nonatomic, assign) uint32_t maxFramerate;
+
+@property(nonatomic, assign) unsigned int qpMax;
+@property(nonatomic, assign) RTCVideoCodecMode mode;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderVP8.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderVP8.h
new file mode 100644
index 00000000..e136a5bd
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderVP8.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderVP8) : NSObject
+
+/* This returns a VP8 encoder that can be returned from a RTCVideoEncoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)vp8Encoder;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderVP9.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderVP9.h
new file mode 100644
index 00000000..8f961ef3
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoEncoderVP9.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderVP9) : NSObject
+
+/* This returns a VP9 encoder that can be returned from a RTCVideoEncoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)vp9Encoder;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoFrame.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoFrame.h
new file mode 100644
index 00000000..f5638d27
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoFrame.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef NS_ENUM(NSInteger, RTCVideoRotation) {
+  RTCVideoRotation_0 = 0,
+  RTCVideoRotation_90 = 90,
+  RTCVideoRotation_180 = 180,
+  RTCVideoRotation_270 = 270,
+};
+
+@protocol RTC_OBJC_TYPE
+(RTCVideoFrameBuffer);
+
+// RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoFrame) : NSObject
+
+/** Width without rotation applied. */
+@property(nonatomic, readonly) int width;
+
+/** Height without rotation applied. */
+@property(nonatomic, readonly) int height;
+@property(nonatomic, readonly) RTCVideoRotation rotation;
+
+/** Timestamp in nanoseconds. */
+@property(nonatomic, readonly) int64_t timeStampNs;
+
+/** Timestamp 90 kHz. */
+@property(nonatomic, assign) int32_t timeStamp;
+
+@property(nonatomic, readonly) id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> buffer;
+
+- (instancetype)init NS_UNAVAILABLE;
+- (instancetype) new NS_UNAVAILABLE;
+
+/** Initialize an RTCVideoFrame from a pixel buffer, rotation, and timestamp.
+ *  Deprecated - initialize with a RTCCVPixelBuffer instead
+ */
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                           rotation:(RTCVideoRotation)rotation
+                        timeStampNs:(int64_t)timeStampNs
+    DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
+
+/** Initialize an RTCVideoFrame from a pixel buffer combined with cropping and
+ *  scaling. Cropping will be applied first on the pixel buffer, followed by
+ *  scaling to the final resolution of scaledWidth x scaledHeight.
+ */
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                        scaledWidth:(int)scaledWidth
+                       scaledHeight:(int)scaledHeight
+                          cropWidth:(int)cropWidth
+                         cropHeight:(int)cropHeight
+                              cropX:(int)cropX
+                              cropY:(int)cropY
+                           rotation:(RTCVideoRotation)rotation
+                        timeStampNs:(int64_t)timeStampNs
+    DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
+
+/** Initialize an RTCVideoFrame from a frame buffer, rotation, and timestamp.
+ */
+- (instancetype)initWithBuffer:(id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>)frameBuffer
+                      rotation:(RTCVideoRotation)rotation
+                   timeStampNs:(int64_t)timeStampNs;
+
+/** Return a frame that is guaranteed to be I420, i.e. it is possible to access
+ *  the YUV data on it.
+ */
+- (RTC_OBJC_TYPE(RTCVideoFrame) *)newI420VideoFrame;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoFrameBuffer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoFrameBuffer.h
new file mode 100644
index 00000000..82d057ee
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoFrameBuffer.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@protocol RTC_OBJC_TYPE
+(RTCI420Buffer);
+
+// RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoFrameBuffer)<NSObject>
+
+    @property(nonatomic, readonly) int width;
+@property(nonatomic, readonly) int height;
+
+- (id<RTC_OBJC_TYPE(RTCI420Buffer)>)toI420;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoRenderer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoRenderer.h
new file mode 100644
index 00000000..0f763295
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoRenderer.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#if TARGET_OS_IPHONE
+#import <UIKit/UIKit.h>
+#endif
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCVideoFrame);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoRenderer)<NSObject>
+
+    /** The size of the frame. */
+    - (void)setSize : (CGSize)size;
+
+/** The frame to be displayed. */
+- (void)renderFrame:(nullable RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+
+@end
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoViewDelegate)
+
+    - (void)videoView : (id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoView didChangeVideoSize
+    : (CGSize)size;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoSource.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoSource.h
new file mode 100644
index 00000000..cdef8b89
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoSource.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMediaSource.h"
+#import "RTCVideoCapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+
+@interface RTC_OBJC_TYPE (RTCVideoSource) : RTC_OBJC_TYPE(RTCMediaSource) <RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Calling this function will cause frames to be scaled down to the
+ * requested resolution. Also, frames will be cropped to match the
+ * requested aspect ratio, and frames will be dropped to match the
+ * requested fps. The requested aspect ratio is orientation agnostic and
+ * will be adjusted to maintain the input orientation, so it doesn't
+ * matter if e.g. 1280x720 or 720x1280 is requested.
+ */
+- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoTrack.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoTrack.h
new file mode 100644
index 00000000..5382b716
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoTrack.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaStreamTrack.h"
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@protocol RTC_OBJC_TYPE
+(RTCVideoRenderer);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@class RTC_OBJC_TYPE(RTCVideoSource);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoTrack) : RTC_OBJC_TYPE(RTCMediaStreamTrack)
+
+/** The video source for this video track. */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCVideoSource) *source;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Register a renderer that will render all frames received on this track. */
+- (void)addRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer;
+
+/** Deregister a renderer. */
+- (void)removeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoViewShading.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoViewShading.h
new file mode 100644
index 00000000..9df30a8f
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCVideoViewShading.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * RTCVideoViewShading provides a way for apps to customize the OpenGL(ES shaders
+ * used in rendering for the RTCEAGLVideoView/RTCNSGLVideoView.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoViewShading)<NSObject>
+
+    /** Callback for I420 frames. Each plane is given as a texture. */
+    - (void)applyShadingForFrameWithWidth : (int)width height : (int)height rotation
+    : (RTCVideoRotation)rotation yPlane : (GLuint)yPlane uPlane : (GLuint)uPlane vPlane
+    : (GLuint)vPlane;
+
+/** Callback for NV12 frames. Each plane is given as a texture. */
+- (void)applyShadingForFrameWithWidth:(int)width
+                               height:(int)height
+                             rotation:(RTCVideoRotation)rotation
+                               yPlane:(GLuint)yPlane
+                              uvPlane:(GLuint)uvPlane;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCYUVPlanarBuffer.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCYUVPlanarBuffer.h
new file mode 100644
index 00000000..be01b915
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/RTCYUVPlanarBuffer.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrameBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Protocol for RTCVideoFrameBuffers containing YUV planar data. */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCYUVPlanarBuffer)<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>
+
+    @property(nonatomic, readonly) int chromaWidth;
+@property(nonatomic, readonly) int chromaHeight;
+@property(nonatomic, readonly) const uint8_t *dataY;
+@property(nonatomic, readonly) const uint8_t *dataU;
+@property(nonatomic, readonly) const uint8_t *dataV;
+@property(nonatomic, readonly) int strideY;
+@property(nonatomic, readonly) int strideU;
+@property(nonatomic, readonly) int strideV;
+
+- (instancetype)initWithWidth:(int)width
+                       height:(int)height
+                        dataY:(const uint8_t *)dataY
+                        dataU:(const uint8_t *)dataU
+                        dataV:(const uint8_t *)dataV;
+- (instancetype)initWithWidth:(int)width height:(int)height;
+- (instancetype)initWithWidth:(int)width
+                       height:(int)height
+                      strideY:(int)strideY
+                      strideU:(int)strideU
+                      strideV:(int)strideV;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/UIDevice+RTCDevice.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/UIDevice+RTCDevice.h
new file mode 100644
index 00000000..a7f655ba
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/UIDevice+RTCDevice.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+typedef NS_ENUM(NSInteger, RTCDeviceType) {
+  RTCDeviceTypeUnknown,
+  RTCDeviceTypeIPhone1G,
+  RTCDeviceTypeIPhone3G,
+  RTCDeviceTypeIPhone3GS,
+  RTCDeviceTypeIPhone4,
+  RTCDeviceTypeIPhone4Verizon,
+  RTCDeviceTypeIPhone4S,
+  RTCDeviceTypeIPhone5GSM,
+  RTCDeviceTypeIPhone5GSM_CDMA,
+  RTCDeviceTypeIPhone5CGSM,
+  RTCDeviceTypeIPhone5CGSM_CDMA,
+  RTCDeviceTypeIPhone5SGSM,
+  RTCDeviceTypeIPhone5SGSM_CDMA,
+  RTCDeviceTypeIPhone6Plus,
+  RTCDeviceTypeIPhone6,
+  RTCDeviceTypeIPhone6S,
+  RTCDeviceTypeIPhone6SPlus,
+  RTCDeviceTypeIPhone7,
+  RTCDeviceTypeIPhone7Plus,
+  RTCDeviceTypeIPhoneSE,
+  RTCDeviceTypeIPhone8,
+  RTCDeviceTypeIPhone8Plus,
+  RTCDeviceTypeIPhoneX,
+  RTCDeviceTypeIPhoneXS,
+  RTCDeviceTypeIPhoneXSMax,
+  RTCDeviceTypeIPhoneXR,
+  RTCDeviceTypeIPhone11,
+  RTCDeviceTypeIPhone11Pro,
+  RTCDeviceTypeIPhone11ProMax,
+  RTCDeviceTypeIPodTouch1G,
+  RTCDeviceTypeIPodTouch2G,
+  RTCDeviceTypeIPodTouch3G,
+  RTCDeviceTypeIPodTouch4G,
+  RTCDeviceTypeIPodTouch5G,
+  RTCDeviceTypeIPodTouch6G,
+  RTCDeviceTypeIPodTouch7G,
+  RTCDeviceTypeIPad,
+  RTCDeviceTypeIPad2Wifi,
+  RTCDeviceTypeIPad2GSM,
+  RTCDeviceTypeIPad2CDMA,
+  RTCDeviceTypeIPad2Wifi2,
+  RTCDeviceTypeIPadMiniWifi,
+  RTCDeviceTypeIPadMiniGSM,
+  RTCDeviceTypeIPadMiniGSM_CDMA,
+  RTCDeviceTypeIPad3Wifi,
+  RTCDeviceTypeIPad3GSM_CDMA,
+  RTCDeviceTypeIPad3GSM,
+  RTCDeviceTypeIPad4Wifi,
+  RTCDeviceTypeIPad4GSM,
+  RTCDeviceTypeIPad4GSM_CDMA,
+  RTCDeviceTypeIPad5,
+  RTCDeviceTypeIPad6,
+  RTCDeviceTypeIPadAirWifi,
+  RTCDeviceTypeIPadAirCellular,
+  RTCDeviceTypeIPadAirWifiCellular,
+  RTCDeviceTypeIPadAir2,
+  RTCDeviceTypeIPadMini2GWifi,
+  RTCDeviceTypeIPadMini2GCellular,
+  RTCDeviceTypeIPadMini2GWifiCellular,
+  RTCDeviceTypeIPadMini3,
+  RTCDeviceTypeIPadMini4,
+  RTCDeviceTypeIPadPro9Inch,
+  RTCDeviceTypeIPadPro12Inch,
+  RTCDeviceTypeIPadPro12Inch2,
+  RTCDeviceTypeIPadPro10Inch,
+  RTCDeviceTypeIPad7Gen10Inch,
+  RTCDeviceTypeIPadPro3Gen11Inch,
+  RTCDeviceTypeIPadPro3Gen12Inch,
+  RTCDeviceTypeIPadMini5Gen,
+  RTCDeviceTypeIPadAir3Gen,
+  RTCDeviceTypeSimulatori386,
+  RTCDeviceTypeSimulatorx86_64,
+};
+
+@interface UIDevice (RTCDevice)
+
++ (RTCDeviceType)deviceType;
++ (BOOL)isIOS11OrLater;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/WebRTC.h b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/WebRTC.h
new file mode 100644
index 00000000..00aac6d9
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Headers/WebRTC.h
@@ -0,0 +1,94 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <WebRTC/RTCCodecSpecificInfo.h>
+#import <WebRTC/RTCEncodedImage.h>
+#import <WebRTC/RTCI420Buffer.h>
+#import <WebRTC/RTCLogging.h>
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCMutableI420Buffer.h>
+#import <WebRTC/RTCMutableYUVPlanarBuffer.h>
+#import <WebRTC/RTCRtpFragmentationHeader.h>
+#import <WebRTC/RTCVideoCapturer.h>
+#import <WebRTC/RTCVideoCodecInfo.h>
+#import <WebRTC/RTCVideoDecoder.h>
+#import <WebRTC/RTCVideoDecoderFactory.h>
+#import <WebRTC/RTCVideoEncoder.h>
+#import <WebRTC/RTCVideoEncoderFactory.h>
+#import <WebRTC/RTCVideoEncoderQpThresholds.h>
+#import <WebRTC/RTCVideoEncoderSettings.h>
+#import <WebRTC/RTCVideoFrame.h>
+#import <WebRTC/RTCVideoFrameBuffer.h>
+#import <WebRTC/RTCVideoRenderer.h>
+#import <WebRTC/RTCYUVPlanarBuffer.h>
+#import <WebRTC/RTCAudioSession.h>
+#import <WebRTC/RTCAudioSessionConfiguration.h>
+#import <WebRTC/RTCCameraVideoCapturer.h>
+#import <WebRTC/RTCFileVideoCapturer.h>
+#import <WebRTC/RTCNetworkMonitor.h>
+#import <WebRTC/RTCMTLVideoView.h>
+#import <WebRTC/RTCEAGLVideoView.h>
+#import <WebRTC/RTCVideoViewShading.h>
+#import <WebRTC/RTCCodecSpecificInfoH264.h>
+#import <WebRTC/RTCDefaultVideoDecoderFactory.h>
+#import <WebRTC/RTCDefaultVideoEncoderFactory.h>
+#import <WebRTC/RTCH264ProfileLevelId.h>
+#import <WebRTC/RTCVideoDecoderFactoryH264.h>
+#import <WebRTC/RTCVideoDecoderH264.h>
+#import <WebRTC/RTCVideoEncoderFactoryH264.h>
+#import <WebRTC/RTCVideoEncoderH264.h>
+#import <WebRTC/RTCCVPixelBuffer.h>
+#import <WebRTC/RTCCameraPreviewView.h>
+#import <WebRTC/RTCDispatcher.h>
+#import <WebRTC/UIDevice+RTCDevice.h>
+#import <WebRTC/RTCAudioSource.h>
+#import <WebRTC/RTCAudioTrack.h>
+#import <WebRTC/RTCConfiguration.h>
+#import <WebRTC/RTCDataChannel.h>
+#import <WebRTC/RTCDataChannelConfiguration.h>
+#import <WebRTC/RTCFieldTrials.h>
+#import <WebRTC/RTCIceCandidate.h>
+#import <WebRTC/RTCIceServer.h>
+#import <WebRTC/RTCLegacyStatsReport.h>
+#import <WebRTC/RTCMediaConstraints.h>
+#import <WebRTC/RTCMediaSource.h>
+#import <WebRTC/RTCMediaStream.h>
+#import <WebRTC/RTCMediaStreamTrack.h>
+#import <WebRTC/RTCMetrics.h>
+#import <WebRTC/RTCMetricsSampleInfo.h>
+#import <WebRTC/RTCPeerConnection.h>
+#import <WebRTC/RTCPeerConnectionFactory.h>
+#import <WebRTC/RTCPeerConnectionFactoryOptions.h>
+#import <WebRTC/RTCRtcpParameters.h>
+#import <WebRTC/RTCRtpCodecParameters.h>
+#import <WebRTC/RTCRtpEncodingParameters.h>
+#import <WebRTC/RTCRtpHeaderExtension.h>
+#import <WebRTC/RTCRtpParameters.h>
+#import <WebRTC/RTCRtpReceiver.h>
+#import <WebRTC/RTCRtpSender.h>
+#import <WebRTC/RTCRtpTransceiver.h>
+#import <WebRTC/RTCDtmfSender.h>
+#import <WebRTC/RTCSSLAdapter.h>
+#import <WebRTC/RTCSessionDescription.h>
+#import <WebRTC/RTCStatisticsReport.h>
+#import <WebRTC/RTCTracing.h>
+#import <WebRTC/RTCCertificate.h>
+#import <WebRTC/RTCCryptoOptions.h>
+#import <WebRTC/RTCVideoSource.h>
+#import <WebRTC/RTCVideoTrack.h>
+#import <WebRTC/RTCVideoCodecConstants.h>
+#import <WebRTC/RTCVideoDecoderVP8.h>
+#import <WebRTC/RTCVideoDecoderVP9.h>
+#import <WebRTC/RTCVideoEncoderVP8.h>
+#import <WebRTC/RTCVideoEncoderVP9.h>
+#import <WebRTC/RTCNativeI420Buffer.h>
+#import <WebRTC/RTCNativeMutableI420Buffer.h>
+#import <WebRTC/RTCCallbackLogger.h>
+#import <WebRTC/RTCFileLogger.h>
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Info.plist b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Info.plist
new file mode 100644
index 00000000..a85d8008
Binary files /dev/null and b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Info.plist differ
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Modules/module.modulemap b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Modules/module.modulemap
new file mode 100644
index 00000000..cd485a4e
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/Modules/module.modulemap
@@ -0,0 +1,6 @@
+framework module WebRTC {
+  umbrella header "WebRTC.h"
+
+  export *
+  module * { export * }
+}
diff --git a/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/WebRTC b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/WebRTC
new file mode 100644
index 00000000..b26dea61
Binary files /dev/null and b/lib/WebRTC.xcframework/ios-arm64_armv7/WebRTC.framework/WebRTC differ
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioSession.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioSession.h
new file mode 100644
index 00000000..f917e327
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioSession.h
@@ -0,0 +1,264 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const kRTCAudioSessionErrorDomain;
+/** Method that requires lock was called without lock. */
+extern NSInteger const kRTCAudioSessionErrorLockRequired;
+/** Unknown configuration error occurred. */
+extern NSInteger const kRTCAudioSessionErrorConfiguration;
+
+@class RTC_OBJC_TYPE(RTCAudioSession);
+@class RTC_OBJC_TYPE(RTCAudioSessionConfiguration);
+
+// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
+// from AVAudioSession and handle them before calling these delegate methods,
+// at which point applications can perform additional processing if required.
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCAudioSessionDelegate)<NSObject>
+
+    @optional
+/** Called on a system notification thread when AVAudioSession starts an
+ *  interruption event.
+ */
+- (void)audioSessionDidBeginInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a system notification thread when AVAudioSession ends an
+ *  interruption event.
+ */
+- (void)audioSessionDidEndInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                   shouldResumeSession:(BOOL)shouldResumeSession;
+
+/** Called on a system notification thread when AVAudioSession changes the
+ *  route.
+ */
+- (void)audioSessionDidChangeRoute:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                            reason:(AVAudioSessionRouteChangeReason)reason
+                     previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
+
+/** Called on a system notification thread when AVAudioSession media server
+ *  terminates.
+ */
+- (void)audioSessionMediaServerTerminated:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a system notification thread when AVAudioSession media server
+ *  restarts.
+ */
+- (void)audioSessionMediaServerReset:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+    didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+
+/** Called on a WebRTC thread when the audio device is notified to begin
+ *  playback or recording.
+ */
+- (void)audioSessionDidStartPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a WebRTC thread when the audio device is notified to stop
+ *  playback or recording.
+ */
+- (void)audioSessionDidStopPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called when the AVAudioSession output volume value changes. */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didChangeOutputVolume:(float)outputVolume;
+
+/** Called when the audio device detects a playout glitch. The argument is the
+ *  number of glitches detected so far in the current audio playout session.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
+
+/** Called when the audio session is about to change the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession willSetActive:(BOOL)active;
+
+/** Called after the audio session sucessfully changed the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession didSetActive:(BOOL)active;
+
+/** Called after the audio session failed to change the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    failedToSetActive:(BOOL)active
+                error:(NSError *)error;
+
+@end
+
+/** This is a protocol used to inform RTCAudioSession when the audio session
+ *  activation state has changed outside of RTCAudioSession. The current known use
+ *  case of this is when CallKit activates the audio session for the application
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCAudioSessionActivationDelegate)<NSObject>
+
+    /** Called when the audio session is activated outside of the app by iOS. */
+    - (void)audioSessionDidActivate : (AVAudioSession *)session;
+
+/** Called when the audio session is deactivated outside of the app by iOS. */
+- (void)audioSessionDidDeactivate:(AVAudioSession *)session;
+
+@end
+
+/** Proxy class for AVAudioSession that adds a locking mechanism similar to
+ *  AVCaptureDevice. This is used to that interleaving configurations between
+ *  WebRTC and the application layer are avoided.
+ *
+ *  RTCAudioSession also coordinates activation so that the audio session is
+ *  activated only once. See |setActive:error:|.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSession) : NSObject <RTC_OBJC_TYPE(RTCAudioSessionActivationDelegate)>
+
+/** Convenience property to access the AVAudioSession singleton. Callers should
+ *  not call setters on AVAudioSession directly, but other method invocations
+ *  are fine.
+ */
+@property(nonatomic, readonly) AVAudioSession *session;
+
+/** Our best guess at whether the session is active based on results of calls to
+ *  AVAudioSession.
+ */
+@property(nonatomic, readonly) BOOL isActive;
+/** Whether RTCAudioSession is currently locked for configuration. */
+@property(nonatomic, readonly) BOOL isLocked;
+
+/** If YES, WebRTC will not initialize the audio unit automatically when an
+ *  audio track is ready for playout or recording. Instead, applications should
+ *  call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
+ *  as soon as an audio track is ready for playout or recording.
+ */
+@property(nonatomic, assign) BOOL useManualAudio;
+
+/** This property is only effective if useManualAudio is YES.
+ *  Represents permission for WebRTC to initialize the VoIP audio unit.
+ *  When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
+ *  stopped and uninitialized. This will stop incoming and outgoing audio.
+ *  When set to YES, WebRTC will initialize and start the audio unit when it is
+ *  needed (e.g. due to establishing an audio connection).
+ *  This property was introduced to work around an issue where if an AVPlayer is
+ *  playing audio while the VoIP audio unit is initialized, its audio would be
+ *  either cut off completely or played at a reduced volume. By preventing
+ *  the audio unit from being initialized until after the audio has completed,
+ *  we are able to prevent the abrupt cutoff.
+ */
+@property(nonatomic, assign) BOOL isAudioEnabled;
+
+// Proxy properties.
+@property(readonly) NSString *category;
+@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
+@property(readonly) NSString *mode;
+@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
+@property(readonly) AVAudioSessionRouteDescription *currentRoute;
+@property(readonly) NSInteger maximumInputNumberOfChannels;
+@property(readonly) NSInteger maximumOutputNumberOfChannels;
+@property(readonly) float inputGain;
+@property(readonly) BOOL inputGainSettable;
+@property(readonly) BOOL inputAvailable;
+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *inputDataSources;
+@property(readonly, nullable) AVAudioSessionDataSourceDescription *inputDataSource;
+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *outputDataSources;
+@property(readonly, nullable) AVAudioSessionDataSourceDescription *outputDataSource;
+@property(readonly) double sampleRate;
+@property(readonly) double preferredSampleRate;
+@property(readonly) NSInteger inputNumberOfChannels;
+@property(readonly) NSInteger outputNumberOfChannels;
+@property(readonly) float outputVolume;
+@property(readonly) NSTimeInterval inputLatency;
+@property(readonly) NSTimeInterval outputLatency;
+@property(readonly) NSTimeInterval IOBufferDuration;
+@property(readonly) NSTimeInterval preferredIOBufferDuration;
+
+/**
+ When YES, calls to -setConfiguration:error: and -setConfiguration:active:error: ignore errors in
+ configuring the audio session's "preferred" attributes (e.g. preferredInputNumberOfChannels).
+ Typically, configurations to preferred attributes are optimizations, and ignoring this type of
+ configuration error allows code flow to continue along the happy path when these optimization are
+ not available. The default value of this property is NO.
+ */
+@property(nonatomic) BOOL ignoresPreferredAttributeConfigurationErrors;
+
+/** Default constructor. */
++ (instancetype)sharedInstance;
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Adds a delegate, which is held weakly. */
+- (void)addDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+/** Removes an added delegate. */
+- (void)removeDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+
+/** Request exclusive access to the audio session for configuration. This call
+ *  will block if the lock is held by another object.
+ */
+- (void)lockForConfiguration;
+/** Relinquishes exclusive access to the audio session. */
+- (void)unlockForConfiguration;
+
+/** If |active|, activates the audio session if it isn't already active.
+ *  Successful calls must be balanced with a setActive:NO when activation is no
+ *  longer required. If not |active|, deactivates the audio session if one is
+ *  active and this is the last balanced call. When deactivating, the
+ *  AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
+ *  AVAudioSession.
+ */
+- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
+
+// The following methods are proxies for the associated methods on
+// AVAudioSession. |lockForConfiguration| must be called before using them
+// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
+
+- (BOOL)setCategory:(NSString *)category
+        withOptions:(AVAudioSessionCategoryOptions)options
+              error:(NSError **)outError;
+- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
+- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
+- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
+- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration error:(NSError **)outError;
+- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
+- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
+- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride error:(NSError **)outError;
+- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort error:(NSError **)outError;
+- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                     error:(NSError **)outError;
+- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                      error:(NSError **)outError;
+@end
+
+@interface RTC_OBJC_TYPE (RTCAudioSession)
+(Configuration)
+
+    /** Applies the configuration to the current session. Attempts to set all
+     *  properties even if previous ones fail. Only the last error will be
+     *  returned.
+     *  |lockForConfiguration| must be called first.
+     */
+    - (BOOL)setConfiguration : (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration error
+    : (NSError **)outError;
+
+/** Convenience method that calls both setConfiguration and setActive.
+ *  |lockForConfiguration| must be called first.
+ */
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioSessionConfiguration.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioSessionConfiguration.h
new file mode 100644
index 00000000..4582b805
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioSessionConfiguration.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN const int kRTCAudioSessionPreferredNumberOfChannels;
+RTC_EXTERN const double kRTCAudioSessionHighPerformanceSampleRate;
+RTC_EXTERN const double kRTCAudioSessionLowComplexitySampleRate;
+RTC_EXTERN const double kRTCAudioSessionHighPerformanceIOBufferDuration;
+RTC_EXTERN const double kRTCAudioSessionLowComplexityIOBufferDuration;
+
+// Struct to hold configuration values.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSessionConfiguration) : NSObject
+
+@property(nonatomic, strong) NSString *category;
+@property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
+@property(nonatomic, strong) NSString *mode;
+@property(nonatomic, assign) double sampleRate;
+@property(nonatomic, assign) NSTimeInterval ioBufferDuration;
+@property(nonatomic, assign) NSInteger inputNumberOfChannels;
+@property(nonatomic, assign) NSInteger outputNumberOfChannels;
+
+/** Initializes configuration to defaults. */
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+/** Returns the current configuration of the audio session. */
++ (instancetype)currentConfiguration;
+/** Returns the configuration that WebRTC needs. */
++ (instancetype)webRTCConfiguration;
+/** Provide a way to override the default configuration. */
++ (void)setWebRTCConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioSource.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioSource.h
new file mode 100644
index 00000000..9f78dcd9
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioSource.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMediaSource.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSource) : RTC_OBJC_TYPE(RTCMediaSource)
+
+- (instancetype)init NS_UNAVAILABLE;
+
+// Sets the volume for the RTCMediaSource. |volume| is a gain value in the range
+// [0, 10].
+// Temporary fix to be able to modify volume of remote audio tracks.
+// TODO(kthelgason): Property stays here temporarily until a proper volume-api
+// is available on the surface exposed by webrtc.
+@property(nonatomic, assign) double volume;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioTrack.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioTrack.h
new file mode 100644
index 00000000..95eb5d3d
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCAudioTrack.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMacros.h"
+#import "RTCMediaStreamTrack.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioSource);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioTrack) : RTC_OBJC_TYPE(RTCMediaStreamTrack)
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** The audio source for this audio track. */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCAudioSource) * source;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCVPixelBuffer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCVPixelBuffer.h
new file mode 100644
index 00000000..17eebd0a
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCVPixelBuffer.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrameBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCVideoFrameBuffer containing a CVPixelBufferRef */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCVPixelBuffer) : NSObject <RTC_OBJC_TYPE(RTCVideoFrameBuffer)>
+
+@property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
+@property(nonatomic, readonly) int cropX;
+@property(nonatomic, readonly) int cropY;
+@property(nonatomic, readonly) int cropWidth;
+@property(nonatomic, readonly) int cropHeight;
+
++ (NSSet<NSNumber *> *)supportedPixelFormats;
+
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer;
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                       adaptedWidth:(int)adaptedWidth
+                      adaptedHeight:(int)adaptedHeight
+                          cropWidth:(int)cropWidth
+                         cropHeight:(int)cropHeight
+                              cropX:(int)cropX
+                              cropY:(int)cropY;
+
+- (BOOL)requiresCropping;
+- (BOOL)requiresScalingToWidth:(int)width height:(int)height;
+- (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
+
+/** The minimum size of the |tmpBuffer| must be the number of bytes returned from the
+ * bufferSizeForCroppingAndScalingToWidth:height: method.
+ * If that size is 0, the |tmpBuffer| may be nil.
+ */
+- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
+        withTempBuffer:(nullable uint8_t *)tmpBuffer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCallbackLogger.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCallbackLogger.h
new file mode 100644
index 00000000..c1aeb825
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCallbackLogger.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCLogging.h"
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef void (^RTCCallbackLoggerMessageHandler)(NSString *message);
+typedef void (^RTCCallbackLoggerMessageAndSeverityHandler)(NSString *message,
+                                                           RTCLoggingSeverity severity);
+
+// This class intercepts WebRTC logs and forwards them to a registered block.
+// This class is not threadsafe.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCallbackLogger) : NSObject
+
+// The severity level to capture. The default is kRTCLoggingSeverityInfo.
+@property(nonatomic, assign) RTCLoggingSeverity severity;
+
+// The callback handler will be called on the same thread that does the
+// logging, so if the logging callback can be slow it may be a good idea
+// to implement dispatching to some other queue.
+- (void)start:(nullable RTCCallbackLoggerMessageHandler)handler;
+- (void)startWithMessageAndSeverityHandler:
+        (nullable RTCCallbackLoggerMessageAndSeverityHandler)handler;
+
+- (void)stop;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCameraPreviewView.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCameraPreviewView.h
new file mode 100644
index 00000000..db9b15a4
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCameraPreviewView.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCMacros.h"
+
+@class AVCaptureSession;
+
+/** RTCCameraPreviewView is a view that renders local video from an
+ *  AVCaptureSession.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCameraPreviewView) : UIView
+
+/** The capture session being rendered in the view. Capture session
+ *  is assigned to AVCaptureVideoPreviewLayer async in the same
+ *  queue that the AVCaptureSession is started/stopped.
+ */
+@property(nonatomic, strong) AVCaptureSession* captureSession;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCameraVideoCapturer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCameraVideoCapturer.h
new file mode 100644
index 00000000..fed5a378
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCameraVideoCapturer.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+// Camera capture that implements RTCVideoCapturer. Delivers frames to a
+// RTCVideoCapturerDelegate (usually RTCVideoSource).
+NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
+@interface RTC_OBJC_TYPE (RTCCameraVideoCapturer) : RTC_OBJC_TYPE(RTCVideoCapturer)
+
+// Capture session that is used for capturing. Valid from initialization to dealloc.
+@property(readonly, nonatomic) AVCaptureSession *captureSession;
+
+// Returns list of available capture devices that support video capture.
++ (NSArray<AVCaptureDevice *> *)captureDevices;
+// Returns list of formats that are supported by this class for this device.
++ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device;
+
+// Returns the most efficient supported output pixel format for this capturer.
+- (FourCharCode)preferredOutputPixelFormat;
+
+// Starts the capture session asynchronously and notifies callback on completion.
+// The device will capture video in the format given in the `format` parameter. If the pixel format
+// in `format` is supported by the WebRTC pipeline, the same pixel format will be used for the
+// output. Otherwise, the format returned by `preferredOutputPixelFormat` will be used.
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps
+             completionHandler:(nullable void (^)(NSError *))completionHandler;
+// Stops the capture session asynchronously and notifies callback on completion.
+- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler;
+
+// Starts the capture session asynchronously.
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps;
+// Stops the capture session asynchronously.
+- (void)stopCapture;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCertificate.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCertificate.h
new file mode 100644
index 00000000..5ac8984d
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCertificate.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCertificate) : NSObject <NSCopying>
+
+/** Private key in PEM. */
+@property(nonatomic, readonly, copy) NSString *private_key;
+
+/** Public key in an x509 cert encoded in PEM. */
+@property(nonatomic, readonly, copy) NSString *certificate;
+
+/**
+ * Initialize an RTCCertificate with PEM strings for private_key and certificate.
+ */
+- (instancetype)initWithPrivateKey:(NSString *)private_key
+                       certificate:(NSString *)certificate NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Generate a new certificate for 're' use.
+ *
+ *  Optional dictionary of parameters. Defaults to KeyType ECDSA if none are
+ *  provided.
+ *  - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
+ */
++ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:(NSDictionary *)params;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCodecSpecificInfo.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCodecSpecificInfo.h
new file mode 100644
index 00000000..5e7800e5
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCodecSpecificInfo.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Implement this protocol to pass codec specific info from the encoder.
+ *  Corresponds to webrtc::CodecSpecificInfo.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCCodecSpecificInfo)<NSObject> @end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCodecSpecificInfoH264.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCodecSpecificInfoH264.h
new file mode 100644
index 00000000..ae3003a1
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCodecSpecificInfoH264.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCodecSpecificInfo.h"
+#import "RTCMacros.h"
+
+/** Class for H264 specific config. */
+typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
+  RTCH264PacketizationModeNonInterleaved = 0,  // Mode 1 - STAP-A, FU-A is allowed
+  RTCH264PacketizationModeSingleNalUnit        // Mode 0 - only single NALU allowed
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCodecSpecificInfoH264) : NSObject <RTC_OBJC_TYPE(RTCCodecSpecificInfo)>
+
+@property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCConfiguration.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCConfiguration.h
new file mode 100644
index 00000000..86eaa6ce
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCConfiguration.h
@@ -0,0 +1,221 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCertificate.h"
+#import "RTCCryptoOptions.h"
+#import "RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCIceServer);
+
+/**
+ * Represents the ice transport policy. This exposes the same states in C++,
+ * which include one more state than what exists in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
+  RTCIceTransportPolicyNone,
+  RTCIceTransportPolicyRelay,
+  RTCIceTransportPolicyNoHost,
+  RTCIceTransportPolicyAll
+};
+
+/** Represents the bundle policy. */
+typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
+  RTCBundlePolicyBalanced,
+  RTCBundlePolicyMaxCompat,
+  RTCBundlePolicyMaxBundle
+};
+
+/** Represents the rtcp mux policy. */
+typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
+
+/** Represents the tcp candidate policy. */
+typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
+  RTCTcpCandidatePolicyEnabled,
+  RTCTcpCandidatePolicyDisabled
+};
+
+/** Represents the candidate network policy. */
+typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) {
+  RTCCandidateNetworkPolicyAll,
+  RTCCandidateNetworkPolicyLowCost
+};
+
+/** Represents the continual gathering policy. */
+typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
+  RTCContinualGatheringPolicyGatherOnce,
+  RTCContinualGatheringPolicyGatherContinually
+};
+
+/** Represents the encryption key type. */
+typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
+  RTCEncryptionKeyTypeRSA,
+  RTCEncryptionKeyTypeECDSA,
+};
+
+/** Represents the chosen SDP semantics for the RTCPeerConnection. */
+typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
+  RTCSdpSemanticsPlanB,
+  RTCSdpSemanticsUnifiedPlan,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCConfiguration) : NSObject
+
+/** If true, allows DSCP codes to be set on outgoing packets, configured using
+ *  networkPriority field of RTCRtpEncodingParameters. Defaults to false.
+ */
+@property(nonatomic, assign) BOOL enableDscp;
+
+/** An array of Ice Servers available to be used by ICE. */
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCIceServer) *> *iceServers;
+
+/** An RTCCertificate for 're' use. */
+@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCertificate) * certificate;
+
+/** Which candidates the ICE agent is allowed to use. The W3C calls it
+ * |iceTransportPolicy|, while in C++ it is called |type|. */
+@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
+
+/** The media-bundling policy to use when gathering ICE candidates. */
+@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
+
+/** The rtcp-mux policy to use when gathering ICE candidates. */
+@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
+@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
+@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
+@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
+
+/** If set to YES, don't gather IPv6 ICE candidates.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL disableIPV6;
+
+/** If set to YES, don't gather IPv6 ICE candidates on Wi-Fi.
+ *  Only intended to be used on specific devices. Certain phones disable IPv6
+ *  when the screen is turned off and it would be better to just disable the
+ *  IPv6 ICE candidates on Wi-Fi in those cases.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL disableIPV6OnWiFi;
+
+/** By default, the PeerConnection will use a limited number of IPv6 network
+ *  interfaces, in order to avoid too many ICE candidate pairs being created
+ *  and delaying ICE completion.
+ *
+ *  Can be set to INT_MAX to effectively disable the limit.
+ */
+@property(nonatomic, assign) int maxIPv6Networks;
+
+/** Exclude link-local network interfaces
+ *  from considertaion for gathering ICE candidates.
+ *  Defaults to NO.
+ */
+@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
+
+@property(nonatomic, assign) int audioJitterBufferMaxPackets;
+@property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
+@property(nonatomic, assign) int iceConnectionReceivingTimeout;
+@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
+
+/** Key type used to generate SSL identity. Default is ECDSA. */
+@property(nonatomic, assign) RTCEncryptionKeyType keyType;
+
+/** ICE candidate pool size as defined in JSEP. Default is 0. */
+@property(nonatomic, assign) int iceCandidatePoolSize;
+
+/** Prune turn ports on the same network to the same turn server.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL shouldPruneTurnPorts;
+
+/** If set to YES, this means the ICE transport should presume TURN-to-TURN
+ *  candidate pairs will succeed, even before a binding response is received.
+ */
+@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
+
+/* This flag is only effective when |continualGatheringPolicy| is
+ * RTCContinualGatheringPolicyGatherContinually.
+ *
+ * If YES, after the ICE transport type is changed such that new types of
+ * ICE candidates are allowed by the new transport type, e.g. from
+ * RTCIceTransportPolicyRelay to RTCIceTransportPolicyAll, candidates that
+ * have been gathered by the ICE transport but not matching the previous
+ * transport type and as a result not observed by PeerConnectionDelegateAdapter,
+ * will be surfaced to the delegate.
+ */
+@property(nonatomic, assign) BOOL shouldSurfaceIceCandidatesOnIceTransportTypeChanged;
+
+/** If set to non-nil, controls the minimal interval between consecutive ICE
+ *  check packets.
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
+
+/** Configure the SDP semantics used by this PeerConnection. Note that the
+ *  WebRTC 1.0 specification requires UnifiedPlan semantics. The
+ *  RTCRtpTransceiver API is only available with UnifiedPlan semantics.
+ *
+ *  PlanB will cause RTCPeerConnection to create offers and answers with at
+ *  most one audio and one video m= section with multiple RTCRtpSenders and
+ *  RTCRtpReceivers specified as multiple a=ssrc lines within the section. This
+ *  will also cause RTCPeerConnection to ignore all but the first m= section of
+ *  the same media type.
+ *
+ *  UnifiedPlan will cause RTCPeerConnection to create offers and answers with
+ *  multiple m= sections where each m= section maps to one RTCRtpSender and one
+ *  RTCRtpReceiver (an RTCRtpTransceiver), either both audio or both
+ *  video. This will also cause RTCPeerConnection) to ignore all but the first a=ssrc
+ *  lines that form a Plan B stream.
+ *
+ *  For users who wish to send multiple audio/video streams and need to stay
+ *  interoperable with legacy WebRTC implementations or use legacy APIs,
+ *  specify PlanB.
+ *
+ *  For all other users, specify UnifiedPlan.
+ */
+@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
+
+/** Actively reset the SRTP parameters when the DTLS transports underneath are
+ *  changed after offer/answer negotiation. This is only intended to be a
+ *  workaround for crbug.com/835958
+ */
+@property(nonatomic, assign) BOOL activeResetSrtpParams;
+
+/** If the remote side support mid-stream codec switches then allow encoder
+ *  switching to be performed.
+ */
+
+@property(nonatomic, assign) BOOL allowCodecSwitching;
+
+/**
+ * Defines advanced optional cryptographic settings related to SRTP and
+ * frame encryption for native WebRTC. Setting this will overwrite any
+ * options set through the PeerConnectionFactory (which is deprecated).
+ */
+@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCryptoOptions) * cryptoOptions;
+
+/**
+ * Time interval between audio RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
+
+/**
+ * Time interval between video RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCryptoOptions.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCryptoOptions.h
new file mode 100644
index 00000000..759a45e0
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCCryptoOptions.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * Objective-C bindings for webrtc::CryptoOptions. This API had to be flattened
+ * as Objective-C doesn't support nested structures.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCryptoOptions) : NSObject
+
+/**
+ * Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
+ * if both sides enable it
+ */
+@property(nonatomic, assign) BOOL srtpEnableGcmCryptoSuites;
+/**
+ * If set to true, the (potentially insecure) crypto cipher
+ * SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
+ * during negotiation. It will only be used if both peers support it and no
+ * other ciphers get preferred.
+ */
+@property(nonatomic, assign) BOOL srtpEnableAes128Sha1_32CryptoCipher;
+/**
+ * If set to true, encrypted RTP header extensions as defined in RFC 6904
+ * will be negotiated. They will only be used if both peers support them.
+ */
+@property(nonatomic, assign) BOOL srtpEnableEncryptedRtpHeaderExtensions;
+
+/**
+ * If set all RtpSenders must have an FrameEncryptor attached to them before
+ * they are allowed to send packets. All RtpReceivers must have a
+ * FrameDecryptor attached to them before they are able to receive packets.
+ */
+@property(nonatomic, assign) BOOL sframeRequireFrameEncryption;
+
+/**
+ * Initializes CryptoOptions with all possible options set explicitly. This
+ * is done when converting from a native RTCConfiguration.crypto_options.
+ */
+- (instancetype)initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
+              srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
+           srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
+                     sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDataChannel.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDataChannel.h
new file mode 100644
index 00000000..2d0661f1
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDataChannel.h
@@ -0,0 +1,132 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AvailabilityMacros.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataBuffer) : NSObject
+
+/** NSData representation of the underlying buffer. */
+@property(nonatomic, readonly) NSData *data;
+
+/** Indicates whether |data| contains UTF-8 or binary data. */
+@property(nonatomic, readonly) BOOL isBinary;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Initialize an RTCDataBuffer from NSData. |isBinary| indicates whether |data|
+ * contains UTF-8 or binary data.
+ */
+- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary;
+
+@end
+
+@class RTC_OBJC_TYPE(RTCDataChannel);
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCDataChannelDelegate)<NSObject>
+
+    /** The data channel state changed. */
+    - (void)dataChannelDidChangeState : (RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
+
+/** The data channel successfully received a data buffer. */
+- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
+    didReceiveMessageWithBuffer:(RTC_OBJC_TYPE(RTCDataBuffer) *)buffer;
+
+@optional
+/** The data channel's |bufferedAmount| changed. */
+- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
+    didChangeBufferedAmount:(uint64_t)amount;
+
+@end
+
+/** Represents the state of the data channel. */
+typedef NS_ENUM(NSInteger, RTCDataChannelState) {
+  RTCDataChannelStateConnecting,
+  RTCDataChannelStateOpen,
+  RTCDataChannelStateClosing,
+  RTCDataChannelStateClosed,
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataChannel) : NSObject
+
+/**
+ * A label that can be used to distinguish this data channel from other data
+ * channel objects.
+ */
+@property(nonatomic, readonly) NSString *label;
+
+/** Whether the data channel can send messages in unreliable mode. */
+@property(nonatomic, readonly) BOOL isReliable DEPRECATED_ATTRIBUTE;
+
+/** Returns whether this data channel is ordered or not. */
+@property(nonatomic, readonly) BOOL isOrdered;
+
+/** Deprecated. Use maxPacketLifeTime. */
+@property(nonatomic, readonly) NSUInteger maxRetransmitTime DEPRECATED_ATTRIBUTE;
+
+/**
+ * The length of the time window (in milliseconds) during which transmissions
+ * and retransmissions may occur in unreliable mode.
+ */
+@property(nonatomic, readonly) uint16_t maxPacketLifeTime;
+
+/**
+ * The maximum number of retransmissions that are attempted in unreliable mode.
+ */
+@property(nonatomic, readonly) uint16_t maxRetransmits;
+
+/**
+ * The name of the sub-protocol used with this data channel, if any. Otherwise
+ * this returns an empty string.
+ */
+@property(nonatomic, readonly) NSString *protocol;
+
+/**
+ * Returns whether this data channel was negotiated by the application or not.
+ */
+@property(nonatomic, readonly) BOOL isNegotiated;
+
+/** Deprecated. Use channelId. */
+@property(nonatomic, readonly) NSInteger streamId DEPRECATED_ATTRIBUTE;
+
+/** The identifier for this data channel. */
+@property(nonatomic, readonly) int channelId;
+
+/** The state of the data channel. */
+@property(nonatomic, readonly) RTCDataChannelState readyState;
+
+/**
+ * The number of bytes of application data that have been queued using
+ * |sendData:| but that have not yet been transmitted to the network.
+ */
+@property(nonatomic, readonly) uint64_t bufferedAmount;
+
+/** The delegate for this data channel. */
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCDataChannelDelegate)> delegate;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Closes the data channel. */
+- (void)close;
+
+/** Attempt to send |data| on this data channel's underlying data transport. */
+- (BOOL)sendData:(RTC_OBJC_TYPE(RTCDataBuffer) *)data;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDataChannelConfiguration.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDataChannelConfiguration.h
new file mode 100644
index 00000000..9459ae0a
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDataChannelConfiguration.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AvailabilityMacros.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataChannelConfiguration) : NSObject
+
+/** Set to YES if ordered delivery is required. */
+@property(nonatomic, assign) BOOL isOrdered;
+
+/** Deprecated. Use maxPacketLifeTime. */
+@property(nonatomic, assign) NSInteger maxRetransmitTimeMs DEPRECATED_ATTRIBUTE;
+
+/**
+ * Max period in milliseconds in which retransmissions will be sent. After this
+ * time, no more retransmissions will be sent. -1 if unset.
+ */
+@property(nonatomic, assign) int maxPacketLifeTime;
+
+/** The max number of retransmissions. -1 if unset. */
+@property(nonatomic, assign) int maxRetransmits;
+
+/** Set to YES if the channel has been externally negotiated and we do not send
+ * an in-band signalling in the form of an "open" message.
+ */
+@property(nonatomic, assign) BOOL isNegotiated;
+
+/** Deprecated. Use channelId. */
+@property(nonatomic, assign) int streamId DEPRECATED_ATTRIBUTE;
+
+/** The id of the data channel. */
+@property(nonatomic, assign) int channelId;
+
+/** Set by the application and opaque to the WebRTC implementation. */
+@property(nonatomic) NSString* protocol;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDefaultVideoDecoderFactory.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDefaultVideoDecoderFactory.h
new file mode 100644
index 00000000..de5a9c46
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDefaultVideoDecoderFactory.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoderFactory.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This decoder factory include support for all codecs bundled with WebRTC. If using custom
+ *  codecs, create custom implementations of RTCVideoEncoderFactory and
+ *  RTCVideoDecoderFactory.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDefaultVideoDecoderFactory) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoderFactory)>
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDefaultVideoEncoderFactory.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDefaultVideoEncoderFactory.h
new file mode 100644
index 00000000..92ab40c9
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDefaultVideoEncoderFactory.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoderFactory.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This encoder factory include support for all codecs bundled with WebRTC. If using custom
+ *  codecs, create custom implementations of RTCVideoEncoderFactory and
+ *  RTCVideoDecoderFactory.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDefaultVideoEncoderFactory) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoderFactory)>
+
+@property(nonatomic, retain) RTC_OBJC_TYPE(RTCVideoCodecInfo) *preferredCodec;
+
++ (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDispatcher.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDispatcher.h
new file mode 100644
index 00000000..e148af6d
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDispatcher.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
+  // Main dispatcher queue.
+  RTCDispatcherTypeMain,
+  // Used for starting/stopping AVCaptureSession, and assigning
+  // capture session to AVCaptureVideoPreviewLayer.
+  RTCDispatcherTypeCaptureSession,
+  // Used for operations on AVAudioSession.
+  RTCDispatcherTypeAudioSession,
+  // Used for operations on NWPathMonitor.
+  RTCDispatcherTypeNetworkMonitor,
+};
+
+/** Dispatcher that asynchronously dispatches blocks to a specific
+ *  shared dispatch queue.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDispatcher) : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Dispatch the block asynchronously on the queue for dispatchType.
+ *  @param dispatchType The queue type to dispatch on.
+ *  @param block The block to dispatch asynchronously.
+ */
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType block:(dispatch_block_t)block;
+
+/** Returns YES if run on queue for the dispatchType otherwise NO.
+ *  Useful for asserting that a method is run on a correct queue.
+ */
++ (BOOL)isOnQueueForType:(RTCDispatcherQueueType)dispatchType;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDtmfSender.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDtmfSender.h
new file mode 100644
index 00000000..0f1b6ba4
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCDtmfSender.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCDtmfSender)<NSObject>
+
+    /**
+     * Returns true if this RTCDtmfSender is capable of sending DTMF. Otherwise
+     * returns false. To be able to send DTMF, the associated RTCRtpSender must be
+     * able to send packets, and a "telephone-event" codec must be negotiated.
+     */
+    @property(nonatomic, readonly) BOOL canInsertDtmf;
+
+/**
+ * Queues a task that sends the DTMF tones. The tones parameter is treated
+ * as a series of characters. The characters 0 through 9, A through D, #, and *
+ * generate the associated DTMF tones. The characters a to d are equivalent
+ * to A to D. The character ',' indicates a delay of 2 seconds before
+ * processing the next character in the tones parameter.
+ *
+ * Unrecognized characters are ignored.
+ *
+ * @param duration The parameter indicates the duration to use for each
+ * character passed in the tones parameter. The duration cannot be more
+ * than 6000 or less than 70 ms.
+ *
+ * @param interToneGap The parameter indicates the gap between tones.
+ * This parameter must be at least 50 ms but should be as short as
+ * possible.
+ *
+ * If InsertDtmf is called on the same object while an existing task for this
+ * object to generate DTMF is still running, the previous task is canceled.
+ * Returns true on success and false on failure.
+ */
+- (BOOL)insertDtmf:(nonnull NSString *)tones
+          duration:(NSTimeInterval)duration
+      interToneGap:(NSTimeInterval)interToneGap;
+
+/** The tones remaining to be played out */
+- (nonnull NSString *)remainingTones;
+
+/**
+ * The current tone duration value. This value will be the value last set via the
+ * insertDtmf method, or the default value of 100 ms if insertDtmf was never called.
+ */
+- (NSTimeInterval)duration;
+
+/**
+ * The current value of the between-tone gap. This value will be the value last set
+ * via the insertDtmf() method, or the default value of 50 ms if insertDtmf() was never
+ * called.
+ */
+- (NSTimeInterval)interToneGap;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCEAGLVideoView.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCEAGLVideoView.h
new file mode 100644
index 00000000..24b26cd6
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCEAGLVideoView.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoRenderer.h"
+#import "RTCVideoViewShading.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCEAGLVideoView);
+
+/**
+ * RTCEAGLVideoView is an RTCVideoRenderer which renders video frames
+ * in its bounds using OpenGLES 2.0 or OpenGLES 3.0.
+ */
+RTC_OBJC_EXPORT
+NS_EXTENSION_UNAVAILABLE_IOS("Rendering not available in app extensions.")
+@interface RTC_OBJC_TYPE (RTCEAGLVideoView) : UIView <RTC_OBJC_TYPE(RTCVideoRenderer)>
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoViewDelegate)> delegate;
+
+- (instancetype)initWithFrame:(CGRect)frame
+                       shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)initWithCoder:(NSCoder *)aDecoder
+                       shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader
+    NS_DESIGNATED_INITIALIZER;
+
+/** @abstract Wrapped RTCVideoRotation, or nil.
+ */
+@property(nonatomic, nullable) NSValue *rotationOverride;
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCEncodedImage.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCEncodedImage.h
new file mode 100644
index 00000000..5fec8a22
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCEncodedImage.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Represents an encoded frame's type. */
+typedef NS_ENUM(NSUInteger, RTCFrameType) {
+  RTCFrameTypeEmptyFrame = 0,
+  RTCFrameTypeAudioFrameSpeech = 1,
+  RTCFrameTypeAudioFrameCN = 2,
+  RTCFrameTypeVideoFrameKey = 3,
+  RTCFrameTypeVideoFrameDelta = 4,
+};
+
+typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
+  RTCVideoContentTypeUnspecified,
+  RTCVideoContentTypeScreenshare,
+};
+
+/** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCEncodedImage) : NSObject
+
+@property(nonatomic, strong) NSData *buffer;
+@property(nonatomic, assign) int32_t encodedWidth;
+@property(nonatomic, assign) int32_t encodedHeight;
+@property(nonatomic, assign) uint32_t timeStamp;
+@property(nonatomic, assign) int64_t captureTimeMs;
+@property(nonatomic, assign) int64_t ntpTimeMs;
+@property(nonatomic, assign) uint8_t flags;
+@property(nonatomic, assign) int64_t encodeStartMs;
+@property(nonatomic, assign) int64_t encodeFinishMs;
+@property(nonatomic, assign) RTCFrameType frameType;
+@property(nonatomic, assign) RTCVideoRotation rotation;
+@property(nonatomic, assign) BOOL completeFrame;
+@property(nonatomic, strong) NSNumber *qp;
+@property(nonatomic, assign) RTCVideoContentType contentType;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCFieldTrials.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCFieldTrials.h
new file mode 100644
index 00000000..7477ad02
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCFieldTrials.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */
+RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweKey;
+RTC_EXTERN NSString * const kRTCFieldTrialAudioForceNoTWCCKey;
+RTC_EXTERN NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey;
+RTC_EXTERN NSString * const kRTCFieldTrialSendSideBweWithOverheadKey;
+RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03AdvertisedKey;
+RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03Key;
+RTC_EXTERN NSString * const kRTCFieldTrialH264HighProfileKey;
+RTC_EXTERN NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey;
+RTC_EXTERN NSString *const kRTCFieldTrialUseNWPathMonitor;
+
+/** The valid value for field trials above. */
+RTC_EXTERN NSString * const kRTCFieldTrialEnabledValue;
+
+/** Initialize field trials using a dictionary mapping field trial keys to their
+ * values. See above for valid keys and values. Must be called before any other
+ * call into WebRTC. See: webrtc/system_wrappers/include/field_trial.h
+ */
+RTC_EXTERN void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials);
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCFileLogger.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCFileLogger.h
new file mode 100644
index 00000000..853e673a
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCFileLogger.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSUInteger, RTCFileLoggerSeverity) {
+  RTCFileLoggerSeverityVerbose,
+  RTCFileLoggerSeverityInfo,
+  RTCFileLoggerSeverityWarning,
+  RTCFileLoggerSeverityError
+};
+
+typedef NS_ENUM(NSUInteger, RTCFileLoggerRotationType) {
+  RTCFileLoggerTypeCall,
+  RTCFileLoggerTypeApp,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+// This class intercepts WebRTC logs and saves them to a file. The file size
+// will not exceed the given maximum bytesize. When the maximum bytesize is
+// reached, logs are rotated according to the rotationType specified.
+// For kRTCFileLoggerTypeCall, logs from the beginning and the end
+// are preserved while the middle section is overwritten instead.
+// For kRTCFileLoggerTypeApp, the oldest log is overwritten.
+// This class is not threadsafe.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCFileLogger) : NSObject
+
+// The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
+@property(nonatomic, assign) RTCFileLoggerSeverity severity;
+
+// The rotation type for this file logger. The default is
+// kRTCFileLoggerTypeCall.
+@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
+
+// Disables buffering disk writes. Should be set before |start|. Buffering
+// is enabled by default for performance.
+@property(nonatomic, assign) BOOL shouldDisableBuffering;
+
+// Default constructor provides default settings for dir path, file size and
+// rotation type.
+- (instancetype)init;
+
+// Create file logger with default rotation type.
+- (instancetype)initWithDirPath:(NSString *)dirPath maxFileSize:(NSUInteger)maxFileSize;
+
+- (instancetype)initWithDirPath:(NSString *)dirPath
+                    maxFileSize:(NSUInteger)maxFileSize
+                   rotationType:(RTCFileLoggerRotationType)rotationType NS_DESIGNATED_INITIALIZER;
+
+// Starts writing WebRTC logs to disk if not already started. Overwrites any
+// existing file(s).
+- (void)start;
+
+// Stops writing WebRTC logs to disk. This method is also called on dealloc.
+- (void)stop;
+
+// Returns the current contents of the logs, or nil if start has been called
+// without a stop.
+- (nullable NSData *)logData;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCFileVideoCapturer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCFileVideoCapturer.h
new file mode 100644
index 00000000..19262c64
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCFileVideoCapturer.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCVideoCapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * Error passing block.
+ */
+typedef void (^RTCFileVideoCapturerErrorBlock)(NSError *error);
+
+/**
+ * Captures buffers from bundled video file.
+ *
+ * See @c RTCVideoCapturer for more info on capturers.
+ */
+RTC_OBJC_EXPORT
+
+NS_CLASS_AVAILABLE_IOS(10)
+@interface RTC_OBJC_TYPE (RTCFileVideoCapturer) : RTC_OBJC_TYPE(RTCVideoCapturer)
+
+/**
+ * Starts asynchronous capture of frames from video file.
+ *
+ * Capturing is not started if error occurs. Underlying error will be
+ * relayed in the errorBlock if one is provided.
+ * Successfully captured video frames will be passed to the delegate.
+ *
+ * @param nameOfFile The name of the bundled video file to be read.
+ * @errorBlock block to be executed upon error.
+ */
+- (void)startCapturingFromFileNamed:(NSString *)nameOfFile
+                            onError:(__nullable RTCFileVideoCapturerErrorBlock)errorBlock;
+
+/**
+ * Immediately stops capture.
+ */
+- (void)stopCapture;
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCH264ProfileLevelId.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCH264ProfileLevelId.h
new file mode 100644
index 00000000..dac7bb56
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCH264ProfileLevelId.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+RTC_OBJC_EXPORT extern NSString *const kRTCVideoCodecH264Name;
+RTC_OBJC_EXPORT extern NSString *const kRTCLevel31ConstrainedHigh;
+RTC_OBJC_EXPORT extern NSString *const kRTCLevel31ConstrainedBaseline;
+RTC_OBJC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh;
+RTC_OBJC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline;
+
+/** H264 Profiles and levels. */
+typedef NS_ENUM(NSUInteger, RTCH264Profile) {
+  RTCH264ProfileConstrainedBaseline,
+  RTCH264ProfileBaseline,
+  RTCH264ProfileMain,
+  RTCH264ProfileConstrainedHigh,
+  RTCH264ProfileHigh,
+};
+
+typedef NS_ENUM(NSUInteger, RTCH264Level) {
+  RTCH264Level1_b = 0,
+  RTCH264Level1 = 10,
+  RTCH264Level1_1 = 11,
+  RTCH264Level1_2 = 12,
+  RTCH264Level1_3 = 13,
+  RTCH264Level2 = 20,
+  RTCH264Level2_1 = 21,
+  RTCH264Level2_2 = 22,
+  RTCH264Level3 = 30,
+  RTCH264Level3_1 = 31,
+  RTCH264Level3_2 = 32,
+  RTCH264Level4 = 40,
+  RTCH264Level4_1 = 41,
+  RTCH264Level4_2 = 42,
+  RTCH264Level5 = 50,
+  RTCH264Level5_1 = 51,
+  RTCH264Level5_2 = 52
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCH264ProfileLevelId) : NSObject
+
+@property(nonatomic, readonly) RTCH264Profile profile;
+@property(nonatomic, readonly) RTCH264Level level;
+@property(nonatomic, readonly) NSString *hexString;
+
+- (instancetype)initWithHexString:(NSString *)hexString;
+- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCI420Buffer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCI420Buffer.h
new file mode 100644
index 00000000..b97f05a5
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCI420Buffer.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCYUVPlanarBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Protocol for RTCYUVPlanarBuffers containing I420 data */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCI420Buffer)<RTC_OBJC_TYPE(RTCYUVPlanarBuffer)> @end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCIceCandidate.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCIceCandidate.h
new file mode 100644
index 00000000..f84843af
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCIceCandidate.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCIceCandidate) : NSObject
+
+/**
+ * If present, the identifier of the "media stream identification" for the media
+ * component this candidate is associated with.
+ */
+@property(nonatomic, readonly, nullable) NSString *sdpMid;
+
+/**
+ * The index (starting at zero) of the media description this candidate is
+ * associated with in the SDP.
+ */
+@property(nonatomic, readonly) int sdpMLineIndex;
+
+/** The SDP string for this candidate. */
+@property(nonatomic, readonly) NSString *sdp;
+
+/** The URL of the ICE server which this candidate is gathered from. */
+@property(nonatomic, readonly, nullable) NSString *serverUrl;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Initialize an RTCIceCandidate from SDP.
+ */
+- (instancetype)initWithSdp:(NSString *)sdp
+              sdpMLineIndex:(int)sdpMLineIndex
+                     sdpMid:(nullable NSString *)sdpMid NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCIceServer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCIceServer.h
new file mode 100644
index 00000000..dd66c61a
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCIceServer.h
@@ -0,0 +1,114 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSUInteger, RTCTlsCertPolicy) {
+  RTCTlsCertPolicySecure,
+  RTCTlsCertPolicyInsecureNoCheck
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCIceServer) : NSObject
+
+/** URI(s) for this server represented as NSStrings. */
+@property(nonatomic, readonly) NSArray<NSString *> *urlStrings;
+
+/** Username to use if this RTCIceServer object is a TURN server. */
+@property(nonatomic, readonly, nullable) NSString *username;
+
+/** Credential to use if this RTCIceServer object is a TURN server. */
+@property(nonatomic, readonly, nullable) NSString *credential;
+
+/**
+ * TLS certificate policy to use if this RTCIceServer object is a TURN server.
+ */
+@property(nonatomic, readonly) RTCTlsCertPolicy tlsCertPolicy;
+
+/**
+  If the URIs in |urls| only contain IP addresses, this field can be used
+  to indicate the hostname, which may be necessary for TLS (using the SNI
+  extension). If |urls| itself contains the hostname, this isn't necessary.
+ */
+@property(nonatomic, readonly, nullable) NSString *hostname;
+
+/** List of protocols to be used in the TLS ALPN extension. */
+@property(nonatomic, readonly) NSArray<NSString *> *tlsAlpnProtocols;
+
+/**
+  List elliptic curves to be used in the TLS elliptic curves extension.
+  Only curve names supported by OpenSSL should be used (eg. "P-256","X25519").
+  */
+@property(nonatomic, readonly) NSArray<NSString *> *tlsEllipticCurves;
+
+- (nonnull instancetype)init NS_UNAVAILABLE;
+
+/** Convenience initializer for a server with no authentication (e.g. STUN). */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, and credentialType.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, and TLS cert policy.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, TLS cert policy and hostname.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(nullable NSString *)hostname;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, TLS cert policy, hostname and ALPN protocols.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(nullable NSString *)hostname
+                  tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, TLS cert policy, hostname, ALPN protocols and
+ * elliptic curves.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(nullable NSString *)hostname
+                  tlsAlpnProtocols:(nullable NSArray<NSString *> *)tlsAlpnProtocols
+                 tlsEllipticCurves:(nullable NSArray<NSString *> *)tlsEllipticCurves
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCLegacyStatsReport.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCLegacyStatsReport.h
new file mode 100644
index 00000000..b3bd12c5
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCLegacyStatsReport.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This does not currently conform to the spec. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCLegacyStatsReport) : NSObject
+
+/** Time since 1970-01-01T00:00:00Z in milliseconds. */
+@property(nonatomic, readonly) CFTimeInterval timestamp;
+
+/** The type of stats held by this object. */
+@property(nonatomic, readonly) NSString *type;
+
+/** The identifier for this object. */
+@property(nonatomic, readonly) NSString *reportId;
+
+/** A dictionary holding the actual stats. */
+@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *values;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCLogging.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCLogging.h
new file mode 100644
index 00000000..754945c8
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCLogging.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+// Subset of rtc::LoggingSeverity.
+typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
+  RTCLoggingSeverityVerbose,
+  RTCLoggingSeverityInfo,
+  RTCLoggingSeverityWarning,
+  RTCLoggingSeverityError,
+  RTCLoggingSeverityNone,
+};
+
+// Wrapper for C++ RTC_LOG(sev) macros.
+// Logs the log string to the webrtc logstream for the given severity.
+RTC_EXTERN void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
+
+// Wrapper for rtc::LogMessage::LogToDebug.
+// Sets the minimum severity to be logged to console.
+RTC_EXTERN void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
+
+// Returns the filename with the path prefix removed.
+RTC_EXTERN NSString* RTCFileName(const char* filePath);
+
+// Some convenience macros.
+
+#define RTCLogString(format, ...)                                           \
+  [NSString stringWithFormat:@"(%@:%d %s): " format, RTCFileName(__FILE__), \
+                             __LINE__, __FUNCTION__, ##__VA_ARGS__]
+
+#define RTCLogFormat(severity, format, ...)                     \
+  do {                                                          \
+    NSString* log_string = RTCLogString(format, ##__VA_ARGS__); \
+    RTCLogEx(severity, log_string);                             \
+  } while (false)
+
+#define RTCLogVerbose(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__)
+
+#define RTCLogInfo(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__)
+
+#define RTCLogWarning(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__)
+
+#define RTCLogError(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__)
+
+#if !defined(NDEBUG)
+#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
+#else
+#define RTCLogDebug(format, ...) \
+  do {                           \
+  } while (false)
+#endif
+
+#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMTLVideoView.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMTLVideoView.h
new file mode 100644
index 00000000..5678112a
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMTLVideoView.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrame.h"
+#import "RTCVideoRenderer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * RTCMTLVideoView is thin wrapper around MTKView.
+ *
+ * It has id<RTCVideoRenderer> property that renders video frames in the view's
+ * bounds using Metal.
+ * NOTE: always check if metal is available on the running device via
+ * RTC_SUPPORTS_METAL macro before initializing this class.
+ */
+NS_CLASS_AVAILABLE_IOS(9)
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMTLVideoView) : UIView<RTC_OBJC_TYPE(RTCVideoRenderer)>
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoViewDelegate)> delegate;
+
+@property(nonatomic) UIViewContentMode videoContentMode;
+
+/** @abstract Enables/disables rendering.
+ */
+@property(nonatomic, getter=isEnabled) BOOL enabled;
+
+/** @abstract Wrapped RTCVideoRotation, or nil.
+ */
+@property(nonatomic, nullable) NSValue* rotationOverride;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMacros.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMacros.h
new file mode 100644
index 00000000..e527ff6b
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMacros.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_BASE_RTCMACROS_H_
+#define SDK_OBJC_BASE_RTCMACROS_H_
+
+// Internal macros used to correctly concatenate symbols.
+#define RTC_SYMBOL_CONCAT_HELPER(a, b) a##b
+#define RTC_SYMBOL_CONCAT(a, b) RTC_SYMBOL_CONCAT_HELPER(a, b)
+
+// RTC_OBJC_TYPE_PREFIX
+//
+// Macro used to prepend a prefix to the API types that are exported with
+// RTC_OBJC_EXPORT.
+//
+// Clients can patch the definition of this macro locally and build
+// WebRTC.framework with their own prefix in case symbol clashing is a
+// problem.
+//
+// This macro must only be defined here and not on via compiler flag to
+// ensure it has a unique value.
+#define RTC_OBJC_TYPE_PREFIX
+
+// RCT_OBJC_TYPE
+//
+// Macro used internally to declare API types. Declaring an API type without
+// using this macro will not include the declared type in the set of types
+// that will be affected by the configurable RTC_OBJC_TYPE_PREFIX.
+#define RTC_OBJC_TYPE(type_name) RTC_SYMBOL_CONCAT(RTC_OBJC_TYPE_PREFIX, type_name)
+
+#define RTC_OBJC_EXPORT __attribute__((visibility("default")))
+
+#if defined(__cplusplus)
+#define RTC_EXTERN extern "C" RTC_OBJC_EXPORT
+#else
+#define RTC_EXTERN extern RTC_OBJC_EXPORT
+#endif
+
+#ifdef __OBJC__
+#define RTC_FWD_DECL_OBJC_CLASS(classname) @class classname
+#else
+#define RTC_FWD_DECL_OBJC_CLASS(classname) typedef struct objc_object classname
+#endif
+
+#endif  // SDK_OBJC_BASE_RTCMACROS_H_
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaConstraints.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaConstraints.h
new file mode 100644
index 00000000..c5baf20c
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaConstraints.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Constraint keys for media sources. */
+/** The value for this key should be a base64 encoded string containing
+ *  the data from the serialized configuration proto.
+ */
+RTC_EXTERN NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
+
+/** Constraint keys for generating offers and answers. */
+RTC_EXTERN NSString *const kRTCMediaConstraintsIceRestart;
+RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveAudio;
+RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveVideo;
+RTC_EXTERN NSString *const kRTCMediaConstraintsVoiceActivityDetection;
+
+/** Constraint values for Boolean parameters. */
+RTC_EXTERN NSString *const kRTCMediaConstraintsValueTrue;
+RTC_EXTERN NSString *const kRTCMediaConstraintsValueFalse;
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaConstraints) : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize with mandatory and/or optional constraints. */
+- (instancetype)
+    initWithMandatoryConstraints:(nullable NSDictionary<NSString *, NSString *> *)mandatory
+             optionalConstraints:(nullable NSDictionary<NSString *, NSString *> *)optional
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaSource.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaSource.h
new file mode 100644
index 00000000..ba19c2a3
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaSource.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSInteger, RTCSourceState) {
+  RTCSourceStateInitializing,
+  RTCSourceStateLive,
+  RTCSourceStateEnded,
+  RTCSourceStateMuted,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaSource) : NSObject
+
+/** The current state of the RTCMediaSource. */
+@property(nonatomic, readonly) RTCSourceState state;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaStream.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaStream.h
new file mode 100644
index 00000000..2d56f15c
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaStream.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@class RTC_OBJC_TYPE(RTCVideoTrack);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaStream) : NSObject
+
+/** The audio tracks in this stream. */
+@property(nonatomic, strong, readonly) NSArray<RTC_OBJC_TYPE(RTCAudioTrack) *> *audioTracks;
+
+/** The video tracks in this stream. */
+@property(nonatomic, strong, readonly) NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *videoTracks;
+
+/** An identifier for this media stream. */
+@property(nonatomic, readonly) NSString *streamId;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Adds the given audio track to this media stream. */
+- (void)addAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack;
+
+/** Adds the given video track to this media stream. */
+- (void)addVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack;
+
+/** Removes the given audio track to this media stream. */
+- (void)removeAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack;
+
+/** Removes the given video track to this media stream. */
+- (void)removeVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaStreamTrack.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaStreamTrack.h
new file mode 100644
index 00000000..2200122c
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMediaStreamTrack.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/**
+ * Represents the state of the track. This exposes the same states in C++.
+ */
+typedef NS_ENUM(NSInteger, RTCMediaStreamTrackState) {
+  RTCMediaStreamTrackStateLive,
+  RTCMediaStreamTrackStateEnded
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN NSString *const kRTCMediaStreamTrackKindAudio;
+RTC_EXTERN NSString *const kRTCMediaStreamTrackKindVideo;
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaStreamTrack) : NSObject
+
+/**
+ * The kind of track. For example, "audio" if this track represents an audio
+ * track and "video" if this track represents a video track.
+ */
+@property(nonatomic, readonly) NSString *kind;
+
+/** An identifier string. */
+@property(nonatomic, readonly) NSString *trackId;
+
+/** The enabled state of the track. */
+@property(nonatomic, assign) BOOL isEnabled;
+
+/** The state of the track. */
+@property(nonatomic, readonly) RTCMediaStreamTrackState readyState;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMetrics.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMetrics.h
new file mode 100644
index 00000000..fddbb27c
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMetrics.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMetricsSampleInfo.h"
+
+/**
+ * Enables gathering of metrics (which can be fetched with
+ * RTCGetAndResetMetrics). Must be called before any other call into WebRTC.
+ */
+RTC_EXTERN void RTCEnableMetrics(void);
+
+/** Gets and clears native histograms. */
+RTC_EXTERN NSArray<RTC_OBJC_TYPE(RTCMetricsSampleInfo) *>* RTCGetAndResetMetrics(void);
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMetricsSampleInfo.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMetricsSampleInfo.h
new file mode 100644
index 00000000..47a877b6
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMetricsSampleInfo.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMetricsSampleInfo) : NSObject
+
+/**
+ * Example of RTCMetricsSampleInfo:
+ * name: "WebRTC.Video.InputFramesPerSecond"
+ * min: 1
+ * max: 100
+ * bucketCount: 50
+ * samples: [29]:2 [30]:1
+ */
+
+/** The name of the histogram. */
+@property(nonatomic, readonly) NSString *name;
+
+/** The minimum bucket value. */
+@property(nonatomic, readonly) int min;
+
+/** The maximum bucket value. */
+@property(nonatomic, readonly) int max;
+
+/** The number of buckets. */
+@property(nonatomic, readonly) int bucketCount;
+
+/** A dictionary holding the samples <value, # of events>. */
+@property(nonatomic, readonly) NSDictionary<NSNumber *, NSNumber *> *samples;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMutableI420Buffer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMutableI420Buffer.h
new file mode 100644
index 00000000..cde72198
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMutableI420Buffer.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCI420Buffer.h"
+#import "RTCMutableYUVPlanarBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Extension of the I420 buffer with mutable data access */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCMutableI420Buffer)<RTC_OBJC_TYPE(RTCI420Buffer), RTC_OBJC_TYPE(RTCMutableYUVPlanarBuffer)> @end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMutableYUVPlanarBuffer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMutableYUVPlanarBuffer.h
new file mode 100644
index 00000000..bd14e3bc
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCMutableYUVPlanarBuffer.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCYUVPlanarBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Extension of the YUV planar data buffer with mutable data access */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCMutableYUVPlanarBuffer)<RTC_OBJC_TYPE(RTCYUVPlanarBuffer)>
+
+    @property(nonatomic, readonly) uint8_t *mutableDataY;
+@property(nonatomic, readonly) uint8_t *mutableDataU;
+@property(nonatomic, readonly) uint8_t *mutableDataV;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCNativeI420Buffer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCNativeI420Buffer.h
new file mode 100644
index 00000000..3afe2090
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCNativeI420Buffer.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCI420Buffer.h"
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCI420Buffer implements the RTCI420Buffer protocol */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCI420Buffer) : NSObject<RTC_OBJC_TYPE(RTCI420Buffer)>
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCNativeMutableI420Buffer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCNativeMutableI420Buffer.h
new file mode 100644
index 00000000..053a10a3
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCNativeMutableI420Buffer.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMutableI420Buffer.h"
+#import "RTCNativeI420Buffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Mutable version of RTCI420Buffer */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMutableI420Buffer) : RTC_OBJC_TYPE(RTCI420Buffer)<RTC_OBJC_TYPE(RTCMutableI420Buffer)>
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCNetworkMonitor.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCNetworkMonitor.h
new file mode 100644
index 00000000..21d22f54
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCNetworkMonitor.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2020 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Listens for NWPathMonitor updates and forwards the results to a C++
+ *  observer.
+ */
+@interface RTCNetworkMonitor : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCPeerConnection.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCPeerConnection.h
new file mode 100644
index 00000000..5401666f
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCPeerConnection.h
@@ -0,0 +1,374 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCConfiguration);
+@class RTC_OBJC_TYPE(RTCDataChannel);
+@class RTC_OBJC_TYPE(RTCDataChannelConfiguration);
+@class RTC_OBJC_TYPE(RTCIceCandidate);
+@class RTC_OBJC_TYPE(RTCMediaConstraints);
+@class RTC_OBJC_TYPE(RTCMediaStream);
+@class RTC_OBJC_TYPE(RTCMediaStreamTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@class RTC_OBJC_TYPE(RTCRtpReceiver);
+@class RTC_OBJC_TYPE(RTCRtpSender);
+@class RTC_OBJC_TYPE(RTCRtpTransceiver);
+@class RTC_OBJC_TYPE(RTCRtpTransceiverInit);
+@class RTC_OBJC_TYPE(RTCSessionDescription);
+@class RTC_OBJC_TYPE(RTCStatisticsReport);
+@class RTC_OBJC_TYPE(RTCLegacyStatsReport);
+
+typedef NS_ENUM(NSInteger, RTCRtpMediaType);
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const kRTCPeerConnectionErrorDomain;
+extern int const kRTCSessionDescriptionErrorCode;
+
+/** Represents the signaling state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCSignalingState) {
+  RTCSignalingStateStable,
+  RTCSignalingStateHaveLocalOffer,
+  RTCSignalingStateHaveLocalPrAnswer,
+  RTCSignalingStateHaveRemoteOffer,
+  RTCSignalingStateHaveRemotePrAnswer,
+  // Not an actual state, represents the total number of states.
+  RTCSignalingStateClosed,
+};
+
+/** Represents the ice connection state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCIceConnectionState) {
+  RTCIceConnectionStateNew,
+  RTCIceConnectionStateChecking,
+  RTCIceConnectionStateConnected,
+  RTCIceConnectionStateCompleted,
+  RTCIceConnectionStateFailed,
+  RTCIceConnectionStateDisconnected,
+  RTCIceConnectionStateClosed,
+  RTCIceConnectionStateCount,
+};
+
+/** Represents the combined ice+dtls connection state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCPeerConnectionState) {
+  RTCPeerConnectionStateNew,
+  RTCPeerConnectionStateConnecting,
+  RTCPeerConnectionStateConnected,
+  RTCPeerConnectionStateDisconnected,
+  RTCPeerConnectionStateFailed,
+  RTCPeerConnectionStateClosed,
+};
+
+/** Represents the ice gathering state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCIceGatheringState) {
+  RTCIceGatheringStateNew,
+  RTCIceGatheringStateGathering,
+  RTCIceGatheringStateComplete,
+};
+
+/** Represents the stats output level. */
+typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
+  RTCStatsOutputLevelStandard,
+  RTCStatsOutputLevelDebug,
+};
+
+@class RTC_OBJC_TYPE(RTCPeerConnection);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCPeerConnectionDelegate)<NSObject>
+
+    /** Called when the SignalingState changed. */
+    - (void)peerConnection
+    : (RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection didChangeSignalingState
+    : (RTCSignalingState)stateChanged;
+
+/** Called when media is received on a new stream from remote peer. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+          didAddStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Called when a remote peer closes a stream.
+ *  This is not called when RTCSdpSemanticsUnifiedPlan is specified.
+ */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+       didRemoveStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Called when negotiation is needed, for example ICE has restarted. */
+- (void)peerConnectionShouldNegotiate:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection;
+
+/** Called any time the IceConnectionState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeIceConnectionState:(RTCIceConnectionState)newState;
+
+/** Called any time the IceGatheringState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeIceGatheringState:(RTCIceGatheringState)newState;
+
+/** New ice candidate has been found. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didGenerateIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate;
+
+/** Called when a group of local Ice candidates have been removed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didRemoveIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+
+/** New data channel has been opened. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didOpenDataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
+
+/** Called when signaling indicates a transceiver will be receiving media from
+ *  the remote endpoint.
+ *  This is only called with RTCSdpSemanticsUnifiedPlan specified.
+ */
+@optional
+/** Called any time the IceConnectionState changes following standardized
+ * transition. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeStandardizedIceConnectionState:(RTCIceConnectionState)newState;
+
+/** Called any time the PeerConnectionState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeConnectionState:(RTCPeerConnectionState)newState;
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didStartReceivingOnTransceiver:(RTC_OBJC_TYPE(RTCRtpTransceiver) *)transceiver;
+
+/** Called when a receiver and its track are created. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+        didAddReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver
+               streams:(NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *)mediaStreams;
+
+/** Called when the receiver and its track are removed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+     didRemoveReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver;
+
+/** Called when the selected ICE candidate pair is changed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeLocalCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)local
+            remoteCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)remote
+             lastReceivedMs:(int)lastDataReceivedMs
+               changeReason:(NSString *)reason;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnection) : NSObject
+
+/** The object that will be notifed about events such as state changes and
+ *  streams being added or removed.
+ */
+@property(nonatomic, weak, nullable) id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)> delegate;
+/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
+ *  |senders| instead.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *localStreams;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * localDescription;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * remoteDescription;
+@property(nonatomic, readonly) RTCSignalingState signalingState;
+@property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
+@property(nonatomic, readonly) RTCPeerConnectionState connectionState;
+@property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
+@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCConfiguration) * configuration;
+
+/** Gets all RTCRtpSenders associated with this peer connection.
+ *  Note: reading this property returns different instances of RTCRtpSender.
+ *  Use isEqual: instead of == to compare RTCRtpSender instances.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpSender) *> *senders;
+
+/** Gets all RTCRtpReceivers associated with this peer connection.
+ *  Note: reading this property returns different instances of RTCRtpReceiver.
+ *  Use isEqual: instead of == to compare RTCRtpReceiver instances.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpReceiver) *> *receivers;
+
+/** Gets all RTCRtpTransceivers associated with this peer connection.
+ *  Note: reading this property returns different instances of
+ *  RTCRtpTransceiver. Use isEqual: instead of == to compare
+ *  RTCRtpTransceiver instances. This is only available with
+ * RTCSdpSemanticsUnifiedPlan specified.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpTransceiver) *> *transceivers;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Sets the PeerConnection's global configuration to |configuration|.
+ *  Any changes to STUN/TURN servers or ICE candidate policy will affect the
+ *  next gathering phase, and cause the next call to createOffer to generate
+ *  new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
+ *  cannot be changed with this method.
+ */
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration;
+
+/** Terminate all media and close the transport. */
+- (void)close;
+
+/** Provide a remote candidate to the ICE Agent. */
+- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate;
+
+/** Remove a group of remote candidates from the ICE Agent. */
+- (void)removeIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+
+/** Add a new media stream to be sent on this peer connection.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  addTrack instead.
+ */
+- (void)addStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Remove the given media stream from this peer connection.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  removeTrack instead.
+ */
+- (void)removeStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Add a new media stream track to be sent on this peer connection, and return
+ *  the newly created RTCRtpSender. The RTCRtpSender will be
+ * associated with the streams specified in the |streamIds| list.
+ *
+ *  Errors: If an error occurs, returns nil. An error can occur if:
+ *  - A sender already exists for the track.
+ *  - The peer connection is closed.
+ */
+- (RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+                                streamIds:(NSArray<NSString *> *)streamIds;
+
+/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
+ *
+ *  With UnifiedPlan semantics, sets sender's track to null and removes the
+ *  send component from the associated RTCRtpTransceiver's direction.
+ *
+ *  Returns YES on success.
+ */
+- (BOOL)removeTrack:(RTC_OBJC_TYPE(RTCRtpSender) *)sender;
+
+/** addTransceiver creates a new RTCRtpTransceiver and adds it to the set of
+ *  transceivers. Adding a transceiver will cause future calls to CreateOffer
+ *  to add a media description for the corresponding transceiver.
+ *
+ *  The initial value of |mid| in the returned transceiver is nil. Setting a
+ *  new session description may change it to a non-nil value.
+ *
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
+ *
+ *  Optionally, an RtpTransceiverInit structure can be specified to configure
+ *  the transceiver from construction. If not specified, the transceiver will
+ *  default to having a direction of kSendRecv and not be part of any streams.
+ *
+ *  These methods are only available when Unified Plan is enabled (see
+ *  RTCConfiguration).
+ */
+
+/** Adds a transceiver with a sender set to transmit the given track. The kind
+ *  of the transceiver (and sender/receiver) will be derived from the kind of
+ *  the track.
+ */
+- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
+    (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track;
+- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)
+    addTransceiverWithTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+                       init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
+
+/** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
+ *  or RTCRtpMediaTypeVideo.
+ */
+- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
+- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType
+                                                      init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)
+                                                               init;
+
+/** Generate an SDP offer. */
+- (void)offerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+          completionHandler:(nullable void (^)(RTC_OBJC_TYPE(RTCSessionDescription) * _Nullable sdp,
+                                               NSError *_Nullable error))completionHandler;
+
+/** Generate an SDP answer. */
+- (void)answerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+           completionHandler:
+               (nullable void (^)(RTC_OBJC_TYPE(RTCSessionDescription) * _Nullable sdp,
+                                  NSError *_Nullable error))completionHandler;
+
+/** Apply the supplied RTCSessionDescription as the local description. */
+- (void)setLocalDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+          completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
+
+/** Apply the supplied RTCSessionDescription as the remote description. */
+- (void)setRemoteDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+           completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
+
+/** Limits the bandwidth allocated for all RTP streams sent by this
+ *  PeerConnection. Nil parameters will be unchanged. Setting
+ * |currentBitrateBps| will force the available bitrate estimate to the given
+ *  value. Returns YES if the parameters were successfully updated.
+ */
+- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
+          currentBitrateBps:(nullable NSNumber *)currentBitrateBps
+              maxBitrateBps:(nullable NSNumber *)maxBitrateBps;
+
+/** Start or stop recording an Rtc EventLog. */
+- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
+- (void)stopRtcEventLog;
+
+@end
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(Media)
+
+    /** Create an RTCRtpSender with the specified kind and media stream ID.
+     *  See RTCMediaStreamTrack.h for available kinds.
+     *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+     *  addTransceiver instead.
+     */
+    - (RTC_OBJC_TYPE(RTCRtpSender) *)senderWithKind : (NSString *)kind streamId
+    : (NSString *)streamId;
+
+@end
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(DataChannel)
+
+    /** Create a new data channel with the given label and configuration. */
+    - (nullable RTC_OBJC_TYPE(RTCDataChannel) *)dataChannelForLabel
+    : (NSString *)label configuration : (RTC_OBJC_TYPE(RTCDataChannelConfiguration) *)configuration;
+
+@end
+
+typedef void (^RTCStatisticsCompletionHandler)(RTC_OBJC_TYPE(RTCStatisticsReport) *);
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(Stats)
+
+    /** Gather stats for the given RTCMediaStreamTrack. If |mediaStreamTrack| is nil
+     *  statistics are gathered for all tracks.
+     */
+    - (void)statsForTrack
+    : (nullable RTC_OBJC_TYPE(RTCMediaStreamTrack) *)mediaStreamTrack statsOutputLevel
+    : (RTCStatsOutputLevel)statsOutputLevel completionHandler
+    : (nullable void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))completionHandler;
+
+/** Gather statistic through the v2 statistics API. */
+- (void)statisticsWithCompletionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+/** Spec-compliant getStats() performing the stats selection algorithm with the
+ *  sender.
+ */
+- (void)statisticsForSender:(RTC_OBJC_TYPE(RTCRtpSender) *)sender
+          completionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+/** Spec-compliant getStats() performing the stats selection algorithm with the
+ *  receiver.
+ */
+- (void)statisticsForReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)receiver
+            completionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+@end
+
+NS_ASSUME_NONNULL_END
\ No newline at end of file
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCPeerConnectionFactory.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCPeerConnectionFactory.h
new file mode 100644
index 00000000..3dcd3b64
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCPeerConnectionFactory.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioSource);
+@class RTC_OBJC_TYPE(RTCAudioTrack);
+@class RTC_OBJC_TYPE(RTCConfiguration);
+@class RTC_OBJC_TYPE(RTCMediaConstraints);
+@class RTC_OBJC_TYPE(RTCMediaStream);
+@class RTC_OBJC_TYPE(RTCPeerConnection);
+@class RTC_OBJC_TYPE(RTCVideoSource);
+@class RTC_OBJC_TYPE(RTCVideoTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions);
+@protocol RTC_OBJC_TYPE
+(RTCPeerConnectionDelegate);
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoderFactory);
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderFactory);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactory) : NSObject
+
+/* Initialize object with default H264 video encoder/decoder factories */
+- (instancetype)init;
+
+/* Initialize object with injectable video encoder/decoder factories */
+- (instancetype)
+    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
+
+/** Initialize an RTCAudioSource with constraints. */
+- (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
+    (nullable RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints;
+
+/** Initialize an RTCAudioTrack with an id. Convenience ctor to use an audio source
+ * with no constraints.
+ */
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithTrackId:(NSString *)trackId;
+
+/** Initialize an RTCAudioTrack with a source and an id. */
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:(RTC_OBJC_TYPE(RTCAudioSource) *)source
+                                               trackId:(NSString *)trackId;
+
+/** Initialize a generic RTCVideoSource. The RTCVideoSource should be
+ * passed to a RTCVideoCapturer implementation, e.g.
+ * RTCCameraVideoCapturer, in order to produce frames.
+ */
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSource;
+
+/** Initialize an RTCVideoTrack with a source and an id. */
+- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
+                                               trackId:(NSString *)trackId;
+
+/** Initialize an RTCMediaStream with an id. */
+- (RTC_OBJC_TYPE(RTCMediaStream) *)mediaStreamWithStreamId:(NSString *)streamId;
+
+/** Initialize an RTCPeerConnection with a configuration, constraints, and
+ *  delegate.
+ */
+- (RTC_OBJC_TYPE(RTCPeerConnection) *)
+    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                           delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+
+/** Set the options to be used for subsequently created RTCPeerConnections */
+- (void)setOptions:(nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options;
+
+/** Start an AecDump recording. This API call will likely change in the future. */
+- (BOOL)startAecDumpWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
+
+/* Stop an active AecDump recording */
+- (void)stopAecDump;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCPeerConnectionFactoryOptions.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCPeerConnectionFactoryOptions.h
new file mode 100644
index 00000000..bfc54a5d
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCPeerConnectionFactoryOptions.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactoryOptions) : NSObject
+
+@property(nonatomic, assign) BOOL disableEncryption;
+
+@property(nonatomic, assign) BOOL disableNetworkMonitor;
+
+@property(nonatomic, assign) BOOL ignoreLoopbackNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreVPNNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreCellularNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreWiFiNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreEthernetNetworkAdapter;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtcpParameters.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtcpParameters.h
new file mode 100644
index 00000000..1bbaedcf
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtcpParameters.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtcpParameters) : NSObject
+
+/** The Canonical Name used by RTCP. */
+@property(nonatomic, readonly, copy) NSString *cname;
+
+/** Whether reduced size RTCP is configured or compound RTCP. */
+@property(nonatomic, assign) BOOL isReducedSize;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpCodecParameters.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpCodecParameters.h
new file mode 100644
index 00000000..a68d9eb8
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpCodecParameters.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN const NSString *const kRTCRtxCodecName;
+RTC_EXTERN const NSString *const kRTCRedCodecName;
+RTC_EXTERN const NSString *const kRTCUlpfecCodecName;
+RTC_EXTERN const NSString *const kRTCFlexfecCodecName;
+RTC_EXTERN const NSString *const kRTCOpusCodecName;
+RTC_EXTERN const NSString *const kRTCIsacCodecName;
+RTC_EXTERN const NSString *const kRTCL16CodecName;
+RTC_EXTERN const NSString *const kRTCG722CodecName;
+RTC_EXTERN const NSString *const kRTCIlbcCodecName;
+RTC_EXTERN const NSString *const kRTCPcmuCodecName;
+RTC_EXTERN const NSString *const kRTCPcmaCodecName;
+RTC_EXTERN const NSString *const kRTCDtmfCodecName;
+RTC_EXTERN const NSString *const kRTCComfortNoiseCodecName;
+RTC_EXTERN const NSString *const kRTCVp8CodecName;
+RTC_EXTERN const NSString *const kRTCVp9CodecName;
+RTC_EXTERN const NSString *const kRTCH264CodecName;
+
+/** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTC_OBJC_TYPE(RTCRtpCodecParameters) */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpCodecParameters) : NSObject
+
+/** The RTP payload type. */
+@property(nonatomic, assign) int payloadType;
+
+/**
+ * The codec MIME subtype. Valid types are listed in:
+ * http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2
+ *
+ * Several supported types are represented by the constants above.
+ */
+@property(nonatomic, readonly, nonnull) NSString *name;
+
+/**
+ * The media type of this codec. Equivalent to MIME top-level type.
+ *
+ * Valid values are kRTCMediaStreamTrackKindAudio and
+ * kRTCMediaStreamTrackKindVideo.
+ */
+@property(nonatomic, readonly, nonnull) NSString *kind;
+
+/** The codec clock rate expressed in Hertz. */
+@property(nonatomic, readonly, nullable) NSNumber *clockRate;
+
+/**
+ * The number of channels (mono=1, stereo=2).
+ * Set to null for video codecs.
+ **/
+@property(nonatomic, readonly, nullable) NSNumber *numChannels;
+
+/** The "format specific parameters" field from the "a=fmtp" line in the SDP */
+@property(nonatomic, readonly, nonnull) NSDictionary *parameters;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpEncodingParameters.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpEncodingParameters.h
new file mode 100644
index 00000000..88b8e49e
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpEncodingParameters.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Corresponds to webrtc::Priority. */
+typedef NS_ENUM(NSInteger, RTCPriority) {
+  RTCPriorityVeryLow,
+  RTCPriorityLow,
+  RTCPriorityMedium,
+  RTCPriorityHigh
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpEncodingParameters) : NSObject
+
+/** The idenfifier for the encoding layer. This is used in simulcast. */
+@property(nonatomic, copy, nullable) NSString *rid;
+
+/** Controls whether the encoding is currently transmitted. */
+@property(nonatomic, assign) BOOL isActive;
+
+/** The maximum bitrate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *maxBitrateBps;
+
+/** The minimum bitrate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *minBitrateBps;
+
+/** The maximum framerate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *maxFramerate;
+
+/** The requested number of temporal layers to use for the encoding, or nil
+ * if the default should be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *numTemporalLayers;
+
+/** Scale the width and height down by this factor for video. If nil,
+ * implementation default scaling factor will be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *scaleResolutionDownBy;
+
+/** The SSRC being used by this encoding. */
+@property(nonatomic, readonly, nullable) NSNumber *ssrc;
+
+/** The relative bitrate priority. */
+@property(nonatomic, assign) double bitratePriority;
+
+/** The relative DiffServ Code Point priority. */
+@property(nonatomic, assign) RTCPriority networkPriority;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
\ No newline at end of file
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpFragmentationHeader.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpFragmentationHeader.h
new file mode 100644
index 00000000..001b4e9d
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpFragmentationHeader.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Information for header. Corresponds to webrtc::RTPFragmentationHeader. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpFragmentationHeader) : NSObject
+
+@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationOffset;
+@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationLength;
+@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationTimeDiff;
+@property(nonatomic, strong) NSArray<NSNumber *> *fragmentationPlType;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpHeaderExtension.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpHeaderExtension.h
new file mode 100644
index 00000000..15be5af5
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpHeaderExtension.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpHeaderExtension) : NSObject
+
+/** The URI of the RTP header extension, as defined in RFC5285. */
+@property(nonatomic, readonly, copy) NSString *uri;
+
+/** The value put in the RTP packet to identify the header extension. */
+@property(nonatomic, readonly) int id;
+
+/** Whether the header extension is encrypted or not. */
+@property(nonatomic, readonly, getter=isEncrypted) BOOL encrypted;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpParameters.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpParameters.h
new file mode 100644
index 00000000..fff6a858
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpParameters.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCRtcpParameters.h"
+#import "RTCRtpCodecParameters.h"
+#import "RTCRtpEncodingParameters.h"
+#import "RTCRtpHeaderExtension.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Corresponds to webrtc::DegradationPreference. */
+typedef NS_ENUM(NSInteger, RTCDegradationPreference) {
+  RTCDegradationPreferenceDisabled,
+  RTCDegradationPreferenceMaintainFramerate,
+  RTCDegradationPreferenceMaintainResolution,
+  RTCDegradationPreferenceBalanced
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpParameters) : NSObject
+
+/** A unique identifier for the last set of parameters applied. */
+@property(nonatomic, copy) NSString *transactionId;
+
+/** Parameters used for RTCP. */
+@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCRtcpParameters) * rtcp;
+
+/** An array containing parameters for RTP header extensions. */
+@property(nonatomic, readonly, copy)
+    NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtension) *> *headerExtensions;
+
+/** The currently active encodings in the order of preference. */
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *encodings;
+
+/** The negotiated set of send codecs in order of preference. */
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpCodecParameters) *> *codecs;
+
+/**
+ * Degradation preference in case of CPU adaptation or constrained bandwidth.
+ * If nil, implementation default degradation preference will be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *degradationPreference;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpReceiver.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpReceiver.h
new file mode 100644
index 00000000..7ab2cfae
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpReceiver.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMediaStreamTrack.h"
+#import "RTCRtpParameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Represents the media type of the RtpReceiver. */
+typedef NS_ENUM(NSInteger, RTCRtpMediaType) {
+  RTCRtpMediaTypeAudio,
+  RTCRtpMediaTypeVideo,
+  RTCRtpMediaTypeData,
+};
+
+@class RTC_OBJC_TYPE(RTCRtpReceiver);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpReceiverDelegate)<NSObject>
+
+    /** Called when the first RTP packet is received.
+     *
+     *  Note: Currently if there are multiple RtpReceivers of the same media type,
+     *  they will all call OnFirstPacketReceived at once.
+     *
+     *  For example, if we create three audio receivers, A/B/C, they will listen to
+     *  the same signal from the underneath network layer. Whenever the first audio packet
+     *  is received, the underneath signal will be fired. All the receivers A/B/C will be
+     *  notified and the callback of the receiver's delegate will be called.
+     *
+     *  The process is the same for video receivers.
+     */
+    - (void)rtpReceiver
+    : (RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver didReceiveFirstPacketForMediaType
+    : (RTCRtpMediaType)mediaType;
+
+@end
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpReceiver)<NSObject>
+
+    /** A unique identifier for this receiver. */
+    @property(nonatomic, readonly) NSString *receiverId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ *
+ *  The WebRTC specification only defines RTCRtpParameters in terms of senders,
+ *  but this API also applies them to receivers, similar to ORTC:
+ *  http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
+ */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpParameters) * parameters;
+
+/** The RTCMediaStreamTrack associated with the receiver.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track;
+
+/** The delegate for this RtpReceiver. */
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCRtpReceiverDelegate)> delegate;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpReceiver) : NSObject <RTC_OBJC_TYPE(RTCRtpReceiver)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpSender.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpSender.h
new file mode 100644
index 00000000..fcdf1998
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpSender.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCDtmfSender.h"
+#import "RTCMacros.h"
+#import "RTCMediaStreamTrack.h"
+#import "RTCRtpParameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpSender)<NSObject>
+
+/** A unique identifier for this sender. */
+@property(nonatomic, readonly) NSString *senderId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ */
+@property(nonatomic, copy) RTC_OBJC_TYPE(RTCRtpParameters) * parameters;
+
+/** The RTCMediaStreamTrack associated with the sender.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, copy, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track;
+
+/** IDs of streams associated with the RTP sender */
+@property(nonatomic, copy) NSArray<NSString *> *streamIds;
+
+/** The RTCDtmfSender accociated with the RTP sender. */
+@property(nonatomic, readonly, nullable) id<RTC_OBJC_TYPE(RTCDtmfSender)> dtmfSender;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpSender) : NSObject <RTC_OBJC_TYPE(RTCRtpSender)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpTransceiver.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpTransceiver.h
new file mode 100644
index 00000000..fd590136
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCRtpTransceiver.h
@@ -0,0 +1,137 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCRtpReceiver.h"
+#import "RTCRtpSender.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const kRTCRtpTransceiverErrorDomain;
+
+/** https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverdirection */
+typedef NS_ENUM(NSInteger, RTCRtpTransceiverDirection) {
+  RTCRtpTransceiverDirectionSendRecv,
+  RTCRtpTransceiverDirectionSendOnly,
+  RTCRtpTransceiverDirectionRecvOnly,
+  RTCRtpTransceiverDirectionInactive,
+  RTCRtpTransceiverDirectionStopped
+};
+
+/** Structure for initializing an RTCRtpTransceiver in a call to
+ *  RTCPeerConnection.addTransceiver.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpTransceiverInit) : NSObject
+
+/** Direction of the RTCRtpTransceiver. See RTCRtpTransceiver.direction. */
+@property(nonatomic) RTCRtpTransceiverDirection direction;
+
+/** The added RTCRtpTransceiver will be added to these streams. */
+@property(nonatomic) NSArray<NSString *> *streamIds;
+
+/** TODO(bugs.webrtc.org/7600): Not implemented. */
+@property(nonatomic) NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *sendEncodings;
+
+@end
+
+@class RTC_OBJC_TYPE(RTCRtpTransceiver);
+
+/** The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the
+ *  WebRTC specification. A transceiver represents a combination of an RTCRtpSender
+ *  and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
+ *  RTCRtpTransceiver is said to be associated with a media description if its
+ *  mid property is non-nil; otherwise, it is said to be disassociated.
+ *  JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
+ *
+ *  Note that RTCRtpTransceivers are only supported when using
+ *  RTCPeerConnection with Unified Plan SDP.
+ *
+ *  WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpTransceiver)<NSObject>
+
+    /** Media type of the transceiver. The sender and receiver will also have this
+     *  type.
+     */
+    @property(nonatomic, readonly) RTCRtpMediaType mediaType;
+
+/** The mid attribute is the mid negotiated and present in the local and
+ *  remote descriptions. Before negotiation is complete, the mid value may be
+ *  nil. After rollbacks, the value may change from a non-nil value to nil.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
+ */
+@property(nonatomic, readonly) NSString *mid;
+
+/** The sender attribute exposes the RTCRtpSender corresponding to the RTP
+ *  media that may be sent with the transceiver's mid. The sender is always
+ *  present, regardless of the direction of media.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
+ */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpSender) * sender;
+
+/** The receiver attribute exposes the RTCRtpReceiver corresponding to the RTP
+ *  media that may be received with the transceiver's mid. The receiver is
+ *  always present, regardless of the direction of media.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
+ */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpReceiver) * receiver;
+
+/** The isStopped attribute indicates that the sender of this transceiver will
+ *  no longer send, and that the receiver will no longer receive. It is true if
+ *  either stop has been called or if setting the local or remote description
+ *  has caused the RTCRtpTransceiver to be stopped.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
+ */
+@property(nonatomic, readonly) BOOL isStopped;
+
+/** The direction attribute indicates the preferred direction of this
+ *  transceiver, which will be used in calls to createOffer and createAnswer.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
+ */
+@property(nonatomic, readonly) RTCRtpTransceiverDirection direction;
+
+/** The currentDirection attribute indicates the current direction negotiated
+ *  for this transceiver. If this transceiver has never been represented in an
+ *  offer/answer exchange, or if the transceiver is stopped, the value is not
+ *  present and this method returns NO.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
+ */
+- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut;
+
+/** The stop method irreversibly stops the RTCRtpTransceiver. The sender of
+ *  this transceiver will no longer send, the receiver will no longer receive.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
+ */
+- (void)stopInternal;
+
+/** An update of directionality does not take effect immediately. Instead,
+ *  future calls to createOffer and createAnswer mark the corresponding media
+ *  descriptions as sendrecv, sendonly, recvonly, or inactive.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
+ */
+- (void)setDirection:(RTCRtpTransceiverDirection)direction error:(NSError **)error;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpTransceiver) : NSObject <RTC_OBJC_TYPE(RTCRtpTransceiver)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCSSLAdapter.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCSSLAdapter.h
new file mode 100644
index 00000000..f68bc5e9
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCSSLAdapter.h
@@ -0,0 +1,20 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/**
+ * Initialize and clean up the SSL library. Failure is fatal. These call the
+ * corresponding functions in webrtc/rtc_base/ssladapter.h.
+ */
+RTC_EXTERN BOOL RTCInitializeSSL(void);
+RTC_EXTERN BOOL RTCCleanupSSL(void);
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCSessionDescription.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCSessionDescription.h
new file mode 100644
index 00000000..8a9479d5
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCSessionDescription.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/**
+ * Represents the session description type. This exposes the same types that are
+ * in C++, which doesn't include the rollback type that is in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCSdpType) {
+  RTCSdpTypeOffer,
+  RTCSdpTypePrAnswer,
+  RTCSdpTypeAnswer,
+  RTCSdpTypeRollback,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCSessionDescription) : NSObject
+
+/** The type of session description. */
+@property(nonatomic, readonly) RTCSdpType type;
+
+/** The SDP string representation of this session description. */
+@property(nonatomic, readonly) NSString *sdp;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize a session description with a type and SDP string. */
+- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp NS_DESIGNATED_INITIALIZER;
+
++ (NSString *)stringForType:(RTCSdpType)type;
+
++ (RTCSdpType)typeForString:(NSString *)string;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCStatisticsReport.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCStatisticsReport.h
new file mode 100644
index 00000000..38d93e87
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCStatisticsReport.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCStatistics);
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** A statistics report. Encapsulates a number of RTCStatistics objects. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCStatisticsReport) : NSObject
+
+/** The timestamp of the report in microseconds since 1970-01-01T00:00:00Z. */
+@property(nonatomic, readonly) CFTimeInterval timestamp_us;
+
+/** RTCStatistics objects by id. */
+@property(nonatomic, readonly) NSDictionary<NSString *, RTC_OBJC_TYPE(RTCStatistics) *> *statistics;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+/** A part of a report (a subreport) covering a certain area. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCStatistics) : NSObject
+
+/** The id of this subreport, e.g. "RTCMediaStreamTrack_receiver_2". */
+@property(nonatomic, readonly) NSString *id;
+
+/** The timestamp of the subreport in microseconds since 1970-01-01T00:00:00Z. */
+@property(nonatomic, readonly) CFTimeInterval timestamp_us;
+
+/** The type of the subreport, e.g. "track", "codec". */
+@property(nonatomic, readonly) NSString *type;
+
+/** The keys and values of the subreport, e.g. "totalFramesDuration = 5.551".
+    The values are either NSNumbers or NSStrings, or NSArrays encapsulating NSNumbers
+    or NSStrings. */
+@property(nonatomic, readonly) NSDictionary<NSString *, NSObject *> *values;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCTracing.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCTracing.h
new file mode 100644
index 00000000..5c66e5a6
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCTracing.h
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+RTC_EXTERN void RTCSetupInternalTracer(void);
+/** Starts capture to specified file. Must be a valid writable path.
+ *  Returns YES if capture starts.
+ */
+RTC_EXTERN BOOL RTCStartInternalCapture(NSString* filePath);
+RTC_EXTERN void RTCStopInternalCapture(void);
+RTC_EXTERN void RTCShutdownInternalTracer(void);
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoCapturer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoCapturer.h
new file mode 100644
index 00000000..a1ffdcf3
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoCapturer.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoFrame.h"
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCVideoCapturer);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoCapturerDelegate)<NSObject> -
+    (void)capturer : (RTC_OBJC_TYPE(RTCVideoCapturer) *)capturer didCaptureVideoFrame
+    : (RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoCapturer) : NSObject
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)> delegate;
+
+- (instancetype)initWithDelegate:(id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoCodecConstants.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoCodecConstants.h
new file mode 100644
index 00000000..03f36e22
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoCodecConstants.h
@@ -0,0 +1,16 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+RTC_OBJC_EXPORT extern NSString* const kRTCVideoCodecVp8Name;
+RTC_OBJC_EXPORT extern NSString* const kRTCVideoCodecVp9Name;
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoCodecInfo.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoCodecInfo.h
new file mode 100644
index 00000000..fa28958f
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoCodecInfo.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Holds information to identify a codec. Corresponds to webrtc::SdpVideoFormat. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoCodecInfo) : NSObject <NSCoding>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+- (instancetype)initWithName:(NSString *)name;
+
+- (instancetype)initWithName:(NSString *)name
+                  parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters
+    NS_DESIGNATED_INITIALIZER;
+
+- (BOOL)isEqualToCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+
+@property(nonatomic, readonly) NSString *name;
+@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *parameters;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoder.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoder.h
new file mode 100644
index 00000000..0fc88c91
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoder.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCodecSpecificInfo.h"
+#import "RTCEncodedImage.h"
+#import "RTCMacros.h"
+#import "RTCVideoEncoderSettings.h"
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Callback block for decoder. */
+typedef void (^RTCVideoDecoderCallback)(RTC_OBJC_TYPE(RTCVideoFrame) * frame);
+
+/** Protocol for decoder implementations. */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoder)<NSObject>
+
+    - (void)setCallback : (RTCVideoDecoderCallback)callback;
+- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores;
+- (NSInteger)releaseDecoder;
+- (NSInteger)decode:(RTC_OBJC_TYPE(RTCEncodedImage) *)encodedImage
+        missingFrames:(BOOL)missingFrames
+    codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
+         renderTimeMs:(int64_t)renderTimeMs;
+- (NSString *)implementationName;
+
+@end
+
+NS_ASSUME_NONNULL_END
\ No newline at end of file
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderFactory.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderFactory.h
new file mode 100644
index 00000000..8d901385
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderFactory.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCodecInfo.h"
+#import "RTCVideoDecoder.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoderFactory)<NSObject>
+
+    - (nullable id<RTC_OBJC_TYPE(RTCVideoDecoder)>)createDecoder
+    : (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)
+    supportedCodecs;  // TODO(andersc): "supportedFormats" instead?
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderFactoryH264.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderFactoryH264.h
new file mode 100644
index 00000000..88bacbbd
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderFactoryH264.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoderFactory.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderFactoryH264) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoderFactory)>
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderH264.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderH264.h
new file mode 100644
index 00000000..a12e4212
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderH264.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderH264) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoder)>
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderVP8.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderVP8.h
new file mode 100644
index 00000000..a118b25e
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderVP8.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderVP8) : NSObject
+
+/* This returns a VP8 decoder that can be returned from a RTCVideoDecoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)vp8Decoder;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderVP9.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderVP9.h
new file mode 100644
index 00000000..b3a17430
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoDecoderVP9.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderVP9) : NSObject
+
+/* This returns a VP9 decoder that can be returned from a RTCVideoDecoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)vp9Decoder;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoder.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoder.h
new file mode 100644
index 00000000..7d1a7afd
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoder.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCodecSpecificInfo.h"
+#import "RTCEncodedImage.h"
+#import "RTCMacros.h"
+#import "RTCRtpFragmentationHeader.h"
+#import "RTCVideoEncoderQpThresholds.h"
+#import "RTCVideoEncoderSettings.h"
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Callback block for encoder. */
+typedef BOOL (^RTCVideoEncoderCallback)(RTC_OBJC_TYPE(RTCEncodedImage) * frame,
+                                        id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)> info,
+                                        RTC_OBJC_TYPE(RTCRtpFragmentationHeader) * header);
+
+/** Protocol for encoder implementations. */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoder)<NSObject>
+
+    - (void)setCallback : (RTCVideoEncoderCallback)callback;
+- (NSInteger)startEncodeWithSettings:(RTC_OBJC_TYPE(RTCVideoEncoderSettings) *)settings
+                       numberOfCores:(int)numberOfCores;
+- (NSInteger)releaseEncoder;
+- (NSInteger)encode:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame
+    codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
+           frameTypes:(NSArray<NSNumber *> *)frameTypes;
+- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate;
+- (NSString *)implementationName;
+
+/** Returns QP scaling settings for encoder. The quality scaler adjusts the resolution in order to
+ *  keep the QP from the encoded images within the given range. Returning nil from this function
+ *  disables quality scaling. */
+- (nullable RTC_OBJC_TYPE(RTCVideoEncoderQpThresholds) *)scalingSettings;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderFactory.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderFactory.h
new file mode 100644
index 00000000..733cb277
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderFactory.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCodecInfo.h"
+#import "RTCVideoEncoder.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCVideoEncoderFactory is an Objective-C version of
+ webrtc::VideoEncoderFactory::VideoEncoderSelector.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderSelector)<NSObject>
+
+    - (void)registerCurrentEncoderInfo : (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+- (nullable RTC_OBJC_TYPE(RTCVideoCodecInfo) *)encoderForBitrate:(NSInteger)bitrate;
+- (nullable RTC_OBJC_TYPE(RTCVideoCodecInfo) *)encoderForBrokenEncoder;
+
+@end
+
+/** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderFactory)<NSObject>
+
+    - (nullable id<RTC_OBJC_TYPE(RTCVideoEncoder)>)createEncoder
+    : (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)
+    supportedCodecs;  // TODO(andersc): "supportedFormats" instead?
+
+@optional
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)implementations;
+- (nullable id<RTC_OBJC_TYPE(RTCVideoEncoderSelector)>)encoderSelector;
+
+@optional
+- (NSArray<RTCVideoCodecInfo *> *)implementations;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderFactoryH264.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderFactoryH264.h
new file mode 100644
index 00000000..45fc4be2
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderFactoryH264.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoderFactory.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderFactoryH264) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoderFactory)>
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderH264.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderH264.h
new file mode 100644
index 00000000..9f4f4c7c
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderH264.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCodecInfo.h"
+#import "RTCVideoEncoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderH264) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoder)>
+
+- (instancetype)initWithCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)codecInfo;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderQpThresholds.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderQpThresholds.h
new file mode 100644
index 00000000..1a6e9e88
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderQpThresholds.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderQpThresholds) : NSObject
+
+- (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high;
+
+@property(nonatomic, readonly) NSInteger low;
+@property(nonatomic, readonly) NSInteger high;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderSettings.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderSettings.h
new file mode 100644
index 00000000..ae792eab
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderSettings.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
+  RTCVideoCodecModeRealtimeVideo,
+  RTCVideoCodecModeScreensharing,
+};
+
+/** Settings for encoder. Corresponds to webrtc::VideoCodec. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderSettings) : NSObject
+
+@property(nonatomic, strong) NSString *name;
+
+@property(nonatomic, assign) unsigned short width;
+@property(nonatomic, assign) unsigned short height;
+
+@property(nonatomic, assign) unsigned int startBitrate;  // kilobits/sec.
+@property(nonatomic, assign) unsigned int maxBitrate;
+@property(nonatomic, assign) unsigned int minBitrate;
+
+@property(nonatomic, assign) uint32_t maxFramerate;
+
+@property(nonatomic, assign) unsigned int qpMax;
+@property(nonatomic, assign) RTCVideoCodecMode mode;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderVP8.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderVP8.h
new file mode 100644
index 00000000..e136a5bd
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderVP8.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderVP8) : NSObject
+
+/* This returns a VP8 encoder that can be returned from a RTCVideoEncoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)vp8Encoder;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderVP9.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderVP9.h
new file mode 100644
index 00000000..8f961ef3
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoEncoderVP9.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderVP9) : NSObject
+
+/* This returns a VP9 encoder that can be returned from a RTCVideoEncoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)vp9Encoder;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoFrame.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoFrame.h
new file mode 100644
index 00000000..f5638d27
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoFrame.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef NS_ENUM(NSInteger, RTCVideoRotation) {
+  RTCVideoRotation_0 = 0,
+  RTCVideoRotation_90 = 90,
+  RTCVideoRotation_180 = 180,
+  RTCVideoRotation_270 = 270,
+};
+
+@protocol RTC_OBJC_TYPE
+(RTCVideoFrameBuffer);
+
+// RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoFrame) : NSObject
+
+/** Width without rotation applied. */
+@property(nonatomic, readonly) int width;
+
+/** Height without rotation applied. */
+@property(nonatomic, readonly) int height;
+@property(nonatomic, readonly) RTCVideoRotation rotation;
+
+/** Timestamp in nanoseconds. */
+@property(nonatomic, readonly) int64_t timeStampNs;
+
+/** Timestamp 90 kHz. */
+@property(nonatomic, assign) int32_t timeStamp;
+
+@property(nonatomic, readonly) id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> buffer;
+
+- (instancetype)init NS_UNAVAILABLE;
+- (instancetype) new NS_UNAVAILABLE;
+
+/** Initialize an RTCVideoFrame from a pixel buffer, rotation, and timestamp.
+ *  Deprecated - initialize with a RTCCVPixelBuffer instead
+ */
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                           rotation:(RTCVideoRotation)rotation
+                        timeStampNs:(int64_t)timeStampNs
+    DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
+
+/** Initialize an RTCVideoFrame from a pixel buffer combined with cropping and
+ *  scaling. Cropping will be applied first on the pixel buffer, followed by
+ *  scaling to the final resolution of scaledWidth x scaledHeight.
+ */
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                        scaledWidth:(int)scaledWidth
+                       scaledHeight:(int)scaledHeight
+                          cropWidth:(int)cropWidth
+                         cropHeight:(int)cropHeight
+                              cropX:(int)cropX
+                              cropY:(int)cropY
+                           rotation:(RTCVideoRotation)rotation
+                        timeStampNs:(int64_t)timeStampNs
+    DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
+
+/** Initialize an RTCVideoFrame from a frame buffer, rotation, and timestamp.
+ */
+- (instancetype)initWithBuffer:(id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>)frameBuffer
+                      rotation:(RTCVideoRotation)rotation
+                   timeStampNs:(int64_t)timeStampNs;
+
+/** Return a frame that is guaranteed to be I420, i.e. it is possible to access
+ *  the YUV data on it.
+ */
+- (RTC_OBJC_TYPE(RTCVideoFrame) *)newI420VideoFrame;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoFrameBuffer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoFrameBuffer.h
new file mode 100644
index 00000000..82d057ee
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoFrameBuffer.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@protocol RTC_OBJC_TYPE
+(RTCI420Buffer);
+
+// RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoFrameBuffer)<NSObject>
+
+    @property(nonatomic, readonly) int width;
+@property(nonatomic, readonly) int height;
+
+- (id<RTC_OBJC_TYPE(RTCI420Buffer)>)toI420;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoRenderer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoRenderer.h
new file mode 100644
index 00000000..0f763295
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoRenderer.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#if TARGET_OS_IPHONE
+#import <UIKit/UIKit.h>
+#endif
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCVideoFrame);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoRenderer)<NSObject>
+
+    /** The size of the frame. */
+    - (void)setSize : (CGSize)size;
+
+/** The frame to be displayed. */
+- (void)renderFrame:(nullable RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+
+@end
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoViewDelegate)
+
+    - (void)videoView : (id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoView didChangeVideoSize
+    : (CGSize)size;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoSource.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoSource.h
new file mode 100644
index 00000000..cdef8b89
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoSource.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMediaSource.h"
+#import "RTCVideoCapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+
+@interface RTC_OBJC_TYPE (RTCVideoSource) : RTC_OBJC_TYPE(RTCMediaSource) <RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Calling this function will cause frames to be scaled down to the
+ * requested resolution. Also, frames will be cropped to match the
+ * requested aspect ratio, and frames will be dropped to match the
+ * requested fps. The requested aspect ratio is orientation agnostic and
+ * will be adjusted to maintain the input orientation, so it doesn't
+ * matter if e.g. 1280x720 or 720x1280 is requested.
+ */
+- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoTrack.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoTrack.h
new file mode 100644
index 00000000..5382b716
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoTrack.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaStreamTrack.h"
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@protocol RTC_OBJC_TYPE
+(RTCVideoRenderer);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@class RTC_OBJC_TYPE(RTCVideoSource);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoTrack) : RTC_OBJC_TYPE(RTCMediaStreamTrack)
+
+/** The video source for this video track. */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCVideoSource) *source;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Register a renderer that will render all frames received on this track. */
+- (void)addRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer;
+
+/** Deregister a renderer. */
+- (void)removeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoViewShading.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoViewShading.h
new file mode 100644
index 00000000..9df30a8f
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCVideoViewShading.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * RTCVideoViewShading provides a way for apps to customize the OpenGL(ES shaders
+ * used in rendering for the RTCEAGLVideoView/RTCNSGLVideoView.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoViewShading)<NSObject>
+
+    /** Callback for I420 frames. Each plane is given as a texture. */
+    - (void)applyShadingForFrameWithWidth : (int)width height : (int)height rotation
+    : (RTCVideoRotation)rotation yPlane : (GLuint)yPlane uPlane : (GLuint)uPlane vPlane
+    : (GLuint)vPlane;
+
+/** Callback for NV12 frames. Each plane is given as a texture. */
+- (void)applyShadingForFrameWithWidth:(int)width
+                               height:(int)height
+                             rotation:(RTCVideoRotation)rotation
+                               yPlane:(GLuint)yPlane
+                              uvPlane:(GLuint)uvPlane;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCYUVPlanarBuffer.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCYUVPlanarBuffer.h
new file mode 100644
index 00000000..be01b915
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/RTCYUVPlanarBuffer.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrameBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Protocol for RTCVideoFrameBuffers containing YUV planar data. */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCYUVPlanarBuffer)<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>
+
+    @property(nonatomic, readonly) int chromaWidth;
+@property(nonatomic, readonly) int chromaHeight;
+@property(nonatomic, readonly) const uint8_t *dataY;
+@property(nonatomic, readonly) const uint8_t *dataU;
+@property(nonatomic, readonly) const uint8_t *dataV;
+@property(nonatomic, readonly) int strideY;
+@property(nonatomic, readonly) int strideU;
+@property(nonatomic, readonly) int strideV;
+
+- (instancetype)initWithWidth:(int)width
+                       height:(int)height
+                        dataY:(const uint8_t *)dataY
+                        dataU:(const uint8_t *)dataU
+                        dataV:(const uint8_t *)dataV;
+- (instancetype)initWithWidth:(int)width height:(int)height;
+- (instancetype)initWithWidth:(int)width
+                       height:(int)height
+                      strideY:(int)strideY
+                      strideU:(int)strideU
+                      strideV:(int)strideV;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/UIDevice+RTCDevice.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/UIDevice+RTCDevice.h
new file mode 100644
index 00000000..a7f655ba
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/UIDevice+RTCDevice.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+typedef NS_ENUM(NSInteger, RTCDeviceType) {
+  RTCDeviceTypeUnknown,
+  RTCDeviceTypeIPhone1G,
+  RTCDeviceTypeIPhone3G,
+  RTCDeviceTypeIPhone3GS,
+  RTCDeviceTypeIPhone4,
+  RTCDeviceTypeIPhone4Verizon,
+  RTCDeviceTypeIPhone4S,
+  RTCDeviceTypeIPhone5GSM,
+  RTCDeviceTypeIPhone5GSM_CDMA,
+  RTCDeviceTypeIPhone5CGSM,
+  RTCDeviceTypeIPhone5CGSM_CDMA,
+  RTCDeviceTypeIPhone5SGSM,
+  RTCDeviceTypeIPhone5SGSM_CDMA,
+  RTCDeviceTypeIPhone6Plus,
+  RTCDeviceTypeIPhone6,
+  RTCDeviceTypeIPhone6S,
+  RTCDeviceTypeIPhone6SPlus,
+  RTCDeviceTypeIPhone7,
+  RTCDeviceTypeIPhone7Plus,
+  RTCDeviceTypeIPhoneSE,
+  RTCDeviceTypeIPhone8,
+  RTCDeviceTypeIPhone8Plus,
+  RTCDeviceTypeIPhoneX,
+  RTCDeviceTypeIPhoneXS,
+  RTCDeviceTypeIPhoneXSMax,
+  RTCDeviceTypeIPhoneXR,
+  RTCDeviceTypeIPhone11,
+  RTCDeviceTypeIPhone11Pro,
+  RTCDeviceTypeIPhone11ProMax,
+  RTCDeviceTypeIPodTouch1G,
+  RTCDeviceTypeIPodTouch2G,
+  RTCDeviceTypeIPodTouch3G,
+  RTCDeviceTypeIPodTouch4G,
+  RTCDeviceTypeIPodTouch5G,
+  RTCDeviceTypeIPodTouch6G,
+  RTCDeviceTypeIPodTouch7G,
+  RTCDeviceTypeIPad,
+  RTCDeviceTypeIPad2Wifi,
+  RTCDeviceTypeIPad2GSM,
+  RTCDeviceTypeIPad2CDMA,
+  RTCDeviceTypeIPad2Wifi2,
+  RTCDeviceTypeIPadMiniWifi,
+  RTCDeviceTypeIPadMiniGSM,
+  RTCDeviceTypeIPadMiniGSM_CDMA,
+  RTCDeviceTypeIPad3Wifi,
+  RTCDeviceTypeIPad3GSM_CDMA,
+  RTCDeviceTypeIPad3GSM,
+  RTCDeviceTypeIPad4Wifi,
+  RTCDeviceTypeIPad4GSM,
+  RTCDeviceTypeIPad4GSM_CDMA,
+  RTCDeviceTypeIPad5,
+  RTCDeviceTypeIPad6,
+  RTCDeviceTypeIPadAirWifi,
+  RTCDeviceTypeIPadAirCellular,
+  RTCDeviceTypeIPadAirWifiCellular,
+  RTCDeviceTypeIPadAir2,
+  RTCDeviceTypeIPadMini2GWifi,
+  RTCDeviceTypeIPadMini2GCellular,
+  RTCDeviceTypeIPadMini2GWifiCellular,
+  RTCDeviceTypeIPadMini3,
+  RTCDeviceTypeIPadMini4,
+  RTCDeviceTypeIPadPro9Inch,
+  RTCDeviceTypeIPadPro12Inch,
+  RTCDeviceTypeIPadPro12Inch2,
+  RTCDeviceTypeIPadPro10Inch,
+  RTCDeviceTypeIPad7Gen10Inch,
+  RTCDeviceTypeIPadPro3Gen11Inch,
+  RTCDeviceTypeIPadPro3Gen12Inch,
+  RTCDeviceTypeIPadMini5Gen,
+  RTCDeviceTypeIPadAir3Gen,
+  RTCDeviceTypeSimulatori386,
+  RTCDeviceTypeSimulatorx86_64,
+};
+
+@interface UIDevice (RTCDevice)
+
++ (RTCDeviceType)deviceType;
++ (BOOL)isIOS11OrLater;
+
+@end
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/WebRTC.h b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/WebRTC.h
new file mode 100644
index 00000000..00aac6d9
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Headers/WebRTC.h
@@ -0,0 +1,94 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <WebRTC/RTCCodecSpecificInfo.h>
+#import <WebRTC/RTCEncodedImage.h>
+#import <WebRTC/RTCI420Buffer.h>
+#import <WebRTC/RTCLogging.h>
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCMutableI420Buffer.h>
+#import <WebRTC/RTCMutableYUVPlanarBuffer.h>
+#import <WebRTC/RTCRtpFragmentationHeader.h>
+#import <WebRTC/RTCVideoCapturer.h>
+#import <WebRTC/RTCVideoCodecInfo.h>
+#import <WebRTC/RTCVideoDecoder.h>
+#import <WebRTC/RTCVideoDecoderFactory.h>
+#import <WebRTC/RTCVideoEncoder.h>
+#import <WebRTC/RTCVideoEncoderFactory.h>
+#import <WebRTC/RTCVideoEncoderQpThresholds.h>
+#import <WebRTC/RTCVideoEncoderSettings.h>
+#import <WebRTC/RTCVideoFrame.h>
+#import <WebRTC/RTCVideoFrameBuffer.h>
+#import <WebRTC/RTCVideoRenderer.h>
+#import <WebRTC/RTCYUVPlanarBuffer.h>
+#import <WebRTC/RTCAudioSession.h>
+#import <WebRTC/RTCAudioSessionConfiguration.h>
+#import <WebRTC/RTCCameraVideoCapturer.h>
+#import <WebRTC/RTCFileVideoCapturer.h>
+#import <WebRTC/RTCNetworkMonitor.h>
+#import <WebRTC/RTCMTLVideoView.h>
+#import <WebRTC/RTCEAGLVideoView.h>
+#import <WebRTC/RTCVideoViewShading.h>
+#import <WebRTC/RTCCodecSpecificInfoH264.h>
+#import <WebRTC/RTCDefaultVideoDecoderFactory.h>
+#import <WebRTC/RTCDefaultVideoEncoderFactory.h>
+#import <WebRTC/RTCH264ProfileLevelId.h>
+#import <WebRTC/RTCVideoDecoderFactoryH264.h>
+#import <WebRTC/RTCVideoDecoderH264.h>
+#import <WebRTC/RTCVideoEncoderFactoryH264.h>
+#import <WebRTC/RTCVideoEncoderH264.h>
+#import <WebRTC/RTCCVPixelBuffer.h>
+#import <WebRTC/RTCCameraPreviewView.h>
+#import <WebRTC/RTCDispatcher.h>
+#import <WebRTC/UIDevice+RTCDevice.h>
+#import <WebRTC/RTCAudioSource.h>
+#import <WebRTC/RTCAudioTrack.h>
+#import <WebRTC/RTCConfiguration.h>
+#import <WebRTC/RTCDataChannel.h>
+#import <WebRTC/RTCDataChannelConfiguration.h>
+#import <WebRTC/RTCFieldTrials.h>
+#import <WebRTC/RTCIceCandidate.h>
+#import <WebRTC/RTCIceServer.h>
+#import <WebRTC/RTCLegacyStatsReport.h>
+#import <WebRTC/RTCMediaConstraints.h>
+#import <WebRTC/RTCMediaSource.h>
+#import <WebRTC/RTCMediaStream.h>
+#import <WebRTC/RTCMediaStreamTrack.h>
+#import <WebRTC/RTCMetrics.h>
+#import <WebRTC/RTCMetricsSampleInfo.h>
+#import <WebRTC/RTCPeerConnection.h>
+#import <WebRTC/RTCPeerConnectionFactory.h>
+#import <WebRTC/RTCPeerConnectionFactoryOptions.h>
+#import <WebRTC/RTCRtcpParameters.h>
+#import <WebRTC/RTCRtpCodecParameters.h>
+#import <WebRTC/RTCRtpEncodingParameters.h>
+#import <WebRTC/RTCRtpHeaderExtension.h>
+#import <WebRTC/RTCRtpParameters.h>
+#import <WebRTC/RTCRtpReceiver.h>
+#import <WebRTC/RTCRtpSender.h>
+#import <WebRTC/RTCRtpTransceiver.h>
+#import <WebRTC/RTCDtmfSender.h>
+#import <WebRTC/RTCSSLAdapter.h>
+#import <WebRTC/RTCSessionDescription.h>
+#import <WebRTC/RTCStatisticsReport.h>
+#import <WebRTC/RTCTracing.h>
+#import <WebRTC/RTCCertificate.h>
+#import <WebRTC/RTCCryptoOptions.h>
+#import <WebRTC/RTCVideoSource.h>
+#import <WebRTC/RTCVideoTrack.h>
+#import <WebRTC/RTCVideoCodecConstants.h>
+#import <WebRTC/RTCVideoDecoderVP8.h>
+#import <WebRTC/RTCVideoDecoderVP9.h>
+#import <WebRTC/RTCVideoEncoderVP8.h>
+#import <WebRTC/RTCVideoEncoderVP9.h>
+#import <WebRTC/RTCNativeI420Buffer.h>
+#import <WebRTC/RTCNativeMutableI420Buffer.h>
+#import <WebRTC/RTCCallbackLogger.h>
+#import <WebRTC/RTCFileLogger.h>
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Info.plist b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Info.plist
new file mode 100644
index 00000000..a85d8008
Binary files /dev/null and b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Info.plist differ
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Modules/module.modulemap b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Modules/module.modulemap
new file mode 100644
index 00000000..cd485a4e
--- /dev/null
+++ b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/Modules/module.modulemap
@@ -0,0 +1,6 @@
+framework module WebRTC {
+  umbrella header "WebRTC.h"
+
+  export *
+  module * { export * }
+}
diff --git a/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/WebRTC b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/WebRTC
new file mode 100644
index 00000000..043372b8
Binary files /dev/null and b/lib/WebRTC.xcframework/ios-i386_x86_64-simulator/WebRTC.framework/WebRTC differ
diff --git a/lib/libWebRTC-LATEST-Universal-Release.a b/lib/libWebRTC-LATEST-Universal-Release.a
deleted file mode 100644
index be44c38f..00000000
Binary files a/lib/libWebRTC-LATEST-Universal-Release.a and /dev/null differ
diff --git a/package-lock.json b/package-lock.json
new file mode 100644
index 00000000..fb2d2317
--- /dev/null
+++ b/package-lock.json
@@ -0,0 +1,13075 @@
+{
+  "name": "cordova-plugin-iosrtc",
+  "version": "8.0.4",
+  "lockfileVersion": 2,
+  "requires": true,
+  "packages": {
+    "": {
+      "name": "cordova-plugin-iosrtc",
+      "version": "8.0.4",
+      "engines": [
+        {
+          "name": "cordova-ios",
+          "version": ">=5.0.1"
+        }
+      ],
+      "license": "MIT",
+      "dependencies": {
+        "debug": "^4.1.1",
+        "domready": "^1.0.8",
+        "html-entities": "^1.2.1",
+        "random-number": "^0.0.7",
+        "xcode": "^3.0.1",
+        "yaeti": "^1.0.2"
+      },
+      "devDependencies": {
+        "browserify": "^14.4.0",
+        "eslint": "7.12.1",
+        "eslint-config-prettier": "6.15.0",
+        "eslint-plugin-prettier": "3.1.4",
+        "gulp": "^4.0.2",
+        "gulp-derequire": "^2.1.1",
+        "gulp-header": "^2.0.9",
+        "gulp-jshint": "^2.0.4",
+        "jshint": "^2.9.4",
+        "jshint-stylish": "~2.2.1",
+        "prettier": "2.1.2",
+        "vinyl-buffer": "^1.0.1",
+        "vinyl-source-stream": "^2.0.0"
+      }
+    },
+    "node_modules/@babel/code-frame": {
+      "version": "7.10.4",
+      "resolved": "https://registry.npmjs.org/@babel/code-frame/-/code-frame-7.10.4.tgz",
+      "integrity": "sha512-vG6SvB6oYEhvgisZNFRmRCUkLz11c7rp+tbNTynGqc6mS1d5ATd/sGyV6W0KZZnXRKMTzZDRgQT3Ou9jhpAfUg==",
+      "dev": true,
+      "dependencies": {
+        "@babel/highlight": "^7.10.4"
+      }
+    },
+    "node_modules/@babel/helper-validator-identifier": {
+      "version": "7.10.4",
+      "resolved": "https://registry.npmjs.org/@babel/helper-validator-identifier/-/helper-validator-identifier-7.10.4.tgz",
+      "integrity": "sha512-3U9y+43hz7ZM+rzG24Qe2mufW5KhvFg/NhnNph+i9mgCtdTCtMJuI1TMkrIUiK7Ix4PYlRF9I5dhqaLYA/ADXw==",
+      "dev": true
+    },
+    "node_modules/@babel/highlight": {
+      "version": "7.10.4",
+      "resolved": "https://registry.npmjs.org/@babel/highlight/-/highlight-7.10.4.tgz",
+      "integrity": "sha512-i6rgnR/YgPEQzZZnbTHHuZdlE8qyoBNalD6F+q4vAFlcMEcqmkoG+mPqJYJCo63qPf74+Y1UZsl3l6f7/RIkmA==",
+      "dev": true,
+      "dependencies": {
+        "@babel/helper-validator-identifier": "^7.10.4",
+        "chalk": "^2.0.0",
+        "js-tokens": "^4.0.0"
+      }
+    },
+    "node_modules/@babel/highlight/node_modules/ansi-styles": {
+      "version": "3.2.1",
+      "resolved": "https://registry.npmjs.org/ansi-styles/-/ansi-styles-3.2.1.tgz",
+      "integrity": "sha512-VT0ZI6kZRdTh8YyJw3SMbYm/u+NqfsAxEpWO0Pf9sq8/e94WxxOpPKx9FR1FlyCtOVDNOQ+8ntlqFxiRc+r5qA==",
+      "dev": true,
+      "dependencies": {
+        "color-convert": "^1.9.0"
+      },
+      "engines": {
+        "node": ">=4"
+      }
+    },
+    "node_modules/@babel/highlight/node_modules/chalk": {
+      "version": "2.4.2",
+      "resolved": "https://registry.npmjs.org/chalk/-/chalk-2.4.2.tgz",
+      "integrity": "sha512-Mti+f9lpJNcwF4tWV8/OrTTtF1gZi+f8FqlyAdouralcFWFQWF2+NgCHShjkCb+IFBLq9buZwE1xckQU4peSuQ==",
+      "dev": true,
+      "dependencies": {
+        "ansi-styles": "^3.2.1",
+        "escape-string-regexp": "^1.0.5",
+        "supports-color": "^5.3.0"
+      },
+      "engines": {
+        "node": ">=4"
+      }
+    },
+    "node_modules/@babel/highlight/node_modules/color-convert": {
+      "version": "1.9.3",
+      "resolved": "https://registry.npmjs.org/color-convert/-/color-convert-1.9.3.tgz",
+      "integrity": "sha512-QfAUtd+vFdAtFQcC8CCyYt1fYWxSqAiK2cSD6zDB8N3cpsEBAvRxp9zOGg6G/SHHJYAT88/az/IuDGALsNVbGg==",
+      "dev": true,
+      "dependencies": {
+        "color-name": "1.1.3"
+      }
+    },
+    "node_modules/@babel/highlight/node_modules/color-name": {
+      "version": "1.1.3",
+      "resolved": "https://registry.npmjs.org/color-name/-/color-name-1.1.3.tgz",
+      "integrity": "sha1-p9BVi9icQveV3UIyj3QIMcpTvCU=",
+      "dev": true
+    },
+    "node_modules/@eslint/eslintrc": {
+      "version": "0.2.1",
+      "resolved": "https://registry.npmjs.org/@eslint/eslintrc/-/eslintrc-0.2.1.tgz",
+      "integrity": "sha512-XRUeBZ5zBWLYgSANMpThFddrZZkEbGHgUdt5UJjZfnlN9BGCiUBrf+nvbRupSjMvqzwnQN0qwCmOxITt1cfywA==",
+      "dev": true,
+      "dependencies": {
+        "ajv": "^6.12.4",
+        "debug": "^4.1.1",
+        "espree": "^7.3.0",
+        "globals": "^12.1.0",
+        "ignore": "^4.0.6",
+        "import-fresh": "^3.2.1",
+        "js-yaml": "^3.13.1",
+        "lodash": "^4.17.19",
+        "minimatch": "^3.0.4",
+        "strip-json-comments": "^3.1.1"
+      },
+      "engines": {
+        "node": "^10.12.0 || >=12.0.0"
+      }
+    },
+    "node_modules/@types/color-name": {
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+        "convert-source-map": "^1.5.0",
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+        "cliui": "^3.2.0",
+        "decamelize": "^1.1.1",
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+}
diff --git a/package.json b/package.json
index 8111f767..53724da6 100644
--- a/package.json
+++ b/package.json
@@ -1,40 +1,58 @@
 {
   "name": "cordova-plugin-iosrtc",
-  "version": "3.2.2",
+  "version": "8.0.4",
   "description": "Cordova iOS plugin exposing the full WebRTC W3C JavaScript APIs",
-  "author": "Iñaki Baz Castillo at eFace2Face, inc. (https://eface2face.com)",
+  "author": "Iñaki Baz Castillo (https://inakibaz.me)",
+  "contributors": [
+    "Saúl Ibarra Corretgé (http://bettercallsaghul.com)",
+    "Iñaki Baz Castillo (https://inakibaz.me)",
+    "Harold Thetiot (https://sylaps.com)"
+  ],
   "license": "MIT",
   "keywords": [
+    "cordova",
     "ecosystem:cordova",
     "webrtc",
-    "ios"
+    "ios",
+    "cordova-ios",
+    "cordova:plugin"
   ],
   "main": "js/iosrtc.js",
-  "homepage": "https://github.com/eface2face/cordova-plugin-iosrtc",
+  "homepage": "https://github.com/cordova-rtc/cordova-plugin-iosrtc",
   "repository": {
     "type": "git",
-    "url": "git+https://github.com/eface2face/cordova-plugin-iosrtc.git"
+    "url": "git+https://github.com/cordova-rtc/cordova-plugin-iosrtc.git"
   },
   "bugs": {
-    "url": "https://github.com/eface2face/cordova-plugin-iosrtc/issues"
+    "url": "https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues"
+  },
+  "scripts": {
+    "lint": "gulp lint && eslint '**/*.js'",
+    "lint-fix": "eslint --fix '**/*.js'",
+    "build": "gulp browserify"
   },
   "dependencies": {
-    "debug": "^2.2.0",
+    "debug": "^4.1.1",
     "domready": "^1.0.8",
+    "html-entities": "^1.2.1",
     "random-number": "^0.0.7",
-    "yaeti": "^0.0.6"
+    "xcode": "^3.0.1",
+    "yaeti": "^1.0.2"
   },
   "devDependencies": {
-    "browserify": "^13.0.1",
-    "gulp": "git+https://github.com/gulpjs/gulp.git#4.0",
-    "gulp-header": "^1.8.7",
-    "gulp-jscs": "^4.0.0",
-    "gulp-jscs-stylish": "^1.4.0",
-    "gulp-jshint": "^2.0.1",
-    "jshint": "^2.9.2",
-    "jshint-stylish": "~2.2.0",
-    "vinyl-buffer": "^1.0.0",
-    "vinyl-source-stream": "^1.1.0"
+    "browserify": "^14.4.0",
+    "eslint": "7.12.1",
+    "eslint-config-prettier": "6.15.0",
+    "eslint-plugin-prettier": "3.1.4",
+    "gulp": "^4.0.2",
+    "gulp-derequire": "^2.1.1",
+    "gulp-header": "^2.0.9",
+    "gulp-jshint": "^2.0.4",
+    "jshint": "^2.9.4",
+    "jshint-stylish": "~2.2.1",
+    "prettier": "2.1.2",
+    "vinyl-buffer": "^1.0.1",
+    "vinyl-source-stream": "^2.0.0"
   },
   "cordova": {
     "id": "cordova-plugin-iosrtc",
@@ -45,7 +63,7 @@
   "engines": [
     {
       "name": "cordova-ios",
-      "version": ">=3.8.0"
+      "version": ">=5.0.1"
     }
   ]
 }
diff --git a/plugin.xml b/plugin.xml
index fb7b4154..61c62b31 100644
--- a/plugin.xml
+++ b/plugin.xml
@@ -1,24 +1,27 @@
 <?xml version="1.0" encoding="UTF-8"?>
 <plugin xmlns="http://apache.org/cordova/ns/plugins/1.0"
 		id="cordova-plugin-iosrtc"
-		version="3.2.2">
+		version="8.0.4">
 
 	<name>iosrtc</name>
 	<description>Cordova iOS plugin exposing the full WebRTC W3C JavaScript APIs</description>
-	<author>Iñaki Baz Castillo at eFace2Face, inc.</author>
+	<author>eFace2Face, Inc. and BasqueVoIPMafia and Cordova-RTC</author>
 	<license>MIT</license>
 	<keywords>webrtc, ios</keywords>
 
 	<info>
 		cordova-plugin-iosrtc enables use of the JavaScript WebRTC APIs (as defined by the W3C) in iOS devices.
-		Learn more at https://github.com/eface2face/cordova-plugin-iosrtc
+		Learn more at https://github.com/cordova-rtc/cordova-plugin-iosrtc
 	</info>
 
 	<engines>
-		<engine name="cordova-ios" version=">=3.8.0" />
+		<engine name="cordova-ios" version=">=4.5.1" />
 	</engines>
 
 	<platform name="ios">
+
+		<hook type="after_prepare" src="extra/hooks/iosrtc-swift-support.js" />
+
 		<config-file target="config.xml" parent="/*">
 			<feature name="iosrtcPlugin">
 				<param name="ios-package" value="iosrtcPlugin" />
@@ -26,43 +29,34 @@
 			</feature>
 		</config-file>
 
-		<js-module src="dist/cordova-plugin-iosrtc.js" name="Plugin">
+		<preference name="MANUAL_INIT_AUDIO_DEVICE" default="FALSE"/>
+
+		<config-file target="*-Info.plist" parent="ManualInitAudioDevice">
+			<string>$MANUAL_INIT_AUDIO_DEVICE</string>
+		</config-file>
+
+		<config-file target="*-Info.plist" parent="UseManualLayoutRenderer">
+			<true/>
+		</config-file>
+
+		<!-- Permission notes -->
+		<config-file parent="NSCameraUsageDescription" target="*-Info.plist">
+			<string>This Application uses your camera to make video calls.</string>
+		</config-file>
+		<config-file parent="NSContactsUsageDescription" target="*-Info.plist">
+			<string>This Application needs access to your contacts in order to be able to call them.</string>
+		</config-file>
+		<config-file parent="NSMicrophoneUsageDescription" target="*-Info.plist">
+			<string>This Application uses your microphone to make calls.</string>
+		</config-file>
+		<config-file parent="NSBluetoothAlwaysUsageDescription" target="*-Info.plist">
+			<string>This Application uses your wireless headphones and microphone to make calls.</string>
+		</config-file>
+
+		<js-module src="www/cordova-plugin-iosrtc.js" name="Plugin">
 			<clobbers target="cordova.plugins.iosrtc" />
 		</js-module>
 
-		<!-- WebRTC ObjetiveC headers -->
-		<header-file src="src/webrtc-headers/RTCAVFoundationVideoSource.h" />
-		<header-file src="src/webrtc-headers/RTCAudioSource.h" />
-		<header-file src="src/webrtc-headers/RTCAudioTrack.h" />
-		<header-file src="src/webrtc-headers/RTCDataChannel.h" />
-		<header-file src="src/webrtc-headers/RTCDTMFSender.h" />
-		<header-file src="src/webrtc-headers/RTCEAGLVideoView.h" />
-		<header-file src="src/webrtc-headers/RTCFileLogger.h" />
-		<header-file src="src/webrtc-headers/RTCI420Frame.h" />
-		<header-file src="src/webrtc-headers/RTCICECandidate.h" />
-		<header-file src="src/webrtc-headers/RTCICEServer.h" />
-		<header-file src="src/webrtc-headers/RTCLogging.h" />
-		<header-file src="src/webrtc-headers/RTCMediaConstraints.h" />
-		<header-file src="src/webrtc-headers/RTCMediaSource.h" />
-		<header-file src="src/webrtc-headers/RTCMediaStream.h" />
-		<header-file src="src/webrtc-headers/RTCMediaStreamTrack.h" />
-		<header-file src="src/webrtc-headers/RTCNSGLVideoView.h" />
-		<header-file src="src/webrtc-headers/RTCOpenGLVideoRenderer.h" />
-		<header-file src="src/webrtc-headers/RTCPair.h" />
-		<header-file src="src/webrtc-headers/RTCPeerConnection.h" />
-		<header-file src="src/webrtc-headers/RTCPeerConnectionDelegate.h" />
-		<header-file src="src/webrtc-headers/RTCPeerConnectionFactory.h" />
-		<header-file src="src/webrtc-headers/RTCPeerConnectionInterface.h" />
-		<header-file src="src/webrtc-headers/RTCSessionDescription.h" />
-		<header-file src="src/webrtc-headers/RTCSessionDescriptionDelegate.h" />
-		<header-file src="src/webrtc-headers/RTCStatsDelegate.h" />
-		<header-file src="src/webrtc-headers/RTCStatsReport.h" />
-		<header-file src="src/webrtc-headers/RTCTypes.h" />
-		<header-file src="src/webrtc-headers/RTCVideoCapturer.h" />
-		<header-file src="src/webrtc-headers/RTCVideoRenderer.h" />
-		<header-file src="src/webrtc-headers/RTCVideoSource.h" />
-		<header-file src="src/webrtc-headers/RTCVideoTrack.h" />
-
 		<!-- Bridging header for using ObjetiveC libraries in Swift -->
 		<header-file src="src/cordova-plugin-iosrtc-Bridging-Header.h" />
 
@@ -80,6 +74,14 @@
 		<source-file src="src/PluginEnumerateDevices.swift" />
 		<source-file src="src/PluginUtils.swift" />
 		<source-file src="src/PluginMediaStreamRenderer.swift" />
+		<source-file src="src/PluginRTCAudioController.swift" />
+		<source-file src="src/PluginRTCVideoCaptureController.swift" />
+		<source-file src="src/PluginRTCRtpTransceiver.swift" />
+		<source-file src="src/PluginRTCRtpReceiver.swift" />
+		<source-file src="src/PluginRTCRtpSender.swift" />
+		<source-file src="src/PluginRTCRtpParameters.swift" />
+		<source-file src="src/PluginRTCRtpCodecParameters.swift" />
+		<source-file src="src/PluginRTCRtpEncodingParameters.swift" />
 
 		<!-- iOS shared dependencies -->
 		<framework src="AVFoundation.framework" />
@@ -99,8 +101,8 @@
 		<framework src="libsqlite3.0.dylib" />
 		<framework src="VideoToolbox.framework" />
 
-		<!-- WebRTC static library -->
-		<source-file src="lib/libWebRTC-LATEST-Universal-Release.a" framework="true" />
+		<!-- WebRTC library -->
+		<framework src="lib/WebRTC.xcframework" custom="true" embed="true" />
 	</platform>
 
 </plugin>
diff --git a/src/PluginEnumerateDevices.swift b/src/PluginEnumerateDevices.swift
index 65b0b95e..b7cdcfa9 100644
--- a/src/PluginEnumerateDevices.swift
+++ b/src/PluginEnumerateDevices.swift
@@ -6,47 +6,159 @@ import AVFoundation
  * Doc: https://developer.apple.com/library/mac/documentation/AVFoundation/Reference/AVCaptureDevice_Class/index.html
  */
 
+struct MediaDeviceInfo {
+	let deviceId, groupId, kind, label: String
+	init(deviceId: String, kind: String, label: String) {
+		self.deviceId = deviceId
+		self.groupId = ""
+		self.kind = kind
+		self.label = label
+	}
+}
+
+// Helper function inserted by Swift 4.2 migrator.
+fileprivate func convertFromAVMediaType(_ input: AVMediaType) -> String {
+	return input.rawValue
+}
 
 class PluginEnumerateDevices {
-	class func call(callback: (data: NSDictionary) -> Void) {
+	class func call(_ callback: (_ data: NSDictionary) -> Void) {
 		NSLog("PluginEnumerateDevices#call()")
 
-		let devices = AVCaptureDevice.devices() as! Array<AVCaptureDevice>
+		let audioDevices: [MediaDeviceInfo] = getAllAudioDevices()
+		let videoDevices: [MediaDeviceInfo] = getAllVideoDevices()
+		let allDevices = videoDevices + audioDevices;
 
 		let json: NSMutableDictionary = [
-			"devices": NSMutableDictionary()
+			"devices": NSMutableArray()
 		]
 
-		for device: AVCaptureDevice in devices {
-			var facing: String
-			let hasAudio = device.hasMediaType(AVMediaTypeAudio)
-			let hasVideo = device.hasMediaType(AVMediaTypeVideo)
-
-			switch device.position {
-			case AVCaptureDevicePosition.Unspecified:
-				facing = "unknown"
-			case AVCaptureDevicePosition.Back:
-				facing = "back"
-			case AVCaptureDevicePosition.Front:
-				facing = "front"
-			}
+		// Casting to NSMutableDictionary
+		for device: MediaDeviceInfo in allDevices {
+			(json["devices"] as! NSMutableArray).add([
+				"deviceId": device.deviceId,
+				"kind": device.kind,
+				"label": device.label,
+				"groupId": device.groupId
+			])
+		}
+
+		print("DEVICES => ", json)
+		callback(json as NSDictionary)
+	}
+}
 
-			NSLog("- device [uniqueID:'%@', localizedName:'%@', facing:%@, audio:%@, video:%@, connected:%@]",
-				String(device.uniqueID), String(device.localizedName), String(facing),
-				String(hasAudio), String(hasVideo), String(device.connected))
+fileprivate func getAllVideoDevices() -> [MediaDeviceInfo] {
+
+	var videoDevicesArr : [MediaDeviceInfo] = []
+	var deviceTypes: [AVCaptureDevice.DeviceType] = [.builtInTelephotoCamera, .builtInWideAngleCamera]
+	if #available(iOS 10.2, *) {
+		deviceTypes.append(.builtInDualCamera)
+
+		// Disabled tp prevent duplicate front camera
+		//if #available(iOS 11.1, *) {
+		//	deviceTypes.append(.builtInTrueDepthCamera)
+		//}
+	}
+
+	let videoDevices: [AVCaptureDevice] = AVCaptureDevice.DiscoverySession.init(
+		deviceTypes: deviceTypes,
+		mediaType: AVMediaType.video,
+		position: AVCaptureDevice.Position.unspecified
+	).devices
+
+	for device: AVCaptureDevice in videoDevices {
+		var facing: String
+		var facingLabel: String;
+		let hasAudio = device.hasMediaType(AVMediaType(rawValue: convertFromAVMediaType(AVMediaType.audio)))
+		let hasVideo = device.hasMediaType(AVMediaType(rawValue: convertFromAVMediaType(AVMediaType.video)))
+
+		switch device.position {
+		case AVCaptureDevice.Position.unspecified:
+			facing = "unknown"
+			facingLabel = "";
+		case AVCaptureDevice.Position.back:
+			facing = "back"
+			facingLabel = "Back Camera"
+		case AVCaptureDevice.Position.front:
+			facing = "front"
+			facingLabel = "Front Camera"
+		}
+
+		if device.isConnected == false || (hasAudio == false && hasVideo == false) {
+			continue
+		}
 
-			if device.connected == false || (hasAudio == false && hasVideo == false) {
-				continue
+		NSLog("- device [uniqueID:'%@', localizedName:'%@', facing:%@, audio:%@, video:%@, connected:%@]",
+			String(device.uniqueID), String(device.localizedName), String(facing),
+			String(hasAudio), String(hasVideo), String(device.isConnected))
+
+		if hasAudio == false {
+			// Add English facingLabel suffix if localizedName does not match for facing detection using label
+			let deviceLabel = device.localizedName.contains(facingLabel) ?
+					device.localizedName : device.localizedName + " (" + facingLabel + ")";
+
+			let device = MediaDeviceInfo(
+				deviceId: device.uniqueID,
+				kind: "videoinput",
+				label: deviceLabel
+			)
+
+			// Put Front devices at beginning of the videoDevicesArr
+			if (facing == "front") {
+				// Simple Swift 4 array unshift
+				let videoDevicesArrFirst : [MediaDeviceInfo] = [device]
+				videoDevicesArr = videoDevicesArrFirst + videoDevicesArr;
+			} else {
+				videoDevicesArr.append(device)
 			}
+		}
+	}
+
+	return videoDevicesArr
+}
+
+// Getter function inserted by get all audio devices connected
+fileprivate func getAllAudioDevices() -> [MediaDeviceInfo] {
+
+	let audioSession: AVAudioSession = AVAudioSession.sharedInstance()
+	var audioDevicesArr : [MediaDeviceInfo] = []
+	let audioInputDevices: [AVAudioSessionPortDescription] = audioSession.availableInputs!
+	var bluetoothDevice: AVAudioSessionPortDescription? = nil
+	var wiredDevice: AVAudioSessionPortDescription? = nil
+	var builtMicDevice: AVAudioSessionPortDescription? = nil
 
-			(json["devices"] as! NSMutableDictionary)[device.uniqueID] = [
-				"deviceId": device.uniqueID,
-				"kind": hasAudio ? "audioinput" : "videoinput",
-				"label": device.localizedName,
-				"facing": facing
-			]
+	for audioInput in audioInputDevices {
+
+		let device = MediaDeviceInfo(
+			deviceId: audioInput.uid,
+			kind: "audioinput",
+			label: audioInput.portName
+		);
+
+		audioDevicesArr.append(device)
+
+		// Initialize audioInputSelected. Default Built-In Microphone
+		if audioInput.portType == AVAudioSession.Port.builtInMic {
+			builtMicDevice = audioInput
+		}
+
+		if audioInput.portType == .bluetoothHFP || audioInput.portType == .bluetoothA2DP {
+			bluetoothDevice = audioInput
+		}
+
+		if audioInput.portType == .usbAudio || audioInput.portType == .headsetMic {
+			wiredDevice = audioInput
 		}
+	}
 
-		callback(data: json)
+	// Initialize audioInputSelected. Priority: [Wired - Wireless - Built-In Microphone]
+	if wiredDevice != nil {
+		PluginRTCAudioController.saveInputAudioDevice(inputDeviceUID: wiredDevice!.uid)
+	} else if bluetoothDevice != nil {
+		PluginRTCAudioController.saveInputAudioDevice(inputDeviceUID: bluetoothDevice!.uid)
+	} else if builtMicDevice != nil {
+		PluginRTCAudioController.saveInputAudioDevice(inputDeviceUID: builtMicDevice!.uid)
 	}
+	return audioDevicesArr
 }
diff --git a/src/PluginGetUserMedia.swift b/src/PluginGetUserMedia.swift
index c3a23504..1c3feb1b 100644
--- a/src/PluginGetUserMedia.swift
+++ b/src/PluginGetUserMedia.swift
@@ -1,188 +1,172 @@
 import Foundation
 import AVFoundation
-
+import UIKit.UIDevice
 
 class PluginGetUserMedia {
-	var rtcPeerConnectionFactory: RTCPeerConnectionFactory
 
+	var rtcPeerConnectionFactory: RTCPeerConnectionFactory
 
 	init(rtcPeerConnectionFactory: RTCPeerConnectionFactory) {
 		NSLog("PluginGetUserMedia#init()")
-
 		self.rtcPeerConnectionFactory = rtcPeerConnectionFactory
+		UIDevice.current.beginGeneratingDeviceOrientationNotifications()
 	}
 
-
 	deinit {
 		NSLog("PluginGetUserMedia#deinit()")
+		UIDevice.current.endGeneratingDeviceOrientationNotifications()
 	}
 
-
 	func call(
-		constraints: NSDictionary,
-		callback: (data: NSDictionary) -> Void,
-		errback: (error: String) -> Void,
-		eventListenerForNewStream: (pluginMediaStream: PluginMediaStream) -> Void
+		_ constraints: NSDictionary,
+		callback: (_ data: NSDictionary) -> Void,
+		errback: (_ error: String) -> Void,
+		eventListenerForNewStream: (_ pluginMediaStream: PluginMediaStream) -> Void
 	) {
+
 		NSLog("PluginGetUserMedia#call()")
 
-		let	audioRequested = constraints.objectForKey("audio") as? Bool ?? false
-		let	videoRequested = constraints.objectForKey("video") as? Bool ?? false
-		let	videoDeviceId = constraints.objectForKey("videoDeviceId") as? String
-		let	videoMinWidth = constraints.objectForKey("videoMinWidth") as? Int ?? 0
-		let	videoMaxWidth = constraints.objectForKey("videoMaxWidth") as? Int ?? 0
-		let	videoMinHeight = constraints.objectForKey("videoMinHeight") as? Int ?? 0
-		let	videoMaxHeight = constraints.objectForKey("videoMaxHeight") as? Int ?? 0
-		let	videoMinFrameRate = constraints.objectForKey("videoMinFrameRate") as? Float ?? 0.0
-		let	videoMaxFrameRate = constraints.objectForKey("videoMaxFrameRate") as? Float ?? 0.0
+		var videoRequested: Bool = false
+		var audioRequested: Bool = false
+
+		if (constraints.object(forKey: "video") != nil) {
+			videoRequested = true
+		}
+
+		if constraints.object(forKey: "audio") != nil {
+			audioRequested = true
+		}
 
 		var rtcMediaStream: RTCMediaStream
 		var pluginMediaStream: PluginMediaStream?
 		var rtcAudioTrack: RTCAudioTrack?
 		var rtcVideoTrack: RTCVideoTrack?
-		var rtcVideoCapturer: RTCVideoCapturer?
 		var rtcVideoSource: RTCVideoSource?
-		var videoDevice: AVCaptureDevice?
-		var mandatoryConstraints: [RTCPair] = []
-		var constraints: RTCMediaConstraints
 
 		if videoRequested == true {
-			switch AVCaptureDevice.authorizationStatusForMediaType(AVMediaTypeVideo) {
-			case AVAuthorizationStatus.NotDetermined:
+			switch AVCaptureDevice.authorizationStatus(for: AVMediaType.video) {
+			case AVAuthorizationStatus.notDetermined:
 				NSLog("PluginGetUserMedia#call() | video authorization: not determined")
-			case AVAuthorizationStatus.Authorized:
+			case AVAuthorizationStatus.authorized:
 				NSLog("PluginGetUserMedia#call() | video authorization: authorized")
-			case AVAuthorizationStatus.Denied:
+			case AVAuthorizationStatus.denied:
 				NSLog("PluginGetUserMedia#call() | video authorization: denied")
-				errback(error: "video denied")
+				errback("video denied")
 				return
-			case AVAuthorizationStatus.Restricted:
+			case AVAuthorizationStatus.restricted:
 				NSLog("PluginGetUserMedia#call() | video authorization: restricted")
-				errback(error: "video restricted")
+				errback("video restricted")
 				return
 			}
 		}
 
 		if audioRequested == true {
-			switch AVCaptureDevice.authorizationStatusForMediaType(AVMediaTypeAudio) {
-			case AVAuthorizationStatus.NotDetermined:
+			switch AVCaptureDevice.authorizationStatus(for: AVMediaType.audio) {
+			case AVAuthorizationStatus.notDetermined:
 				NSLog("PluginGetUserMedia#call() | audio authorization: not determined")
-			case AVAuthorizationStatus.Authorized:
+			case AVAuthorizationStatus.authorized:
 				NSLog("PluginGetUserMedia#call() | audio authorization: authorized")
-			case AVAuthorizationStatus.Denied:
+			case AVAuthorizationStatus.denied:
 				NSLog("PluginGetUserMedia#call() | audio authorization: denied")
-				errback(error: "audio denied")
+				errback("audio denied")
 				return
-			case AVAuthorizationStatus.Restricted:
+			case AVAuthorizationStatus.restricted:
 				NSLog("PluginGetUserMedia#call() | audio authorization: restricted")
-				errback(error: "audio restricted")
+				errback("audio restricted")
 				return
 			}
 		}
 
-		rtcMediaStream = self.rtcPeerConnectionFactory.mediaStreamWithLabel(NSUUID().UUIDString)
+		rtcMediaStream = self.rtcPeerConnectionFactory.mediaStream(withStreamId: UUID().uuidString)
 
-		if videoRequested == true {
-			// No specific video device requested.
-			if videoDeviceId == nil {
-				NSLog("PluginGetUserMedia#call() | video requested (device not specified)")
-
-				for device: AVCaptureDevice in (AVCaptureDevice.devicesWithMediaType(AVMediaTypeVideo) as! Array<AVCaptureDevice>) {
-					if device.position == AVCaptureDevicePosition.Front {
-						videoDevice = device
-						break
-					}
-				}
-			}
+		if videoRequested {
 
-			// Video device specified.
-			else {
-				NSLog("PluginGetUserMedia#call() | video requested (specified device id: '%@')", String(videoDeviceId!))
+			NSLog("PluginGetUserMedia#call() | video requested")
 
-				for device: AVCaptureDevice in (AVCaptureDevice.devicesWithMediaType(AVMediaTypeVideo) as! Array<AVCaptureDevice>) {
-					if device.uniqueID == videoDeviceId {
-						videoDevice = device
-						break
-					}
-				}
-			}
+			rtcVideoSource = self.rtcPeerConnectionFactory.videoSource()
 
-			if videoDevice == nil {
-				NSLog("PluginGetUserMedia#call() | video requested but no suitable device found")
+			rtcVideoTrack = self.rtcPeerConnectionFactory.videoTrack(with: rtcVideoSource!, trackId: UUID().uuidString)
 
-				errback(error: "no suitable camera device found")
-				return
+			// Handle legacy plugin instance or video: true
+			var videoConstraints : NSDictionary = [:];
+			if (!(constraints.object(forKey: "video") is Bool)) {
+			   videoConstraints = constraints.object(forKey: "video") as! NSDictionary
 			}
 
-			NSLog("PluginGetUserMedia#call() | chosen video device: %@", String(videoDevice!))
+			NSLog("PluginGetUserMedia#call() | chosen video constraints: %@", videoConstraints)
 
-			rtcVideoCapturer = RTCVideoCapturer(deviceName: videoDevice!.localizedName)
+// Ignore Simulator cause does not support Camera
+#if !targetEnvironment(simulator)
+			let videoCapturer: RTCCameraVideoCapturer = RTCCameraVideoCapturer(delegate: rtcVideoSource!)
+			let videoCaptureController: PluginRTCVideoCaptureController = PluginRTCVideoCaptureController(capturer: videoCapturer)
+			rtcVideoTrack!.videoCaptureController = videoCaptureController
 
-			if videoMinWidth > 0 {
-				NSLog("PluginGetUserMedia#call() | adding media constraint [minWidth:%@]", String(videoMinWidth))
-				mandatoryConstraints.append(RTCPair(key: "minWidth", value: String(videoMinWidth)))
-			}
-			if videoMaxWidth > 0 {
-				NSLog("PluginGetUserMedia#call() | adding media constraint [maxWidth:%@]", String(videoMaxWidth))
-				mandatoryConstraints.append(RTCPair(key: "maxWidth", value: String(videoMaxWidth)))
-			}
-			if videoMinHeight > 0 {
-				NSLog("PluginGetUserMedia#call() | adding media constraint [minHeight:%@]", String(videoMinHeight))
-				mandatoryConstraints.append(RTCPair(key: "minHeight", value: String(videoMinHeight)))
-			}
-			if videoMaxHeight > 0 {
-				NSLog("PluginGetUserMedia#call() | adding media constraint [maxHeight:%@]", String(videoMaxHeight))
-				mandatoryConstraints.append(RTCPair(key: "maxHeight", value: String(videoMaxHeight)))
-			}
-			if videoMinFrameRate > 0 {
-				NSLog("PluginGetUserMedia#call() | adding media constraint [videoMinFrameRate:%@]", String(videoMinFrameRate))
-				mandatoryConstraints.append(RTCPair(key: "minFrameRate", value: String(videoMinFrameRate)))
-			}
-			if videoMaxFrameRate > 0 {
-				NSLog("PluginGetUserMedia#call() | adding media constraint [videoMaxFrameRate:%@]", String(videoMaxFrameRate))
-				mandatoryConstraints.append(RTCPair(key: "maxFrameRate", value: String(videoMaxFrameRate)))
+			let constraintsSatisfied = videoCaptureController.setConstraints(constraints: videoConstraints)
+			if (!constraintsSatisfied) {
+				errback("constraints not satisfied")
+				return
 			}
 
-			constraints = RTCMediaConstraints(
-				mandatoryConstraints: mandatoryConstraints,
-				optionalConstraints: []
-			)
-
-			rtcVideoSource = self.rtcPeerConnectionFactory.videoSourceWithCapturer(rtcVideoCapturer,
-				constraints: constraints
-			)
+			let captureStarted = videoCaptureController.startCapture()
+			if (!captureStarted) {
+				errback("constraints failed")
+				return
+			}
+#endif
 
 			// If videoSource state is "ended" it means that constraints were not satisfied so
 			// invoke the given errback.
-			if (rtcVideoSource!.state == RTCSourceStateEnded) {
+			if (rtcVideoSource!.state == RTCSourceState.ended) {
 				NSLog("PluginGetUserMedia() | rtcVideoSource.state is 'ended', constraints not satisfied")
 
-				errback(error: "constraints not satisfied")
+				errback("constraints not satisfied")
 				return
 			}
 
-			rtcVideoTrack = self.rtcPeerConnectionFactory.videoTrackWithID(NSUUID().UUIDString,
-				source: rtcVideoSource
-			)
+			if let device = rtcVideoTrack!.videoCaptureController?.device {
+				rtcVideoTrack!.capabilities["deviceId"] = device.uniqueID
+			}
 
-			rtcMediaStream.addVideoTrack(rtcVideoTrack)
+			rtcMediaStream.addVideoTrack(rtcVideoTrack!)
 		}
 
 		if audioRequested == true {
+
 			NSLog("PluginGetUserMedia#call() | audio requested")
 
-			rtcAudioTrack = self.rtcPeerConnectionFactory.audioTrackWithID(NSUUID().UUIDString)
+			// Handle legacy plugin instance or audio: true
+			var audioConstraints : NSDictionary = [:];
+			if (!(constraints.object(forKey: "audio") is Bool)) {
+			   audioConstraints = constraints.object(forKey: "audio") as! NSDictionary
+			}
+
+			NSLog("PluginGetUserMedia#call() | chosen audio constraints: %@", audioConstraints)
 
+
+			var audioDeviceId = audioConstraints.object(forKey: "deviceId") as? String
+			if(audioDeviceId == nil && audioConstraints.object(forKey: "deviceId") != nil){
+				let audioId = audioConstraints.object(forKey: "deviceId") as! NSDictionary
+				audioDeviceId = audioId.object(forKey: "exact") as? String
+				if(audioDeviceId == nil){
+					audioDeviceId = audioId.object(forKey: "ideal") as? String
+				}
+			}
+
+			rtcAudioTrack = self.rtcPeerConnectionFactory.audioTrack(withTrackId: UUID().uuidString)
 			rtcMediaStream.addAudioTrack(rtcAudioTrack!)
+
+			if (audioDeviceId != nil) {
+				PluginRTCAudioController.saveInputAudioDevice(inputDeviceUID: audioDeviceId!)
+			}
 		}
 
 		pluginMediaStream = PluginMediaStream(rtcMediaStream: rtcMediaStream)
 		pluginMediaStream!.run()
 
 		// Let the plugin store it in its dictionary.
-		eventListenerForNewStream(pluginMediaStream: pluginMediaStream!)
+		eventListenerForNewStream(pluginMediaStream!)
 
-		callback(data: [
+		callback([
 			"stream": pluginMediaStream!.getJSON()
 		])
 	}
diff --git a/src/PluginMediaStream.swift b/src/PluginMediaStream.swift
index 50b14e11..93764962 100644
--- a/src/PluginMediaStream.swift
+++ b/src/PluginMediaStream.swift
@@ -1,34 +1,43 @@
 import Foundation
 
+class PluginMediaStream : NSObject {
 
-class PluginMediaStream : NSObject, RTCMediaStreamDelegate {
 	var rtcMediaStream: RTCMediaStream
 	var id: String
 	var audioTracks: [String : PluginMediaStreamTrack] = [:]
 	var videoTracks: [String : PluginMediaStreamTrack] = [:]
-	var eventListener: ((data: NSDictionary) -> Void)?
-	var eventListenerForAddTrack: ((pluginMediaStreamTrack: PluginMediaStreamTrack) -> Void)?
-	var eventListenerForRemoveTrack: ((id: String) -> Void)?
-
+	var eventListener: ((_ data: NSDictionary) -> Void)?
+	var eventListenerForAddTrack: ((_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Void)?
+	var eventListenerForRemoveTrack: ((_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Void)?
 
 	/**
 	 * Constructor for pc.onaddstream event and getUserMedia().
 	 */
-	init(rtcMediaStream: RTCMediaStream) {
+	init(rtcMediaStream: RTCMediaStream, streamId: String? = nil) {
 		NSLog("PluginMediaStream#init()")
 
-		self.rtcMediaStream = rtcMediaStream
-		// ObjC API does not provide id property, so let's set a random one.
-		self.id = rtcMediaStream.label + "-" + NSUUID().UUIDString
+		self.rtcMediaStream = rtcMediaStream;
+
+		if (streamId == nil) {
+			// Handle possible duplicate remote trackId with  janus or short duplicate name
+			// See: https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/432
+			if (rtcMediaStream.streamId.count < 36) {
+				self.id = rtcMediaStream.streamId + "_" + UUID().uuidString;
+			} else {
+				self.id = rtcMediaStream.streamId;
+			}
+		} else {
+			self.id = streamId!;
+		}
 
-		for track: RTCMediaStreamTrack in (self.rtcMediaStream.audioTracks as! Array<RTCMediaStreamTrack>) {
+		for track: RTCMediaStreamTrack in (self.rtcMediaStream.audioTracks as Array<RTCMediaStreamTrack>) {
 			let pluginMediaStreamTrack = PluginMediaStreamTrack(rtcMediaStreamTrack: track)
 
 			pluginMediaStreamTrack.run()
 			self.audioTracks[pluginMediaStreamTrack.id] = pluginMediaStreamTrack
 		}
 
-		for track: RTCMediaStreamTrack in (self.rtcMediaStream.videoTracks as! Array<RTCMediaStreamTrack>) {
+		for track: RTCMediaStreamTrack in (self.rtcMediaStream.videoTracks as Array<RTCMediaStreamTrack>) {
 			let pluginMediaStreamTrack = PluginMediaStreamTrack(rtcMediaStreamTrack: track)
 
 			pluginMediaStreamTrack.run()
@@ -36,18 +45,29 @@ class PluginMediaStream : NSObject, RTCMediaStreamDelegate {
 		}
 	}
 
-
 	deinit {
 		NSLog("PluginMediaStream#deinit()")
+		stop();
 	}
 
-
 	func run() {
 		NSLog("PluginMediaStream#run()")
-
-		self.rtcMediaStream.delegate = self
 	}
 
+	func stop() {
+		NSLog("PluginMediaStream#stop()")
+
+		for (_, track) in audioTracks {
+			if(self.eventListenerForRemoveTrack != nil) {
+				self.eventListenerForRemoveTrack!(track)
+			}
+		}
+		for (_, track) in videoTracks {
+			if(self.eventListenerForRemoveTrack != nil) {
+				self.eventListenerForRemoveTrack!(track)
+			}
+		}
+	}
 
 	func getJSON() -> NSDictionary {
 		let json: NSMutableDictionary = [
@@ -67,11 +87,10 @@ class PluginMediaStream : NSObject, RTCMediaStreamDelegate {
 		return json as NSDictionary
 	}
 
-
 	func setListener(
-		eventListener: (data: NSDictionary) -> Void,
-		eventListenerForAddTrack: ((pluginMediaStreamTrack: PluginMediaStreamTrack) -> Void)?,
-		eventListenerForRemoveTrack: ((id: String) -> Void)?
+		_ eventListener: @escaping (_ data: NSDictionary) -> Void,
+		eventListenerForAddTrack: ((_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Void)?,
+		eventListenerForRemoveTrack: ((_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Void)?
 	) {
 		NSLog("PluginMediaStream#setListener()")
 
@@ -80,146 +99,89 @@ class PluginMediaStream : NSObject, RTCMediaStreamDelegate {
 		self.eventListenerForRemoveTrack = eventListenerForRemoveTrack
 	}
 
+	func hasTrack(_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Bool {
+		if pluginMediaStreamTrack.kind == "audio" {
+			return self.audioTracks[pluginMediaStreamTrack.id] != nil;
+		} else if pluginMediaStreamTrack.kind == "video" {
+			return self.videoTracks[pluginMediaStreamTrack.id] != nil;
+		} else {
+			return false
+		}
+	}
 
-	func addTrack(pluginMediaStreamTrack: PluginMediaStreamTrack) -> Bool {
+	func addTrack(_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Bool {
 		NSLog("PluginMediaStream#addTrack()")
 
 		if pluginMediaStreamTrack.kind == "audio" {
-			if self.rtcMediaStream.addAudioTrack(pluginMediaStreamTrack.rtcMediaStreamTrack as! RTCAudioTrack) {
-				NSLog("PluginMediaStream#addTrack() | audio track added")
-				self.audioTracks[pluginMediaStreamTrack.id] = pluginMediaStreamTrack
-				return true
-			} else {
-				NSLog("PluginMediaStream#addTrack() | ERROR: audio track not added")
-				return false
-			}
+			self.rtcMediaStream.addAudioTrack(pluginMediaStreamTrack.rtcMediaStreamTrack as! RTCAudioTrack)
+			NSLog("PluginMediaStream#addTrack() | audio track added")
+			self.audioTracks[pluginMediaStreamTrack.id] = pluginMediaStreamTrack
 		} else if pluginMediaStreamTrack.kind == "video" {
-			if self.rtcMediaStream.addVideoTrack(pluginMediaStreamTrack.rtcMediaStreamTrack as! RTCVideoTrack) {
-				NSLog("PluginMediaStream#addTrack() | video track added")
-				self.videoTracks[pluginMediaStreamTrack.id] = pluginMediaStreamTrack
-				return true
-			} else {
-				NSLog("PluginMediaStream#addTrack() | ERROR: video track not added")
-				return false
-			}
+			self.rtcMediaStream.addVideoTrack(pluginMediaStreamTrack.rtcMediaStreamTrack as! RTCVideoTrack)
+			NSLog("PluginMediaStream#addTrack() | video track added")
+			self.videoTracks[pluginMediaStreamTrack.id] = pluginMediaStreamTrack
+		} else {
+			return false
 		}
 
-		return false
+		onAddTrack(pluginMediaStreamTrack)
+		return true
 	}
 
-
-	func removeTrack(pluginMediaStreamTrack: PluginMediaStreamTrack) -> Bool {
+	func removeTrack(_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Bool {
 		NSLog("PluginMediaStream#removeTrack()")
 
 		if pluginMediaStreamTrack.kind == "audio" {
 			self.audioTracks[pluginMediaStreamTrack.id] = nil
-			if self.rtcMediaStream.removeAudioTrack(pluginMediaStreamTrack.rtcMediaStreamTrack as! RTCAudioTrack) {
-				NSLog("PluginMediaStream#removeTrack() | audio track removed")
-				return true
-			} else {
-				NSLog("PluginMediaStream#removeTrack() | ERROR: audio track not removed")
-				return false
-			}
+			self.rtcMediaStream.removeAudioTrack(pluginMediaStreamTrack.rtcMediaStreamTrack as! RTCAudioTrack)
+			NSLog("PluginMediaStream#removeTrack() | audio track removed")
 		} else if pluginMediaStreamTrack.kind == "video" {
 			self.videoTracks[pluginMediaStreamTrack.id] = nil
-			if self.rtcMediaStream.removeVideoTrack(pluginMediaStreamTrack.rtcMediaStreamTrack as! RTCVideoTrack) {
-				NSLog("PluginMediaStream#removeTrack() | video track removed")
-				return true
-			} else {
-				NSLog("PluginMediaStream#removeTrack() | ERROR: video track not removed")
-				return false
-			}
+			self.rtcMediaStream.removeVideoTrack(pluginMediaStreamTrack.rtcMediaStreamTrack as! RTCVideoTrack)
+			NSLog("PluginMediaStream#removeTrack() | video track removed")
+		} else {
+			return false
 		}
 
-		return false
+		onRemoveTrack(pluginMediaStreamTrack)
+		return true
 	}
 
+	func onAddTrack(_ track: PluginMediaStreamTrack) {
+		NSLog("PluginMediaStream | OnAddTrack [label:%@]", String(track.id))
 
-	/**
-	 * Methods inherited from RTCMediaStreamDelegate.
-	 */
-
-
-	func OnAddAudioTrack(rtcMediaStream: RTCMediaStream!, track: RTCMediaStreamTrack!) {
-		NSLog("PluginMediaStream | OnAddAudioTrack [label:%@]", String(track.label))
-
-		let pluginMediaStreamTrack = PluginMediaStreamTrack(rtcMediaStreamTrack: track)
-
-		pluginMediaStreamTrack.run()
-		self.audioTracks[pluginMediaStreamTrack.id] = pluginMediaStreamTrack
-
-		if self.eventListener != nil {
-			self.eventListenerForAddTrack!(pluginMediaStreamTrack: pluginMediaStreamTrack)
-
-			self.eventListener!(data: [
-				"type": "addtrack",
-				"track": pluginMediaStreamTrack.getJSON()
-			])
-		}
-	}
-
-
-	func OnAddVideoTrack(rtcMediaStream: RTCMediaStream!, track: RTCMediaStreamTrack!) {
-		NSLog("PluginMediaStream | OnAddVideoTrack [label:%@]", String(track.label))
-
-		let pluginMediaStreamTrack = PluginMediaStreamTrack(rtcMediaStreamTrack: track)
-
-		pluginMediaStreamTrack.run()
-		self.videoTracks[pluginMediaStreamTrack.id] = pluginMediaStreamTrack
+		track.run()
 
 		if self.eventListener != nil {
-			self.eventListenerForAddTrack!(pluginMediaStreamTrack: pluginMediaStreamTrack)
+			self.eventListenerForAddTrack!(track)
 
-			self.eventListener!(data: [
+			self.eventListener!([
 				"type": "addtrack",
-				"track": pluginMediaStreamTrack.getJSON()
-			])
-		}
-	}
-
-
-	func OnRemoveAudioTrack(rtcMediaStream: RTCMediaStream!, track: RTCMediaStreamTrack!) {
-		NSLog("PluginMediaStream | OnRemoveAudioTrack [label:%@]", String(track.label))
-
-		// It may happen that track was removed due to user action (removeTrack()).
-		if self.audioTracks[track.label] == nil {
-			return
-		}
-
-		self.audioTracks[track.label] = nil
-
-		if self.eventListener != nil {
-			self.eventListenerForRemoveTrack!(id: track.label)
-
-			self.eventListener!(data: [
-				"type": "removetrack",
-				"track": [
-					"id": track.label,
-					"kind": "audio"
-				]
+				"track": track.getJSON()
 			])
 		}
 	}
 
-
-	func OnRemoveVideoTrack(rtcMediaStream: RTCMediaStream!, track: RTCMediaStreamTrack!) {
-		NSLog("PluginMediaStream | OnRemoveVideoTrack [label:%@]", String(track.label))
+	func onRemoveTrack(_ track: PluginMediaStreamTrack) {
+		NSLog("PluginMediaStream | OnRemoveTrack [label:%@]", String(track.id))
 
 		// It may happen that track was removed due to user action (removeTrack()).
-		if self.videoTracks[track.label] == nil {
+		if self.audioTracks[track.id] != nil {
+			self.audioTracks[track.id] = nil
+		} else if self.videoTracks[track.id] != nil {
+			self.videoTracks[track.id] = nil
+		} else {
 			return
 		}
 
-		self.videoTracks[track.label] = nil
-
 		if self.eventListener != nil {
-			self.eventListenerForRemoveTrack!(id: track.label)
+			self.eventListenerForRemoveTrack!(track)
 
-			self.eventListener!(data: [
+			self.eventListener!([
 				"type": "removetrack",
 				"track": [
-					"id": track.label,
-					"kind": "video"
+					"id": track.id,
+					"kind": track.kind
 				]
 			])
 		}
diff --git a/src/PluginMediaStreamRenderer.swift b/src/PluginMediaStreamRenderer.swift
index 79dc7a08..de227637 100644
--- a/src/PluginMediaStreamRenderer.swift
+++ b/src/PluginMediaStreamRenderer.swift
@@ -1,58 +1,90 @@
 import Foundation
 import AVFoundation
 
-
 class PluginMediaStreamRenderer : NSObject, RTCEAGLVideoViewDelegate {
+
+	var id: String
+	var eventListener: (_ data: NSDictionary) -> Void
+	var closed: Bool
+
 	var webView: UIView
-	var eventListener: (data: NSDictionary) -> Void
 	var elementView: UIView
-	var videoView: RTCEAGLVideoView
 	var pluginMediaStream: PluginMediaStream?
+
+	var videoView: RTCEAGLVideoView
 	var rtcAudioTrack: RTCAudioTrack?
 	var rtcVideoTrack: RTCVideoTrack?
-
+    var pluginVideoTrack: PluginMediaStreamTrack?
 
 	init(
 		webView: UIView,
-		eventListener: (data: NSDictionary) -> Void
+		eventListener: @escaping (_ data: NSDictionary) -> Void
 	) {
 		NSLog("PluginMediaStreamRenderer#init()")
 
+		// Open Renderer
+		self.id = UUID().uuidString;
+		self.closed = false
+
 		// The browser HTML view.
 		self.webView = webView
 		self.eventListener = eventListener
+
+		let useManualLayoutRenderer = Bundle.main.object(forInfoDictionaryKey: "UseManualLayoutRenderer") as? Bool ?? false
+
 		// The video element view.
-		self.elementView = UIView()
+		let guide = webView.safeAreaLayoutGuide;
+		self.elementView = useManualLayoutRenderer
+			? UIView(frame: CGRect(x: 0.0, y: guide.layoutFrame.minY, width: guide.layoutFrame.width, height: guide.layoutFrame.height))
+			: UIView()
+
 		// The effective video view in which the the video stream is shown.
 		// It's placed over the elementView.
 		self.videoView = RTCEAGLVideoView()
+		self.videoView.isUserInteractionEnabled = false
 
-		self.elementView.userInteractionEnabled = false
-		self.elementView.hidden = true
-		self.elementView.backgroundColor = UIColor.blackColor()
+		self.elementView.isUserInteractionEnabled = false
+		self.elementView.isHidden = true
+		self.elementView.backgroundColor = UIColor.black
 		self.elementView.addSubview(self.videoView)
 		self.elementView.layer.masksToBounds = true
-
-		self.videoView.userInteractionEnabled = false
+		self.elementView.translatesAutoresizingMaskIntoConstraints = false
 
 		// Place the video element view inside the WebView's superview
-		self.webView.superview?.addSubview(self.elementView)
+		self.webView.addSubview(self.elementView)
+		self.webView.isOpaque = false
+		self.webView.backgroundColor = UIColor.clear
+
+		// https://stackoverflow.com/questions/46317061/use-safe-area-layout-programmatically
+		// https://developer.apple.com/documentation/uikit/uiview/2891102-safearealayoutguide
+		// https://developer.apple.com/documentation/uikit/
+		let view = self.elementView;
+		if !useManualLayoutRenderer {
+			if #available(iOS 11.0, *) {
+				view.topAnchor.constraint(equalTo: guide.topAnchor).isActive = true
+				view.bottomAnchor.constraint(equalTo: guide.bottomAnchor).isActive = true
+				view.leftAnchor.constraint(equalTo: guide.leftAnchor).isActive = true
+				view.rightAnchor.constraint(equalTo: guide.rightAnchor).isActive = true
+			} else {
+				NSLayoutConstraint(item: view, attribute: .top, relatedBy: .equal, toItem: webView, attribute: .top, multiplier: 1.0, constant: 0).isActive = true
+				NSLayoutConstraint(item: view, attribute: .bottom, relatedBy: .equal, toItem: webView, attribute: .bottom, multiplier: 1.0, constant: 0).isActive = true
+				NSLayoutConstraint(item: view, attribute: .leading, relatedBy: .equal, toItem: webView, attribute: .leading, multiplier: 1.0, constant: 0).isActive = true
+				NSLayoutConstraint(item: view, attribute: .trailing, relatedBy: .equal, toItem: webView, attribute: .trailing, multiplier: 1.0, constant: 0).isActive = true
+			}
+		}
 	}
 
-
 	deinit {
 		NSLog("PluginMediaStreamRenderer#deinit()")
 	}
 
-
 	func run() {
 		NSLog("PluginMediaStreamRenderer#run()")
 
 		self.videoView.delegate = self
 	}
 
-
-	func render(pluginMediaStream: PluginMediaStream) {
+	func render(_ pluginMediaStream: PluginMediaStream) {
 		NSLog("PluginMediaStreamRenderer#render()")
 
 		if self.pluginMediaStream != nil {
@@ -69,16 +101,18 @@ class PluginMediaStreamRenderer : NSObject, RTCEAGLVideoViewDelegate {
 
 		// Take the first video track.
 		for (_, track) in pluginMediaStream.videoTracks {
+            self.pluginVideoTrack = track
 			self.rtcVideoTrack = track.rtcMediaStreamTrack as? RTCVideoTrack
 			break
 		}
 
+        
 		if self.rtcVideoTrack != nil {
-			self.rtcVideoTrack!.addRenderer(self.videoView)
+			self.rtcVideoTrack!.add(self.videoView)
+            self.pluginVideoTrack?.registerRender(render: self)
 		}
 	}
 
-
 	func mediaStreamChanged() {
 		NSLog("PluginMediaStreamRenderer#mediaStreamChanged()")
 
@@ -86,11 +120,13 @@ class PluginMediaStreamRenderer : NSObject, RTCEAGLVideoViewDelegate {
 			return
 		}
 
+        let oldPluginVideoTrack: PluginMediaStreamTrack? = self.pluginVideoTrack
 		let oldRtcVideoTrack: RTCVideoTrack? = self.rtcVideoTrack
-
+        
 		self.rtcAudioTrack = nil
 		self.rtcVideoTrack = nil
-
+        self.pluginVideoTrack = nil
+        
 		// Take the first audio track.
 		for (_, track) in self.pluginMediaStream!.audioTracks {
 			self.rtcAudioTrack = track.rtcMediaStreamTrack as? RTCAudioTrack
@@ -99,104 +135,112 @@ class PluginMediaStreamRenderer : NSObject, RTCEAGLVideoViewDelegate {
 
 		// Take the first video track.
 		for (_, track) in pluginMediaStream!.videoTracks {
+            self.pluginVideoTrack = track
 			self.rtcVideoTrack = track.rtcMediaStreamTrack as? RTCVideoTrack
 			break
 		}
 
 		// If same video track as before do nothing.
 		if oldRtcVideoTrack != nil && self.rtcVideoTrack != nil &&
-			oldRtcVideoTrack!.label == self.rtcVideoTrack!.label {
+			oldRtcVideoTrack!.trackId == self.rtcVideoTrack!.trackId {
 			NSLog("PluginMediaStreamRenderer#mediaStreamChanged() | same video track as before")
 		}
 
 		// Different video track.
 		else if oldRtcVideoTrack != nil && self.rtcVideoTrack != nil &&
-			oldRtcVideoTrack!.label != self.rtcVideoTrack!.label {
+			oldRtcVideoTrack!.trackId != self.rtcVideoTrack!.trackId {
 			NSLog("PluginMediaStreamRenderer#mediaStreamChanged() | has a new video track")
-
-			oldRtcVideoTrack!.removeRenderer(self.videoView)
-			self.rtcVideoTrack!.addRenderer(self.videoView)
+            oldPluginVideoTrack?.unregisterRender(render: self)
+			oldRtcVideoTrack!.remove(self.videoView)
+            self.pluginVideoTrack?.registerRender(render: self)
+			self.rtcVideoTrack!.add(self.videoView)
 		}
 
 		// Did not have video but now it has.
 		else if oldRtcVideoTrack == nil && self.rtcVideoTrack != nil {
 			NSLog("PluginMediaStreamRenderer#mediaStreamChanged() | video track added")
-
-			self.rtcVideoTrack!.addRenderer(self.videoView)
+            if oldPluginVideoTrack != nil{
+                oldPluginVideoTrack?.unregisterRender(render: self)
+            }
+            self.pluginVideoTrack?.registerRender(render: self)
+			self.rtcVideoTrack!.add(self.videoView)
 		}
 
 		// Had video but now it has not.
 		else if oldRtcVideoTrack != nil && self.rtcVideoTrack == nil {
 			NSLog("PluginMediaStreamRenderer#mediaStreamChanged() | video track removed")
-
-			oldRtcVideoTrack!.removeRenderer(self.videoView)
+            oldPluginVideoTrack?.unregisterRender(render: self)
+			oldRtcVideoTrack!.remove(self.videoView)
 		}
 	}
 
-
-	func refresh(data: NSDictionary) {
-		let elementLeft = data.objectForKey("elementLeft") as? Float ?? 0
-		let elementTop = data.objectForKey("elementTop") as? Float ?? 0
-		let elementWidth = data.objectForKey("elementWidth") as? Float ?? 0
-		let elementHeight = data.objectForKey("elementHeight") as? Float ?? 0
-		var videoViewWidth = data.objectForKey("videoViewWidth") as? Float ?? 0
-		var videoViewHeight = data.objectForKey("videoViewHeight") as? Float ?? 0
-		let visible = data.objectForKey("visible") as? Bool ?? true
-		let opacity = data.objectForKey("opacity") as? Float ?? 1
-		let zIndex = data.objectForKey("zIndex") as? Float ?? 0
-		let mirrored = data.objectForKey("mirrored") as? Bool ?? false
-		let clip = data.objectForKey("clip") as? Bool ?? true
-		let borderRadius = data.objectForKey("borderRadius") as? Float ?? 0
+	func refresh(_ data: NSDictionary) {
+
+		let elementLeft = data.object(forKey: "elementLeft") as? Double ?? 0
+		let elementTop = data.object(forKey: "elementTop") as? Double ?? 0
+		let elementWidth = data.object(forKey: "elementWidth") as? Double ?? 0
+		let elementHeight = data.object(forKey: "elementHeight") as? Double ?? 0
+		var videoViewWidth = data.object(forKey: "videoViewWidth") as? Double ?? 0
+		var videoViewHeight = data.object(forKey: "videoViewHeight") as? Double ?? 0
+		let visible = data.object(forKey: "visible") as? Bool ?? true
+		let opacity = data.object(forKey: "opacity") as? Double ?? 1
+		let zIndex = data.object(forKey: "zIndex") as? Double ?? 0
+		let mirrored = data.object(forKey: "mirrored") as? Bool ?? false
+		let clip = data.object(forKey: "clip") as? Bool ?? true
+		let borderRadius = data.object(forKey: "borderRadius") as? Double ?? 0
+		let backgroundColor = data.object(forKey: "backgroundColor") as? String ?? "0,0,0"
 
 		NSLog("PluginMediaStreamRenderer#refresh() [elementLeft:%@, elementTop:%@, elementWidth:%@, elementHeight:%@, videoViewWidth:%@, videoViewHeight:%@, visible:%@, opacity:%@, zIndex:%@, mirrored:%@, clip:%@, borderRadius:%@]",
 			String(elementLeft), String(elementTop), String(elementWidth), String(elementHeight),
 			String(videoViewWidth), String(videoViewHeight), String(visible), String(opacity), String(zIndex),
 			String(mirrored), String(clip), String(borderRadius))
 
-		let videoViewLeft: Float = (elementWidth - videoViewWidth) / 2
-		let videoViewTop: Float = (elementHeight - videoViewHeight) / 2
+		let videoViewLeft: Double = (elementWidth - videoViewWidth) / 2
+		let videoViewTop: Double = (elementHeight - videoViewHeight) / 2
 
-		self.elementView.frame = CGRectMake(
-			CGFloat(elementLeft),
-			CGFloat(elementTop),
-			CGFloat(elementWidth),
-			CGFloat(elementHeight)
+		// use SAI to adjust the top/left corner from the client
+		self.elementView.frame = CGRect(
+			x: CGFloat(elementLeft + self.webView.safeAreaInsets.left),
+			y: CGFloat(elementTop + self.webView.safeAreaInsets.top),
+			width: CGFloat(elementWidth),
+			height: CGFloat(elementHeight)
 		)
 
 		// NOTE: Avoid a zero-size UIView for the video (the library complains).
 		if videoViewWidth == 0 || videoViewHeight == 0 {
 			videoViewWidth = 1
 			videoViewHeight = 1
-			self.videoView.hidden = true
+			self.videoView.isHidden = true
 		} else {
-			self.videoView.hidden = false
+			self.videoView.isHidden = false
 		}
 
-		self.videoView.frame = CGRectMake(
-			CGFloat(videoViewLeft),
-			CGFloat(videoViewTop),
-			CGFloat(videoViewWidth),
-			CGFloat(videoViewHeight)
+		self.videoView.frame = CGRect(
+			x: CGFloat(videoViewLeft),
+			y: CGFloat(videoViewTop),
+			width: CGFloat(videoViewWidth),
+			height: CGFloat(videoViewHeight)
 		)
 
 		if visible {
-			self.elementView.hidden = false
+			self.elementView.isHidden = false
 		} else {
-			self.elementView.hidden = true
+			self.elementView.isHidden = true
 		}
 
 		self.elementView.alpha = CGFloat(opacity)
 		self.elementView.layer.zPosition = CGFloat(zIndex)
 
-                // if the zIndex is 0 (the default) bring the view to the top, last one wins
-                if zIndex == 0 {
-			self.webView.superview?.bringSubviewToFront(self.elementView)
-                }
+		// if the zIndex is 0 (the default) bring the view to the top, last one wins
+		if zIndex == 0 {
+			self.webView.bringSubviewToFront(self.elementView)
+			//self.webView?.bringSubview(toFront: self.elementView)
+		}
 
 		if !mirrored {
-			self.elementView.transform = CGAffineTransformIdentity
+			self.elementView.transform = CGAffineTransform.identity
 		} else {
-			self.elementView.transform = CGAffineTransformMakeScale(-1.0, 1.0)
+			self.elementView.transform = CGAffineTransform(scaleX: -1.0, y: 1.0)
 		}
 
 		if clip {
@@ -206,45 +250,68 @@ class PluginMediaStreamRenderer : NSObject, RTCEAGLVideoViewDelegate {
 		}
 
 		self.elementView.layer.cornerRadius = CGFloat(borderRadius)
+		let rgb = backgroundColor.components(separatedBy: ",").map{ CGFloat(($0 as NSString).floatValue) / 256.0 }
+		let color = UIColor(red: rgb[0], green: rgb[1], blue: rgb[2], alpha: 1)
+		self.elementView.backgroundColor = color
+	}
+
+	func save(callback: (_ data: String) -> Void,
+			  errback: (_ error: String) -> Void) {
+		//NSLog("PluginMediaStreamRenderer#save()")
+		UIGraphicsBeginImageContextWithOptions(videoView.bounds.size, videoView.isOpaque, 0.0)
+		videoView.drawHierarchy(in: videoView.bounds, afterScreenUpdates: false)
+		let snapshotImageFromMyView = UIGraphicsGetImageFromCurrentImageContext()
+		UIGraphicsEndImageContext()
+		let imageData = snapshotImageFromMyView?.jpegData(compressionQuality: 1.0)
+		let strBase64 = imageData?.base64EncodedString(options: .lineLength64Characters)
+		
+		callback(strBase64!);
 	}
 
+	func stop() {
+		NSLog("PluginMediaStreamRenderer | video stop")
+
+		self.eventListener([
+			"type": "videostop"
+		])
+	}
 
 	func close() {
 		NSLog("PluginMediaStreamRenderer#close()")
-
+		self.closed = true
 		self.reset()
 		self.elementView.removeFromSuperview()
 	}
 
-
 	/**
 	 * Private API.
 	 */
 
-
-	private func reset() {
+	fileprivate func reset() {
 		NSLog("PluginMediaStreamRenderer#reset()")
 
 		if self.rtcVideoTrack != nil {
-			self.rtcVideoTrack!.removeRenderer(self.videoView)
+			self.rtcVideoTrack!.remove(self.videoView)
 		}
-
+        if self.pluginVideoTrack != nil {
+            self.pluginVideoTrack?.unregisterRender(render: self)
+        }
+        self.pluginVideoTrack = nil
 		self.pluginMediaStream = nil
 		self.rtcAudioTrack = nil
 		self.rtcVideoTrack = nil
 	}
 
-
 	/**
 	 * Methods inherited from RTCEAGLVideoViewDelegate.
 	 */
 
+	func videoView(_ videoView: RTCVideoRenderer, didChangeVideoSize size: CGSize) {
 
-	func videoView(videoView: RTCEAGLVideoView!, didChangeVideoSize size: CGSize) {
 		NSLog("PluginMediaStreamRenderer | video size changed [width:%@, height:%@]",
-			String(size.width), String(size.height))
+			String(describing: size.width), String(describing: size.height))
 
-		self.eventListener(data: [
+		self.eventListener([
 			"type": "videoresize",
 			"size": [
 				"width": Int(size.width),
@@ -253,4 +320,43 @@ class PluginMediaStreamRenderer : NSObject, RTCEAGLVideoViewDelegate {
 		])
 	}
 
+	func videoView(_ videoView: RTCVideoRenderer, didChange frame: RTCVideoFrame?) {
+
+		// TODO save from frame buffer instead of renderer
+		/*
+		let i420: RTCI420BufferProtocol = frame!.buffer.toI420()
+		let YPtr: UnsafePointer<UInt8> = i420.dataY
+		let UPtr: UnsafePointer<UInt8> = i420.dataU
+		let VPtr: UnsafePointer<UInt8> = i420.dataV
+		let YSize: Int = Int(frame!.width * frame!.height)
+		let USize: Int = Int(YSize / 4)
+		let VSize: Int = Int(YSize / 4)
+		var frameSize:Int32 = Int32(YSize + USize + VSize)
+		var width: Int16 = Int16(frame!.width)
+		var height: Int16 = Int16(frame!.height)
+		var rotation: Int16 = Int16(frame!.rotation.rawValue)
+		var timestamp: Int32 = Int32(frame!.timeStamp)
+
+		// head + body
+		// head: type(2B)+len(4B)+width(2B)+height(2B)+rotation(2B)+timestamp(4B)
+		// body: data(len)
+		let headSize:Int32 = 16
+		let dataSize:Int32 = headSize + frameSize
+		let pduData: NSMutableData? = NSMutableData(length: Int(dataSize))
+
+		let headPtr = pduData!.mutableBytes
+		var pduType:UInt16 = 0x2401
+		memcpy(headPtr, &pduType, 2)
+		memcpy(headPtr+2, &frameSize, 4)
+		memcpy(headPtr+2+4, &width, 2)
+		memcpy(headPtr+2+4+2, &height, 2)
+		memcpy(headPtr+2+4+2+2, &rotation, 2)
+		//memcpy(headPtr+2+4+2+2+2, &timestamp, 4)
+
+		let bodyPtr = pduData!.mutableBytes + Int(headSize)
+		memcpy(bodyPtr, YPtr, YSize)
+		memcpy(bodyPtr + YSize, UPtr, USize);
+		memcpy(bodyPtr + YSize + USize, VPtr, VSize);
+		*/
+	}
 }
diff --git a/src/PluginMediaStreamTrack.swift b/src/PluginMediaStreamTrack.swift
index ba28346a..44d4f424 100644
--- a/src/PluginMediaStreamTrack.swift
+++ b/src/PluginMediaStreamTrack.swift
@@ -1,118 +1,149 @@
 import Foundation
 
 
-class PluginMediaStreamTrack : NSObject, RTCMediaStreamTrackDelegate {
+class PluginMediaStreamTrack : NSObject {
 	var rtcMediaStreamTrack: RTCMediaStreamTrack
+	var originalId: String
 	var id: String
 	var kind: String
-	var eventListener: ((data: NSDictionary) -> Void)?
+	var eventListener: ((_ data: NSDictionary) -> Void)?
 	var eventListenerForEnded: (() -> Void)?
 	var lostStates = Array<String>()
+	var renders: [String : PluginMediaStreamRenderer]
 
-
-	init(rtcMediaStreamTrack: RTCMediaStreamTrack) {
+	init(rtcMediaStreamTrack: RTCMediaStreamTrack, trackId: String? = nil) {
 		NSLog("PluginMediaStreamTrack#init()")
 
 		self.rtcMediaStreamTrack = rtcMediaStreamTrack
-		self.id = rtcMediaStreamTrack.label  // NOTE: No "id" property provided.
+
+		if (trackId == nil) {
+			self.originalId = rtcMediaStreamTrack.trackId;
+			// Handle possible duplicate remote trackId with  janus or short duplicate name
+			// See: https://github.com/cordova-rtc/cordova-plugin-iosrtc/issues/432
+			if (rtcMediaStreamTrack.trackId.count<36) {
+				self.id = rtcMediaStreamTrack.trackId + "_" + UUID().uuidString;
+			} else {
+				self.id = rtcMediaStreamTrack.trackId;
+			}
+		} else {
+			self.originalId = String();
+			self.id = trackId!;
+		}
+
 		self.kind = rtcMediaStreamTrack.kind
+		self.renders = [:]
 	}
 
-
 	deinit {
 		NSLog("PluginMediaStreamTrack#deinit()")
+		stop()
 	}
 
-
 	func run() {
 		NSLog("PluginMediaStreamTrack#run() [kind:%@, id:%@]", String(self.kind), String(self.id))
-
-		self.rtcMediaStreamTrack.delegate = self
 	}
 
+	func getReadyState() -> String {
+		switch self.rtcMediaStreamTrack.readyState  {
+		case RTCMediaStreamTrackState.live:
+			return "live"
+		case RTCMediaStreamTrackState.ended:
+			return "ended"
+		default:
+			return "ended"
+		}
+	}
 
-	func getJSON() -> NSDictionary {
+	func getJSON() -> [String: Any] {
 		return [
 			"id": self.id,
 			"kind": self.kind,
-			"label": self.rtcMediaStreamTrack.label,
-			"enabled": self.rtcMediaStreamTrack.isEnabled() ? true : false,
-			"readyState": PluginRTCTypes.mediaStreamTrackStates[self.rtcMediaStreamTrack.state().rawValue] as String!
+			"trackId": self.rtcMediaStreamTrack.trackId,
+			"enabled": self.rtcMediaStreamTrack.isEnabled ? true : false,
+			"capabilities": self.rtcMediaStreamTrack.capabilities,
+			"readyState": self.getReadyState()
 		]
 	}
 
-
 	func setListener(
-		eventListener: (data: NSDictionary) -> Void,
-		eventListenerForEnded: () -> Void
+		_ eventListener: @escaping (_ data: NSDictionary) -> Void,
+		eventListenerForEnded: @escaping () -> Void
 	) {
+		if(self.eventListener != nil){
+			NSLog("PluginMediaStreamTrack#setListener():Error Listener already Set [kind:%@, id:%@]", String(self.kind), String(self.id));
+			return;
+		}
+
 		NSLog("PluginMediaStreamTrack#setListener() [kind:%@, id:%@]", String(self.kind), String(self.id))
 
 		self.eventListener = eventListener
 		self.eventListenerForEnded = eventListenerForEnded
 
 		for readyState in self.lostStates {
-			self.eventListener!(data: [
+			self.eventListener!([
 				"type": "statechange",
 				"readyState": readyState,
-				"enabled": self.rtcMediaStreamTrack.isEnabled() ? true : false
+				"enabled": self.rtcMediaStreamTrack.isEnabled ? true : false
 			])
 
 			if readyState == "ended" {
-				self.eventListenerForEnded!()
+				if(self.eventListenerForEnded != nil) {
+					self.eventListenerForEnded!()
+				}
 			}
 		}
 		self.lostStates.removeAll()
 	}
 
-
-	func setEnabled(value: Bool) {
+	func setEnabled(_ value: Bool) {
 		NSLog("PluginMediaStreamTrack#setEnabled() [kind:%@, id:%@, value:%@]",
 			String(self.kind), String(self.id), String(value))
 
-		self.rtcMediaStreamTrack.setEnabled(value)
+		if (self.rtcMediaStreamTrack.isEnabled != value) {
+			self.rtcMediaStreamTrack.isEnabled = value
+			if (value) {
+				self.rtcMediaStreamTrack.videoCaptureController?.startCapture()
+			}else {
+				self.rtcMediaStreamTrack.videoCaptureController?.stopCapture()
+			}
+		}
 	}
 
-
-	// TODO: No way to stop the track.
-	// Check https://github.com/eface2face/cordova-plugin-iosrtc/issues/140
-	func stop() {
-		NSLog("PluginMediaStreamTrack#stop() [kind:%@, id:%@]", String(self.kind), String(self.id))
-
-		NSLog("PluginMediaStreamTrack#stop() | stop() not implemented (see: https://github.com/eface2face/cordova-plugin-iosrtc/issues/140")
-
-		// NOTE: There is no setState() anymore
-		// self.rtcMediaStreamTrack.setState(RTCTrackStateEnded)
-
-		// Let's try setEnabled(false), but it also fails.
-		self.rtcMediaStreamTrack.setEnabled(false)
+	func switchCamera() {
+		self.rtcMediaStreamTrack.videoCaptureController?.switchCamera()
 	}
 
+	func registerRender(render: PluginMediaStreamRenderer) {
+		if let exist = self.renders[render.id] {
+			_ = exist
+		} else {
+			self.renders[render.id] = render
+		}
+	}
 
-	/**
-	 * Methods inherited from RTCMediaStreamTrackDelegate.
-	 */
-
+	func unregisterRender(render: PluginMediaStreamRenderer) {
+		self.renders.removeValue(forKey: render.id);
+	}
 
-	func mediaStreamTrackDidChange(rtcMediaStreamTrack: RTCMediaStreamTrack!) {
-		let state_str = PluginRTCTypes.mediaStreamTrackStates[self.rtcMediaStreamTrack.state().rawValue] as String!
+	func stop() {
+		NSLog("PluginMediaStreamTrack#stop() [kind:%@, id:%@]", String(self.kind), String(self.id))
 
-		NSLog("PluginMediaStreamTrack | state changed [kind:%@, id:%@, state:%@, enabled:%@]",
-			String(self.kind), String(self.id), String(state_str), String(self.rtcMediaStreamTrack.isEnabled()))
+		self.rtcMediaStreamTrack.videoCaptureController?.stopCapture();
 
-		if self.eventListener != nil {
-			self.eventListener!(data: [
+		// Let's try setEnabled(false), but it also fails.
+		self.rtcMediaStreamTrack.isEnabled = false
+		// eventListener could be null if the track is never used
+		if(self.eventListener != nil){
+			self.eventListener!([
 				"type": "statechange",
-				"readyState": state_str,
-				"enabled": self.rtcMediaStreamTrack.isEnabled() ? true : false
+				"readyState": "ended",
+				"enabled": self.rtcMediaStreamTrack.isEnabled ? true : false
 			])
+		}
 
-			if self.rtcMediaStreamTrack.state().rawValue == RTCTrackStateEnded.rawValue {
-				self.eventListenerForEnded!()
-			}
-		} else {
-			// It may happen that the eventListener is not yet set, so store the lost states.
-			self.lostStates.append(state_str)
+		for (_, render) in self.renders {
+			render.stop()
 		}
+		self.renders.removeAll();
 	}
 }
diff --git a/src/PluginRTCAudioController.swift b/src/PluginRTCAudioController.swift
new file mode 100644
index 00000000..b5e700fb
--- /dev/null
+++ b/src/PluginRTCAudioController.swift
@@ -0,0 +1,205 @@
+//
+//  AudioOutputController.swift
+//  StreamingDemo
+//
+//  Created by ows on 6/11/17.
+//
+//
+
+import Foundation
+import WebRTC
+import AVFoundation
+
+class PluginRTCAudioController {
+
+	static private var audioCategory : AVAudioSession.Category = AVAudioSession.Category.playAndRecord
+
+	static private var audioCategoryOptions : AVAudioSession.CategoryOptions = [
+		AVAudioSession.CategoryOptions.mixWithOthers,
+		AVAudioSession.CategoryOptions.allowBluetooth,
+		AVAudioSession.CategoryOptions.allowAirPlay,
+		AVAudioSession.CategoryOptions.allowBluetoothA2DP
+	]
+
+	/*
+	 This mode is intended for Voice over IP (VoIP) apps and can only be used with the playAndRecord category. When this mode is used, the device’s tonal equalization is optimized for voice and the set of allowable audio routes is reduced to only those appropriate for voice chat.
+
+	  See: https://developer.apple.com/documentation/avfoundation/avaudiosession/mode/1616455-voicechat
+	 */
+	static private var audioMode = AVAudioSession.Mode.voiceChat
+	static private var audioModeDefault : AVAudioSession.Mode = AVAudioSession.Mode.default
+
+	static private var audioInputSelected: AVAudioSessionPortDescription? = nil
+
+	//
+	// Audio Input
+	//
+
+	static func initAudioDevices() -> Void {
+
+		PluginRTCAudioController.setCategory()
+
+		do {
+			let audioSession: AVAudioSession = AVAudioSession.sharedInstance()
+			try audioSession.setActive(true)
+		} catch  {
+			print("Error messing with audio session: \(error)")
+		}
+	}
+
+	static func setCategory() -> Void {
+		// Enable speaker
+		NSLog("PluginRTCAudioController#setCategory()")
+
+		do {
+			let audioSession: AVAudioSession = AVAudioSession.sharedInstance()
+			try audioSession.setCategory(
+				PluginRTCAudioController.audioCategory,
+				mode: PluginRTCAudioController.audioMode,
+				options: PluginRTCAudioController.audioCategoryOptions
+			)
+		} catch {
+			NSLog("PluginRTCAudioController#setCategory() | ERROR \(error)")
+		};
+	}
+
+	// Setter function inserted by save specific audio device
+	static func saveInputAudioDevice(inputDeviceUID: String) -> Void {
+		let audioSession: AVAudioSession = AVAudioSession.sharedInstance()
+		if let audioInput: AVAudioSessionPortDescription = audioSession.availableInputs!.first(where: { $0.uid == inputDeviceUID }) {
+			PluginRTCAudioController.audioInputSelected = audioInput
+		} else {
+			NSLog("PluginRTCAudioController#saveInputAudioDevice() | ERROR invalid deviceId \(inputDeviceUID)")
+			PluginRTCAudioController.audioInputSelected = audioSession.availableInputs!.first
+		}
+	}
+
+	// Setter function inserted by set specific audio device
+	static func restoreInputOutputAudioDevice() -> Void {
+		let audioSession: AVAudioSession = AVAudioSession.sharedInstance()
+
+		do {
+			try audioSession.setPreferredInput(PluginRTCAudioController.audioInputSelected)
+		} catch {
+			NSLog("PluginRTCAudioController:restoreInputOutputAudioDevice: Error setting audioSession preferred input.")
+		}
+
+		PluginRTCAudioController.setOutputSpeakerIfNeed(enabled: speakerEnabled);
+	}
+
+	static func setOutputSpeakerIfNeed(enabled: Bool) {
+
+		speakerEnabled = enabled
+
+		let audioSession: AVAudioSession = AVAudioSession.sharedInstance()
+		let currentRoute = audioSession.currentRoute
+
+		if currentRoute.outputs.count != 0 {
+			for description in currentRoute.outputs {
+				if (
+					description.portType == AVAudioSession.Port.headphones ||
+						description.portType == AVAudioSession.Port.bluetoothA2DP ||
+							description.portType == AVAudioSession.Port.carAudio ||
+								description.portType == AVAudioSession.Port.airPlay ||
+									description.portType == AVAudioSession.Port.lineOut
+				) {
+					NSLog("PluginRTCAudioController#setOutputSpeakerIfNeed() | external audio output plugged in -> do nothing")
+				} else {
+					NSLog("PluginRTCAudioController#setOutputSpeakerIfNeed() | external audio pulled out")
+
+					if (speakerEnabled) {
+						do {
+							try audioSession.overrideOutputAudioPort(AVAudioSession.PortOverride.speaker)
+						} catch {
+							NSLog("PluginRTCAudioController#setOutputSpeakerIfNeed() | ERROR \(error)")
+						};
+					}
+				}
+			}
+		} else {
+			NSLog("PluginRTCAudioController#setOutputSpeakerIfNeed() | requires connection to device")
+		}
+	}
+
+	static func selectAudioOutputSpeaker() {
+		// Enable speaker
+		NSLog("PluginRTCAudioController#selectAudioOutputSpeaker()")
+
+		speakerEnabled = true;
+
+		setCategory()
+
+		do {
+			let audioSession: AVAudioSession = AVAudioSession.sharedInstance()
+			try audioSession.overrideOutputAudioPort(AVAudioSession.PortOverride.speaker)
+		} catch {
+			NSLog("PluginRTCAudioController#selectAudioOutputSpeaker() | ERROR \(error)")
+		};
+	}
+
+	static func selectAudioOutputEarpiece() {
+		// Disable speaker, switched to default
+		NSLog("PluginRTCAudioController#selectAudioOutputEarpiece()")
+
+		speakerEnabled = false;
+
+		setCategory()
+
+		do {
+			let audioSession: AVAudioSession = AVAudioSession.sharedInstance()
+			try audioSession.overrideOutputAudioPort(AVAudioSession.PortOverride.none)
+		} catch {
+			NSLog("PluginRTCAudioController#selectAudioOutputEarpiece() | ERROR \(error)")
+		};
+	}
+
+    static func setDefaultAudioOutput(isSpeaker: Bool) {
+        NSLog("PluginRTCAudioController#setDefaultAudioOutput() | isSpeaker \(isSpeaker)")
+        
+    	speakerEnabled = isSpeaker
+
+        let audioConfiguration = RTCAudioSessionConfiguration()
+        audioConfiguration.category = PluginRTCAudioController.audioCategory.rawValue;
+        audioConfiguration.categoryOptions = PluginRTCAudioController.audioCategoryOptions
+        audioConfiguration.mode = PluginRTCAudioController.audioMode.rawValue
+
+        if (speakerEnabled) {
+            audioConfiguration.categoryOptions.insert(AVAudioSession.CategoryOptions.defaultToSpeaker)
+        }
+
+        RTCAudioSessionConfiguration.setWebRTC(audioConfiguration)
+    }
+	//
+	// Audio Output
+	//
+
+	static private var speakerEnabled: Bool = false
+
+	init() {
+		let shouldManualInit = Bundle.main.object(forInfoDictionaryKey: "ManualInitAudioDevice") as? String
+
+		if(shouldManualInit == "FALSE") {
+			PluginRTCAudioController.initAudioDevices()
+		}
+
+		NotificationCenter.default.addObserver(
+			self,
+			selector: #selector(self.audioRouteChangeListener(_:)),
+			name: AVAudioSession.routeChangeNotification,
+			object: nil)
+	}
+
+	@objc dynamic fileprivate func audioRouteChangeListener(_ notification:Notification) {
+		let audioRouteChangeReason = notification.userInfo![AVAudioSessionRouteChangeReasonKey] as! UInt
+
+		switch audioRouteChangeReason {
+		case AVAudioSession.RouteChangeReason.newDeviceAvailable.rawValue:
+			NSLog("PluginRTCAudioController#audioRouteChangeListener() | headphone plugged in")
+		case AVAudioSession.RouteChangeReason.oldDeviceUnavailable.rawValue:
+			NSLog("PluginRTCAudioController#audioRouteChangeListener() | headphone pulled out -> restore state speakerEnabled: %@", PluginRTCAudioController.speakerEnabled ? "true" : "false")
+			PluginRTCAudioController.setOutputSpeakerIfNeed(enabled: PluginRTCAudioController.speakerEnabled)
+		default:
+			break
+		}
+	}
+}
diff --git a/src/PluginRTCDTMFSender.swift b/src/PluginRTCDTMFSender.swift
index edea9280..48890af2 100644
--- a/src/PluginRTCDTMFSender.swift
+++ b/src/PluginRTCDTMFSender.swift
@@ -1,62 +1,70 @@
 import Foundation
 
 
-class PluginRTCDTMFSender : NSObject, RTCDTMFSenderDelegate {
-	var rtcDTMFSender: RTCDTMFSender?
-	var eventListener: ((data: NSDictionary) -> Void)?
-
+class PluginRTCDTMFSender : NSObject {
+	var rtcRtpSender: RTCRtpSender?
+	var eventListener: ((_ data: NSDictionary) -> Void)?
 
 	/**
-	* Constructor for pc.createDTMFSender().
-	*/
+	 * Constructor for pc.createDTMFSender().
+	 */
 	init(
 		rtcPeerConnection: RTCPeerConnection,
 		track: RTCMediaStreamTrack,
-		eventListener: (data: NSDictionary) -> Void
+		streamId: String,
+		eventListener: @escaping (_ data: NSDictionary) -> Void
 		) {
 		NSLog("PluginRTCDTMFSender#init()")
 
 		self.eventListener = eventListener
-		self.rtcDTMFSender = rtcPeerConnection.createDTMFSenderForTrack(track as? RTCAudioTrack)
 
-		if self.rtcDTMFSender == nil {
+		if !rtcPeerConnection.senders.isEmpty {
+			self.rtcRtpSender = rtcPeerConnection.senders[0]
+		} else {
+			// TODO check if new rtcRtpSender can be used one Unified-Plan merged
+			//self.rtcRtpSender = rtcPeerConnection.add(track, streamIds: [streamId]);
+		}
+
+		if self.rtcRtpSender == nil {
 			NSLog("PluginRTCDTMFSender#init() | rtcPeerConnection.createDTMFSenderForTrack() failed")
 			return
 		}
 	}
 
-
 	deinit {
 		NSLog("PluginRTCDTMFSender#deinit()")
 	}
 
-
 	func run() {
 		NSLog("PluginRTCDTMFSender#run()")
-
-		self.rtcDTMFSender!.delegate = self
 	}
 
-
-	func insertDTMF(tones: String, duration: Int, interToneGap: Int) {
+	func insertDTMF(_ tones: String, duration: TimeInterval, interToneGap: TimeInterval) {
 		NSLog("PluginRTCDTMFSender#insertDTMF()")
 
-		let result = self.rtcDTMFSender!.insertDTMF(tones, withDuration: duration, andInterToneGap: interToneGap)
+		let dtmfSender = self.rtcRtpSender?.dtmfSender
+
+		if dtmfSender == nil {
+			NSLog("RTCDTMFSender#insertDTMF() | RTCDTMFSender.insertDTMF() failed")
+			return
+		}
+
+		let durationMs = duration / 100
+		let interToneGapMs = interToneGap / 100
+		let result = dtmfSender!.insertDtmf(tones, duration: durationMs, interToneGap: interToneGapMs)
 		if !result {
 			NSLog("PluginRTCDTMFSender#indertDTMF() | RTCDTMFSender#indertDTMF() failed")
 		}
 	}
 
-
 	/**
-	* Methods inherited from RTCDTMFSenderDelegate.
-	*/
-
-	func toneChange(tone: String) {
+	 * Methods inherited from RTCDTMFSenderDelegate.
+	 */
+	func toneChange(_ tone: String) {
 		NSLog("PluginRTCDTMFSender | tone change [tone:%@]", tone)
 
 		if self.eventListener != nil {
-			self.eventListener!(data: [
+			self.eventListener!([
 				"type": "tonechange",
 				"tone": tone
 			])
diff --git a/src/PluginRTCDataChannel.swift b/src/PluginRTCDataChannel.swift
index 2eab64db..a3201627 100644
--- a/src/PluginRTCDataChannel.swift
+++ b/src/PluginRTCDataChannel.swift
@@ -1,14 +1,36 @@
 import Foundation
 
+// FIXME: comparison operators with optionals were removed from the Swift Standard Libary.
+// Consider refactoring the code to use the non-optional operators.
+fileprivate func < <T : Comparable>(lhs: T?, rhs: T?) -> Bool {
+	switch (lhs, rhs) {
+	case let (l?, r?):
+		return l < r
+	case (nil, _?):
+		return true
+	default:
+		return false
+	}
+}
+
+// FIXME: comparison operators with optionals were removed from the Swift Standard Libary.
+// Consider refactoring the code to use the non-optional operators.
+fileprivate func > <T : Comparable>(lhs: T?, rhs: T?) -> Bool {
+	switch (lhs, rhs) {
+	case let (l?, r?):
+		return l > r
+	default:
+		return rhs < lhs
+	}
+}
 
 class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 	var rtcDataChannel: RTCDataChannel?
-	var eventListener: ((data: NSDictionary) -> Void)?
-	var eventListenerForBinaryMessage: ((data: NSData) -> Void)?
+	var eventListener: ((_ data: NSDictionary) -> Void)?
+	var eventListenerForBinaryMessage: ((_ data: Data) -> Void)?
 	var lostStates = Array<String>()
 	var lostMessages = Array<RTCDataBuffer>()
 
-
 	/**
 	 * Constructor for pc.createDataChannel().
 	 */
@@ -16,27 +38,27 @@ class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 		rtcPeerConnection: RTCPeerConnection,
 		label: String,
 		options: NSDictionary?,
-		eventListener: (data: NSDictionary) -> Void,
-		eventListenerForBinaryMessage: (data: NSData) -> Void
+		eventListener: @escaping (_ data: NSDictionary) -> Void,
+		eventListenerForBinaryMessage: @escaping (_ data: Data) -> Void
 	) {
 		NSLog("PluginRTCDataChannel#init()")
 
 		self.eventListener = eventListener
 		self.eventListenerForBinaryMessage = eventListenerForBinaryMessage
 
-		let rtcDataChannelInit = RTCDataChannelInit()
+		let rtcDataChannelInit = RTCDataChannelConfiguration.init()
 
-		if options?.objectForKey("ordered") != nil {
-			rtcDataChannelInit.isOrdered = options!.objectForKey("ordered") as! Bool
+		if options?.object(forKey: "ordered") != nil {
+			rtcDataChannelInit.isOrdered = options!.object(forKey: "ordered") as! Bool
 		}
 
-		if options?.objectForKey("maxPacketLifeTime") != nil {
+		if options?.object(forKey: "maxPacketLifeTime") != nil {
 			// TODO: rtcDataChannel.maxRetransmitTime always reports 0.
-			rtcDataChannelInit.maxRetransmitTimeMs = options!.objectForKey("maxPacketLifeTime") as! Int
+			rtcDataChannelInit.maxRetransmitTimeMs = options!.object(forKey: "maxPacketLifeTime") as! Int
 		}
 
-		if options?.objectForKey("maxRetransmits") != nil {
-			rtcDataChannelInit.maxRetransmits = options!.objectForKey("maxRetransmits") as! Int
+		if options?.object(forKey: "maxRetransmits") != nil {
+			rtcDataChannelInit.maxRetransmits = options!.object(forKey: "maxRetransmits") as! Int32
 		}
 
 		// TODO: error: expected member name following '.'
@@ -44,21 +66,19 @@ class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 		// if options?.objectForKey("protocol") != nil {
 			// rtcDataChannelInit.protocol = options!.objectForKey("protocol") as! String
 		// }
-		if options?.objectForKey("protocol") != nil {
-			rtcDataChannelInit.`protocol` = options!.objectForKey("protocol") as! String
+		if options?.object(forKey: "protocol") != nil {
+			rtcDataChannelInit.`protocol` = options!.object(forKey: "protocol") as! String
 		}
 
-		if options?.objectForKey("negotiated") != nil {
-			rtcDataChannelInit.isNegotiated = options!.objectForKey("negotiated") as! Bool
+		if options?.object(forKey: "negotiated") != nil {
+			rtcDataChannelInit.isNegotiated = options!.object(forKey: "negotiated") as! Bool
 		}
 
-		if options?.objectForKey("id") != nil {
-			rtcDataChannelInit.streamId = options!.objectForKey("id") as! Int
+		if options?.object(forKey: "id") != nil {
+			rtcDataChannelInit.channelId = options!.object(forKey: "id") as! Int32
 		}
 
-		self.rtcDataChannel = rtcPeerConnection.createDataChannelWithLabel(label,
-			config: rtcDataChannelInit
-		)
+		self.rtcDataChannel = rtcPeerConnection.dataChannel(forLabel: label, configuration: rtcDataChannelInit)
 
 		if self.rtcDataChannel == nil {
 			NSLog("PluginRTCDataChannel#init() | rtcPeerConnection.createDataChannelWithLabel() failed")
@@ -66,27 +86,25 @@ class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 		}
 
 		// Report definitive data to update the JS instance.
-		self.eventListener!(data: [
+		self.eventListener!([
 			"type": "new",
 			"channel": [
 				"ordered": self.rtcDataChannel!.isOrdered,
-				"maxPacketLifeTime": self.rtcDataChannel!.maxRetransmitTime,
+				"maxPacketLifeTime": self.rtcDataChannel!.maxPacketLifeTime,
 				"maxRetransmits": self.rtcDataChannel!.maxRetransmits,
 				"protocol": self.rtcDataChannel!.`protocol`,
 				"negotiated": self.rtcDataChannel!.isNegotiated,
-				"id": self.rtcDataChannel!.streamId,
-				"readyState": PluginRTCTypes.dataChannelStates[self.rtcDataChannel!.state.rawValue] as String!,
+				"id": self.rtcDataChannel!.channelId,
+				"readyState": PluginRTCDataChannel.stateToString(state: self.rtcDataChannel!.readyState),
 				"bufferedAmount": self.rtcDataChannel!.bufferedAmount
 			]
 		])
 	}
 
-
 	deinit {
 		NSLog("PluginRTCDataChannel#deinit()")
 	}
 
-
 	/**
 	 * Constructor for pc.ondatachannel event.
 	 */
@@ -96,7 +114,6 @@ class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 		self.rtcDataChannel = rtcDataChannel
 	}
 
-
 	func run() {
 		NSLog("PluginRTCDataChannel#run()")
 
@@ -104,15 +121,14 @@ class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 
 		//if data channel is created after there is a connection,
 		// we need to dispatch its current state.
-		if (self.rtcDataChannel?.state.rawValue > 0) {
-			channelDidChangeState(self.rtcDataChannel);
+		if (self.rtcDataChannel?.readyState != RTCDataChannelState.connecting) {
+			dataChannelDidChangeState(self.rtcDataChannel!);
 		}
 	}
 
-
 	func setListener(
-		eventListener: (data: NSDictionary) -> Void,
-		eventListenerForBinaryMessage: (data: NSData) -> Void
+		_ eventListener: @escaping (_ data: NSDictionary) -> Void,
+		eventListenerForBinaryMessage: @escaping (_ data: Data) -> Void
 	) {
 		NSLog("PluginRTCDataChannel#setListener()")
 
@@ -120,7 +136,7 @@ class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 		self.eventListenerForBinaryMessage = eventListenerForBinaryMessage
 
 		for readyState in self.lostStates {
-			self.eventListener!(data: [
+			self.eventListener!([
 				"type": "statechange",
 				"readyState": readyState
 			])
@@ -133,15 +149,14 @@ class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 		self.lostMessages.removeAll()
 	}
 
-
 	func sendString(
-		data: String,
-		callback: (data: NSDictionary) -> Void
+		_ data: String,
+		callback: (_ data: NSDictionary) -> Void
 	) {
 		NSLog("PluginRTCDataChannel#sendString()")
 
 		let buffer = RTCDataBuffer(
-			data: (data.dataUsingEncoding(NSUTF8StringEncoding))!,
+			data: (data.data(using: String.Encoding.utf8))!,
 			isBinary: false
 		)
 
@@ -151,10 +166,9 @@ class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 		}
 	}
 
-
 	func sendBinary(
-		data: NSData,
-		callback: (data: NSDictionary) -> Void
+		_ data: Data,
+		callback: (_ data: NSDictionary) -> Void
 	) {
 		NSLog("PluginRTCDataChannel#sendBinary()")
 
@@ -169,26 +183,41 @@ class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 		}
 	}
 
-
 	func close() {
 		NSLog("PluginRTCDataChannel#close()")
 
 		self.rtcDataChannel!.close()
 	}
 
+	static func stateToString(state: RTCDataChannelState) -> String {
+		switch state {
+		case RTCDataChannelState.connecting:
+			return "connecting"
+		case RTCDataChannelState.open:
+			return "open"
+		case RTCDataChannelState.closing:
+			return "closing"
+		case RTCDataChannelState.closed:
+			return "closed"
+		}
+	}
+
+	func getState() -> String {
+		return PluginRTCDataChannel.stateToString(state: self.rtcDataChannel!.readyState)
+	}
 
 	/**
 	 * Methods inherited from RTCDataChannelDelegate.
 	 */
 
+	/** The data channel state changed. */
+	func dataChannelDidChangeState(_ channel: RTCDataChannel) {
+		let state_str = self.getState()
 
-	func channelDidChangeState(channel: RTCDataChannel!) {
-		let state_str = PluginRTCTypes.dataChannelStates[self.rtcDataChannel!.state.rawValue] as String!
-
-		NSLog("PluginRTCDataChannel | state changed [state:%@]", String(state_str))
+		NSLog("PluginRTCDataChannel | state changed [state:%@]", String(describing: state_str))
 
 		if self.eventListener != nil {
-			self.eventListener!(data: [
+			self.eventListener!([
 				"type": "statechange",
 				"readyState": state_str
 			])
@@ -198,52 +227,53 @@ class PluginRTCDataChannel : NSObject, RTCDataChannelDelegate {
 		}
 	}
 
-
-	func channel(channel: RTCDataChannel!, didReceiveMessageWithBuffer buffer: RTCDataBuffer!) {
+	/** The data channel successfully received a data buffer. */
+	func dataChannel(_ channel: RTCDataChannel, didReceiveMessageWith buffer: RTCDataBuffer) {
 		if !buffer.isBinary {
 			NSLog("PluginRTCDataChannel | utf8 message received")
 
 			if self.eventListener != nil {
-				self.emitReceivedMessage(buffer!)
+				self.emitReceivedMessage(buffer)
 			} else {
 				// It may happen that the eventListener is not yet set, so store the lost messages.
-				self.lostMessages.append(buffer!)
+				self.lostMessages.append(buffer)
 			}
 		} else {
 			NSLog("PluginRTCDataChannel | binary message received")
 
 			if self.eventListenerForBinaryMessage != nil {
-				self.emitReceivedMessage(buffer!)
+				self.emitReceivedMessage(buffer)
 			} else {
 				// It may happen that the eventListener is not yet set, so store the lost messages.
-				self.lostMessages.append(buffer!)
+				self.lostMessages.append(buffer)
 			}
 		}
 	}
 
-	func channel(channel: RTCDataChannel!, didChangeBufferedAmount amount: UInt) {
+	/** The data channel's |bufferedAmount| changed. */
+	func dataChannel(_ channel: RTCDataChannel, didChangeBufferedAmount amount: UInt64) {
 		NSLog("PluginRTCDataChannel | didChangeBufferedAmount %d", amount)
 
-		self.eventListener!(data: [
+		self.eventListener!([
 			"type": "bufferedamount",
 			"bufferedAmount": amount
 			])
 
 	}
 
-	func emitReceivedMessage(buffer: RTCDataBuffer) {
+	func emitReceivedMessage(_ buffer: RTCDataBuffer) {
 		if !buffer.isBinary {
-			let string = NSString(
+			let message = NSString(
 				data: buffer.data,
-				encoding: NSUTF8StringEncoding
+				encoding: String.Encoding.utf8.rawValue
 			)
 
-			self.eventListener!(data: [
+			self.eventListener!([
 				"type": "message",
-				"message": string as! String
+				"message": message! as String
 			])
 		} else {
-			self.eventListenerForBinaryMessage!(data: buffer.data)
+			self.eventListenerForBinaryMessage!(buffer.data)
 		}
 	}
 }
diff --git a/src/PluginRTCPeerConnection.swift b/src/PluginRTCPeerConnection.swift
index 170d325c..ecd2c8c6 100644
--- a/src/PluginRTCPeerConnection.swift
+++ b/src/PluginRTCPeerConnection.swift
@@ -1,7 +1,8 @@
 import Foundation
 
+class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate {
+	private let queue: DispatchQueue
 
-class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate, RTCSessionDescriptionDelegate, RTCStatsDelegate {
 	var rtcPeerConnectionFactory: RTCPeerConnectionFactory
 	var rtcPeerConnection: RTCPeerConnection!
 	var pluginRTCPeerConnectionConfig: PluginRTCPeerConnectionConfig
@@ -10,226 +11,295 @@ class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate, RTCSessionD
 	var pluginRTCDataChannels: [Int : PluginRTCDataChannel] = [:]
 	// PluginRTCDTMFSender dictionary.
 	var pluginRTCDTMFSenders: [Int : PluginRTCDTMFSender] = [:]
-	var eventListener: (data: NSDictionary) -> Void
-	var eventListenerForAddStream: (pluginMediaStream: PluginMediaStream) -> Void
-	var eventListenerForRemoveStream: (id: String) -> Void
-	var onCreateDescriptionSuccessCallback: ((rtcSessionDescription: RTCSessionDescription) -> Void)!
-	var onCreateDescriptionFailureCallback: ((error: NSError) -> Void)!
+	// PluginRTCRtpTransceiver dictionary.
+	var pluginRTCRtpTransceivers: [Int : PluginRTCRtpTransceiver] = [:]
+	var eventListener: (_ data: NSDictionary) -> Void
+	var eventListenerForAddStream: (_ pluginMediaStream: PluginMediaStream) -> Void
+	var eventListenerForRemoveStream: (_ pluginMediaStream: PluginMediaStream) -> Void
+	var eventListenerForAddTrack: (_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Void
+	var eventListenerForRemoveTrack: (_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Void
+
 	var onSetDescriptionSuccessCallback: (() -> Void)!
-	var onSetDescriptionFailureCallback: ((error: NSError) -> Void)!
-	var onGetStatsCallback: ((array: NSArray) -> Void)!
+	var onSetDescriptionFailureCallback: ((_ error: Error) -> Void)!
+
+	var onGetStatsCallback: ((_ array: NSArray) -> Void)!
+
+	var pluginMediaStreams: [String : PluginMediaStream]! = [:]
+	var pluginMediaTracks: [String : PluginMediaStreamTrack]! = [:]
+
+	var pluginRTCRtpSenders: [Int : PluginRTCRtpSender] = [:]
+	var pluginRTCRtpReceivers: [Int : PluginRTCRtpReceiver] = [:]
+
+	var isAudioInputSelected: Bool = false
 
 	init(
+		queue: DispatchQueue,
 		rtcPeerConnectionFactory: RTCPeerConnectionFactory,
 		pcConfig: NSDictionary?,
 		pcConstraints: NSDictionary?,
-		eventListener: (data: NSDictionary) -> Void,
-		eventListenerForAddStream: (pluginMediaStream: PluginMediaStream) -> Void,
-		eventListenerForRemoveStream: (id: String) -> Void
+		eventListener: @escaping (_ data: NSDictionary) -> Void,
+		eventListenerForAddStream: @escaping (_ pluginMediaStream: PluginMediaStream) -> Void,
+		eventListenerForRemoveStream: @escaping (_ pluginMediaStream: PluginMediaStream) -> Void,
+		eventListenerForAddTrack: @escaping (_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Void,
+		eventListenerForRemoveTrack: @escaping (_ pluginMediaStreamTrack: PluginMediaStreamTrack) -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#init()")
-
+		self.queue = queue
 		self.rtcPeerConnectionFactory = rtcPeerConnectionFactory
 		self.pluginRTCPeerConnectionConfig = PluginRTCPeerConnectionConfig(pcConfig: pcConfig)
 		self.pluginRTCPeerConnectionConstraints = PluginRTCPeerConnectionConstraints(pcConstraints: pcConstraints)
 		self.eventListener = eventListener
 		self.eventListenerForAddStream = eventListenerForAddStream
 		self.eventListenerForRemoveStream = eventListenerForRemoveStream
+		self.eventListenerForAddTrack = eventListenerForAddTrack
+		self.eventListenerForRemoveTrack = eventListenerForRemoveTrack
 	}
 
-
 	deinit {
 		NSLog("PluginRTCPeerConnection#deinit()")
 		self.pluginRTCDTMFSenders = [:]
+        self.pluginRTCRtpTransceivers = [:]
 	}
 
-
 	func run() {
 		NSLog("PluginRTCPeerConnection#run()")
 
-		self.rtcPeerConnection = self.rtcPeerConnectionFactory.peerConnectionWithICEServers(
-			self.pluginRTCPeerConnectionConfig.getIceServers(),
+		self.rtcPeerConnection = self.rtcPeerConnectionFactory.peerConnection(
+			with: self.pluginRTCPeerConnectionConfig.getConfiguration(),
 			constraints: self.pluginRTCPeerConnectionConstraints.getConstraints(),
 			delegate: self
 		)
 	}
 
-
 	func createOffer(
-		options: NSDictionary?,
-		callback: (data: NSDictionary) -> Void,
-		errback: (error: NSError) -> Void
+		_ options: NSDictionary?,
+		callback: @escaping (_ data: NSDictionary) -> Void,
+		errback: @escaping (_ error: Error) -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#createOffer()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
 		let pluginRTCPeerConnectionConstraints = PluginRTCPeerConnectionConstraints(pcConstraints: options)
 
-		self.onCreateDescriptionSuccessCallback = { (rtcSessionDescription: RTCSessionDescription) -> Void in
+
+		let onCreateLocalDescriptionSuccessCallback = { (rtcSessionDescription: RTCSessionDescription) -> Void in
 			NSLog("PluginRTCPeerConnection#createOffer() | success callback")
 
 			let data = [
-				"type": rtcSessionDescription.type,
-				"sdp": rtcSessionDescription.description
-			]
+				"type": RTCSessionDescription.string(for: rtcSessionDescription.type),
+				"sdp": rtcSessionDescription.sdp,
+				"transceivers": self.getTransceiversJSON()
+			] as [String : Any]
 
-			callback(data: data)
+			callback(data as NSDictionary)
 		}
 
-		self.onCreateDescriptionFailureCallback = { (error: NSError) -> Void in
-			NSLog("PluginRTCPeerConnection#createOffer() | failure callback: %@", String(error))
+		let onCreateLocalDescriptionFailureCallback = { (error: Error) -> Void in
+			NSLog("PluginRTCPeerConnection#createOffer() | failure callback: %@", String(describing: error))
 
-			errback(error: error)
+			errback(error)
 		}
 
-		self.rtcPeerConnection.createOfferWithDelegate(self,
-			constraints: pluginRTCPeerConnectionConstraints.getConstraints())
+		self.rtcPeerConnection.offer(for: pluginRTCPeerConnectionConstraints.getConstraints(), completionHandler: {
+			(sdp: RTCSessionDescription?, error: Error?) in
+			if (error == nil) {
+				onCreateLocalDescriptionSuccessCallback(sdp!);
+			} else {
+				onCreateLocalDescriptionFailureCallback(error!);
+			}
+		})
 	}
 
 
 	func createAnswer(
-		options: NSDictionary?,
-		callback: (data: NSDictionary) -> Void,
-		errback: (error: NSError) -> Void
+		_ options: NSDictionary?,
+		callback: @escaping (_ data: NSDictionary) -> Void,
+		errback: @escaping (_ error: Error) -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#createAnswer()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
 		let pluginRTCPeerConnectionConstraints = PluginRTCPeerConnectionConstraints(pcConstraints: options)
 
-		self.onCreateDescriptionSuccessCallback = { (rtcSessionDescription: RTCSessionDescription) -> Void in
+		let onCreateRemoteDescriptionSuccessCallback = { (rtcSessionDescription: RTCSessionDescription) -> Void in
 			NSLog("PluginRTCPeerConnection#createAnswer() | success callback")
 
 			let data = [
-				"type": rtcSessionDescription.type,
-				"sdp": rtcSessionDescription.description
-			]
+				"type": RTCSessionDescription.string(for: rtcSessionDescription.type),
+				"sdp": rtcSessionDescription.sdp,
+				"transceivers": self.getTransceiversJSON()
+			] as [String : Any]
 
-			callback(data: data)
+			callback(data as NSDictionary)
 		}
 
-		self.onCreateDescriptionFailureCallback = { (error: NSError) -> Void in
-			NSLog("PluginRTCPeerConnection#createAnswer() | failure callback: %@", String(error))
+		let onCreateRemoteDescriptionFailureCallback = { (error: Error) -> Void in
+			NSLog("PluginRTCPeerConnection#createAnswer() | failure callback: %@", String(describing: error))
 
-			errback(error: error)
+			errback(error)
 		}
 
-		self.rtcPeerConnection.createAnswerWithDelegate(self,
-			constraints: pluginRTCPeerConnectionConstraints.getConstraints())
+		self.rtcPeerConnection.answer(for: pluginRTCPeerConnectionConstraints.getConstraints(), completionHandler: {
+			(sdp: RTCSessionDescription?, error: Error?) in
+			if (error == nil) {
+				onCreateRemoteDescriptionSuccessCallback(sdp!)
+			} else {
+				onCreateRemoteDescriptionFailureCallback(error!)
+			}
+		})
 	}
 
-
 	func setLocalDescription(
-		desc: NSDictionary,
-		callback: (data: NSDictionary) -> Void,
-		errback: (error: NSError) -> Void
+		_ desc: NSDictionary,
+		callback: @escaping (_ data: NSDictionary) -> Void,
+		errback: @escaping (_ error: Error) -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#setLocalDescription()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
-		let type = desc.objectForKey("type") as? String ?? ""
-		let sdp = desc.objectForKey("sdp") as? String ?? ""
-		let rtcSessionDescription = RTCSessionDescription(type: type, sdp: sdp)
+		let type = desc.object(forKey: "type") as? String ?? ""
+		let sdp = desc.object(forKey: "sdp") as? String ?? ""
+		let sdpType = RTCSessionDescription.type(for: type)
+		let rtcSessionDescription = RTCSessionDescription(type: sdpType, sdp: sdp)
 
 		self.onSetDescriptionSuccessCallback = { [unowned self] () -> Void in
 			NSLog("PluginRTCPeerConnection#setLocalDescription() | success callback")
-
+			var descType = ""
+			var descSDP = ""
+			if self.rtcPeerConnection.localDescription != nil {
+				descType = RTCSessionDescription.string(for: self.rtcPeerConnection.localDescription!.type)
+				descSDP = self.rtcPeerConnection.localDescription!.sdp
+			}
 			let data = [
-				"type": self.rtcPeerConnection.localDescription.type,
-				"sdp": self.rtcPeerConnection.localDescription.description
-			]
+				"type": descType,
+				"sdp": descSDP,
+				"transceivers": self.getTransceiversJSON()
+			] as [String : Any]
 
-			callback(data: data)
+			callback(data as NSDictionary)
 		}
 
-		self.onSetDescriptionFailureCallback = { (error: NSError) -> Void in
-			NSLog("PluginRTCPeerConnection#setLocalDescription() | failure callback: %@", String(error))
+		self.onSetDescriptionFailureCallback = { (error: Error) -> Void in
+			NSLog("PluginRTCPeerConnection#setLocalDescription() | failure callback: %@ description %@",
+						String(describing: error), desc)
 
-			errback(error: error)
+			errback(error)
 		}
 
-		self.rtcPeerConnection.setLocalDescriptionWithDelegate(self,
-			sessionDescription: rtcSessionDescription
-		)
+		// TODO: Would transceivers with mid=null get their mid's assigned after
+		 // setLocalDescription call? Investigate...
+		 //
+		 // Reference:
+		 // Something I overlooked last year is that transceiver.mid is null before setLocalDescription. We avoided 
+		 // that problem above by establishing the connection ahead of sending anything, but this makes mid useless 
+		 // for correlating in the initial negotiation.
+		 // https://blog.mozilla.org/webrtc/rtcrtptransceiver-explored/
+
+		self.rtcPeerConnection.setLocalDescription(rtcSessionDescription, completionHandler: {
+			(error: Error?) in
+			self.queue.async {
+				if (error == nil) {
+					self.onSetDescriptionSuccessCallback();
+				} else {
+					self.onSetDescriptionFailureCallback(error!);
+				}
+			}
+		})
 	}
 
 
 	func setRemoteDescription(
-		desc: NSDictionary,
-		callback: (data: NSDictionary) -> Void,
-		errback: (error: NSError) -> Void
+		_ desc: NSDictionary,
+		callback: @escaping (_ data: NSDictionary) -> Void,
+		errback: @escaping (_ error: Error) -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#setRemoteDescription()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
-		let type = desc.objectForKey("type") as? String ?? ""
-		let sdp = desc.objectForKey("sdp") as? String ?? ""
-		let rtcSessionDescription = RTCSessionDescription(type: type, sdp: sdp)
+		let type = desc.object(forKey: "type") as? String ?? ""
+		let sdp = desc.object(forKey: "sdp") as? String ?? ""
+		let sdpType = RTCSessionDescription.type(for: type)
+		let rtcSessionDescription = RTCSessionDescription(type: sdpType, sdp: sdp)
 
 		self.onSetDescriptionSuccessCallback = { [unowned self] () -> Void in
 			NSLog("PluginRTCPeerConnection#setRemoteDescription() | success callback")
 
 			let data = [
-				"type": self.rtcPeerConnection.remoteDescription.type,
-				"sdp": self.rtcPeerConnection.remoteDescription.description
-			]
+				"type": RTCSessionDescription.string(for: self.rtcPeerConnection.remoteDescription!.type),
+				"sdp": self.rtcPeerConnection.remoteDescription!.sdp,
+				"transceivers": self.getTransceiversJSON()
+            ] as NSDictionary
 
-			callback(data: data)
+			callback(data)
 		}
 
-		self.onSetDescriptionFailureCallback = { (error: NSError) -> Void in
-			NSLog("PluginRTCPeerConnection#setRemoteDescription() | failure callback: %@", String(error))
+		self.onSetDescriptionFailureCallback = { (error: Error) -> Void in
+			NSLog("PluginRTCPeerConnection#setRemoteDescription() | failure callback: %@", String(describing: error))
 
-			errback(error: error)
+			errback(error)
 		}
 
-		self.rtcPeerConnection.setRemoteDescriptionWithDelegate(self,
-			sessionDescription: rtcSessionDescription
-		)
+		self.rtcPeerConnection.setRemoteDescription(rtcSessionDescription, completionHandler: {
+			(error: Error?) in
+			self.queue.async {
+				if (error == nil) {
+					self.onSetDescriptionSuccessCallback();
+				} else {
+					self.onSetDescriptionFailureCallback(error!);
+				}
+			}
+		})
 	}
 
-
 	func addIceCandidate(
-		candidate: NSDictionary,
-		callback: (data: NSDictionary) -> Void,
+		_ candidate: NSDictionary,
+		callback: (_ data: NSDictionary) -> Void,
 		errback: () -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#addIceCandidate()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
-		let sdpMid = candidate.objectForKey("sdpMid") as? String ?? ""
-		let sdpMLineIndex = candidate.objectForKey("sdpMLineIndex") as? Int ?? 0
-		let candidate = candidate.objectForKey("candidate") as? String ?? ""
+		let sdpMid = candidate.object(forKey: "sdpMid") as? String ?? ""
+		let sdpMLineIndex = candidate.object(forKey: "sdpMLineIndex") as? Int32 ?? 0
+		let candidate = candidate.object(forKey: "candidate") as? String ?? ""
 
-		let result: Bool = self.rtcPeerConnection.addICECandidate(RTCICECandidate(
-			mid: sdpMid,
-			index: sdpMLineIndex,
-			sdp: candidate
+		self.rtcPeerConnection!.add(RTCIceCandidate(
+			sdp: candidate,
+			sdpMLineIndex: sdpMLineIndex,
+			sdpMid: sdpMid
 		))
 
-		var data: NSDictionary
+		// TODO detect RTCIceCandidate failure
+		let result = true
+
+		// TODO check if it still needed or moved elsewhere
+		if !self.isAudioInputSelected {
+			PluginRTCAudioController.restoreInputOutputAudioDevice()
+			self.isAudioInputSelected = true
+		}
 
 		if result == true {
+			var data: NSDictionary
 			if self.rtcPeerConnection.remoteDescription != nil {
 				data = [
 					"remoteDescription": [
-						"type": self.rtcPeerConnection.remoteDescription.type,
-						"sdp": self.rtcPeerConnection.remoteDescription.description
+						"type": RTCSessionDescription.string(for: self.rtcPeerConnection.remoteDescription!.type),
+						"sdp": self.rtcPeerConnection.remoteDescription!.sdp
 					]
 				]
 			} else {
@@ -238,45 +308,255 @@ class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate, RTCSessionD
 				]
 			}
 
-			callback(data: data)
+			callback(data)
 		} else {
 			errback()
 		}
 	}
 
-
-	func addStream(pluginMediaStream: PluginMediaStream) -> Bool {
+	func addStream(_ pluginMediaStream: PluginMediaStream) -> Bool {
 		NSLog("PluginRTCPeerConnection#addStream()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return false
 		}
 
-		return self.rtcPeerConnection.addStream(pluginMediaStream.rtcMediaStream)
-	}
+		if (IsUnifiedPlan()) {
 
+			var streamAdded : Bool = false;
+			let streamId = pluginMediaStream.rtcMediaStream.streamId;
+			for (_, pluginMediaTrack) in pluginMediaStream.audioTracks {
+				let pluginRTCRtpTransceiverId = self.createIdFor(collection: self.pluginRTCRtpTransceivers)
+				let pluginRTCRtpReceiverId = self.createIdFor(collection: self.pluginRTCRtpReceivers)
+				let pluginRTCRtpSenderId = self.createIdFor(collection: self.pluginRTCRtpSenders)
+				let pluginRTCRtpSender = self.addTrack(pluginMediaTrack, pluginRTCRtpTransceiverId, pluginRTCRtpReceiverId, pluginRTCRtpSenderId, [streamId])
+				streamAdded = streamAdded && (pluginRTCRtpSender != nil)
+			}
+
+			for (_, pluginMediaTrack) in pluginMediaStream.videoTracks {
+				let pluginRTCRtpTransceiverId = self.createIdFor(collection: self.pluginRTCRtpTransceivers)
+				let pluginRTCRtpReceiverId = self.createIdFor(collection: self.pluginRTCRtpReceivers)
+				let pluginRTCRtpSenderId = self.createIdFor(collection: self.pluginRTCRtpSenders)
+				let pluginRTCRtpSender = self.addTrack(pluginMediaTrack, pluginRTCRtpTransceiverId, pluginRTCRtpReceiverId, pluginRTCRtpSenderId, [streamId])
+				streamAdded = streamAdded && (pluginRTCRtpSender != nil)
+			}
 
-	func removeStream(pluginMediaStream: PluginMediaStream) {
+			return streamAdded;
+
+		} else {
+			self.rtcPeerConnection.add(pluginMediaStream.rtcMediaStream)
+		}
+
+		return true
+	}
+
+	func removeStream(_ pluginMediaStream: PluginMediaStream) {
 		NSLog("PluginRTCPeerConnection#removeStream()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
+			return
+		}
+
+		if (IsUnifiedPlan()) {
+			var trackIdsToRemove: [String] = []
+			for (_, pluginMediaStreamTrack) in pluginMediaStream.audioTracks {
+				trackIdsToRemove.append(pluginMediaStreamTrack.rtcMediaStreamTrack.trackId)
+			}
+			for (_, pluginMediaStreamTrack) in pluginMediaStream.videoTracks {
+				trackIdsToRemove.append(pluginMediaStreamTrack.rtcMediaStreamTrack.trackId)
+			}
+
+			for trackId in trackIdsToRemove {
+				for (_, pluginRTCRtpSender) in self.pluginRTCRtpSenders {
+					if pluginRTCRtpSender.rtpSender.track?.trackId == trackId {
+						self.removeTrack(pluginRTCRtpSender)
+						break
+					}
+				}
+			}
+		} else {
+			self.rtcPeerConnection.remove(pluginMediaStream.rtcMediaStream)
+		}
+	}
+
+	func IsUnifiedPlan() -> Bool {
+		return rtcPeerConnection.configuration.sdpSemantics == RTCSdpSemantics.unifiedPlan;
+	}
+
+	func addTrack(
+		_ pluginMediaTrack: PluginMediaStreamTrack,
+		_ pluginRTCRtpTransceiverId: Int,
+		_ pluginRTCRtpReceiverId: Int,
+		_ pluginRTCRtpSenderId: Int,
+		_ streamIds: [String]
+	) -> PluginRTCRtpSender? {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
+			return nil
+		}
+
+		let rtcMediaStreamTrack = pluginMediaTrack.rtcMediaStreamTrack;
+		var pluginRTCRtpSender = self.pluginRTCRtpSenders[pluginRTCRtpSenderId];
+		NSLog("PluginRTCPeerConnection#addTrack() trackId=%@ rtcId=%d streamIds %@",
+				pluginMediaTrack.id, pluginRTCRtpSenderId, streamIds);
+		if (pluginRTCRtpSender == nil) {
+			let rtcRtpSender: RTCRtpSender? = self.rtcPeerConnection.add(rtcMediaStreamTrack, streamIds: streamIds)
+			if let rtcRtpSender = rtcRtpSender {
+				let pluginRTCRtpTransceiver = self.updateTransceivers(rtcRtpSender: rtcRtpSender)
+				self.updateTransceiverId(pluginRTCRtpTransceiver!, pluginRTCRtpTransceiverId, pluginRTCRtpReceiverId, pluginRTCRtpSenderId)
+				pluginRTCRtpSender = pluginRTCRtpTransceiver!.pluginRTCRtpSender
+			} else {
+				NSLog("PluginRTCPeerConnection#addTrack() | RTCPeerConnection#add() failed")
+			}
+		}
+
+		return pluginRTCRtpSender;
+	}
+
+	func removeTrack(_ pluginRTCRtpSender: PluginRTCRtpSender) {
+		NSLog("PluginRTCPeerConnection#removeTrack()")
+
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
-		self.rtcPeerConnection.removeStream(pluginMediaStream.rtcMediaStream)
+		self.rtcPeerConnection.removeTrack(pluginRTCRtpSender.rtpSender)
+	}
+
+	func addTransceiver(
+		with: PluginMediaStreamTrack?,
+		of: RTCRtpMediaType?,
+		options: NSDictionary?,
+		transceiverId: Int,
+		senderId: Int,
+		receiverId: Int,
+		receiverTrackId: String,
+		callback: (_ data: NSDictionary) -> Void
+	) {
+		NSLog("PluginRTCPeerConnection#addTransceiver()")
+
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
+			return
+		}
+
+		var pluginRTCRtpTransceiver: PluginRTCRtpTransceiver
+
+		if with != nil {
+			pluginRTCRtpTransceiver = PluginRTCRtpTransceiver(
+				rtcPeerConnection: self.rtcPeerConnection,
+                mediaStreamTrack: with!.rtcMediaStreamTrack,
+				options: options,
+				transceiverId: transceiverId,
+				senderId: senderId,
+				receiverId: receiverId
+        	)
+		} else if of != nil {
+			pluginRTCRtpTransceiver = PluginRTCRtpTransceiver(
+				rtcPeerConnection: self.rtcPeerConnection,
+                mediaType: of!,
+				options: options,
+				transceiverId: transceiverId,
+				senderId: senderId,
+				receiverId: receiverId
+        	)
+		} else {
+			NSLog("PluginRTCPeerConnection#addTransceiver() | error: no valid track or type")
+			return
+		}
+
+		// NOTE: Creates native track in case it's not already existing.
+        self.getPluginMediaStreamTrack(pluginRTCRtpTransceiver.rtcRtpTransceiver!.receiver.track, trackId: receiverTrackId)
+
+		self.updateTransceivers(pluginRTCRtpTransceiver: pluginRTCRtpTransceiver)
+
+		let response: NSDictionary = [
+			"transceivers": self.getTransceiversJSON()
+		]
+
+		callback(response)
+	}
+	
+	func createIdFor(collection: [Int : Any]) -> Int {
+		var newId: Int
+		repeat {
+			newId = Int.random(in: 10000..<100000)
+		} while (collection[newId] != nil)
+		return newId
+	}
+
+	func updateTransceiverId(
+		_ pluginRTCRtpTransceiver: PluginRTCRtpTransceiver,
+		_ pluginRTCRtpTransceiverId: Int,
+		_ pluginRTCRtpReceiverId: Int,
+		_ pluginRTCRtpSenderId: Int
+	) {
+		self.pluginRTCRtpTransceivers.removeValue(forKey: pluginRTCRtpTransceiver.id)
+		pluginRTCRtpTransceiver.id = pluginRTCRtpTransceiverId
+		self.pluginRTCRtpTransceivers[pluginRTCRtpTransceiverId] = pluginRTCRtpTransceiver
+
+		self.pluginRTCRtpReceivers.removeValue(forKey: pluginRTCRtpTransceiver.pluginRTCRtpReceiver!.id)
+		pluginRTCRtpTransceiver.pluginRTCRtpReceiver!.id = pluginRTCRtpReceiverId
+		self.pluginRTCRtpReceivers[pluginRTCRtpReceiverId] = pluginRTCRtpTransceiver.pluginRTCRtpReceiver
+
+		self.pluginRTCRtpSenders.removeValue(forKey: pluginRTCRtpTransceiver.pluginRTCRtpSender!.id)
+		pluginRTCRtpTransceiver.pluginRTCRtpSender!.id = pluginRTCRtpSenderId
+		self.pluginRTCRtpSenders[pluginRTCRtpSenderId] = pluginRTCRtpTransceiver.pluginRTCRtpSender
+	}
+
+	func updateTransceivers(pluginRTCRtpTransceiver: PluginRTCRtpTransceiver) {
+		self.pluginRTCRtpTransceivers[pluginRTCRtpTransceiver.id] = pluginRTCRtpTransceiver
+		if pluginRTCRtpTransceiver.pluginRTCRtpSender != nil {
+			self.pluginRTCRtpSenders[pluginRTCRtpTransceiver.pluginRTCRtpSender!.id] = pluginRTCRtpTransceiver.pluginRTCRtpSender
+		}
+		if pluginRTCRtpTransceiver.pluginRTCRtpReceiver != nil {
+			self.pluginRTCRtpReceivers[pluginRTCRtpTransceiver.pluginRTCRtpReceiver!.id] = pluginRTCRtpTransceiver.pluginRTCRtpReceiver
+		}
+	}
+
+	func updateTransceivers(rtcRtpSender: RTCRtpSender) -> PluginRTCRtpTransceiver? {
+		for rtcRtpTransceiver in self.rtcPeerConnection.transceivers {
+			if rtcRtpTransceiver.sender == rtcRtpSender || rtcRtpTransceiver.sender.senderId == rtcRtpSender.senderId {
+				return self.updateTransceivers(rtcRtpTransceiver: rtcRtpTransceiver)
+			}
+		}
+		// NOTE: should not happen
+		return nil
 	}
 
+	func updateTransceivers(rtcRtpTransceiver: RTCRtpTransceiver) -> PluginRTCRtpTransceiver {
+		var senderReceiverToTransceiver: [String:PluginRTCRtpTransceiver] = [:]
+		for (_, pluginRTCRtpTransceiver) in self.pluginRTCRtpTransceivers {
+			// NOTE: this test should be enough
+			if pluginRTCRtpTransceiver.rtcRtpTransceiver == rtcRtpTransceiver {
+				return pluginRTCRtpTransceiver
+			}
+			let senderReceiverKey = pluginRTCRtpTransceiver.rtcRtpTransceiver!.sender.senderId + "::" + pluginRTCRtpTransceiver.rtcRtpTransceiver!.receiver.receiverId
+			senderReceiverToTransceiver[senderReceiverKey] = pluginRTCRtpTransceiver
+		}
+
+		// test by senderId and receiverId
+		let key = rtcRtpTransceiver.sender.senderId + "::" + rtcRtpTransceiver.receiver.receiverId
+		let existingPluginRTCRtpTransceiver = senderReceiverToTransceiver[key]
+		if existingPluginRTCRtpTransceiver != nil {
+			return existingPluginRTCRtpTransceiver!
+		}
+
+		// new transceiver found
+		let pluginRTCRtpTransceiver = PluginRTCRtpTransceiver(rtcRtpTransceiver)
+		self.updateTransceivers(pluginRTCRtpTransceiver: pluginRTCRtpTransceiver)
+		NSLog("PluginRTCPeerConnection#updateTransceivers() | New transceiver created id=\(pluginRTCRtpTransceiver.id)")
+		return pluginRTCRtpTransceiver
+	}
 
 	func createDataChannel(
-		dcId: Int,
+		_ dcId: Int,
 		label: String,
 		options: NSDictionary?,
-		eventListener: (data: NSDictionary) -> Void,
-		eventListenerForBinaryMessage: (data: NSData) -> Void
+		eventListener: @escaping (_ data: NSDictionary) -> Void,
+		eventListenerForBinaryMessage: @escaping (_ data: Data) -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#createDataChannel()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
@@ -295,11 +575,10 @@ class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate, RTCSessionD
 		pluginRTCDataChannel.run()
 	}
 
-
 	func RTCDataChannel_setListener(
-		dcId: Int,
-		eventListener: (data: NSDictionary) -> Void,
-		eventListenerForBinaryMessage: (data: NSData) -> Void
+		_ dcId: Int,
+		eventListener: @escaping (_ data: NSDictionary) -> Void,
+		eventListenerForBinaryMessage: @escaping (_ data: Data) -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#RTCDataChannel_setListener()")
 
@@ -317,19 +596,20 @@ class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate, RTCSessionD
 
 
 	func createDTMFSender(
-		dsId: Int,
+		_ dsId: Int,
 		track: PluginMediaStreamTrack,
-		eventListener: (data: NSDictionary) -> Void
+		eventListener: @escaping (_ data: NSDictionary) -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#createDTMFSender()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
 		let pluginRTCDTMFSender = PluginRTCDTMFSender(
-			rtcPeerConnection: rtcPeerConnection,
+			rtcPeerConnection: self.rtcPeerConnection,
 			track: track.rtcMediaStreamTrack,
+			streamId: String(dsId), //TODO
 			eventListener: eventListener
 		)
 
@@ -341,46 +621,61 @@ class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate, RTCSessionD
 	}
 
 	func getStats(
-		pluginMediaStreamTrack: PluginMediaStreamTrack?,
-		callback: (data: NSArray) -> Void,
-		errback: (error: NSError) -> Void
+		_ pluginMediaStreamTrack: PluginMediaStreamTrack?,
+		callback: @escaping (_ data: [[String:Any]]) -> Void,
+		errback: (_ error: NSError) -> Void
 	) {
-		NSLog("PluginRTCPeerConnection#getStats()")
+//		NSLog("PluginRTCPeerConnection#getStats()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
-		self.onGetStatsCallback = { (array: NSArray) -> Void in
-			callback(data: array)
-		}
-
-		if !self.rtcPeerConnection.getStatsWithDelegate(self,
-			mediaStreamTrack: pluginMediaStreamTrack?.rtcMediaStreamTrack,
-			statsOutputLevel: RTCStatsOutputLevelStandard) {
-				errback(error: NSError(domain: "Cannot get peer connection stats.", code: -1, userInfo: nil))
-		}
-    }
+		self.rtcPeerConnection.stats(for: pluginMediaStreamTrack?.rtcMediaStreamTrack, statsOutputLevel: RTCStatsOutputLevel.standard, completionHandler: { (stats: [RTCLegacyStatsReport]) in
+			var data: [[String:Any]] = []
+			for i in 0 ..< stats.count {
+				let report: RTCLegacyStatsReport = stats[i]
+				data.append([
+					"reportId" : report.reportId,
+					"type" : report.type,
+					"timestamp" : report.timestamp,
+					"values" : report.values
+				])
+			}
+//			NSLog("Stats:\n %@", data)
+			callback(data)
+		})
+	}
 
 	func close() {
 		NSLog("PluginRTCPeerConnection#close()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
+		for (_, pluginMediaStream) in self.pluginMediaStreams {
+			self.eventListenerForRemoveStream(pluginMediaStream)
+		}
+
+		for (_, pluginMediaTrack) in self.pluginMediaTracks {
+			self.eventListenerForRemoveTrack(pluginMediaTrack)
+		}
+
+		self.pluginMediaTracks = [:];
+		self.pluginMediaStreams = [:];
+
 		self.rtcPeerConnection.close()
 	}
 
-
 	func RTCDataChannel_sendString(
-		dcId: Int,
+		_ dcId: Int,
 		data: String,
-		callback: (data: NSDictionary) -> Void
+		callback: (_ data: NSDictionary) -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#RTCDataChannel_sendString()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
@@ -395,13 +690,13 @@ class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate, RTCSessionD
 
 
 	func RTCDataChannel_sendBinary(
-		dcId: Int,
-		data: NSData,
-		callback: (data: NSDictionary) -> Void
+		_ dcId: Int,
+		data: Data,
+		callback: (_ data: NSDictionary) -> Void
 	) {
 		NSLog("PluginRTCPeerConnection#RTCDataChannel_sendBinary()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
@@ -415,10 +710,10 @@ class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate, RTCSessionD
 	}
 
 
-	func RTCDataChannel_close(dcId: Int) {
+	func RTCDataChannel_close(_ dcId: Int) {
 		NSLog("PluginRTCPeerConnection#RTCDataChannel_close()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
@@ -436,14 +731,14 @@ class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate, RTCSessionD
 
 
 	func RTCDTMFSender_insertDTMF(
-		dsId: Int,
+		_ dsId: Int,
 		tones: String,
-		duration: Int,
-		interToneGap: Int
+		duration: Double,
+		interToneGap: Double
 	) {
 		NSLog("PluginRTCPeerConnection#RTCDTMFSender_insertDTMF()")
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
+		if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
 			return
 		}
 
@@ -452,215 +747,383 @@ class PluginRTCPeerConnection : NSObject, RTCPeerConnectionDelegate, RTCSessionD
 			return
 		}
 
-		pluginRTCDTMFSender!.insertDTMF(tones, duration: duration, interToneGap: interToneGap)
+		pluginRTCDTMFSender!.insertDTMF(tones, duration: duration as TimeInterval, interToneGap: interToneGap as TimeInterval)
 	}
 
-
 	/**
 	 * Methods inherited from RTCPeerConnectionDelegate.
 	 */
 
+	private func getPluginMediaStream(_ stream: RTCMediaStream?) -> PluginMediaStream? {
 
-	func peerConnection(peerConnection: RTCPeerConnection!,
-		signalingStateChanged newState: RTCSignalingState) {
-		let state_str = PluginRTCTypes.signalingStates[newState.rawValue] as String!
+		if (stream == nil) {
+			return nil;
+		}
 
-		NSLog("PluginRTCPeerConnection | onsignalingstatechange [signalingState:%@]", String(state_str))
+		var currentMediaStream : PluginMediaStream? = nil;
 
-		self.eventListener(data: [
-			"type": "signalingstatechange",
-			"signalingState": state_str
-		])
-	}
+		for (_, pluginMediaStream) in self.pluginMediaStreams {
+			if (pluginMediaStream.rtcMediaStream.streamId == stream!.streamId) {
+				currentMediaStream = pluginMediaStream;
+				break;
+			}
+		}
 
+		if (currentMediaStream == nil) {
 
-	func peerConnection(peerConnection: RTCPeerConnection!,
-		iceGatheringChanged newState: RTCICEGatheringState) {
-		let state_str = PluginRTCTypes.iceGatheringStates[newState.rawValue] as String!
+			currentMediaStream = PluginMediaStream(rtcMediaStream: stream!)
 
-		NSLog("PluginRTCPeerConnection | onicegatheringstatechange [iceGatheringState:%@]", String(state_str))
+			currentMediaStream!.run()
 
-		self.eventListener(data: [
-			"type": "icegatheringstatechange",
-			"iceGatheringState": state_str
-		])
+			// Let the plugin store it in its dictionary.
+			self.pluginMediaStreams[currentMediaStream!.id] = currentMediaStream;
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
-			return
+			// Fixes issue #576
+			self.eventListenerForAddStream(currentMediaStream!)
 		}
 
-		// Emit an empty candidate if iceGatheringState is "complete".
-		if newState.rawValue == RTCICEGatheringComplete.rawValue && self.rtcPeerConnection.localDescription != nil {
-			self.eventListener(data: [
-				"type": "icecandidate",
-				// NOTE: Cannot set null as value.
-				"candidate": false,
-				"localDescription": [
-					"type": self.rtcPeerConnection.localDescription.type,
-					"sdp": self.rtcPeerConnection.localDescription.description
-				]
-			])
-		}
+		return currentMediaStream;
 	}
 
 
-	func peerConnection(peerConnection: RTCPeerConnection!,
-		gotICECandidate candidate: RTCICECandidate!) {
-		NSLog("PluginRTCPeerConnection | onicecandidate [sdpMid:%@, sdpMLineIndex:%@, candidate:%@]",
-			String(candidate.sdpMid), String(candidate.sdpMLineIndex), String(candidate.sdp))
+	private func getPluginMediaStreamTrack(_ track: RTCMediaStreamTrack?, trackId: String?) -> PluginMediaStreamTrack? {
+		if (track == nil) {
+			return nil;
+		}
 
-		if self.rtcPeerConnection.signalingState.rawValue == RTCSignalingClosed.rawValue {
-			return
+		var currentMediaStreamTrack : PluginMediaStreamTrack? = nil;
+
+		for (_, pluginMediaTrack) in self.pluginMediaTracks {
+			if (pluginMediaTrack.rtcMediaStreamTrack.trackId == track!.trackId) {
+				currentMediaStreamTrack = pluginMediaTrack;
+				break;
+			}
 		}
 
-		self.eventListener(data: [
-			"type": "icecandidate",
-			"candidate": [
-				"sdpMid": candidate.sdpMid,
-				"sdpMLineIndex": candidate.sdpMLineIndex,
-				"candidate": candidate.sdp
-			],
-			"localDescription": [
-				"type": self.rtcPeerConnection.localDescription.type,
-				"sdp": self.rtcPeerConnection.localDescription.description
-			]
-		])
-	}
+		if (currentMediaStreamTrack == nil) {
+
+			currentMediaStreamTrack = PluginMediaStreamTrack(rtcMediaStreamTrack: track!, trackId: trackId)
 
+			currentMediaStreamTrack!.run()
 
-	func peerConnection(peerConnection: RTCPeerConnection!,
-		iceConnectionChanged newState: RTCICEConnectionState) {
-		let state_str = PluginRTCTypes.iceConnectionStates[newState.rawValue] as String!
+			// Let the plugin store it in its dictionary.
+			self.pluginMediaTracks[currentMediaStreamTrack!.id] = currentMediaStreamTrack;
 
-		NSLog("PluginRTCPeerConnection | oniceconnectionstatechange [iceConnectionState:%@]", String(state_str))
+			// Fixes issue #576
+			self.eventListenerForAddTrack(currentMediaStreamTrack!)
+		}
 
-		self.eventListener(data: [
-			"type": "iceconnectionstatechange",
-			"iceConnectionState": state_str
-		])
+		return currentMediaStreamTrack;
 	}
 
+	func getTransceiversJSON() -> [[String: Any]] {
+        if (!IsUnifiedPlan()) {
+          NSLog("PluginRTCPeerConnection#getTransceiversJSON() | transceiers is not available when using plan-b")
+          return [];
+        }
+        
+		return self.rtcPeerConnection.transceivers.map({ (transceiver: RTCRtpTransceiver) -> [String: Any] in
+//			let receiverTrack = self.getPluginMediaStreamTrack(transceiver.receiver.track, trackId: nil);
+//			let senderTrack = self.getPluginMediaStreamTrack(transceiver.sender.track, trackId: nil);
+			let transceiverHolder = self.updateTransceivers(rtcRtpTransceiver: transceiver)
+			var transceiverDesc = transceiverHolder.getJSON()
+
+			if var receiver = transceiverDesc["receiver"] as? [String: Any] {
+				if var track = receiver["track"] as? [String: Any] {
+					if let id = track["id"] as? String {
+						let pluginTrack = self.pluginMediaTracks.first { (key: String, value: PluginMediaStreamTrack) in
+							return value.originalId == id
+						}
+						if let pluginTrack = pluginTrack {
+							track["id"] = pluginTrack.value.id
+						}
+					}
+					receiver["track"] = track
+				}
+				transceiverDesc["receiver"] = receiver
+			}
 
-	func peerConnection(rtcPeerConnection: RTCPeerConnection!,
-		addedStream rtcMediaStream: RTCMediaStream!) {
-		NSLog("PluginRTCPeerConnection | onaddstream")
+			if var sender = transceiverDesc["sender"] as? [String: Any] {
+				if var track = sender["track"] as? [String: Any] {
+					if let id = track["id"] as? String {
+						let pluginTrack = self.pluginMediaTracks.first { (key: String, value: PluginMediaStreamTrack) in
+							return value.originalId == id
+						}
+						if let pluginTrack = pluginTrack {
+							track["id"] = pluginTrack.value.id
+						}
+					}
+					sender["track"] = track
+				}
+				transceiverDesc["sender"] = sender
+			}
 
-		let pluginMediaStream = PluginMediaStream(rtcMediaStream: rtcMediaStream)
+			return transceiverDesc
+		})
+	}
 
-		pluginMediaStream.run()
+	/** Called when media is received on a new stream from remote peer. */
+	func peerConnection(_ peerConnection: RTCPeerConnection, didAdd stream: RTCMediaStream) {
+		self.queue.async {
+			NSLog("PluginRTCPeerConnection | onaddstream")
 
-		// Let the plugin store it in its dictionary.
-		self.eventListenerForAddStream(pluginMediaStream: pluginMediaStream)
+			let pluginMediaStream = self.getPluginMediaStream(stream);
 
-		// Fire the 'addstream' event so the JS will create a new MediaStream.
-		self.eventListener(data: [
-			"type": "addstream",
-			"stream": pluginMediaStream.getJSON()
-		])
+			// Fire the 'addstream' event so the JS will create a new MediaStream.
+			self.eventListener([
+				"type": "addstream",
+				"streamId": pluginMediaStream!.id,
+				"stream": pluginMediaStream!.getJSON(),
+				"transceivers": self.getTransceiversJSON()
+			])
+		}
 	}
 
+	/** Called when a remote peer closes a stream. */
+	func peerConnection(_ peerConnection: RTCPeerConnection, didRemove stream: RTCMediaStream) {
+		self.queue.async {
+			NSLog("PluginRTCPeerConnection | onremovestream")
+
+			let pluginMediaStream = self.getPluginMediaStream(stream);
 
-	func peerConnection(rtcPeerConnection: RTCPeerConnection!,
-		removedStream rtcMediaStream: RTCMediaStream!) {
-		NSLog("PluginRTCPeerConnection | onremovestream")
+			self.eventListenerForRemoveStream(pluginMediaStream!)
 
-		// Let the plugin remove it from its dictionary.
-		self.eventListenerForRemoveStream(id: rtcMediaStream.label)
+			// Let the plugin remove it from its dictionary.
+			self.pluginMediaStreams[pluginMediaStream!.id] = nil;
 
-		self.eventListener(data: [
-			"type": "removestream",
-			"streamId": rtcMediaStream.label  // NOTE: No "id" property yet.
-		])
+			self.eventListener([
+				"type": "removestream",
+				"streamId": pluginMediaStream!.id,
+				"transceivers": self.getTransceiversJSON()
+			])
+		}
 	}
 
+	/** New track as been added. */
+	func peerConnection(_ peerConnection: RTCPeerConnection, didAdd rtpReceiver: RTCRtpReceiver, streams:[RTCMediaStream]) {
+		self.queue.async {
+			NSLog("PluginRTCPeerConnection | ontrack")
+
+			let pluginMediaTrack = self.getPluginMediaStreamTrack(rtpReceiver.track, trackId: nil);
 
-	func peerConnectionOnRenegotiationNeeded(peerConnection: RTCPeerConnection!) {
-		NSLog("PluginRTCPeerConnection | onnegotiationeeded")
+			// Add stream only if available in case of Unified-Plan of track event without stream
+			// TODO investigate why no stream sometimes with Unified-Plan and confirm that expexted behavior.
 
-		self.eventListener(data: [
-			"type": "negotiationneeded"
-		])
+			if (streams.isEmpty) {
+				self.eventListener([
+					"type": "track",
+					"track": pluginMediaTrack!.getJSON(),
+					"transceivers": self.getTransceiversJSON()
+				])
+			} else {
+				let pluginMediaStream = self.getPluginMediaStream(streams[0]);
+
+				// Check if pluginMediaStream had already the new track, otherwise add new track
+				if (pluginMediaStream!.hasTrack(pluginMediaTrack!) == false) {
+					pluginMediaStream!.addTrack(pluginMediaTrack!);
+				}
+
+				self.eventListener([
+					"type": "track",
+					"track": pluginMediaTrack!.getJSON(),
+					"streamId": pluginMediaStream!.id,
+					"stream": pluginMediaStream!.getJSON(),
+					"transceivers": self.getTransceiversJSON()
+				])
+			}
+		}
 	}
 
+	/** Called when the SignalingState changed. */
 
-	func peerConnection(peerConnection: RTCPeerConnection!,
-		didOpenDataChannel rtcDataChannel: RTCDataChannel!) {
-		NSLog("PluginRTCPeerConnection | ondatachannel")
+	// TODO: remove on M75
+	// This was already fixed in M-75, but note that "Issue 740501: RTCPeerConnection.onnegotiationneeded can sometimes fire multiple times in a row" was a prerequisite of Perfect Negotiation as well.
+	// https://stackoverflow.com/questions/48963787/failed-to-set-local-answer-sdp-called-in-wrong-state-kstable
+	// https://bugs.chromium.org/p/chromium/issues/detail?id=740501
+	// https://bugs.chromium.org/p/chromium/issues/detail?id=980872
+	var isNegotiating = false;
 
-		let dcId = PluginUtils.randomInt(10000, max:99999)
-		let pluginRTCDataChannel = PluginRTCDataChannel(
-			rtcDataChannel: rtcDataChannel
-		)
+	func peerConnection(_ peerConnection: RTCPeerConnection, didChange stateChanged: RTCSignalingState) {
+		self.queue.async {
+			let state_str = PluginRTCTypes.signalingStates[stateChanged.rawValue] as String?
 
-		// Store the pluginRTCDataChannel into the dictionary.
-		self.pluginRTCDataChannels[dcId] = pluginRTCDataChannel
+			NSLog("PluginRTCPeerConnection | onsignalingstatechange [signalingState:%@]", String(describing: state_str))
 
-		// Run it.
-		pluginRTCDataChannel.run()
+			self.isNegotiating = (state_str != "stable")
 
-		// Fire the 'datachannel' event so the JS will create a new RTCDataChannel.
-		self.eventListener(data: [
-			"type": "datachannel",
-			"channel": [
-				"dcId": dcId,
-				"label": rtcDataChannel.label,
-				"ordered": rtcDataChannel.isOrdered,
-				"maxPacketLifeTime": rtcDataChannel.maxRetransmitTime,
-				"maxRetransmits": rtcDataChannel.maxRetransmits,
-				"protocol": rtcDataChannel.`protocol`,
-				"negotiated": rtcDataChannel.isNegotiated,
-				"id": rtcDataChannel.streamId,
-				"readyState": PluginRTCTypes.dataChannelStates[rtcDataChannel.state.rawValue] as String!,
-				"bufferedAmount": rtcDataChannel.bufferedAmount
-			]
-		])
+			self.eventListener([
+				"type": "signalingstatechange",
+				"signalingState": state_str!
+			])
+		}
 	}
 
+	/** Called when negotiation is needed, for example ICE has restarted. */
+	func peerConnectionShouldNegotiate(_ peerConnection: RTCPeerConnection) {
+		self.queue.async {
+			NSLog("PluginRTCPeerConnection | onnegotiationeeded")
 
-	/**
-	 * Methods inherited from RTCSessionDescriptionDelegate.
-	 */
-
+			if (!self.IsUnifiedPlan() && self.isNegotiating) {
+			  NSLog("PluginRTCPeerConnection#addStream() | signalingState is stable skip nested negotiations when using plan-b")
+			  return;
+			}
 
-	func peerConnection(rtcPeerConnection: RTCPeerConnection!,
-		didCreateSessionDescription rtcSessionDescription: RTCSessionDescription!, error: NSError!) {
-		if error == nil {
-			self.onCreateDescriptionSuccessCallback(rtcSessionDescription: rtcSessionDescription)
-		} else {
-			self.onCreateDescriptionFailureCallback(error: error)
+			self.eventListener([
+				"type": "negotiationneeded"
+			])
 		}
 	}
 
+	/** Called any time the IceConnectionState changes. */
+	func peerConnection(_ peerConnection: RTCPeerConnection, didChange newState: RTCIceConnectionState) {
+		self.queue.async {
+			let state_str = PluginRTCTypes.iceConnectionStates[newState.rawValue]
 
-	func peerConnection(peerConnection: RTCPeerConnection!,
-		didSetSessionDescriptionWithError error: NSError!) {
-		if error == nil {
-			self.onSetDescriptionSuccessCallback()
-		} else {
-			self.onSetDescriptionFailureCallback(error: error)
+			NSLog("PluginRTCPeerConnection | oniceconnectionstatechange [iceConnectionState:%@]", String(describing: state_str))
+
+			self.eventListener([
+				"type": "iceconnectionstatechange",
+				"iceConnectionState": state_str as Any
+			])
 		}
 	}
 
-	/**
-	* Methods inherited from RTCStatsDelegate
-	*/
+	/** Called any time the IceGatheringState changes. */
+	func peerConnection(_ peerConnection: RTCPeerConnection, didChange newState: RTCIceGatheringState) {
+		self.queue.async {
+			let state_str = PluginRTCTypes.iceGatheringStates[newState.rawValue]
 
-	func peerConnection(peerConnection: RTCPeerConnection!,
-		didGetStats stats: [AnyObject]!) {
+			NSLog("PluginRTCPeerConnection | onicegatheringstatechange [iceGatheringState:%@]", String(describing: state_str))
 
-		var jsStats = [NSDictionary]()
+			self.eventListener([
+				"type": "icegatheringstatechange",
+				"iceGatheringState": state_str as Any
+			])
 
-		for stat in stats as NSArray {
-			var jsValues = Dictionary<String,String>()
+			if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
+				return
+			}
 
-			for pair in stat.values as! [RTCPair] {
-				jsValues[pair.key] = pair.value;
+			// Emit an empty candidate if iceGatheringState is "complete".
+			if newState.rawValue == RTCIceGatheringState.complete.rawValue && self.rtcPeerConnection.localDescription != nil {
+				self.eventListener([
+					"type": "icecandidate",
+					// NOTE: Cannot set null as value.
+					"candidate": false,
+					"localDescription": [
+						"type": RTCSessionDescription.string(for: self.rtcPeerConnection.localDescription!.type),
+						"sdp": self.rtcPeerConnection.localDescription!.sdp
+					] as [String : Any]
+				])
 			}
+		}
+	}
 
-			jsStats.append(["reportId": stat.reportId, "type": stat.type, "timestamp": stat.timestamp, "values": jsValues]);
+	/** New ice candidate has been found. */
+	func peerConnection(_ peerConnection: RTCPeerConnection, didGenerate candidate: RTCIceCandidate) {
+		self.queue.async {
+			NSLog("PluginRTCPeerConnection | onicecandidate [sdpMid:%@, sdpMLineIndex:%@, candidate:%@]",
+				  String(candidate.sdpMid!), String(candidate.sdpMLineIndex), String(candidate.sdp))
 
+			if self.rtcPeerConnection.signalingState == RTCSignalingState.closed {
+				return
+			}
+
+			self.eventListener([
+				"type": "icecandidate",
+				"candidate": [
+					"sdpMid": candidate.sdpMid as Any,
+					"sdpMLineIndex": candidate.sdpMLineIndex,
+					"candidate": candidate.sdp
+				],
+				"localDescription": [
+					"type": RTCSessionDescription.string(for: self.rtcPeerConnection.localDescription!.type),
+					"sdp": self.rtcPeerConnection.localDescription!.sdp
+				] as [String : Any]
+			])
 		}
+	}
+
+	/** Called when a group of local Ice candidates have been removed. */
+	func peerConnection(_ peerConnection: RTCPeerConnection, didRemove candidates: [RTCIceCandidate]) {
+		NSLog("PluginRTCPeerConnection | removeicecandidates")
+	}
 
-		self.onGetStatsCallback(array: jsStats);
+	/** New data channel has been opened. */
+	func peerConnection(_ peerConnection: RTCPeerConnection, didOpen rtcDataChannel: RTCDataChannel) {
+		self.queue.async {
+			NSLog("PluginRTCPeerConnection | ondatachannel")
+
+			let dcId = PluginUtils.randomInt(10000, max:99999)
+			let pluginRTCDataChannel = PluginRTCDataChannel(
+				rtcDataChannel: rtcDataChannel
+			)
+
+			// Store the pluginRTCDataChannel into the dictionary.
+			self.pluginRTCDataChannels[dcId] = pluginRTCDataChannel
+
+			// Run it.
+			pluginRTCDataChannel.run()
+
+			// Fire the 'datachannel' event so the JS will create a new RTCDataChannel.
+			self.eventListener([
+				"type": "datachannel",
+				"channel": [
+					"dcId": dcId,
+					"label": rtcDataChannel.label,
+					"ordered": rtcDataChannel.isOrdered,
+					"maxPacketLifeTime": rtcDataChannel.maxPacketLifeTime,
+					"maxRetransmits": rtcDataChannel.maxRetransmits,
+					"protocol": rtcDataChannel.`protocol`,
+					"negotiated": rtcDataChannel.isNegotiated,
+					"id": rtcDataChannel.channelId,
+					"readyState": PluginRTCTypes.dataChannelStates[rtcDataChannel.readyState.rawValue] as Any,
+					"bufferedAmount": rtcDataChannel.bufferedAmount
+				]
+			])
+		}
 	}
+    
+    // TODO: Is this event required at all?
+    /* Called when transceier will start receiving data. */
+    /* func peerConnection(_ peerConnection: RTCPeerConnection, didStartReceivingOn transceiver: RTCRtpTransceiver) {
+        NSLog("PluginRTCPeerConnection | didStartReceivingOn")
+        
+        // NOTE: Is this a new transceiver or was it created already?
+        var existingTransceiver: PluginRTCRtpTransceiver? = nil
+        
+		// TODO: Is it correct reusing transceiver instance with same mid?
+        for (_, pluginTransceiver) in pluginRTCRtpTransceivers {
+            if (transceiver.mid == pluginTransceiver.rtcRtpTransceiver!.mid) {
+                existingTransceiver = pluginTransceiver
+                break
+            }
+        }
+        
+        if (existingTransceiver == nil) {
+            NSLog("PluginRTCPeerConnection | Info: No existing transceiver matching mid: %@", transceiver.mid)
+
+			let tcId = PluginUtils.randomInt(10000, max: 99999)
+			let pluginRTCRtpTransceiver = PluginRTCRtpTransceiver(transceiver)
+
+			self.pluginRTCRtpTransceivers[tcId] = pluginRTCRtpTransceiver
+
+			var currentDirection = RTCRtpTransceiverDirection.inactive
+            transceiver.currentDirection(&currentDirection)
+
+			self.eventListener([
+				"type": "transceiver",
+				"transceiver": [
+                    "tcId": tcId,
+                    "mid": transceiver.mid,
+					"currentDirection": PluginRTCRtpTransceiver.directionToString(currentDirection),
+                	"direction": PluginRTCRtpTransceiver.directionToString(transceiver.direction),
+                	"stopped": transceiver.isStopped
+				]
+			])
+        } else {
+			// TODO: Can this situation happen?
+            NSLog("PluginRTCPeerConnection | Info: Found existing transceiver matching mid: %@", transceiver.mid)
+        }
+    } */
 }
diff --git a/src/PluginRTCPeerConnectionConfig.swift b/src/PluginRTCPeerConnectionConfig.swift
index 28b9088a..1f08a207 100644
--- a/src/PluginRTCPeerConnectionConfig.swift
+++ b/src/PluginRTCPeerConnectionConfig.swift
@@ -2,45 +2,161 @@ import Foundation
 
 
 class PluginRTCPeerConnectionConfig {
-	private var iceServers: [RTCICEServer] = []
 
+	fileprivate let allowedSdpSemantics = [
+		"plan-b": RTCSdpSemantics.planB,
+		"unified-plan": RTCSdpSemantics.unifiedPlan
+	]
+
+	fileprivate let allowedBundlePolicy = [
+		"balanced": RTCBundlePolicy.balanced,
+		"max-compat": RTCBundlePolicy.maxCompat,
+		"max-bundle": RTCBundlePolicy.maxBundle
+	]
+
+	fileprivate let allowedIceTransportPolicy = [
+		"all": RTCIceTransportPolicy.all,
+		"relay": RTCIceTransportPolicy.relay
+	]
+
+	fileprivate let allowedRtcpMuxPolicy = [
+		"require": RTCRtcpMuxPolicy.require,
+		"negotiate": RTCRtcpMuxPolicy.negotiate,
+	]
+
+	fileprivate var rtcConfiguration: RTCConfiguration
 
 	init(pcConfig: NSDictionary?) {
 		NSLog("PluginRTCPeerConnectionConfig#init()")
 
-		let iceServers = pcConfig?.objectForKey("iceServers") as? [NSDictionary]
+		self.rtcConfiguration = RTCConfiguration()
 
-		if iceServers == nil {
-			return
-		}
+		/*
+		 Since Chrome 65, this has been an experimental feature that you can opt-in to by passing
+		 sdpSemantics:'unified-plan' to the RTCPeerConnection constructor, with transceivers added in M69.
+		 History: https://crbug.com/799030. Canary/Dev experiments making Unified Plan the default behavior started in 71.
+		 The target is to release Unified Plan by default in 72 Stable. For more information, see https://webrtc.org/web-apis/chrome/unified-plan/.
+		 See: https://www.chromestatus.com/feature/5723303167655936
+		*/
+
+		let sdpSemanticsConfig = pcConfig?.object(forKey: "sdpSemantics") as? String;
+		let sdpSemantics = (sdpSemanticsConfig != nil && allowedSdpSemantics[sdpSemanticsConfig!] != nil) ?
+			allowedSdpSemantics[sdpSemanticsConfig!] : RTCSdpSemantics.unifiedPlan
+
+		rtcConfiguration.sdpSemantics = sdpSemantics!;
+
+		/*
+		 Specifies how to handle negotiation of candidates when the remote peer is not compatible with the SDP BUNDLE standard.
+		 This must be one of the values from the enum RTCBundlePolicy. If this value isn't included in the dictionary, "balanced" is assumed.
+
+		 "balanced"    On BUNDLE-aware connections, the ICE agent should gather candidates for all of the media types in use (audio, video, and data). Otherwise, the ICE agent should only negotiate one audio and video track on separate transports.
+		 "max-compat"    The ICE agent should gather candidates for each track, using separate transports to negotiate all media tracks for connections which aren't BUNDLE-compatible.
+		 "max-bundle"    The ICE agent should gather candidates for just one track. If the connection isn't BUNDLE-compatible, then the ICE agent should negotiate just one media track.
+		 */
+
+		let bundlePolicyConfig = pcConfig?.object(forKey: "bundlePolicy") as? String;
+		let bundlePolicy = (bundlePolicyConfig != nil && allowedBundlePolicy[bundlePolicyConfig!] != nil) ?
+								allowedBundlePolicy[bundlePolicyConfig!] : RTCBundlePolicy.balanced
+
+		rtcConfiguration.bundlePolicy = bundlePolicy!;
+
+		/*
+		 An Array of objects of type RTCCertificate which are used by the connection for authentication.
+		 If this property isn't specified, a set of certificates is generated automatically for each RTCPeerConnection instance.
+		 Although only one certificate is used by a given connection, providing certificates for multiple algorithms may improve
+		 the odds of successfully connecting in some circumstances. See Using certificates below for additional information.
+		 This configuration option cannot be changed after it is first specified; once the certificates have been set,
+		  this property is ignored in future calls to RTCPeerConnection.setConfiguration().
+		 */
+		// TODO certificates
+
+		/*
+		 An unsigned 16-bit integer value which specifies the size of the prefetched ICE candidate pool.
+		 The default value is 0 (meaning no candidate prefetching will occur).
+		 You may find in some cases that connections can be established more quickly by allowing the ICE agent
+		 to start fetching ICE candidates before you start trying to connect, so that they're already available
+		 for inspection when RTCPeerConnection.setLocalDescription() is called.
+		 Changing the size of the ICE candidate pool may trigger the beginning of ICE gathering.
+		 */
+
+		let iceCandidatePoolSizeConfig = pcConfig?.object(forKey: "iceCandidatePoolSize") as? Int32;
+		rtcConfiguration.iceCandidatePoolSize = iceCandidatePoolSizeConfig != nil ? iceCandidatePoolSizeConfig! : 0;
 
-		for iceServer: NSDictionary in iceServers! {
-			let url = iceServer.objectForKey("url") as? String
-			let username = iceServer.objectForKey("username") as? String ?? ""
-			let password = iceServer.objectForKey("credential") as? String ?? ""
+		/*
+		 The current ICE transport policy; this must be one of the values from the RTCIceTransportPolicy enum.
+		 If this isn't specified, "all" is assumed.
 
-			if (url != nil) {
-				NSLog("PluginRTCPeerConnectionConfig#init() | adding ICE server [url:'%@', username:'%@', password:'******']",
-					String(url!), String(username))
+		 - "all"    All ICE candidates will be considered.
+		 - "public"    Only ICE candidates with public IP addresses will be considered. Removed from the specification's May 13, 2016 working draft
+		 - "relay"     Only ICE candidates whose IP addresses are being relayed, such as those being passed through a TURN server, will be considered.
+		 */
 
-				self.iceServers.append(RTCICEServer(
-					URI: NSURL(string: url!),
-					username: username,
-					password: password
-				))
+		let iceTransportPolicyConfig = pcConfig?.object(forKey: "iceTransportPolicy") as? String;
+		let iceTransportPolicy = (iceTransportPolicyConfig != nil && allowedIceTransportPolicy[iceTransportPolicyConfig!] != nil) ?
+									allowedIceTransportPolicy[iceTransportPolicyConfig!] : RTCIceTransportPolicy.all
+
+		rtcConfiguration.iceTransportPolicy = iceTransportPolicy!;
+
+		/*
+		 The RTCP mux policy to use when gathering ICE candidates, in order to support non-multiplexed RTCP.
+		 The value must be one of those from the RTCRtcpMuxPolicy enum. The default is "require".
+
+		 - "negotiate"    Instructs the ICE agent to gather both RTP and RTCP candidates. If the remote peer can multiplex RTCP,
+						 then RTCP candidates are multiplexed atop the corresponding RTP candidates. Otherwise, both the RTP and RTCP candidates are returned, separately.
+		 - "require"    Tells the ICE agent to gather ICE candidates for only RTP, and to multiplex RTCP atop them.
+						 If the remote peer doesn't support RTCP multiplexing, then session negotiation fails.
+		 */
+
+		let rtcpMuxPolicyConfig = pcConfig?.object(forKey: "rtcpMuxPolicy") as? String;
+		let rtcpMuxPolicy = (rtcpMuxPolicyConfig != nil && allowedRtcpMuxPolicy[rtcpMuxPolicyConfig!] != nil) ?
+								allowedRtcpMuxPolicy[rtcpMuxPolicyConfig!] : RTCRtcpMuxPolicy.require
+
+		rtcConfiguration.rtcpMuxPolicy = rtcpMuxPolicy!;
+
+		/*
+		 A DOMString which specifies the target peer identity for the RTCPeerConnection.
+		 If this value is set (it defaults to null), the RTCPeerConnection will not connect to a remote peer
+		 unless it can successfully authenticate with the given name.
+		 */
+		// TODO peerIdentity
+
+		/*
+		An array of RTCIceServer objects, each describing one server which may be used by the ICE agent;
+		these are typically STUN and/or TURN servers. If this isn't specified, the connection attempt
+		will be made with no STUN or TURN server available, which limits the connection to local peers.
+		*/
+		let iceServers = pcConfig?.object(forKey: "iceServers") as? [NSDictionary]
+
+		if iceServers != nil {
+			for iceServer: NSDictionary in iceServers! {
+				let urlsConfig = (iceServer.object(forKey: "url") != nil ?
+						iceServer.object(forKey: "url") : iceServer.object(forKey: "urls"))
+
+				let urls = urlsConfig is String ? [urlsConfig as? String ?? ""] : urlsConfig as? [String] ?? nil;
+				let username = iceServer.object(forKey: "username") as? String ?? ""
+				let password = iceServer.object(forKey: "credential") as? String ?? ""
+
+				if (urls != nil && urls!.count > 0) {
+					NSLog("PluginRTCPeerConnectionConfig#init() | adding ICE server [url:'%@', username:'%@', password:'******']",
+						  String(urls![0]), String(username))
+
+					self.rtcConfiguration.iceServers.append(RTCIceServer(
+						urlStrings: urls!,
+						username: username,
+						credential: password
+					))
+				}
 			}
 		}
 	}
 
-
 	deinit {
 		NSLog("PluginRTCPeerConnectionConfig#deinit()")
 	}
 
+	func getConfiguration() -> RTCConfiguration {
+		NSLog("PluginRTCPeerConnectionConfig#getConfiguration()")
 
-	func getIceServers() -> [RTCICEServer] {
-		NSLog("PluginRTCPeerConnectionConfig#getIceServers()")
-
-		return self.iceServers
+		return self.rtcConfiguration
 	}
 }
diff --git a/src/PluginRTCPeerConnectionConstraints.swift b/src/PluginRTCPeerConnectionConstraints.swift
index ed5f8ccd..2d239676 100644
--- a/src/PluginRTCPeerConnectionConstraints.swift
+++ b/src/PluginRTCPeerConnectionConstraints.swift
@@ -2,51 +2,132 @@ import Foundation
 
 
 class PluginRTCPeerConnectionConstraints {
-	private var constraints: RTCMediaConstraints
-
+	fileprivate var constraints: RTCMediaConstraints
 
 	init(pcConstraints: NSDictionary?) {
 		NSLog("PluginRTCPeerConnectionConstraints#init()")
+		var mandatoryConstraints: [String : String] = [:]
+		var optionalConstraints: [String : String] = [:]
 
-		if pcConstraints == nil {
-			self.constraints = RTCMediaConstraints()
-			return
-		}
-
-		var	offerToReceiveAudio = pcConstraints?.objectForKey("offerToReceiveAudio") as? Bool
-		var	offerToReceiveVideo = pcConstraints?.objectForKey("offerToReceiveVideo") as? Bool
-
-		if offerToReceiveAudio == nil && offerToReceiveVideo == nil {
-			self.constraints = RTCMediaConstraints()
-			return
+		// Handle constraints at the root of pcConstraints.mandatory
+		let _mandatoryConstraints = pcConstraints?.object(forKey: "mandatory") as? NSDictionary
+		if _mandatoryConstraints != nil {
+			mandatoryConstraints = PluginRTCPeerConnectionConstraints.parseConstraints(constraints: _mandatoryConstraints!)
 		}
 
-		if offerToReceiveAudio == nil {
-			offerToReceiveAudio = false
+		// Handle constraints at the root of pcConstraints.optional
+		let _optionalConstraints = pcConstraints?.object(forKey: "optional") as? NSArray
+		if _optionalConstraints != nil {
+			optionalConstraints = PluginRTCPeerConnectionConstraints.parseOptionalConstraints(optionalConstraints: _optionalConstraints!)
 		}
 
-		if offerToReceiveVideo == nil {
-			offerToReceiveVideo = false
+		// Handle constraints at the root of pcConstraints
+		if pcConstraints != nil && _optionalConstraints == nil && _mandatoryConstraints == nil  {
+			mandatoryConstraints = PluginRTCPeerConnectionConstraints.parseConstraints(constraints: pcConstraints!)
 		}
 
-		NSLog("PluginRTCPeerConnectionConstraints#init() | [offerToReceiveAudio:%@, offerToReceiveVideo:%@]",
-			String(offerToReceiveAudio!), String(offerToReceiveVideo!))
+		NSLog("PluginRTCPeerConnectionConstraints#init() | [mandatoryConstraints:%@, optionalConstraints:%@]",
+			mandatoryConstraints, optionalConstraints)
 
-		self.constraints = RTCMediaConstraints(
-			mandatoryConstraints: [
-				RTCPair(key: "OfferToReceiveAudio", value: offerToReceiveAudio == true ? "true" : "false"),
-				RTCPair(key: "OfferToReceiveVideo", value: offerToReceiveVideo == true ? "true" : "false")
-			],
-			optionalConstraints: []
+		self.constraints = RTCMediaConstraints.init(
+			mandatoryConstraints: mandatoryConstraints,
+			optionalConstraints: optionalConstraints
 		)
 	}
 
+	// Custom proprietary constrains in libwebrtc
+	fileprivate static let allowedOptionalConstraints : Array = [
+		// Temporary pseudo-constraints used to enable DTLS-SRT
+		"DtlsSrtpKeyAgreement",
+		// Constraint to enable IPv6
+		"googIPv6",
+		// Constraint to enable combined audio+video bandwidth estimation.
+		"googImprovedWifiBwe",
+		// Temporary pseudo-constraint for enabling DSCP
+		"googDscp",
+		// If enabled, will lower outgoing video quality and video resolution
+		"googCpuOveruseDetection",
+		// Min CPU load (percents), used in pair with
+		"googCpuUnderuseThreshold",
+		// Max CPU (percents), used in pair with googCpuOveruseDetection
+		"googCpuOveruseThreshold",
+		// Cause video track to be automatically disabled if the detected bandwidth drops below "min bitrate"
+		"googSuspendBelowMinBitrate",
+		//"googScreencastMinBitrate"
+	]
+
+	fileprivate static func parseOptionalConstraints(optionalConstraints: NSArray) -> [String : String] {
+		var _constraints : [String : String] = [:]
+
+		for optionalConstraint in optionalConstraints {
+			if (optionalConstraint is NSDictionary) {
+				for (key, value) in optionalConstraint as! NSDictionary {
+					let finalKey: String = key as! String;
+
+					// Parse only allowedOptionalConstraints
+					if (allowedOptionalConstraints.firstIndex(of: finalKey) != nil) {
+					   if value is Int32 {
+						   _constraints[finalKey] = String(describing: value);
+					   } else if value is Bool {
+						   _constraints[finalKey] = value as! Bool ? "true" : "false"
+					   } else if value is String {
+						   _constraints[finalKey] = (value as! String)
+					   }
+					}
+				}
+			}
+		}
+
+		return _constraints;
+	}
+
+	// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createOffer#RTCOfferOptions_dictionary
+	// TODO TODO voiceActivityDetection This option defaults to true
+	fileprivate static let allowedConstraints : Array = [
+		"IceRestart",
+		"OfferToReceiveVideo",
+		"OfferToReceiveAudio",
+		"voiceActivityDetection"
+	]
+
+	fileprivate static func parseConstraints(constraints: NSDictionary) -> [String : String] {
+		var _constraints : [String : String] = [:]
+
+		for (key, value) in constraints {
+			var finalValue: String,
+				finalKey: String = key as! String;
+
+			if value is Bool {
+				finalValue = value as! Bool ? "true" : "false"
+			} else if value is String {
+				finalValue = value as! String
+			} else {
+				continue
+			}
+
+			// Handle Spec for offerToReceiveAudio|offerToReceiveVideo but
+			// libwebrtc still use OfferToReceiveAudio|OfferToReceiveVideo
+			if (finalKey == "offerToReceiveAudio") {
+				finalKey =  "OfferToReceiveAudio";
+			} else if (finalKey == "offerToReceiveVideo") {
+				finalKey =  "OfferToReceiveVideo";
+			} else if (finalKey == "iceRestart") {
+				finalKey =  "IceRestart";
+			}
+
+			// Filter to avoid injection
+			if (allowedConstraints.firstIndex(of: finalKey) != nil) {
+				_constraints[finalKey] = finalValue
+			}
+		}
+
+		return _constraints;
+	}
 
 	deinit {
 		NSLog("PluginRTCPeerConnectionConstraints#deinit()")
 	}
 
-
 	func getConstraints() -> RTCMediaConstraints {
 		NSLog("PluginRTCPeerConnectionConstraints#getConstraints()")
 
diff --git a/src/PluginRTCRtpCodecParameters.swift b/src/PluginRTCRtpCodecParameters.swift
new file mode 100644
index 00000000..14c76fe6
--- /dev/null
+++ b/src/PluginRTCRtpCodecParameters.swift
@@ -0,0 +1,21 @@
+import Foundation
+
+class PluginRTCRtpCodecParameters : NSObject {
+	var rtcRtpCodecParameters: RTCRtpCodecParameters
+
+	init(_ rtcRtpCodecParameters: RTCRtpCodecParameters) {
+		self.rtcRtpCodecParameters = rtcRtpCodecParameters
+	}
+
+	func getJSON() -> NSDictionary {
+		return [
+			"kind": self.rtcRtpCodecParameters.kind,
+			"name": self.rtcRtpCodecParameters.name,
+			"payloadType": self.rtcRtpCodecParameters.payloadType,
+			"mimeType": NSString(format: "%@/%@", self.rtcRtpCodecParameters.name, self.rtcRtpCodecParameters.kind),
+			"clockRate": (self.rtcRtpCodecParameters.clockRate ?? nil) as Any,
+			"channels": (self.rtcRtpCodecParameters.numChannels ?? nil) as Any,
+			"sdpFmtpLine": self.rtcRtpCodecParameters.parameters
+		]
+	}
+}
diff --git a/src/PluginRTCRtpEncodingParameters.swift b/src/PluginRTCRtpEncodingParameters.swift
new file mode 100644
index 00000000..93bd4df8
--- /dev/null
+++ b/src/PluginRTCRtpEncodingParameters.swift
@@ -0,0 +1,19 @@
+import Foundation
+
+class PluginRTCRtpEncodingParameters : NSObject {
+	var rtcRtpEncodingParameters: RTCRtpEncodingParameters
+	
+	init(_ rtcRtpEncodingParameters: RTCRtpEncodingParameters) {
+		self.rtcRtpEncodingParameters = rtcRtpEncodingParameters
+	}
+		
+	func getJSON() -> NSDictionary {
+		return [
+			"rid": self.rtcRtpEncodingParameters.rid,
+			"active": self.rtcRtpEncodingParameters.isActive,
+			"maxBitrate": self.rtcRtpEncodingParameters.maxBitrateBps,
+			"minBitrate": self.rtcRtpEncodingParameters.minBitrateBps,
+			"scaleResolutionDownBy": self.rtcRtpEncodingParameters.scaleResolutionDownBy
+		]
+	}
+}
diff --git a/src/PluginRTCRtpParameters.swift b/src/PluginRTCRtpParameters.swift
new file mode 100644
index 00000000..7e1e5637
--- /dev/null
+++ b/src/PluginRTCRtpParameters.swift
@@ -0,0 +1,22 @@
+import Foundation
+
+class PluginRTCRtpParameters : NSObject {
+	var rtcRtpParameters: RTCRtpParameters
+
+	init(_ rtcRtpParameters: RTCRtpParameters) {
+		self.rtcRtpParameters = rtcRtpParameters
+	}
+
+	func getJSON() -> NSDictionary {
+		let codecs = self.rtcRtpParameters.codecs.map { (codecParams: RTCRtpCodecParameters) -> NSDictionary in
+			return PluginRTCRtpCodecParameters(codecParams).getJSON()
+		}
+		let encodings = self.rtcRtpParameters.encodings.map { (encodingParams: RTCRtpEncodingParameters) -> NSDictionary in
+			return PluginRTCRtpEncodingParameters(encodingParams).getJSON()
+		}
+		return [
+			"codecs": codecs,
+			"encodings": encodings
+		]
+	}
+}
diff --git a/src/PluginRTCRtpReceiver.swift b/src/PluginRTCRtpReceiver.swift
new file mode 100644
index 00000000..51d87adc
--- /dev/null
+++ b/src/PluginRTCRtpReceiver.swift
@@ -0,0 +1,28 @@
+import Foundation
+
+class PluginRTCRtpReceiver : NSObject {
+	var rtpReceiver: RTCRtpReceiver
+	var id: Int
+	
+	init(_ rtpReceiver: RTCRtpReceiver, _ id: Int) {
+		self.rtpReceiver = rtpReceiver
+		self.id = id != 0 ? id : Int.random(in: 0...10000)
+    }
+	
+	func getJSON() -> [String: Any] {
+		let track = self.rtpReceiver.track != nil ? [
+			"id": self.rtpReceiver.track!.trackId,
+			   "kind": self.rtpReceiver.track!.kind,
+			   "readyState": self.rtpReceiver.track!.readyState == RTCMediaStreamTrackState.live ? "live" : "ended",
+			   "enabled": self.rtpReceiver.track!.isEnabled,
+			   "label": self.rtpReceiver.track!.trackId
+		] : nil
+		return [
+			"id": self.id,
+			"receiverId": self.rtpReceiver.receiverId,
+			"params": PluginRTCRtpParameters(self.rtpReceiver.parameters).getJSON(),
+			"track": track
+		]
+	}
+
+}
diff --git a/src/PluginRTCRtpSender.swift b/src/PluginRTCRtpSender.swift
new file mode 100644
index 00000000..4527824b
--- /dev/null
+++ b/src/PluginRTCRtpSender.swift
@@ -0,0 +1,78 @@
+import Foundation
+
+class PluginRTCRtpSender : NSObject {
+	var rtpSender: RTCRtpSender
+	var id: Int
+
+	init(_ rtpSender: RTCRtpSender, _ id: Int) {
+		self.rtpSender = rtpSender
+		self.id = id != 0 ? id : Int.random(in: 0...10000)
+	}
+
+	func setParameters(_ params: [String : NSObject],
+		_ callback: @escaping (_ data: NSDictionary) -> Void,
+		_ errback: @escaping (_ error: Error) -> Void
+	) {
+		let oldParameters : RTCRtpParameters! = rtpSender.parameters
+		let newParameters : RTCRtpParameters! = RTCRtpParameters()
+		newParameters.transactionId = oldParameters.transactionId
+
+		let encodings = params["encodings"] as? [[String:Any]]
+		if encodings != nil {
+			for e in encodings! {
+				let p : RTCRtpEncodingParameters = RTCRtpEncodingParameters()
+				if e["maxBitrate"] as? NSNumber != nil {
+					p.maxBitrateBps = e["maxBitrate"] as? NSNumber
+				}
+				if e["scaleResolutionDownBy"] as? NSNumber != nil {
+					p.scaleResolutionDownBy = e["scaleResolutionDownBy"] as? NSNumber
+				}
+				newParameters.encodings.append(p)
+			}
+		}
+
+		rtpSender.parameters = newParameters
+
+		var newCodecs : [NSDictionary] = []
+		for c in newParameters.codecs {
+			let codec = PluginRTCRtpCodecParameters(c)
+			newCodecs.append(codec.getJSON())
+		}
+
+		var newEncodings : [NSDictionary] = []
+		for e in newParameters.encodings {
+			let encoding = PluginRTCRtpEncodingParameters(e)
+			newEncodings.append(encoding.getJSON())
+		}
+
+		let newHeaderExtensions : [NSDictionary] = []
+
+		let result : [String : Any] = [
+			"codecs": newCodecs,
+			"encodings": newEncodings,
+			"headerExtensions": newHeaderExtensions
+		]
+		callback(result as NSDictionary)
+	}
+	
+	func replaceTrack(_ pluginMediaStreamTrack: PluginMediaStreamTrack?) {
+		let rtcMediaStreamTrack = pluginMediaStreamTrack?.rtcMediaStreamTrack
+		self.rtpSender.track = rtcMediaStreamTrack
+	}
+
+	func getJSON() -> [String: Any] {
+		let track = self.rtpSender.track != nil ? [
+			"id": self.rtpSender.track!.trackId,
+			   "kind": self.rtpSender.track!.kind,
+			   "readyState": self.rtpSender.track!.readyState == RTCMediaStreamTrackState.live ? "live" : "ended",
+			   "enabled": self.rtpSender.track!.isEnabled,
+			   "label": self.rtpSender.track!.trackId
+		] : nil
+		return [
+			"id": self.id,
+			"senderId": self.rtpSender.senderId,
+			"params": PluginRTCRtpParameters(self.rtpSender.parameters).getJSON(),
+			"track": track
+		]
+	}
+}
diff --git a/src/PluginRTCRtpTransceiver.swift b/src/PluginRTCRtpTransceiver.swift
new file mode 100644
index 00000000..796b0726
--- /dev/null
+++ b/src/PluginRTCRtpTransceiver.swift
@@ -0,0 +1,170 @@
+import Foundation
+
+class PluginRTCRtpTransceiver : NSObject {
+    // NOTE: ID used to reference this native transceiver from JS.
+    var id: Int
+    var rtcRtpTransceiver: RTCRtpTransceiver?
+	var pluginRTCRtpReceiver: PluginRTCRtpReceiver?
+	var pluginRTCRtpSender: PluginRTCRtpSender?
+    
+    init(_ rtcRtpTransceiver: RTCRtpTransceiver) {
+        NSLog("PluginRTCRtpTransceiver#init(rtcRtpTransceiver)")
+        
+        // TODO: Using random ID could cause conflicts.
+        self.id = Int.random(in: 0...10000)
+        self.rtcRtpTransceiver = rtcRtpTransceiver
+		self.pluginRTCRtpSender = PluginRTCRtpSender(rtcRtpTransceiver.sender, 0)
+		self.pluginRTCRtpReceiver = PluginRTCRtpReceiver(rtcRtpTransceiver.receiver, 0)
+        
+        super.init()
+    }
+
+    init(
+        rtcPeerConnection: RTCPeerConnection,
+        mediaType: RTCRtpMediaType,
+        options: NSDictionary?,
+		transceiverId: Int,
+		senderId: Int,
+		receiverId: Int
+    ) {
+        NSLog("PluginRTCRtpTransceiver#init(mediaType)")
+        
+        let rtcRtpTransceiverInit = PluginRTCRtpTransceiver.initFromOptionsDictionary(options)
+
+		self.id = transceiverId != 0 ? transceiverId : Int.random(in: 0...10000)
+        self.rtcRtpTransceiver = rtcPeerConnection.addTransceiver(of: mediaType, init: rtcRtpTransceiverInit)
+
+        if self.rtcRtpTransceiver == nil {
+			NSLog("PluginRTCRtpTransceiver#init(mediaType) | rtcPeerConnection.addTransceiver() failed")
+		} else {
+			if self.rtcRtpTransceiver?.sender != nil {
+				self.pluginRTCRtpSender = PluginRTCRtpSender(self.rtcRtpTransceiver!.sender, senderId)
+			}
+			if self.rtcRtpTransceiver?.receiver != nil {
+				self.pluginRTCRtpReceiver = PluginRTCRtpReceiver(self.rtcRtpTransceiver!.receiver, receiverId)
+			}
+		}
+        
+        super.init()
+    }
+
+    init(
+        rtcPeerConnection: RTCPeerConnection, 
+        mediaStreamTrack: RTCMediaStreamTrack, 
+        options: NSDictionary?,
+		transceiverId: Int,
+		senderId: Int,
+		receiverId: Int
+    ) {
+        NSLog("PluginRTCRtpTransceiver#init(mediaStreamTrack)")
+        
+        let rtcRtpTransceiverInit = PluginRTCRtpTransceiver.initFromOptionsDictionary(options)
+
+		self.id = transceiverId != 0 ? transceiverId : Int.random(in: 0...10000)
+        self.rtcRtpTransceiver = rtcPeerConnection.addTransceiver(with: mediaStreamTrack, init: rtcRtpTransceiverInit)
+
+        if self.rtcRtpTransceiver == nil {
+			NSLog("PluginRTCRtpTransceiver#init(mediaStream) | rtcPeerConnection.addTransceiver() failed")
+		} else {
+			if self.rtcRtpTransceiver?.sender != nil {
+				self.pluginRTCRtpSender = PluginRTCRtpSender(self.rtcRtpTransceiver!.sender, senderId)
+			}
+			if self.rtcRtpTransceiver?.receiver != nil {
+				self.pluginRTCRtpReceiver = PluginRTCRtpReceiver(self.rtcRtpTransceiver!.receiver, receiverId)
+			}
+		}
+
+        super.init()
+    }
+
+    deinit {
+		NSLog("PluginRTCRtpTransceiver#deinit()")
+	}
+
+    func stop() {
+        self.rtcRtpTransceiver!.stopInternal()
+    }
+    
+    func setDirection(direction: String) {
+        self.rtcRtpTransceiver?.setDirection(PluginRTCRtpTransceiver.stringToDirection(direction), error: nil)
+    }
+    
+    static func stringToDirection(_ direction: String) -> RTCRtpTransceiverDirection {
+        switch direction {
+        case "inactive":
+            return RTCRtpTransceiverDirection.inactive
+        case "recvonly":
+            return RTCRtpTransceiverDirection.recvOnly
+        case "sendonly":
+            return RTCRtpTransceiverDirection.sendOnly
+        case "sendrecv":
+            return RTCRtpTransceiverDirection.sendRecv
+        case "stopped":
+            return RTCRtpTransceiverDirection.stopped
+        default:
+            NSLog("PluginRTCRtpTransceiver#stringToDirection() | Unrecognized direction value: @%", direction)
+            return RTCRtpTransceiverDirection.inactive
+        }
+    }
+    
+    static func directionToString(_ direction: RTCRtpTransceiverDirection) -> String {
+        switch direction {
+        case RTCRtpTransceiverDirection.inactive:
+            return "inactive"
+        case RTCRtpTransceiverDirection.recvOnly:
+            return "recvonly"
+        case RTCRtpTransceiverDirection.sendOnly:
+            return "sendonly"
+        case RTCRtpTransceiverDirection.sendRecv:
+            return "sendrecv"
+        case RTCRtpTransceiverDirection.stopped:
+            return "stopped"
+        }
+    }
+
+    static func initFromOptionsDictionary(_ options: NSDictionary?) -> RTCRtpTransceiverInit {
+        let rtcRtpTransceiverInit = RTCRtpTransceiverInit();
+
+        if options?.object(forKey: "direction") != nil {
+            let direction = options!.object(forKey: "direction") as! String
+
+            rtcRtpTransceiverInit.direction = PluginRTCRtpTransceiver.stringToDirection(direction)
+        }
+
+        if options?.object(forKey: "streams") != nil {
+            let streamIds = options!.object(forKey: "streams") as! [String]
+
+            rtcRtpTransceiverInit.streamIds = streamIds
+        }
+
+        if options?.object(forKey: "sendEncodings") != nil {
+            let encodings = options!.object(forKey: "sendEncodings") as! [NSDictionary]
+            rtcRtpTransceiverInit.sendEncodings = encodings.map({ (encoding: NSDictionary) -> RTCRtpEncodingParameters in
+                let encodingParameters = RTCRtpEncodingParameters()
+                encodingParameters.isActive = encoding["active"] as? Bool ?? true
+                encodingParameters.maxBitrateBps = encoding["maxBitrate"] as? NSNumber
+                encodingParameters.maxFramerate = encoding["maxFramerate"] as? NSNumber
+                encodingParameters.rid = encoding["rid"] as? String
+                encodingParameters.scaleResolutionDownBy = encoding["scaleResolutionDownBy"] as? NSNumber
+                return encodingParameters
+            })
+        }
+
+        return rtcRtpTransceiverInit
+    }
+
+	func getJSON() -> [String: Any] {
+		var currentDirection = RTCRtpTransceiverDirection.inactive
+		self.rtcRtpTransceiver?.currentDirection(&currentDirection)
+
+		return [
+			"id": self.id,
+			"mid": self.rtcRtpTransceiver?.mid,
+			"stopped": self.rtcRtpTransceiver?.isStopped,
+			"direction": PluginRTCRtpTransceiver.directionToString(rtcRtpTransceiver!.direction),
+			"currentDirection": PluginRTCRtpTransceiver.directionToString(currentDirection),
+			"receiver": self.pluginRTCRtpReceiver?.getJSON(),
+			"sender": self.pluginRTCRtpSender?.getJSON()
+		]
+	}
+}
diff --git a/src/PluginRTCTypes.swift b/src/PluginRTCTypes.swift
index e8a1d763..9ea30394 100644
--- a/src/PluginRTCTypes.swift
+++ b/src/PluginRTCTypes.swift
@@ -1,38 +1,39 @@
 struct PluginRTCTypes {
 	static let signalingStates = [
-		RTCSignalingStable.rawValue:             "stable",
-		RTCSignalingHaveLocalOffer.rawValue:     "have-local-offer",
-		RTCSignalingHaveLocalPrAnswer.rawValue:  "have-local-pranswer",
-		RTCSignalingHaveRemoteOffer.rawValue:    "have-remote-offer",
-		RTCSignalingHaveRemotePrAnswer.rawValue: "have-remote-pranswer",
-		RTCSignalingClosed.rawValue:             "closed"
+		RTCSignalingState.stable.rawValue:             "stable",
+		RTCSignalingState.haveLocalOffer.rawValue:     "have-local-offer",
+		RTCSignalingState.haveLocalPrAnswer.rawValue:  "have-local-pranswer",
+		RTCSignalingState.haveRemoteOffer.rawValue:    "have-remote-offer",
+		RTCSignalingState.haveRemotePrAnswer.rawValue: "have-remote-pranswer",
+		RTCSignalingState.closed.rawValue:             "closed"
 	]
 
 	static let iceGatheringStates = [
-		RTCICEGatheringNew.rawValue:            "new",
-		RTCICEGatheringGathering.rawValue:      "gathering",
-		RTCICEGatheringComplete.rawValue:       "complete"
+		RTCIceGatheringState.new.rawValue:            "new",
+		RTCIceGatheringState.gathering.rawValue:      "gathering",
+		RTCIceGatheringState.complete.rawValue:       "complete"
 	]
 
 	static let iceConnectionStates = [
-		RTCICEConnectionNew.rawValue:            "new",
-		RTCICEConnectionChecking.rawValue:       "checking",
-		RTCICEConnectionConnected.rawValue:      "connected",
-		RTCICEConnectionCompleted.rawValue:      "completed",
-		RTCICEConnectionFailed.rawValue:         "failed",
-		RTCICEConnectionDisconnected.rawValue:   "disconnected",
-		RTCICEConnectionClosed.rawValue:         "closed"
+		RTCIceConnectionState.new.rawValue:            "new",
+		RTCIceConnectionState.checking.rawValue:       "checking",
+		RTCIceConnectionState.connected.rawValue:      "connected",
+		RTCIceConnectionState.completed.rawValue:      "completed",
+		RTCIceConnectionState.failed.rawValue:         "failed",
+		RTCIceConnectionState.disconnected.rawValue:   "disconnected",
+		RTCIceConnectionState.closed.rawValue:         "closed"
 	]
 
 	static let dataChannelStates = [
-		kRTCDataChannelStateConnecting.rawValue: "connecting",
-		kRTCDataChannelStateOpen.rawValue:       "open",
-		kRTCDataChannelStateClosing.rawValue:    "closing",
-		kRTCDataChannelStateClosed.rawValue:     "closed"
+		RTCDataChannelState.connecting.rawValue: "connecting",
+		RTCDataChannelState.open.rawValue:       "open",
+		RTCDataChannelState.closing.rawValue:    "closing",
+		RTCDataChannelState.closed.rawValue:     "closed"
 	]
 
 	static let mediaStreamTrackStates = [
-		RTCTrackStateLive.rawValue:              "live",
-		RTCTrackStateEnded.rawValue:             "ended"
+		RTCMediaStreamTrackState.live.rawValue:		"live",
+		RTCMediaStreamTrackState.ended.rawValue:	"ended"
 	]
+
 }
diff --git a/src/PluginRTCVideoCaptureController.swift b/src/PluginRTCVideoCaptureController.swift
new file mode 100644
index 00000000..05597ce3
--- /dev/null
+++ b/src/PluginRTCVideoCaptureController.swift
@@ -0,0 +1,637 @@
+import Foundation
+import AVFoundation
+
+import func ObjectiveC.objc_getAssociatedObject
+import func ObjectiveC.objc_setAssociatedObject
+import enum ObjectiveC.objc_AssociationPolicy
+
+public func objc_getx<TargetObject: AnyObject, AssociatedObject: AnyObject>
+	(object getObject: @autoclosure () -> AssociatedObject,
+	 associatedTo target:TargetObject,
+	 withConstPtrKey ptr:UnsafeRawPointer)
+	-> AssociatedObject
+{
+	var object = objc_getAssociatedObject(target, ptr) as? AssociatedObject
+	if object == nil {
+		object = getObject()
+		objc_setAssociatedObject(target, ptr, object, objc_AssociationPolicy.OBJC_ASSOCIATION_RETAIN_NONATOMIC)
+	}
+	return object!
+}
+
+/* Expose videoCaptureController on RTCMediaStreamTrack */
+extension RTCMediaStreamTrack {
+
+	class PropClass {
+		var videoCaptureController: PluginRTCVideoCaptureController?
+		let capabilities = NSMutableDictionary()
+	}
+
+	var _propClass : PropClass {
+		get {
+			let key: UnsafeRawPointer! = UnsafeRawPointer.init(bitPattern:self.hashValue)
+			return objc_getx(object: PropClass(), associatedTo: self, withConstPtrKey: key)
+		}
+	}
+
+	var videoCaptureController: PluginRTCVideoCaptureController? {
+		get {
+			return _propClass.videoCaptureController
+		}
+		set {
+			_propClass.videoCaptureController = newValue
+		}
+	}
+
+	var capabilities: NSMutableDictionary {
+		get {
+			return _propClass.capabilities
+		}
+	}
+}
+
+class PluginRTCVideoCaptureController : NSObject {
+
+	private let DEFAULT_HEIGHT : Int32 = 480
+	private let DEFAULT_WIDTH : Int32 = 640
+	private let DEFAULT_FPS : Int = 15
+	private let DEFAULT_ASPECT_RATIO : Float32 = 4/3
+	private let FACING_MODE_USER : String = "user";
+	private let FACING_MODE_ENV : String = "environment";
+
+	var capturer: RTCCameraVideoCapturer
+	var isCapturing: Bool = false;
+	
+	// Default to the front camera.
+	var device: AVCaptureDevice?
+	var deviceFormat: AVCaptureDevice.Format?
+	var deviceFrameRate: Int?
+	
+	var constraints: NSDictionary = [:]
+	
+	init(capturer: RTCCameraVideoCapturer) {
+		self.capturer = capturer
+	}
+	
+	/*
+	// See: https://www.w3.org/TR/mediacapture-streams/#media-track-constraints
+	dictionary MediaTrackConstraintSet {
+	 // Supported
+	 ConstrainDOMString deviceId;
+	 ConstrainDOMString facingMode;
+	 ConstrainULong     width;
+	 ConstrainULong     height;
+	 ConstrainDouble    aspectRatio;
+	 ConstrainDouble    frameRate;
+	 ConstrainULong     sampleRate;
+	 // Not Supported
+	 ConstrainDOMString resizeMode;
+	 ConstrainULong     sampleSize;
+	 ConstrainBoolean   echoCancellation;
+	 ConstrainBoolean   autoGainControl;
+	 ConstrainBoolean   noiseSuppression;
+	 ConstrainDouble    latency;
+	 ConstrainULong     channelCount;
+	 ConstrainDOMString groupId;
+	};
+	 
+	 // typedef ([Clamp] unsigned long or ConstrainULongRange) ConstrainULong;
+	 dictionary ULongRange {
+		[Clamp] unsigned long max;
+		[Clamp] unsigned long min;
+	 };
+
+	 dictionary ConstrainULongRange : ULongRange {
+		  [Clamp] unsigned long exact;
+		  [Clamp] unsigned long ideal;
+	 };
+	 
+	 // See: https://www.w3.org/TR/mediacapture-streams/#dom-doublerange
+	 // typedef (double or ConstrainDoubleRange) ConstrainDouble;
+	 dictionary DoubleRange {
+		double max;
+		double min;
+	 };
+	 
+	 dictionary ConstrainDoubleRange : DoubleRange {
+		double exact;
+		double ideal;
+	 };
+	 
+	 // typedef (boolean or ConstrainBooleanParameters) ConstrainBoolean;
+	 dictionary ConstrainBooleanParameters {
+		boolean exact;
+		boolean ideal;
+	 };
+	 
+	 // typedef (DOMString or sequence<DOMString> or ConstrainDOMStringParameters) ConstrainDOMString;
+	 dictionary ConstrainDOMStringParameters {
+		(DOMString or sequence<DOMString>) exact;
+		(DOMString or sequence<DOMString>) ideal;
+	 };
+	*/
+	func setConstraints(constraints: NSDictionary) -> Bool {
+		
+		self.constraints = constraints;
+		
+		self.device = findDevice()
+		
+		return device != nil;
+	}
+	
+	func startCapture() -> Bool {
+		
+		// Stop previous capture in case of setConstraints, followed by startCapture
+		// aka future applyConstraints
+		if (isCapturing) {
+			stopCapture();
+		}
+		
+		if (device == nil) {
+			NSLog("PluginRTCVideoCaptureController#startCapture No matching device found for constraints %@", constraints);
+			return false;
+		}
+		
+		if (deviceFormat == nil) {
+			NSLog("PluginRTCVideoCaptureController#startCapture No valid formats for device %@", device!);
+			return false;
+		}
+		
+		// TODO: completionHandler with DispatchSemaphore
+		capturer.startCapture(
+			with: device!,
+			format: deviceFormat!,
+			fps: deviceFrameRate!
+		)
+		isCapturing = true
+		
+		NSLog("PluginRTCVideoCaptureController#startCapture Capture started, device:%@, format:%@", device!, deviceFormat!);
+		
+		return true;
+	}
+	
+	func stopCapture() {
+		// TODO: stopCaptureWithCompletionHandler with DispatchSemaphore
+		if (isCapturing) {
+			capturer.stopCapture()
+			isCapturing = false
+			
+			NSLog("PluginRTCVideoCaptureController#stopCapture Capture stopped");
+		}
+	}
+	
+	fileprivate func findDevice() -> AVCaptureDevice? {
+		
+		var device : AVCaptureDevice?;
+		
+		// facingMode ConstrainDOMString NSDictionary
+		// Check the video contraints: examine facingMode and deviceId
+		// and pick a default if neither are specified.
+		var position : AVCaptureDevice.Position = AVCaptureDevice.Position.unspecified;
+		
+		let facingMode = self.getConstrainDOMStringValue(constraint: "facingMode"),
+			facingModeRequired = self.isConstrainDOMStringExact(constraint: "facingMode");
+		
+		if (facingMode.count > 0) {
+			let isfacingModeEnv = facingMode == self.FACING_MODE_ENV,
+				isfacingModeUser = facingMode == self.FACING_MODE_USER;
+		 
+			position = isfacingModeEnv ? AVCaptureDevice.Position.back :
+				isfacingModeUser ? AVCaptureDevice.Position.front : AVCaptureDevice.Position.unspecified;
+			
+			if (facingModeRequired && position == AVCaptureDevice.Position.unspecified) {
+				NSLog("PluginRTCVideoCaptureController#findDevice facingMode fail exact requirement");
+				return nil;
+			}
+			
+			NSLog("PluginRTCVideoCaptureController#findDevice facingMode:%@ env:%@ user:%@", facingMode,
+				  isfacingModeEnv ? "YES" : "NO",
+				  isfacingModeUser ? "YES" : "NO"
+			);
+		}
+		
+		// deviceId ConstrainDOMString NSDictionary
+		let deviceId = self.getConstrainDOMStringValue(constraint: "deviceId");
+		if (deviceId.count > 0) {
+			device = AVCaptureDevice(uniqueID: deviceId)
+			if (device != nil && !device!.isConnected) {
+				device = nil
+			}
+			
+			if (device == nil && self.isConstrainDOMStringExact(constraint: "deviceId")) {
+				NSLog("PluginRTCVideoCaptureController#findDevice deviceId fail exact requirement");
+				return nil;
+			}
+		}
+
+		if (device == nil) {
+			device = findDeviceForPosition(position: position);
+		}
+		
+		NSLog("PluginRTCVideoCaptureController#findDevice device:%@", device!);
+		
+		// Check facingMode requirements
+		if (facingModeRequired && (
+			device == nil ||
+				facingMode == "environment" && device?.position != AVCaptureDevice.Position.back ||
+					facingMode == "user" && device?.position != AVCaptureDevice.Position.front
+		)) {
+
+			NSLog("PluginRTCVideoCaptureController#findDevice facingMode fail exact requirement");
+			return nil;
+		}
+		
+		deviceFormat = findFormatForDevice(device:device!);
+		
+		if (deviceFormat == nil) {
+			return nil;
+		}
+		
+		let frameRateRange: NSDictionary = getConstraintDoubleValues(constraint: "frameRate", defaultValue: self.DEFAULT_FPS),
+			minFrameRate = frameRateRange.object(forKey: "min") as! Int,
+			maxFrameRate = frameRateRange.object(forKey: "max") as! Int
+			
+		if (minFrameRate > 0 || maxFrameRate > 0) {
+			let formatFrameRateRange: AVFrameRateRange = deviceFormat!.videoSupportedFrameRateRanges[0]
+		 
+			// Fail if frameRate is exact and formatFrameRateRange is not capable
+			if (
+				isConstrainRangeExact(constraint: "frameRate") && (
+					Int(formatFrameRateRange.maxFrameRate) < maxFrameRate ||
+						Int(formatFrameRateRange.minFrameRate) > minFrameRate
+				)
+			) {
+				NSLog("PluginRTCVideoCaptureController#findDevice frameRate fail exact requirement");
+				return nil;
+			}
+			
+			deviceFrameRate = Int(maxFrameRate > 0 ? maxFrameRate : minFrameRate)
+			
+			NSLog("PluginRTCVideoCaptureController#findDevice deviceFrameRate:%i", deviceFrameRate!);
+			
+		// Apply default deviceFrameRate
+		} else {
+			deviceFrameRate = DEFAULT_FPS;
+		}
+		
+		return device;
+	}
+	
+	fileprivate func findDeviceForPosition(position: AVCaptureDevice.Position) -> AVCaptureDevice? {
+		let captureDevices: NSArray = RTCCameraVideoCapturer.captureDevices() as NSArray
+		for device: Any in captureDevices {
+			let avDevice = device as! AVCaptureDevice
+			if (
+				avDevice.position == position ||
+					// Default to front if unspecified
+					(avDevice.position == AVCaptureDevice.Position.front && position == AVCaptureDevice.Position.unspecified)
+			) {
+				return avDevice
+			}
+		}
+	
+		// TODO fail on no match ?
+		return captureDevices.firstObject as? AVCaptureDevice
+	}
+	
+	func switchCamera() -> Bool {
+		
+		if (self.capturer.captureSession.isRunning) {
+			self.capturer.stopCapture()
+		}
+
+		self.device = self.findAlternativeDevicePosition(currentDevice: self.device)
+		
+		self.deviceFormat = self.findFormatForDevice(device: self.device!)
+		
+		return self.startCapture()
+	}
+	
+	fileprivate func findAlternativeDevicePosition(currentDevice: AVCaptureDevice?) -> AVCaptureDevice? {
+		let captureDevices: NSArray = RTCCameraVideoCapturer.captureDevices() as NSArray
+		for device: Any in captureDevices {
+			let avDevice = device as! AVCaptureDevice
+			if (avDevice.position != currentDevice!.position) {
+				return avDevice
+			}
+		}
+	
+		return nil
+	}
+	
+	
+	// TODO use in isConstrainExact and findFormatForDevice
+	/*
+	 For string valued constraints, we define "==" below to be true if one of the values in the sequence is exactly the same as the value being compared against.
+
+	 1. We define the fitness distance between a settings dictionary and a constraint set CS as the sum, for each member (represented by a constraintName and constraintValue pair) present in CS, of the following values:
+
+	 2. If constraintName is not supported by the browser, the fitness distance is 0.
+
+	 3. If the constraint is required (constraintValue either contains one or more members named 'min', 'max', or 'exact', or is itself a bare value and bare values are to be treated as 'exact'), and the settings dictionary's value for the constraint does not satisfy the constraint, the fitness distance is positive infinity.
+
+	 4. If the constraint is not required, and does not apply for this type of device, the fitness distance is 0 (that is, the constraint does not influence the fitness distance).
+
+	 5. If no ideal value is specified (constraintValue either contains no member named 'ideal', or, if bare values are to be treated as 'ideal', isn't a bare value), the fitness distance is 0.
+	 
+	 6. For all positive numeric non-required constraints (such as height, width, frameRate, aspectRatio, sampleRate and sampleSize), the fitness distance is the result of the formula
+	 (actual == ideal) ? 0 : |actual - ideal| / max(|actual|, |ideal|)
+	 
+	 7. For all string and enum non-required constraints (e.g. deviceId, groupId, facingMode, resizeMode, echoCancellation), the fitness distance is the result of the formula
+	 (actual == ideal) ? 0 : 1
+	*/
+	
+	fileprivate func findFormatForDevice(device: AVCaptureDevice) -> AVCaptureDevice.Format? {
+		
+		var selectedFormat: AVCaptureDevice.Format? = nil
+		let formats: NSArray = RTCCameraVideoCapturer.supportedFormats(for: device) as NSArray
+		
+		var widthRange: NSDictionary = getConstraintDoubleValues(constraint: "width", defaultValue: 0),
+			minWidth = widthRange.object(forKey: "min") as! Int32,
+			maxWidth = widthRange.object(forKey: "max") as! Int32
+			
+		var heightRange: NSDictionary = getConstraintDoubleValues(constraint: "height", defaultValue: 0),
+			minHeight = heightRange.object(forKey: "min") as! Int32,
+			maxHeight = heightRange.object(forKey: "max") as! Int32
+		
+		let aspectRatioRange: NSDictionary = getConstraintLongValues(constraint: "aspectRatio", defaultValue: 0.0),
+			minAspectRatio = aspectRatioRange.object(forKey: "min") as! Float32,
+			maxAspectRatio = aspectRatioRange.object(forKey: "max") as! Float32
+			
+		let frameRateRange: NSDictionary = getConstraintDoubleValues(constraint: "frameRate", defaultValue: self.DEFAULT_FPS),
+			minFrameRate = frameRateRange.object(forKey: "min") as! Float64,
+			maxFrameRate = frameRateRange.object(forKey: "max") as! Float64
+		
+		// If aspectRatioRange > 0
+		if (minAspectRatio > 0 || maxAspectRatio  > 0) {
+
+			// and only one dimension is provided compute missing dimension
+			if (((minWidth > 0 || maxWidth > 0) && minHeight == 0 && maxHeight == 0)) {
+				minHeight = Int32(Float32(minWidth) / (minAspectRatio > 0 ? minAspectRatio : maxAspectRatio));
+				maxHeight = Int32(Float32(maxWidth) / (minAspectRatio > 0 ? minAspectRatio : maxAspectRatio));
+			} else if (((minHeight > 0 || maxHeight > 0) && minWidth == 0 && maxWidth == 0)) {
+				minWidth = Int32(Float32(maxHeight) * (minAspectRatio > 0 ? minAspectRatio : maxAspectRatio));
+				maxWidth = Int32(Float32(maxHeight) * (minAspectRatio > 0 ? minAspectRatio : maxAspectRatio));
+			}
+			
+			NSLog("PluginRTCVideoCaptureController#findFormatForDevice contraints using aspectRatioRange - width:%i/%i, height:%i/%i, aspectRatio: %f/%f", minWidth, maxWidth, minHeight, maxHeight, minAspectRatio, maxAspectRatio);
+			
+		// If no aspectRatioRange and no dimension is provided use default dimension
+		} else if (minHeight == 0 && maxHeight == 0 && minWidth == 0 && maxWidth == 0)  {
+			minHeight = self.DEFAULT_HEIGHT;
+			maxHeight = self.DEFAULT_HEIGHT;
+			maxWidth = self.DEFAULT_WIDTH;
+			maxWidth = self.DEFAULT_WIDTH;
+			
+			NSLog("PluginRTCVideoCaptureController#findFormatForDevice contraints using default - width:%i/%i, height:%i/%i, aspectRatio: %f/%f", minWidth, maxWidth, minHeight, maxHeight, minAspectRatio, maxAspectRatio);
+		}
+		
+		for format: Any in formats {
+			
+			let devFormat: AVCaptureDevice.Format = format as! AVCaptureDevice.Format
+			let pixelFormat: FourCharCode = CMFormatDescriptionGetMediaSubType(devFormat.formatDescription)
+		
+			let dimension: CMVideoDimensions = CMVideoFormatDescriptionGetDimensions(devFormat.formatDescription)
+			
+			let aspectRatio : Float32 = (Float32(dimension.width) / Float32(dimension.height));
+			
+			let frameRateRanges: [AVFrameRateRange] = devFormat.videoSupportedFrameRateRanges
+			let frameRates = frameRateRanges[0];
+			
+			NSLog("PluginRTCVideoCaptureController#findFormatForDevice device format - width:%i, height:%i, aspectRatio: %f, frameRateRanges:%f/%f", dimension.width, dimension.height, aspectRatio, frameRates.minFrameRate, frameRates.maxFrameRate);
+			
+			
+			if (
+				(minAspectRatio == 0 && maxAspectRatio == 0) || (
+					(minAspectRatio == 0 || checkDoubleIsEqual(fromFloat: aspectRatio, toFloat: minAspectRatio)) ||
+								(maxAspectRatio == 0 || checkDoubleIsEqual(fromFloat: aspectRatio, toFloat: maxAspectRatio))
+				)
+			) {
+				//selectedFormat = devFormat
+			} else {
+					
+				// Skip next tests
+				//NSLog("Bad aspectRatio");
+				continue
+			}
+			
+			// dimension.height and dimension.width Matches
+			if (
+				((maxHeight == 0 || dimension.height <= maxHeight) && (minHeight == 0 || dimension.height >= minHeight)) &&
+						((maxWidth == 0 || dimension.width <= maxWidth) && (minWidth == 0 || dimension.width >= minWidth))
+			) {
+				//NSLog("dimensions %i/%i, aspectRatio: %f",  dimension.width,  dimension.height, aspectRatio);
+			} else {
+				
+				// Skip next tests
+				//NSLog("Bad dimensions");
+				continue
+			}
+			
+			if (
+				(minFrameRate == 0 && maxFrameRate == 0) || (
+					(maxFrameRate == 0 || frameRates.maxFrameRate >= maxFrameRate) &&
+						(minFrameRate  == 0 || frameRates.minFrameRate <= minFrameRate)
+					)
+			) {
+				//selectedFormat = devFormat
+			} else {
+				
+				// Skip next tests
+				//NSLog("Bad frameRate");
+				continue
+			}
+			
+			if (pixelFormat != self.capturer.preferredOutputPixelFormat()) {
+				
+				// Skip next tests
+				//NSLog("Bad pixelFormat");
+				continue
+			}
+			
+			selectedFormat = devFormat;
+			
+			// TODO Handle isConstrainExact for required constraint
+		}
+
+		if (selectedFormat != nil) {
+
+			let dimension: CMVideoDimensions = CMVideoFormatDescriptionGetDimensions(selectedFormat!.formatDescription)
+			let aspectRatio : Float32 = (Float32(dimension.width) / Float32(dimension.height));
+			let frameRateRanges: [AVFrameRateRange] = selectedFormat!.videoSupportedFrameRateRanges
+			let frameRates = frameRateRanges[0];
+			
+			// TODO Apply frameRates via minFrameRate/maxFrameRate
+			
+			NSLog("PluginRTCVideoCaptureController#findFormatForDevice selected format - width:%i, height:%i, aspectRatio: %f, frameRateRanges:%f/%f", dimension.width, dimension.height, aspectRatio, frameRates.minFrameRate, frameRates.maxFrameRate);
+		}
+	
+		return selectedFormat
+	}
+	
+	//
+	// constraints parsers
+	//
+	
+	fileprivate func getConstrainDOMStringValue(constraint: String) -> String {
+		var finalValue: String = "";
+		let constraints = self.constraints;
+		let value = constraints.object(forKey: constraint);
+		
+		if value is String {
+			finalValue = value as! String
+		} else if value is NSDictionary {
+			let value = value as! NSDictionary;
+			if (value.object(forKey: "exact") != nil) {
+				finalValue = value.object(forKey: "exact") as! String
+			} else if (value.object(forKey: "ideal") != nil) {
+				finalValue = value.object(forKey: "ideal") as! String
+			}
+		}
+		
+		return finalValue;
+	}
+	
+	// If the constraint is required (constraintValue either contains one or more members named 'min', 'max', or 'exact',
+	// or is itself a bare value and bare values are to be treated as 'exact'), and the settings dictionary's value for
+	// the constraint does not satisfy the constraint, the fitness distance is positive infinity.
+	
+	private let NON_REQUIRE_CONSTRAINTS = ["height", "width", "frameRate", "aspectRatio", "sampleRate", "sampleSize"]
+	private let REQUIRE_CONSTRAINTS = ["deviceId", "groupId", "facingMode", "resizeMode", "echoCancellation"]
+	
+	fileprivate func isConstrainDOMStringExact(constraint: String) -> Bool {
+		return isConstrainExact(constraint: constraint);
+	}
+	
+	fileprivate func isConstrainExact(constraint: String) -> Bool {
+		var isRequired: Bool = false;
+		let constraints = self.constraints;
+		let value = constraints.object(forKey: constraint);
+		
+		if value is String {
+			isRequired = (value as! String).count > 0;
+		} else if value is NSNumber {
+			isRequired = (value as! NSNumber).floatValue > 0;
+		} else if value is NSDictionary {
+			let value = value as! NSDictionary;
+			if (value.object(forKey: "exact") != nil) {
+				isRequired = (value.object(forKey: "exact") as! String).count > 0
+			} else if (value.object(forKey: "ideal") != nil) {
+				isRequired = false;
+			}
+		}
+		
+		return isRequired;
+	}
+	
+	
+	fileprivate func getConstrainDoubleIdealValue(constraint: String) -> Int {
+		return getConstrainIdealValue(constraint: constraint) as! Int;
+	}
+	
+	fileprivate func getConstraintLongIdealValue(constraint: String) -> Float32 {
+		return getConstrainIdealValue(constraint: constraint).floatValue;
+	}
+	
+	fileprivate func getConstrainIdealValue(constraint: String) -> NSNumber {
+		var idealValue: NSNumber = -1;
+		let constraints = self.constraints;
+		let value = constraints.object(forKey: constraint);
+		
+		if value is NSDictionary {
+			let value = value as! NSDictionary;
+			if (value.object(forKey: "ideal") != nil) {
+				idealValue = value.object(forKey: "ideal") as! NSNumber;
+			}
+		}
+		
+		return idealValue;
+	}
+	
+	// Convert NSNumber to Float32 (e.g 1.7777777777777777 to 1.777778)
+	fileprivate func NSNumberToFloat32(number: NSNumber) -> Float32 {
+		return number.floatValue;
+		//return Float32(String(format: "%.7f", number.floatValue))!;
+	}
+	
+	// Handle (e.g Compare 1.7777777777777777 with 1.777778)
+	func checkDoubleIsEqual(fromFloat: Float32, toFloat : Float32, includingNumberOfFractionalDigits : Int = 6) -> Bool {
+
+	   let denominator : Float32 = pow(10.0, Float32(includingNumberOfFractionalDigits))
+	   let maximumDifference : Float32 = 1.0 / denominator
+	   let realDifference : Float32 = abs(fromFloat - toFloat)
+
+	   if realDifference >= maximumDifference {
+		   return false
+	   } else {
+		   return true
+	   }
+	  }
+	
+	fileprivate func getConstraintDoubleValues(constraint: String, defaultValue: Int) -> NSDictionary {
+		return getConstrainRangeValues(constraint: constraint, defaultValue: NSNumber(value: defaultValue));
+	}
+	
+	fileprivate func getConstraintLongValues(constraint: String, defaultValue: Float) -> NSDictionary {
+		let finalValue = getConstrainRangeValues(constraint: constraint, defaultValue: NSNumber(value: defaultValue));
+		
+		return [
+			"min": NSNumberToFloat32(number: finalValue.object(forKey: "min") as! NSNumber),
+			"max": NSNumberToFloat32(number: finalValue.object(forKey: "max") as! NSNumber)
+		];
+	}
+	
+	fileprivate func getConstrainRangeValues(constraint: String, defaultValue: NSNumber) -> NSDictionary {
+		let constraints = self.constraints;
+		let finalValue: NSMutableDictionary = [:];
+		finalValue.setValue(defaultValue, forKey: "min")
+		finalValue.setValue(defaultValue, forKey: "max")
+		
+		let value = constraints.object(forKey: constraint);
+		
+		if value is NSNumber {
+			finalValue.setValue(value, forKey: "min")
+			finalValue.setValue(value, forKey: "max")
+		} else if value is NSDictionary {
+			let value = value as! NSDictionary;
+			if (value.object(forKey: "exact") != nil) {
+				let value = value.object(forKey: "exact") as! NSNumber
+				finalValue.setValue(value, forKey: "min")
+				finalValue.setValue(value, forKey: "max")
+			} else if (value.object(forKey: "ideal") != nil) {
+				let value = value.object(forKey: "ideal") as! NSNumber
+				finalValue.setValue(value, forKey: "min")
+				finalValue.setValue(value, forKey: "max")
+			}
+			// Note: We do not support (exact|ideal) + min/min at root, min/max will take over for now.
+			if (value.object(forKey: "min") != nil) {
+				let value = value.object(forKey: "min") as! NSNumber
+				finalValue.setValue(value, forKey: "min")
+			}
+			if (value.object(forKey: "max") != nil) {
+				let value = value.object(forKey: "max") as! NSNumber
+				finalValue.setValue(value, forKey: "max")
+			}
+		}
+		
+		return finalValue;
+	}
+	
+	fileprivate func isConstrainRangeExact(constraint: String) -> Bool {
+		let constraints = self.constraints;
+		var isRequired: Bool = false;
+		let value = constraints.object(forKey: constraint);
+		
+		if value is Int32 {
+			isRequired = (value as! Int32) > 0;
+		} else if value is NSDictionary {
+			if (constraints.object(forKey: "exact") != nil) {
+				isRequired = true;
+			} else if (constraints.object(forKey: "ideal") != nil) {
+				isRequired = false;
+			}
+		}
+		
+		return isRequired;
+	}
+}
diff --git a/src/PluginUtils.swift b/src/PluginUtils.swift
index 0802360c..8d161ec8 100644
--- a/src/PluginUtils.swift
+++ b/src/PluginUtils.swift
@@ -2,31 +2,7 @@ import Foundation
 
 
 class PluginUtils {
-	class func randomInt(min: Int, max: Int) -> Int {
+	class func randomInt(_ min: Int, max: Int) -> Int {
 		return Int(arc4random_uniform(UInt32(max - min))) + min
 	}
-
-	// class func randomString(len: Int) -> String {
-	// 	let letters: NSString = "abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ0123456789"
-	// 	var randomString: NSMutableString = NSMutableString(capacity: len)
-
-	// 	for (var i = 0; i < len; i++) {
-	// 			var length = UInt32(letters.length)
-	// 			var rand = arc4random_uniform(length)
-	// 			randomString.appendFormat("%C", letters.characterAtIndex(Int(rand)))
-	// 	}
-
-	// 	return randomString as String
-	// }
-
-	// class func delay(delay: Double, closure: () -> ()) {
-	// 	dispatch_after(
-	// 		dispatch_time(
-	// 			DISPATCH_TIME_NOW,
-	// 			Int64(delay * Double(NSEC_PER_SEC))
-	// 		),
-	// 		dispatch_get_main_queue(),
-	// 		closure
-	// 	)
-	// }
 }
diff --git a/src/cordova-plugin-iosrtc-Bridging-Header.h b/src/cordova-plugin-iosrtc-Bridging-Header.h
index 4e7efb7f..661de540 100644
--- a/src/cordova-plugin-iosrtc-Bridging-Header.h
+++ b/src/cordova-plugin-iosrtc-Bridging-Header.h
@@ -5,30 +5,27 @@
 
 #import <Cordova/CDVPlugin.h>
 
-// #import "RTCAudioSource.h"
-#import "RTCAudioTrack.h"
-#import "RTCDataChannel.h"
-#import "RTCDTMFSender.h"
-#import "RTCEAGLVideoView.h"
-// #import "RTCI420Frame.h"
-#import "RTCICECandidate.h"
-#import "RTCICEServer.h"
-#import "RTCMediaConstraints.h"
-// #import "RTCMediaSource.h"
-#import "RTCMediaStream.h"
-#import "RTCMediaStreamTrack.h"
-// #import "RTCNSGLVideoView.h"
-// #import "RTCOpenGLVideoRenderer.h"
-#import "RTCPair.h"
-#import "RTCPeerConnection.h"
-#import "RTCPeerConnectionDelegate.h"
-#import "RTCPeerConnectionFactory.h"
-#import "RTCSessionDescription.h"
-#import "RTCSessionDescriptionDelegate.h"
-#import "RTCStatsDelegate.h"
-#import "RTCStatsReport.h"
-#import "RTCTypes.h"
-#import "RTCVideoCapturer.h"
-#import "RTCVideoRenderer.h"
-#import "RTCVideoSource.h"
-#import "RTCVideoTrack.h"
+#import <WebRTC/RTCAudioSource.h>
+#import <WebRTC/RTCAudioTrack.h>
+#import <WebRTC/RTCDataChannel.h>
+#import <WebRTC/RTCDtmfSender.h>
+#import <WebRTC/RTCEAGLVideoView.h>
+#import <WebRTC/RTCIceCandidate.h>
+#import <WebRTC/RTCIceServer.h>
+#import <WebRTC/RTCMediaConstraints.h>
+#import <WebRTC/RTCMediaSource.h>
+#import <WebRTC/RTCMediaStream.h>
+#import <WebRTC/RTCMediaStreamTrack.h>
+#import <WebRTC/RTCPeerConnection.h>
+#import <WebRTC/RTCPeerConnectionFactory.h>
+#import <WebRTC/RTCSessionDescription.h>
+#import <WebRTC/RTCVideoCapturer.h>
+#import <WebRTC/RTCVideoRenderer.h>
+#import <WebRTC/RTCVideoSource.h>
+#import <WebRTC/RTCVideoTrack.h>
+#import <WebRTC/RTCRtpReceiver.h>
+
+// Expose RTCEAGLVideoViewDelegate to Swift
+RTC_OBJC_EXPORT
+@protocol RTCEAGLVideoViewDelegate <RTCVideoViewDelegate>
+@end
diff --git a/src/iosrtcPlugin.swift b/src/iosrtcPlugin.swift
index b4078eaa..8a5434e2 100644
--- a/src/iosrtcPlugin.swift
+++ b/src/iosrtcPlugin.swift
@@ -17,74 +17,107 @@ class iosrtcPlugin : CDVPlugin {
 	// PluginMediaStreamRenderer dictionary.
 	var pluginMediaStreamRenderers: [Int : PluginMediaStreamRenderer]!
 	// Dispatch queue for serial operations.
-	var queue: dispatch_queue_t!
+	var queue: DispatchQueue!
+	// Auto selecting output speaker
+	var audioOutputController: PluginRTCAudioController!
 
 
 	// This is just called if <param name="onload" value="true" /> in plugin.xml.
-	override func pluginInitialize() {
+	@objc(pluginInitialize) override func pluginInitialize() {
 		NSLog("iosrtcPlugin#pluginInitialize()")
 
 		// Make the web view transparent
-		self.webView!.opaque = false
-		self.webView!.backgroundColor = UIColor.clearColor()
+		self.webView!.isOpaque = false
+		self.webView!.backgroundColor = UIColor.clear
 
 		pluginMediaStreams = [:]
 		pluginMediaStreamTracks = [:]
 		pluginMediaStreamRenderers = [:]
-		queue = dispatch_queue_create("cordova-plugin-iosrtc", DISPATCH_QUEUE_SERIAL)
+		queue = DispatchQueue(label: "cordova-plugin-iosrtc", attributes: [])
 		pluginRTCPeerConnections = [:]
 
 		// Initialize DTLS stuff.
-		RTCPeerConnectionFactory.initializeSSL()
+		RTCInitializeSSL()
+		//RTCSetMinDebugLogLevel(RTCLoggingSeverity.warning)
 
 		// Create a RTCPeerConnectionFactory.
-		self.rtcPeerConnectionFactory = RTCPeerConnectionFactory()
+		self.initPeerConnectionFactory();
 
 		// Create a PluginGetUserMedia instance.
 		self.pluginGetUserMedia = PluginGetUserMedia(
 			rtcPeerConnectionFactory: rtcPeerConnectionFactory
 		)
+
+		// Create a PluginRTCAudioController instance.
+		self.audioOutputController = PluginRTCAudioController()
 	}
 
+	private func initPeerConnectionFactory() {
+		let encoderFactory = RTCDefaultVideoEncoderFactory()
+		let decoderFactory = RTCDefaultVideoDecoderFactory()
+		encoderFactory.preferredCodec = getSupportedVideoEncoder(factory: encoderFactory)
 
-	override func onReset() {
-		NSLog("iosrtcPlugin#onReset() | doing nothing")
+		self.rtcPeerConnectionFactory = RTCPeerConnectionFactory(
+			encoderFactory: encoderFactory,
+			decoderFactory: decoderFactory
+		)
 	}
 
+	private func getSupportedVideoEncoder(factory: RTCDefaultVideoEncoderFactory) -> RTCVideoCodecInfo {
+		let supportedCodecs: [RTCVideoCodecInfo] = RTCDefaultVideoEncoderFactory.supportedCodecs()
+		if supportedCodecs.contains(RTCVideoCodecInfo.init(name: kRTCH264CodecName)){
+			return RTCVideoCodecInfo.init(name: kRTCH264CodecName)
+		} else if supportedCodecs.contains(RTCVideoCodecInfo.init(name: kRTCVp9CodecName)) {
+			return RTCVideoCodecInfo.init(name: kRTCVp9CodecName)
+		} else {
+			return RTCVideoCodecInfo.init(name: kRTCVp8CodecName)
+		}
+	}
 
-	override func onAppTerminate() {
-		NSLog("iosrtcPlugin#onAppTerminate() | doing nothing")
+	@objc(onReset) override func onReset() {
+		NSLog("iosrtcPlugin#onReset() | cleanup")
+		cleanup();
 	}
 
+	@objc(onAppTerminate) override func onAppTerminate() {
+		NSLog("iosrtcPlugin#onAppTerminate() | cleanup")
+		cleanup();
+	}
 
-	func new_RTCPeerConnection(command: CDVInvokedUrlCommand) {
+	@objc(new_RTCPeerConnection:) func new_RTCPeerConnection(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#new_RTCPeerConnection()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
+		let pcId = command.argument(at: 0) as! Int
 		var pcConfig: NSDictionary?
 		var pcConstraints: NSDictionary?
 
-		if command.argumentAtIndex(1) != nil {
-			pcConfig = command.argumentAtIndex(1) as? NSDictionary
+		if command.argument(at: 1) != nil {
+			pcConfig = command.argument(at: 1) as? NSDictionary
 		}
 
-		if command.argumentAtIndex(2) != nil {
-			pcConstraints = command.argumentAtIndex(2) as? NSDictionary
+		if command.argument(at: 2) != nil {
+			pcConstraints = command.argument(at: 2) as? NSDictionary
 		}
 
 		let pluginRTCPeerConnection = PluginRTCPeerConnection(
+			queue: self.queue,
 			rtcPeerConnectionFactory: self.rtcPeerConnectionFactory,
 			pcConfig: pcConfig,
 			pcConstraints: pcConstraints,
 			eventListener: { (data: NSDictionary) -> Void in
-				let result = CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+				let result = CDVPluginResult(
+					status: CDVCommandStatus_OK,
+					messageAs: data as? [AnyHashable: Any]
+				)
 
 				// Allow more callbacks.
-				result.setKeepCallbackAsBool(true);
-				self.emit(command.callbackId, result: result)
+				result?.setKeepCallbackAs(true);
+				self.emit(command.callbackId, result: result!)
 			},
 			eventListenerForAddStream: self.saveMediaStream,
-			eventListenerForRemoveStream: self.deleteMediaStream
+			eventListenerForRemoveStream: self.deleteMediaStream,
+			eventListenerForAddTrack: self.saveMediaStreamTrack,
+			eventListenerForRemoveTrack: self.deleteMediaStreamTrack
 		)
 
 		// Store the pluginRTCPeerConnection into the dictionary.
@@ -94,15 +127,14 @@ class iosrtcPlugin : CDVPlugin {
 		pluginRTCPeerConnection.run()
 	}
 
-
-	func RTCPeerConnection_createOffer(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_createOffer:) func RTCPeerConnection_createOffer(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_createOffer()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
+		let pcId = command.argument(at: 0) as! Int
 		var options: NSDictionary?
 
-		if command.argumentAtIndex(1) != nil {
-			options = command.argumentAtIndex(1) as? NSDictionary
+		if command.argument(at: 1) != nil {
+			options = command.argument(at: 1) as? NSDictionary
 		}
 
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
@@ -112,31 +144,33 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.createOffer(options,
 				callback: { (data: NSDictionary) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+						result: CDVPluginResult(
+							status: CDVCommandStatus_OK,
+							messageAs: data as? [AnyHashable: Any]
+						)
 					)
 				},
-				errback: { (error: NSError) -> Void in
+				errback: { (error: Error) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAsString: error.localizedDescription)
+						result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAs: error.localizedDescription)
 					)
 				}
 			)
 		}
 	}
 
-
-	func RTCPeerConnection_createAnswer(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_createAnswer:) func RTCPeerConnection_createAnswer(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_createAnswer()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
+		let pcId = command.argument(at: 0) as! Int
 		var options: NSDictionary?
 
-		if command.argumentAtIndex(1) != nil {
-			options = command.argumentAtIndex(1) as? NSDictionary
+		if command.argument(at: 1) != nil {
+			options = command.argument(at: 1) as? NSDictionary
 		}
 
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
@@ -146,28 +180,30 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.createAnswer(options,
 				callback: { (data: NSDictionary) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+						result: CDVPluginResult(
+							status: CDVCommandStatus_OK,
+							messageAs: data as? [AnyHashable: Any]
+						)
 					)
 				},
-				errback: { (error: NSError) -> Void in
+				errback: { (error: Error) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAsString: error.localizedDescription)
+						result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAs: error.localizedDescription)
 					)
 				}
 			)
 		}
 	}
 
-
-	func RTCPeerConnection_setLocalDescription(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_setLocalDescription:) func RTCPeerConnection_setLocalDescription(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_setLocalDescription()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let desc = command.argumentAtIndex(1) as! NSDictionary
+		let pcId = command.argument(at: 0) as! Int
+		let desc = command.argument(at: 1) as! NSDictionary
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 
 		if pluginRTCPeerConnection == nil {
@@ -175,28 +211,30 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.setLocalDescription(desc,
 				callback: { (data: NSDictionary) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+						result: CDVPluginResult(
+							status: CDVCommandStatus_OK,
+							messageAs: data as? [AnyHashable: Any]
+						)
 					)
 				},
-				errback: { (error: NSError) -> Void in
+				errback: { (error: Error) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAsString: error.localizedDescription)
+						result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAs: error.localizedDescription)
 					)
 				}
 			)
 		}
 	}
 
-
-	func RTCPeerConnection_setRemoteDescription(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_setRemoteDescription:) func RTCPeerConnection_setRemoteDescription(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_setRemoteDescription()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let desc = command.argumentAtIndex(1) as! NSDictionary
+		let pcId = command.argument(at: 0) as! Int
+		let desc = command.argument(at: 1) as! NSDictionary
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 
 		if pluginRTCPeerConnection == nil {
@@ -204,28 +242,35 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.setRemoteDescription(desc,
 				callback: { (data: NSDictionary) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+						result: CDVPluginResult(
+							status: CDVCommandStatus_OK,
+							messageAs: data as? [AnyHashable: Any]
+						)
 					)
 				},
-				errback: { (error: NSError) -> Void in
+				errback: { (error: Error) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAsString: error.localizedDescription)
+						result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAs: error.localizedDescription)
 					)
 				}
 			)
 		}
 	}
 
-
-	func RTCPeerConnection_addIceCandidate(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_addIceCandidate:) func RTCPeerConnection_addIceCandidate(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_addIceCandidate()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let candidate = command.argumentAtIndex(1) as! NSDictionary
+		if command.argument(at: 1) == nil {
+			NSLog("iosrtcPlugin#RTCPeerConnection_addIceCandidate() | ERROR: pluginRTCPeerConnection argument is NIL")
+			return;
+		}
+
+		let pcId = command.argument(at: 0) as! Int
+		let candidate = command.argument(at: 1) as! NSDictionary
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 
 		if pluginRTCPeerConnection == nil {
@@ -233,11 +278,14 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.addIceCandidate(candidate,
 				callback: { (data: NSDictionary) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+						result: CDVPluginResult(
+							status: CDVCommandStatus_OK,
+							messageAs: data as? [AnyHashable: Any]
+						)
 					)
 				},
 				errback: { () -> Void in
@@ -249,12 +297,11 @@ class iosrtcPlugin : CDVPlugin {
 		}
 	}
 
-
-	func RTCPeerConnection_addStream(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_addStream:) func RTCPeerConnection_addStream(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_addStream()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let streamId = command.argumentAtIndex(1) as! String
+		let pcId = command.argument(at: 0) as! Int
+		let streamId = command.argument(at: 1) as! String
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 		let pluginMediaStream = self.pluginMediaStreams[streamId]
 
@@ -268,19 +315,18 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection, weak pluginMediaStream] in
+		self.queue.async { [weak pluginRTCPeerConnection, weak pluginMediaStream] in
 			if pluginRTCPeerConnection?.addStream(pluginMediaStream!) == true {
 				self.saveMediaStream(pluginMediaStream!)
 			}
 		}
 	}
 
-
-	func RTCPeerConnection_removeStream(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_removeStream:) func RTCPeerConnection_removeStream(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_removeStream()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let streamId = command.argumentAtIndex(1) as! String
+		let pcId = command.argument(at: 0) as! Int
+		let streamId = command.argument(at: 1) as! String
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 		let pluginMediaStream = self.pluginMediaStreams[streamId]
 
@@ -294,22 +340,233 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection, weak pluginMediaStream] in
+		self.queue.async { [weak pluginRTCPeerConnection, weak pluginMediaStream] in
 			pluginRTCPeerConnection?.removeStream(pluginMediaStream!)
 		}
 	}
 
+	@objc(RTCPeerConnection_addTrack:) func RTCPeerConnection_addTrack(_ command: CDVInvokedUrlCommand) {
+
+		let pcId = command.argument(at: 0) as! Int
+		let trackId = command.argument(at: 1) as! String
+		let pluginRTCRtpTransceiverId = command.argument(at: 2) as! Int
+		let pluginRTCRtpReceiverId = command.argument(at: 3) as! Int
+		let pluginRTCRtpSenderId = command.argument(at: 4) as! Int
+		var streamIds : [String] = [];
+		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
+		let pluginMediaStreamTrack = self.pluginMediaStreamTracks[trackId]
+
+		if pluginRTCPeerConnection == nil {
+			NSLog("iosrtcPlugin#RTCPeerConnection_addTrack() | ERROR: pluginRTCPeerConnection with pcId=%@ does not exist", String(pcId))
+			return;
+		}
+
+		if command.argument(at: 5) != nil {
+			let streamId = command.argument(at: 5) as! String
+			let pluginMediaStream = self.pluginMediaStreams[streamId]
+
+			if pluginMediaStream == nil {
+				NSLog("iosrtcPlugin#RTCPeerConnection_addTrack() | ERROR: pluginMediaStream with id=%@ does not exist", String(streamId))
+				return;
+			}
+
+			streamIds.append(pluginMediaStream!.id)
+			self.saveMediaStream(pluginMediaStream!)
+		}
+
+		if pluginMediaStreamTrack == nil {
+			NSLog("iosrtcPlugin#RTCPeerConnection_addTrack() | ERROR: pluginMediaStreamTrack with id=\(trackId) does not exist")
+			return;
+		}
+
+		self.queue.async { [weak pluginRTCPeerConnection, weak pluginMediaStreamTrack] in
+			let pluginRTCRtpSender = pluginRTCPeerConnection?.addTrack(pluginMediaStreamTrack!, pluginRTCRtpTransceiverId, pluginRTCRtpReceiverId, pluginRTCRtpSenderId, streamIds)
+			self.emit(command.callbackId,
+				result: CDVPluginResult(
+					status: CDVCommandStatus_OK,
+					messageAs: pluginRTCRtpSender?.getJSON() as? [AnyHashable: Any]
+				)
+			)
+		}
+	}
+
+	@objc(RTCPeerConnection_removeTrack:) func RTCPeerConnection_removeTrack(_ command: CDVInvokedUrlCommand) {
+		let pcId = command.argument(at: 0) as! Int
+		let pluginRTCRtpSenderId = command.argument(at: 1) as! Int
+		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
+
+		if pluginRTCPeerConnection == nil {
+			NSLog("iosrtcPlugin#RTCPeerConnection_removeTrack() | ERROR: pluginRTCPeerConnection with pcId=%@ does not exist", String(pcId))
+			return;
+		}
+
+		self.queue.async { [weak pluginRTCPeerConnection] in
+			let pluginRTCRtpSender = pluginRTCPeerConnection?.pluginRTCRtpSenders[pluginRTCRtpSenderId]
+			if pluginRTCRtpSender == nil {
+				NSLog("iosrtcPlugin#RTCPeerConnection_removeTrack() | ERROR: pluginRTCRtpSender with id=\(pluginRTCRtpSenderId) does not exist")
+				return;
+			}
+			pluginRTCPeerConnection?.removeTrack(pluginRTCRtpSender!)
+			// TODO remove only if not used by other stream
+			// self.deleteMediaStreamTrack(pluginMediaStreamTrack!)
+		}
+	}
+
+	@objc(RTCPeerConnection_addTransceiver:) func RTCPeerConnection_addTransceiver(_ command: CDVInvokedUrlCommand) {
+		NSLog("iosrtcPlugin#RTCPeerConnection_addTransceiver()")
 
-	func RTCPeerConnection_createDataChannel(command: CDVInvokedUrlCommand) {
+		let pcId = command.argument(at: 0) as! Int
+		let trackIdOrMediaType = command.argument(at: 1) as? String
+
+		var options: NSDictionary? = nil
+		if command.argument(at: 2) != nil {
+			options = command.argument(at: 2) as? NSDictionary
+		}
+
+		let transceiverId = command.argument(at: 3) as! Int
+		let senderId = command.argument(at: 4) as! Int
+		let receiverId = command.argument(at: 5) as! Int
+		let receiverTrackId = command.argument(at: 6) as? String
+
+		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
+
+		if pluginRTCPeerConnection == nil {
+			NSLog("iosrtcPlugin#RTCPeerConnection_addTransceiver() | ERROR: pluginRTCPeerConnection with pcId=%@ does not exist", String(pcId))
+			return;
+		}
+
+		var pluginMediaStreamTrack: PluginMediaStreamTrack? = nil
+		var mediaType: RTCRtpMediaType? = nil
+
+		if trackIdOrMediaType == "video" {
+			mediaType = RTCRtpMediaType.video
+		} else if trackIdOrMediaType == "audio" {
+			mediaType = RTCRtpMediaType.audio
+		} else {
+			pluginMediaStreamTrack = self.pluginMediaStreamTracks[trackIdOrMediaType!]
+
+			if pluginMediaStreamTrack == nil {
+            	NSLog("iosrtcPlugin#RTCPeerConnection_addTransceiver() | ERROR: pluginMediaStreamTrack with id=\(trackIdOrMediaType!) does not exist")
+				return;
+			}
+		}
+
+		self.queue.async { [weak pluginRTCPeerConnection, weak pluginMediaStreamTrack] in
+            let callback = { (data: NSDictionary) -> Void in
+                let result = CDVPluginResult(
+                    status: CDVCommandStatus_OK,
+                    messageAs: data as? [AnyHashable: Any]
+                )
+
+                result!.setKeepCallbackAs(true);
+                self.emit(command.callbackId, result: result!)
+            }
+
+			pluginRTCPeerConnection!.addTransceiver(
+				with: pluginMediaStreamTrack,
+				of: mediaType,
+				options: options,
+				transceiverId: transceiverId,
+				senderId: senderId,
+				receiverId: receiverId,
+				receiverTrackId: receiverTrackId!,
+				callback: callback
+            )
+		}
+	}
+
+	@objc(RTCPeerConnection_RTCRtpTransceiver_setDirection:) func RTCPeerConnection_RTCRtpTransceiver_setDirection(_ command: CDVInvokedUrlCommand) {
+		NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpTransceiver_setDirection()")
+
+		let pcId = command.argument(at: 0) as! Int
+		let tcId = command.argument(at: 1) as! Int
+		let direction = command.argument(at: 2) as! String
+
+        let pluginRTCPeerConnection = pluginRTCPeerConnections[pcId]
+
+        if pluginRTCPeerConnection == nil {
+            NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpTransceiver_setDirection() | ERROR: pluginRTCPeerConnection with pcId=%@ does not exist", String(pcId))
+            return;
+        }
+
+        let pluginRTCRtpTransceiver = pluginRTCPeerConnection!.pluginRTCRtpTransceivers[tcId]
+
+        if pluginRTCRtpTransceiver == nil {
+            NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpTransceiver_setDirection() | ERROR: pluginRTCRtpTransceiver with id=\(tcId) does not exist")
+            return;
+        }
+
+		self.queue.async { [weak pluginRTCPeerConnection, weak pluginRTCRtpTransceiver] in
+            let callback = { (data: NSDictionary) -> Void in
+                let result = CDVPluginResult(
+                    status: CDVCommandStatus_OK,
+                    messageAs: data as? [AnyHashable: Any]
+                )
+
+                result!.setKeepCallbackAs(true);
+                self.emit(command.callbackId, result: result!)
+            }
+
+			pluginRTCRtpTransceiver!.setDirection(direction: direction)
+
+			let response: NSDictionary = [
+				"transceivers": pluginRTCPeerConnection!.getTransceiversJSON()
+			]
+
+			callback(response)
+		}
+	}
+
+	@objc(RTCPeerConnection_RTCRtpTransceiver_stop:) func RTCPeerConnection_RTCRtpTransceiver_stop(_ command: CDVInvokedUrlCommand) {
+		NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpTransceiver_stop()")
+
+        let pcId = command.argument(at: 0) as! Int
+        let tcId = command.argument(at: 1) as! Int
+
+        let pluginRTCPeerConnection = pluginRTCPeerConnections[pcId]
+
+        if pluginRTCPeerConnection == nil {
+            NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpTransceiver_stop() | ERROR: pluginRTCPeerConnection with pcId=%@ does not exist", String(pcId))
+            return;
+        }
+
+		self.queue.async { [weak pluginRTCPeerConnection] in
+			let pluginRTCRtpTransceiver = pluginRTCPeerConnection!.pluginRTCRtpTransceivers[tcId]
+			if pluginRTCRtpTransceiver == nil {
+				NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpTransceiver_stop() | ERROR: pluginRTCRtpTransceiver with id=\(tcId) does not exist")
+				return;
+			}
+
+            let callback = { (data: NSDictionary) -> Void in
+                let result = CDVPluginResult(
+                    status: CDVCommandStatus_OK,
+                    messageAs: data as? [AnyHashable: Any]
+                )
+
+                result!.setKeepCallbackAs(true);
+                self.emit(command.callbackId, result: result!)
+            }
+
+			pluginRTCRtpTransceiver!.stop()
+
+			let response: NSDictionary = [
+				"transceivers": pluginRTCPeerConnection!.getTransceiversJSON()
+			]
+
+			callback(response)
+		}
+	}
+
+	@objc(RTCPeerConnection_createDataChannel:) func RTCPeerConnection_createDataChannel(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_createDataChannel()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let dcId = command.argumentAtIndex(1) as! Int
-		let label = command.argumentAtIndex(2) as! String
+		let pcId = command.argument(at: 0) as! Int
+		let dcId = command.argument(at: 1) as! Int
+		let label = command.argument(at: 2) as! String
 		var options: NSDictionary?
 
-		if command.argumentAtIndex(3) != nil {
-			options = command.argumentAtIndex(3) as? NSDictionary
+		if command.argument(at: 3) != nil {
+			options = command.argument(at: 3) as? NSDictionary
 		}
 
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
@@ -319,32 +576,35 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.createDataChannel(dcId,
 				label: label,
 				options: options,
 				eventListener: { (data: NSDictionary) -> Void in
-					let result = CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+					let result = CDVPluginResult(
+						status: CDVCommandStatus_OK,
+						messageAs: data as? [AnyHashable: Any]
+					)
 
 					// Allow more callbacks.
-					result.setKeepCallbackAsBool(true);
-					self.emit(command.callbackId, result: result)
+					result!.setKeepCallbackAs(true);
+					self.emit(command.callbackId, result: result!)
 				},
-				eventListenerForBinaryMessage: { (data: NSData) -> Void in
+				eventListenerForBinaryMessage: { (data: Data) -> Void in
 					let result = CDVPluginResult(status: CDVCommandStatus_OK, messageAsArrayBuffer: data)
 
 					// Allow more callbacks.
-					result.setKeepCallbackAsBool(true);
-					self.emit(command.callbackId, result: result)
+					result!.setKeepCallbackAs(true);
+					self.emit(command.callbackId, result: result!)
 				}
 			)
 		}
 	}
 
-	func RTCPeerConnection_getStats(command: CDVInvokedUrlCommand) {
-		NSLog("iosrtcPlugin#RTCPeerConnection_getStats()")
+	@objc(RTCPeerConnection_getStats:) func RTCPeerConnection_getStats(_ command: CDVInvokedUrlCommand) {
+//		NSLog("iosrtcPlugin#RTCPeerConnection_getStats()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
+		let pcId = command.argument(at: 0) as! Int
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 
 		if pluginRTCPeerConnection == nil {
@@ -354,8 +614,8 @@ class iosrtcPlugin : CDVPlugin {
 
 		var pluginMediaStreamTrack: PluginMediaStreamTrack?
 
-		if command.argumentAtIndex(1) != nil {
-			let trackId = command.argumentAtIndex(1) as! String
+		if command.argument(at: 1) != nil {
+			let trackId = command.argument(at: 1) as! String
 			pluginMediaStreamTrack = self.pluginMediaStreamTracks[trackId]
 
 			if pluginMediaStreamTrack == nil {
@@ -364,26 +624,32 @@ class iosrtcPlugin : CDVPlugin {
 			}
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection, weak pluginMediaStreamTrack] in
+		self.queue.async { [weak pluginRTCPeerConnection, weak pluginMediaStreamTrack] in
 			pluginRTCPeerConnection?.getStats(pluginMediaStreamTrack,
-				callback: { (array: NSArray) -> Void in
+				callback: { (array: [[String:Any]]) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_OK, messageAsArray: array as [AnyObject])
+						result: CDVPluginResult(
+							status: CDVCommandStatus_OK,
+							messageAs: array as [AnyObject]
 						)
+					)
 				},
 				errback: { (error: NSError) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAsString: error.localizedDescription)
+						result: CDVPluginResult(
+							status: CDVCommandStatus_ERROR,
+							messageAs: error.localizedDescription
 						)
+					)
 				}
 			)
 		}
 	}
 
-	func RTCPeerConnection_close(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_close:) func RTCPeerConnection_close(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_close()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
+		let pcId = command.argument(at: 0) as! Int
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 
 		if pluginRTCPeerConnection == nil {
@@ -391,7 +657,7 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			if pluginRTCPeerConnection != nil {
 				pluginRTCPeerConnection!.close()
 			}
@@ -401,12 +667,11 @@ class iosrtcPlugin : CDVPlugin {
 		}
 	}
 
-
-	func RTCPeerConnection_RTCDataChannel_setListener(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_RTCDataChannel_setListener:) func RTCPeerConnection_RTCDataChannel_setListener(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_RTCDataChannel_setListener()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let dcId = command.argumentAtIndex(1) as! Int
+		let pcId = command.argument(at: 0) as! Int
+		let dcId = command.argument(at: 1) as! Int
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 
 		if pluginRTCPeerConnection == nil {
@@ -414,33 +679,35 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.RTCDataChannel_setListener(dcId,
 				eventListener: { (data: NSDictionary) -> Void in
-					let result = CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+					let result = CDVPluginResult(
+						status: CDVCommandStatus_OK,
+						messageAs: data as? [AnyHashable: Any]
+					)
 
 					// Allow more callbacks.
-					result.setKeepCallbackAsBool(true);
-					self.emit(command.callbackId, result: result)
+					result!.setKeepCallbackAs(true);
+					self.emit(command.callbackId, result: result!)
 				},
-				eventListenerForBinaryMessage: { (data: NSData) -> Void in
+				eventListenerForBinaryMessage: { (data: Data) -> Void in
 					let result = CDVPluginResult(status: CDVCommandStatus_OK, messageAsArrayBuffer: data)
 
 					// Allow more callbacks.
-					result.setKeepCallbackAsBool(true);
-					self.emit(command.callbackId, result: result)
+					result!.setKeepCallbackAs(true);
+					self.emit(command.callbackId, result: result!)
 				}
 			)
 		}
 	}
 
-
-	func RTCPeerConnection_RTCDataChannel_sendString(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_RTCDataChannel_sendString:) func RTCPeerConnection_RTCDataChannel_sendString(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_RTCDataChannel_sendString()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let dcId = command.argumentAtIndex(1) as! Int
-		let data = command.argumentAtIndex(2) as! String
+		let pcId = command.argument(at: 0) as! Int
+		let dcId = command.argument(at: 1) as! Int
+		let data = command.argument(at: 2) as! String
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 
 		if pluginRTCPeerConnection == nil {
@@ -448,25 +715,27 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.RTCDataChannel_sendString(dcId,
 				data: data,
 				callback: { (data: NSDictionary) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+						result: CDVPluginResult(
+							status: CDVCommandStatus_OK,
+							messageAs: data as? [AnyHashable: Any]
+						)
 					)
 				}
 			)
 		}
 	}
 
-
-	func RTCPeerConnection_RTCDataChannel_sendBinary(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_RTCDataChannel_sendBinary:) func RTCPeerConnection_RTCDataChannel_sendBinary(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_RTCDataChannel_sendBinary()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let dcId = command.argumentAtIndex(1) as! Int
-		let data = command.argumentAtIndex(2) as! NSData
+		let pcId = command.argument(at: 0) as! Int
+		let dcId = command.argument(at: 1) as! Int
+		let data = command.argument(at: 2) as! Data
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 
 		if pluginRTCPeerConnection == nil {
@@ -474,24 +743,26 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.RTCDataChannel_sendBinary(dcId,
 				data: data,
 				callback: { (data: NSDictionary) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+						result: CDVPluginResult(
+							status: CDVCommandStatus_OK,
+							messageAs: data as? [AnyHashable: Any]
+						)
 					)
 				}
 			)
 		}
 	}
 
-
-	func RTCPeerConnection_RTCDataChannel_close(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_RTCDataChannel_close:) func RTCPeerConnection_RTCDataChannel_close(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_RTCDataChannel_close()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let dcId = command.argumentAtIndex(1) as! Int
+		let pcId = command.argument(at: 0) as! Int
+		let dcId = command.argument(at: 1) as! Int
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 
 		if pluginRTCPeerConnection == nil {
@@ -499,18 +770,17 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.RTCDataChannel_close(dcId)
 		}
 	}
 
-
-	func RTCPeerConnection_createDTMFSender(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_createDTMFSender:) func RTCPeerConnection_createDTMFSender(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_createDTMFSender()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let dsId = command.argumentAtIndex(1) as! Int
-		let trackId = command.argumentAtIndex(2) as! String
+		let pcId = command.argument(at: 0) as! Int
+		let dsId = command.argument(at: 1) as! Int
+		let trackId = command.argument(at: 2) as! String
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 		let pluginMediaStreamTrack = self.pluginMediaStreamTracks[trackId]
 
@@ -525,29 +795,31 @@ class iosrtcPlugin : CDVPlugin {
 		}
 
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.createDTMFSender(dsId,
 				track: pluginMediaStreamTrack!,
 				eventListener: { (data: NSDictionary) -> Void in
-					let result = CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+					let result = CDVPluginResult(
+						status: CDVCommandStatus_OK,
+						messageAs: data as? [AnyHashable: Any]
+					)
 
 					// Allow more callbacks.
-					result.setKeepCallbackAsBool(true);
-					self.emit(command.callbackId, result: result)
+					result!.setKeepCallbackAs(true);
+					self.emit(command.callbackId, result: result!)
 				}
 			)
 		}
 	}
 
-
-	func RTCPeerConnection_RTCDTMFSender_insertDTMF(command: CDVInvokedUrlCommand) {
+	@objc(RTCPeerConnection_RTCDTMFSender_insertDTMF:) func RTCPeerConnection_RTCDTMFSender_insertDTMF(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#RTCPeerConnection_RTCDTMFSender_insertDTMF()")
 
-		let pcId = command.argumentAtIndex(0) as! Int
-		let dsId = command.argumentAtIndex(1) as! Int
-		let tones = command.argumentAtIndex(2) as! String
-		let duration = command.argumentAtIndex(3) as! Int
-		let interToneGap = command.argumentAtIndex(4) as! Int
+		let pcId = command.argument(at: 0) as! Int
+		let dsId = command.argument(at: 1) as! Int
+		let tones = command.argument(at: 2) as! String
+		let duration = command.argument(at: 3) as! Double
+		let interToneGap = command.argument(at: 4) as! Double
 		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
 
 		if pluginRTCPeerConnection == nil {
@@ -555,7 +827,7 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginRTCPeerConnection] in
+		self.queue.async { [weak pluginRTCPeerConnection] in
 			pluginRTCPeerConnection?.RTCDTMFSender_insertDTMF(dsId,
 				tones: tones,
 				duration: duration,
@@ -564,11 +836,26 @@ class iosrtcPlugin : CDVPlugin {
 		}
 	}
 
+	@objc(MediaStream_init:) func MediaStream_init(_ command: CDVInvokedUrlCommand) {
+		NSLog("iosrtcPlugin#MediaStream_init()")
 
-	func MediaStream_setListener(command: CDVInvokedUrlCommand) {
+		let streamId = command.argument(at: 0) as! String
+
+		if self.pluginMediaStreams[streamId] == nil {
+			let rtcMediaStream : RTCMediaStream = self.rtcPeerConnectionFactory.mediaStream(withStreamId: streamId)
+			let pluginMediaStream = PluginMediaStream(rtcMediaStream: rtcMediaStream, streamId: streamId)
+			pluginMediaStream.run()
+
+			self.saveMediaStream(pluginMediaStream)
+		} else {
+			NSLog("iosrtcPlugin#MediaStream_init() | ERROR: pluginMediaStream with id=%@ already exist", String(streamId))
+		}
+	}
+
+	@objc(MediaStream_setListener:) func MediaStream_setListener(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#MediaStream_setListener()")
 
-		let id = command.argumentAtIndex(0) as! String
+		let id = command.argument(at: 0) as! String
 		let pluginMediaStream = self.pluginMediaStreams[id]
 
 		if pluginMediaStream == nil {
@@ -576,15 +863,18 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginMediaStream] in
+		self.queue.async { [weak pluginMediaStream] in
 			// Set the eventListener.
 			pluginMediaStream?.setListener(
 				{ (data: NSDictionary) -> Void in
-					let result = CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+					let result = CDVPluginResult(
+						status: CDVCommandStatus_OK,
+						messageAs: data as? [AnyHashable: Any]
+					)
 
 					// Allow more callbacks.
-					result.setKeepCallbackAsBool(true);
-					self.emit(command.callbackId, result: result)
+					result!.setKeepCallbackAs(true);
+					self.emit(command.callbackId, result: result!)
 				},
 				eventListenerForAddTrack: self.saveMediaStreamTrack,
 				eventListenerForRemoveTrack: self.deleteMediaStreamTrack
@@ -592,15 +882,14 @@ class iosrtcPlugin : CDVPlugin {
 		}
 	}
 
+	@objc(MediaStream_addTrack:) func MediaStream_addTrack(_ command: CDVInvokedUrlCommand) {
 
-	func MediaStream_addTrack(command: CDVInvokedUrlCommand) {
-		NSLog("iosrtcPlugin#MediaStream_addTrack()")
 
-		let id = command.argumentAtIndex(0) as! String
-		let trackId = command.argumentAtIndex(1) as! String
+		let id = command.argument(at: 0) as! String
+		let trackId = command.argument(at: 1) as! String
 		let pluginMediaStream = self.pluginMediaStreams[id]
 		let pluginMediaStreamTrack = self.pluginMediaStreamTracks[trackId]
-
+		NSLog("iosrtcPlugin#MediaStream_addTrack() trackId=%@ " ,trackId)
 		if pluginMediaStream == nil {
 			NSLog("iosrtcPlugin#MediaStream_addTrack() | ERROR: pluginMediaStream with id=%@ does not exist", String(id))
 			return
@@ -611,17 +900,16 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginMediaStream, weak pluginMediaStreamTrack] in
+		self.queue.async { [weak pluginMediaStream, weak pluginMediaStreamTrack] in
 			pluginMediaStream?.addTrack(pluginMediaStreamTrack!)
 		}
 	}
 
-
-	func MediaStream_removeTrack(command: CDVInvokedUrlCommand) {
+	@objc(MediaStream_removeTrack:) func MediaStream_removeTrack(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#MediaStream_removeTrack()")
 
-		let id = command.argumentAtIndex(0) as! String
-		let trackId = command.argumentAtIndex(1) as! String
+		let id = command.argument(at: 0) as! String
+		let trackId = command.argument(at: 1) as! String
 		let pluginMediaStream = self.pluginMediaStreams[id]
 		let pluginMediaStreamTrack = self.pluginMediaStreamTracks[trackId]
 
@@ -635,16 +923,20 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginMediaStream, weak pluginMediaStreamTrack] in
+		self.queue.async { [weak pluginMediaStream, weak pluginMediaStreamTrack] in
 			pluginMediaStream?.removeTrack(pluginMediaStreamTrack!)
+
+			// TODO only stop if no more pluginMediaStream attached only
+			// currently pluginMediaStreamTrack can be attached to more than one pluginMediaStream
+			// use track.stop() or stream.stop() to stop tracks
+			//pluginMediaStreamTrack?.stop()
 		}
 	}
 
-
-	func MediaStream_release(command: CDVInvokedUrlCommand) {
+	@objc(MediaStream_release:) func MediaStream_release(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#MediaStream_release()")
 
-		let id = command.argumentAtIndex(0) as! String
+		let id = command.argument(at: 0) as! String
 		let pluginMediaStream = self.pluginMediaStreams[id]
 
 		if pluginMediaStream == nil {
@@ -655,11 +947,48 @@ class iosrtcPlugin : CDVPlugin {
 		self.pluginMediaStreams[id] = nil
 	}
 
+	@objc(MediaStreamTrack_clone:) func MediaStreamTrack_clone(_ command: CDVInvokedUrlCommand) {
 
-	func MediaStreamTrack_setListener(command: CDVInvokedUrlCommand) {
+
+		let existingTrackId = command.argument(at: 0) as! String
+		let newTrackId = command.argument(at: 1) as! String
+		let pluginMediaStreamTrack = self.pluginMediaStreamTracks[existingTrackId]
+		NSLog("iosrtcPlugin#MediaStreamTrack_clone() existing=%@ new=%@", existingTrackId, newTrackId)
+		if pluginMediaStreamTrack == nil {
+			NSLog("iosrtcPlugin#MediaStreamTrack_clone() | ERROR: pluginMediaStreamTrack with id=%@ does not exist", String(existingTrackId))
+			return;
+		}
+
+		if self.pluginMediaStreams[newTrackId] == nil {
+			var rtcMediaStreamTrack = self.pluginMediaStreamTracks[existingTrackId]!.rtcMediaStreamTrack;
+			// twilio uses the sdp local description to map the track ids to the media id.
+			// if the original rtcMediaStreamTrack is not cloned, the rtcPeerConnection
+			// will not add the track and as such will not be found by Twilio.
+			// it is unable to do the mapping and find track and thus
+			// will not publish the local track.
+			if pluginMediaStreamTrack?.kind == "video" {
+				if let rtcVideoTrack = rtcMediaStreamTrack as? RTCVideoTrack{
+					NSLog("iosrtcPlugin#MediaStreamTrack_clone() cloning video source");
+					rtcMediaStreamTrack = self.rtcPeerConnectionFactory.videoTrack(with: rtcVideoTrack.source, trackId: newTrackId);
+				}
+			} else if pluginMediaStreamTrack?.kind == "audio" {
+				if let rtcAudioTrack = rtcMediaStreamTrack as? RTCAudioTrack{
+					NSLog("iosrtcPlugin#MediaStreamTrack_clone() cloning audio source");
+					rtcMediaStreamTrack = self.rtcPeerConnectionFactory.audioTrack(with: rtcAudioTrack.source, trackId: newTrackId);
+				}
+			}
+			let newPluginMediaStreamTrack = PluginMediaStreamTrack(rtcMediaStreamTrack: rtcMediaStreamTrack, trackId: newTrackId)
+
+			self.saveMediaStreamTrack(newPluginMediaStreamTrack)
+		} else {
+			NSLog("iosrtcPlugin#MediaStreamTrack_clone() | ERROR: pluginMediaStreamTrack with id=%@ already exist", String(newTrackId))
+		}
+	}
+
+	@objc(MediaStreamTrack_setListener:) func MediaStreamTrack_setListener(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#MediaStreamTrack_setListener()")
 
-		let id = command.argumentAtIndex(0) as! String
+		let id = command.argument(at: 0) as! String
 		let pluginMediaStreamTrack = self.pluginMediaStreamTracks[id]
 
 		if pluginMediaStreamTrack == nil {
@@ -667,30 +996,32 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginMediaStreamTrack] in
+		self.queue.async { [weak pluginMediaStreamTrack] in
 			// Set the eventListener.
 			pluginMediaStreamTrack?.setListener(
 				{ (data: NSDictionary) -> Void in
-					let result = CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+					let result = CDVPluginResult(
+						status: CDVCommandStatus_OK,
+						messageAs: data as? [AnyHashable: Any]
+					)
 
 					// Allow more callbacks.
-					result.setKeepCallbackAsBool(true);
-					self.emit(command.callbackId, result: result)
+					result!.setKeepCallbackAs(true);
+					self.emit(command.callbackId, result: result!)
 				},
 				eventListenerForEnded: { () -> Void in
 					// Remove the track from the container.
-					self.pluginMediaStreamTracks[pluginMediaStreamTrack!.id] = nil
+					self.deleteMediaStreamTrack(pluginMediaStreamTrack!);
 				}
 			)
 		}
 	}
 
-
-	func MediaStreamTrack_setEnabled(command: CDVInvokedUrlCommand) {
+	@objc(MediaStreamTrack_setEnabled:) func MediaStreamTrack_setEnabled(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#MediaStreamTrack_setEnabled()")
 
-		let id = command.argumentAtIndex(0) as! String
-		let value = command.argumentAtIndex(1) as! Bool
+		let id = command.argument(at: 0) as! String
+		let value = command.argument(at: 1) as! Bool
 		let pluginMediaStreamTrack = self.pluginMediaStreamTracks[id]
 
 		if pluginMediaStreamTrack == nil {
@@ -698,16 +1029,15 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) {[weak pluginMediaStreamTrack] in
+		self.queue.async {[weak pluginMediaStreamTrack] in
 			pluginMediaStreamTrack?.setEnabled(value)
 		}
 	}
 
-
-	func MediaStreamTrack_stop(command: CDVInvokedUrlCommand) {
+	@objc(MediaStreamTrack_stop:) func MediaStreamTrack_stop(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#MediaStreamTrack_stop()")
 
-		let id = command.argumentAtIndex(0) as! String
+		let id = command.argument(at: 0) as! String
 		let pluginMediaStreamTrack = self.pluginMediaStreamTracks[id]
 
 		if pluginMediaStreamTrack == nil {
@@ -715,25 +1045,27 @@ class iosrtcPlugin : CDVPlugin {
 			return;
 		}
 
-		dispatch_async(self.queue) { [weak pluginMediaStreamTrack] in
+		self.queue.async { [weak pluginMediaStreamTrack] in
 			pluginMediaStreamTrack?.stop()
 		}
 	}
 
-
-	func new_MediaStreamRenderer(command: CDVInvokedUrlCommand) {
+	@objc(new_MediaStreamRenderer:) func new_MediaStreamRenderer(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#new_MediaStreamRenderer()")
 
-		let id = command.argumentAtIndex(0) as! Int
+		let id = command.argument(at: 0) as! Int
 
 		let pluginMediaStreamRenderer = PluginMediaStreamRenderer(
 			webView: self.webView!,
 			eventListener: { (data: NSDictionary) -> Void in
-				let result = CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+				let result = CDVPluginResult(
+					status: CDVCommandStatus_OK,
+					messageAs: data as? [AnyHashable: Any]
+				)
 
 				// Allow more callbacks.
-				result.setKeepCallbackAsBool(true);
-				self.emit(command.callbackId, result: result)
+				result?.setKeepCallbackAs(true);
+				self.emit(command.callbackId, result: result!)
 			}
 		)
 
@@ -744,12 +1076,11 @@ class iosrtcPlugin : CDVPlugin {
 		pluginMediaStreamRenderer.run()
 	}
 
-
-	func MediaStreamRenderer_render(command: CDVInvokedUrlCommand) {
+	@objc(MediaStreamRenderer_render:) func MediaStreamRenderer_render(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#MediaStreamRenderer_render()")
 
-		let id = command.argumentAtIndex(0) as! Int
-		let streamId = command.argumentAtIndex(1) as! String
+		let id = command.argument(at: 0) as! Int
+		let streamId = command.argument(at: 1) as! String
 		let pluginMediaStreamRenderer = self.pluginMediaStreamRenderers[id]
 		let pluginMediaStream = self.pluginMediaStreams[streamId]
 
@@ -766,11 +1097,10 @@ class iosrtcPlugin : CDVPlugin {
 		pluginMediaStreamRenderer!.render(pluginMediaStream!)
 	}
 
-
-	func MediaStreamRenderer_mediaStreamChanged(command: CDVInvokedUrlCommand) {
+	@objc(MediaStreamRenderer_mediaStreamChanged:) func MediaStreamRenderer_mediaStreamChanged(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#MediaStreamRenderer_mediaStreamChanged()")
 
-		let id = command.argumentAtIndex(0) as! Int
+		let id = command.argument(at: 0) as! Int
 		let pluginMediaStreamRenderer = self.pluginMediaStreamRenderers[id]
 
 		if pluginMediaStreamRenderer == nil {
@@ -781,12 +1111,11 @@ class iosrtcPlugin : CDVPlugin {
 		pluginMediaStreamRenderer!.mediaStreamChanged()
 	}
 
-
-	func MediaStreamRenderer_refresh(command: CDVInvokedUrlCommand) {
+	@objc(MediaStreamRenderer_refresh:) func MediaStreamRenderer_refresh(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#MediaStreamRenderer_refresh()")
 
-		let id = command.argumentAtIndex(0) as! Int
-		let data = command.argumentAtIndex(1) as! NSDictionary
+		let id = command.argument(at: 0) as! Int
+		let data = command.argument(at: 1) as! NSDictionary
 		let pluginMediaStreamRenderer = self.pluginMediaStreamRenderers[id]
 
 		if pluginMediaStreamRenderer == nil {
@@ -797,11 +1126,43 @@ class iosrtcPlugin : CDVPlugin {
 		pluginMediaStreamRenderer!.refresh(data)
 	}
 
+	@objc(MediaStreamRenderer_save:) func MediaStreamRenderer_save(_ command: CDVInvokedUrlCommand) {
+		//NSLog("iosrtcPlugin#MediaStreamRenderer_save()")
+
+		let id = command.argument(at: 0) as! Int
+		let pluginMediaStreamRenderer = self.pluginMediaStreamRenderers[id]
+
+		if pluginMediaStreamRenderer == nil {
+			NSLog("iosrtcPlugin#MediaStreamRenderer_save() | ERROR: pluginMediaStreamRenderer with id=%@ does not exist", String(id))
+			return;
+		}
+
+		// Perform the task on a background queue.
+		DispatchQueue.global().async {
+			pluginMediaStreamRenderer!.save(
+				callback: { (data: String) -> Void in
+					DispatchQueue.main.async {
+						self.emit(command.callbackId,
+							result: CDVPluginResult(
+								status: CDVCommandStatus_OK,
+								messageAs: data
+							)
+						)
+					}
+				},
+				errback: { (error: String) -> Void in
+					self.emit(command.callbackId,
+						result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAs: error)
+					)
+				}
+			)
+		}
+	}
 
-	func MediaStreamRenderer_close(command: CDVInvokedUrlCommand) {
+	@objc(MediaStreamRenderer_close:) func MediaStreamRenderer_close(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#MediaStreamRenderer_close()")
 
-		let id = command.argumentAtIndex(0) as! Int
+		let id = command.argument(at: 0) as! Int
 		let pluginMediaStreamRenderer = self.pluginMediaStreamRenderers[id]
 
 		if pluginMediaStreamRenderer == nil {
@@ -815,66 +1176,128 @@ class iosrtcPlugin : CDVPlugin {
 		self.pluginMediaStreamRenderers[id] = nil
 	}
 
-
-	func getUserMedia(command: CDVInvokedUrlCommand) {
+	@objc(getUserMedia:) func getUserMedia(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#getUserMedia()")
 
-		let constraints = command.argumentAtIndex(0) as! NSDictionary
+		let constraints = command.argument(at: 0) as! NSDictionary
 
 		self.pluginGetUserMedia.call(constraints,
 			callback: { (data: NSDictionary) -> Void in
 				self.emit(command.callbackId,
-					result: CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+					result: CDVPluginResult(
+						status: CDVCommandStatus_OK,
+						messageAs: data as? [AnyHashable: Any]
+					)
 				)
 			},
 			errback: { (error: String) -> Void in
 				self.emit(command.callbackId,
-					result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAsString: error)
+					result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAs: error)
 				)
 			},
 			eventListenerForNewStream: self.saveMediaStream
 		)
 	}
 
-
-	func enumerateDevices(command: CDVInvokedUrlCommand) {
+	@objc(enumerateDevices:) func enumerateDevices(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#enumerateDevices()")
 
-		dispatch_async(self.queue) {
+		self.queue.async {
 			PluginEnumerateDevices.call(
 				{ (data: NSDictionary) -> Void in
 					self.emit(command.callbackId,
-						result: CDVPluginResult(status: CDVCommandStatus_OK, messageAsDictionary: data as [NSObject : AnyObject])
+						result: CDVPluginResult(
+							status: CDVCommandStatus_OK,
+							messageAs: data as? [AnyHashable: Any]
+						)
 					)
 				}
 			)
 		}
 	}
 
+	@objc(RTCRequestPermission:) func RTCRequestPermission(_ command: CDVInvokedUrlCommand) {
+		DispatchQueue.main.async {
+			var status: Bool = false
+			let audioRequested: Bool = CBool(command.arguments[0] as! Bool)
+			let videoRequested: Bool = CBool(command.arguments[1] as! Bool)
+
+			if videoRequested == true {
+				switch AVCaptureDevice.authorizationStatus(for: AVMediaType.video) {
+				case AVAuthorizationStatus.notDetermined:
+					NSLog("PluginGetUserMedia#call() | video authorization: not determined")
+					AVCaptureDevice.requestAccess(for:.video) { granted in
+						status = granted
+					}
+				case AVAuthorizationStatus.restricted:
+					NSLog("PluginGetUserMedia#call() | video authorization: restricted")
+				case AVAuthorizationStatus.denied:
+					NSLog("PluginGetUserMedia#call() | video authorization: denied")
+				case AVAuthorizationStatus.authorized:
+					NSLog("PluginGetUserMedia#call() | video authorization: authorized")
+					status = true
+				}
+			}
 
-	func selectAudioOutputEarpiece(command: CDVInvokedUrlCommand) {
-		NSLog("iosrtcPlugin#selectAudioOutputEarpiece()")
+			if audioRequested == true {
+				switch AVCaptureDevice.authorizationStatus(for: AVMediaType.audio) {
+				case AVAuthorizationStatus.notDetermined:
+					NSLog("PluginGetUserMedia#call() | audio authorization: not determined")
+					AVCaptureDevice.requestAccess(for:.audio) { granted in
+						status = granted
+					}
+				case AVAuthorizationStatus.restricted:
+					NSLog("PluginGetUserMedia#call() | audio authorization: restricted")
+				case AVAuthorizationStatus.denied:
+					NSLog("PluginGetUserMedia#call() | audio authorization: denied")
+				case AVAuthorizationStatus.authorized:
+					NSLog("PluginGetUserMedia#call() | audio authorization: authorized")
+					status = true
+				}
+			}
 
-		do {
-			try AVAudioSession.sharedInstance().overrideOutputAudioPort(AVAudioSessionPortOverride.None)
-		} catch {
-			NSLog("iosrtcPlugin#selectAudioOutputEarpiece() | ERROR: %@", String(error))
-		};
+			if (status) {
+				self.emit(command.callbackId,result: CDVPluginResult(status: CDVCommandStatus_OK))
+			} else {
+				self.emit(command.callbackId,result: CDVPluginResult(status: CDVCommandStatus_ERROR))
+			}
+		}
+	}
+
+	@objc(initAudioDevices:) func initAudioDevices(_ command: CDVInvokedUrlCommand) {
+		NSLog("iosrtcPlugin#initAudioDevices()")
+
+		PluginRTCAudioController.initAudioDevices()
 	}
 
+	@objc(RTCTurnOnSpeaker:) func RTCTurnOnSpeaker(_ command: CDVInvokedUrlCommand) {
+		DispatchQueue.main.async {
+			let isTurnOn: Bool = CBool(command.arguments[0] as! Bool)
+			PluginRTCAudioController.setOutputSpeakerIfNeed(enabled: isTurnOn)
+			self.emit(command.callbackId, result: CDVPluginResult(status: CDVCommandStatus_OK))
+		}
+	}
 
-	func selectAudioOutputSpeaker(command: CDVInvokedUrlCommand) {
+	@objc(selectAudioOutputEarpiece:) func selectAudioOutputEarpiece(_ command: CDVInvokedUrlCommand) {
+		NSLog("iosrtcPlugin#selectAudioOutputEarpiece()")
+
+		PluginRTCAudioController.selectAudioOutputEarpiece()
+	}
+
+	@objc(selectAudioOutputSpeaker:) func selectAudioOutputSpeaker(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#selectAudioOutputSpeaker()")
 
-		do {
-			try AVAudioSession.sharedInstance().overrideOutputAudioPort(AVAudioSessionPortOverride.Speaker)
-		} catch {
-			NSLog("iosrtcPlugin#selectAudioOutputSpeaker() | ERROR: %@", String(error))
-		};
+		PluginRTCAudioController.selectAudioOutputSpeaker()
 	}
 
+	@objc(setDefaultAudioOutput:) func setDefaultAudioOutput(_ command: CDVInvokedUrlCommand) {
+		NSLog("iosrtcPlugin#setDefaultAudioOutput()")
+		let isSpeaker: Bool = CBool(command.arguments[0] as! Bool)
 
-	func dump(command: CDVInvokedUrlCommand) {
+        PluginRTCAudioController.setDefaultAudioOutput(isSpeaker: isSpeaker)
+	}
+
+	@objc(dump:) func dump(_ command: CDVInvokedUrlCommand) {
 		NSLog("iosrtcPlugin#dump()")
 
 		for (id, _) in self.pluginRTCPeerConnections {
@@ -894,53 +1317,169 @@ class iosrtcPlugin : CDVPlugin {
 		}
 	}
 
-
 	/**
 	 * Private API.
 	 */
 
-
-	private func emit(callbackId: String, result: CDVPluginResult) {
-		dispatch_async(dispatch_get_main_queue()) {
-			self.commandDelegate!.sendPluginResult(result, callbackId: callbackId)
+	fileprivate func emit(_ callbackId: String, result: CDVPluginResult) {
+		DispatchQueue.main.async {
+			self.commandDelegate!.send(result, callbackId: callbackId)
 		}
 	}
 
-
-	private func saveMediaStream(pluginMediaStream: PluginMediaStream) {
+	fileprivate func saveMediaStream(_ pluginMediaStream: PluginMediaStream) {
 		if self.pluginMediaStreams[pluginMediaStream.id] == nil {
 			self.pluginMediaStreams[pluginMediaStream.id] = pluginMediaStream
 		} else {
+			NSLog("iosrtcPlugin#saveMediaStream() | WARN: PluginMediaStream already exist [id:%@]", String(pluginMediaStream.id))
 			return;
 		}
 
 		// Store its PluginMediaStreamTracks' into the dictionary.
-		for (id, track) in pluginMediaStream.audioTracks {
-			if self.pluginMediaStreamTracks[id] == nil {
-				self.pluginMediaStreamTracks[id] = track
-			}
+		for (_, pluginMediaStreamTrack) in pluginMediaStream.audioTracks {
+			saveMediaStreamTrack(pluginMediaStreamTrack);
 		}
-		for (id, track) in pluginMediaStream.videoTracks {
-			if self.pluginMediaStreamTracks[id] == nil {
-				self.pluginMediaStreamTracks[id] = track
-			}
+
+		for (_, pluginMediaStreamTrack) in pluginMediaStream.videoTracks {
+			saveMediaStreamTrack(pluginMediaStreamTrack);
 		}
 	}
 
+	fileprivate func deleteMediaStream(_ pluginMediaStream: PluginMediaStream) {
+		if (self.pluginMediaStreams[pluginMediaStream.id] != nil) {
+			self.pluginMediaStreams[pluginMediaStream.id] = nil
 
-	private func deleteMediaStream(id: String) {
-		self.pluginMediaStreams[id] = nil
+			// deinit should call stop by itself
+			//pluginMediaStream.stop();
+		}
 	}
 
-
-	private func saveMediaStreamTrack(pluginMediaStreamTrack: PluginMediaStreamTrack) {
+	fileprivate func saveMediaStreamTrack(_ pluginMediaStreamTrack: PluginMediaStreamTrack) {
 		if self.pluginMediaStreamTracks[pluginMediaStreamTrack.id] == nil {
 			self.pluginMediaStreamTracks[pluginMediaStreamTrack.id] = pluginMediaStreamTrack
 		}
 	}
 
+	fileprivate func deleteMediaStreamTrack(_ pluginMediaStreamTrack: PluginMediaStreamTrack) {
+		if (self.pluginMediaStreamTracks[pluginMediaStreamTrack.id] != nil) {
+			self.pluginMediaStreamTracks[pluginMediaStreamTrack.id] = nil
 
-	private func deleteMediaStreamTrack(id: String) {
-		self.pluginMediaStreamTracks[id] = nil
+			// deinit should call stop by itself
+			//pluginMediaStreamTrack.stop();
+		}
 	}
+
+	fileprivate func cleanup() {
+
+		// Close all RTCPeerConnections
+		for (pcId, pluginRTCPeerConnection) in self.pluginRTCPeerConnections {
+			pluginRTCPeerConnection.close()
+			self.pluginRTCPeerConnections[pcId] = nil;
+		}
+
+		// Close all StreamRenderers
+		for (id, pluginMediaStreamRenderer) in self.pluginMediaStreamRenderers {
+			pluginMediaStreamRenderer.close()
+			self.pluginMediaStreamRenderers[id] = nil;
+		}
+
+		// Close All MediaStream
+		for (streamId, pluginMediaStream) in self.pluginMediaStreams {
+			// Store its PluginMediaStreamTracks' into the dictionary.
+			for (trackId, pluginMediaStreamTrack) in pluginMediaStream.audioTracks {
+				pluginMediaStream.removeTrack(pluginMediaStreamTrack);
+				deleteMediaStreamTrack(pluginMediaStreamTrack);
+			}
+
+			for (trackId, pluginMediaStreamTrack) in pluginMediaStream.videoTracks {
+				pluginMediaStream.removeTrack(pluginMediaStreamTrack);
+				deleteMediaStreamTrack(pluginMediaStreamTrack);
+			}
+
+			deleteMediaStream(pluginMediaStream);
+		}
+
+		// Close All MediaStreamTracks without MediaStream
+		for (trackId, pluginMediaStreamTrack) in self.pluginMediaStreamTracks {
+			deleteMediaStreamTrack(pluginMediaStreamTrack);
+		}
+	}
+
+	@objc(RTCPeerConnection_RTCRtpSender_setParameters:) func RTCPeerConnection_RTCRtpSender_setParameters(_ command: CDVInvokedUrlCommand) {
+		NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpSender_setParameters()")
+
+		let pcId = command.argument(at: 0) as! Int
+		let senderId = command.argument(at: 1) as! Int
+		let params = command.argument(at: 2) as! [String : NSObject]
+
+		let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId]
+
+		if pluginRTCPeerConnection == nil {
+			NSLog("iosrtcPlugin#RTCRtpSender_setParameters() | ERROR: pluginRTCPeerConnection with pcId=%@ does not exist", String(pcId))
+			return;
+		}
+
+		self.queue.async { [weak pluginRTCPeerConnection] in
+			let pluginRTCRptSender = pluginRTCPeerConnection!.pluginRTCRtpSenders[senderId]
+			if pluginRTCRptSender == nil {
+				NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpSender_setParameters() | ERROR: pluginRTCRptSender with id=%@ does not exist", String(senderId))
+				return;
+			}
+
+			let callback: (NSDictionary) -> Void = { data in
+				self.emit(command.callbackId,
+					result: CDVPluginResult(
+						status: CDVCommandStatus_OK,
+						messageAs: data as? [AnyHashable: Any]
+					)
+				)
+			}
+			let errback: (Error) -> Void = { error in
+				self.emit(command.callbackId,
+					result: CDVPluginResult(status: CDVCommandStatus_ERROR, messageAs: error.localizedDescription)
+				)
+			}
+			pluginRTCRptSender!.setParameters(params, callback, errback);
+		}
+	}
+
+	@objc(RTCPeerConnection_RTCRtpSender_replaceTrack:) func RTCPeerConnection_RTCRtpSender_replaceTrack(_ command: CDVInvokedUrlCommand) {
+        NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpSender_replaceTrack()")
+
+        guard let pcId = command.argument(at: 0) as? Int,
+              let senderId = command.argument(at: 1) as? Int else {
+            NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpSender_replaceTrack() | ERROR: Invalid pcId or senderId")
+            return
+        }
+
+        let trackId: String? = command.argument(at: 2) as? String
+
+        guard let pluginRTCPeerConnection = self.pluginRTCPeerConnections[pcId] else {
+            NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpSender_replaceTrack() | ERROR: pluginRTCPeerConnection with pcId=%@ does not exist", String(pcId))
+            return
+        }
+
+        self.queue.async { [weak pluginRTCPeerConnection] in
+            guard let pluginRTCRptSender = pluginRTCPeerConnection?.pluginRTCRtpSenders[senderId] else {
+                NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpSender_replaceTrack() | ERROR: Unable to find sender")
+                return
+            }
+
+            if let trackId = trackId, let pluginMediaStreamTrack = self.pluginMediaStreamTracks[trackId] {
+                pluginRTCRptSender.replaceTrack(pluginMediaStreamTrack)
+                self.emit(command.callbackId,
+                          result: CDVPluginResult(
+                            status: CDVCommandStatus_OK,
+                            messageAs: [
+                                "track": pluginMediaStreamTrack.getJSON()
+                            ]))
+            } else {
+                NSLog("iosrtcPlugin#RTCPeerConnection_RTCRtpSender_replaceTrack() | ERROR: Unable to find track")
+                self.emit(command.callbackId,
+                          result: CDVPluginResult(
+                            status: CDVCommandStatus_ERROR,
+                            messageAs: "Cannot find native RTCRtpSender track"))
+            }
+        }
+    }
 }
diff --git a/src/webrtc-headers/RTCAVFoundationVideoSource.h b/src/webrtc-headers/RTCAVFoundationVideoSource.h
deleted file mode 100644
index fcc8f1f2..00000000
--- a/src/webrtc-headers/RTCAVFoundationVideoSource.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * libjingle
- * Copyright 2015 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import "RTCVideoSource.h"
-
-@class AVCaptureSession;
-@class RTCMediaConstraints;
-@class RTCPeerConnectionFactory;
-
-// RTCAVFoundationVideoSource is a video source that uses
-// webrtc::AVFoundationVideoCapturer. We do not currently provide a wrapper for
-// that capturer because cricket::VideoCapturer is not ref counted and we cannot
-// guarantee its lifetime. Instead, we expose its properties through the ref
-// counted video source interface.
-@interface RTCAVFoundationVideoSource : RTCVideoSource
-
-- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
-                    constraints:(RTCMediaConstraints*)constraints;
-
-// Returns whether rear-facing camera is available for use.
-@property(nonatomic, readonly) BOOL canUseBackCamera;
-// Switches the camera being used (either front or back).
-@property(nonatomic, assign) BOOL useBackCamera;
-// Returns the active capture session.
-@property(nonatomic, readonly) AVCaptureSession* captureSession;
-
-@end
diff --git a/src/webrtc-headers/RTCAudioSource.h b/src/webrtc-headers/RTCAudioSource.h
deleted file mode 100644
index c0d38fe7..00000000
--- a/src/webrtc-headers/RTCAudioSource.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import "RTCMediaSource.h"
-
-// RTCAudioSource is an ObjectiveC wrapper for AudioSourceInterface.  It is
-// used as the source for one or more RTCAudioTrack objects.
-@interface RTCAudioSource : RTCMediaSource
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__(
-    (unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCAudioTrack.h b/src/webrtc-headers/RTCAudioTrack.h
deleted file mode 100644
index 48c27831..00000000
--- a/src/webrtc-headers/RTCAudioTrack.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import "RTCMediaStreamTrack.h"
-
-// RTCAudioTrack is an ObjectiveC wrapper for AudioTrackInterface.
-@interface RTCAudioTrack : RTCMediaStreamTrack
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__(
-    (unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCDTMFSender.h b/src/webrtc-headers/RTCDTMFSender.h
deleted file mode 100644
index 7e1f675d..00000000
--- a/src/webrtc-headers/RTCDTMFSender.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * libjingle
- * Copyright 2014 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-#import "RTCAudioTrack.h"
-
-NS_ASSUME_NONNULL_BEGIN
-
-@class RTCDTMFSender;
-// Protocol for receving tone change events.
-@protocol RTCDTMFSenderDelegate<NSObject>
-
-// Called when a DTMF tone is played out.
-- (void)toneChange:(NSString*)tone;
-
-@end
-
-// ObjectiveC wrapper for a DtmfSender object.
-// See webrtc/api/dtmfsenderinterface.h
-@interface RTCDTMFSender : NSObject
-
-@property(nonatomic, readonly) BOOL canInsertDTMF;
-@property(nonatomic, readonly) NSString* toneBuffer;
-@property(nonatomic, readonly) NSInteger duration;
-@property(nonatomic, readonly) NSInteger interToneGap;
-@property(nonatomic, weak) id<RTCDTMFSenderDelegate> delegate;
-// The track associated with this DTMF sender. This property
-// returns a copy of the RTCMediaStreamTrack
-@property(nonatomic, copy, nullable) RTCAudioTrack *track;
-
-- (BOOL)insertDTMF:(NSString*)tones withDuration:(NSInteger)duration andInterToneGap:(NSInteger)interToneGap;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__((
-    unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
-
-NS_ASSUME_NONNULL_END
diff --git a/src/webrtc-headers/RTCDataChannel.h b/src/webrtc-headers/RTCDataChannel.h
deleted file mode 100644
index 24a46f65..00000000
--- a/src/webrtc-headers/RTCDataChannel.h
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * libjingle
- * Copyright 2014 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// ObjectiveC wrapper for a DataChannelInit object.
-@interface RTCDataChannelInit : NSObject
-
-// Set to YES if ordered delivery is required
-@property(nonatomic) BOOL isOrdered;
-// Max period in milliseconds in which retransmissions will be sent. After this
-// time, no more retransmissions will be sent. -1 if unset.
-@property(nonatomic) NSInteger maxRetransmitTimeMs;
-// The max number of retransmissions. -1 if unset.
-@property(nonatomic) NSInteger maxRetransmits;
-// Set to YES if the channel has been externally negotiated and we do not send
-// an in-band signalling in the form of an "open" message
-@property(nonatomic) BOOL isNegotiated;
-// The stream id, or SID, for SCTP data channels. -1 if unset.
-@property(nonatomic) NSInteger streamId;
-// Set by the application and opaque to the WebRTC implementation.
-@property(nonatomic) NSString* protocol;
-
-@end
-
-// ObjectiveC wrapper for a DataBuffer object.
-@interface RTCDataBuffer : NSObject
-
-@property(nonatomic, readonly) NSData* data;
-@property(nonatomic, readonly) BOOL isBinary;
-
-- (instancetype)initWithData:(NSData*)data isBinary:(BOOL)isBinary;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__((
-    unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
-
-// Keep in sync with webrtc::DataChannelInterface::DataState
-typedef enum {
-  kRTCDataChannelStateConnecting,
-  kRTCDataChannelStateOpen,
-  kRTCDataChannelStateClosing,
-  kRTCDataChannelStateClosed
-} RTCDataChannelState;
-
-@class RTCDataChannel;
-// Protocol for receving data channel state and message events.
-@protocol RTCDataChannelDelegate<NSObject>
-
-// Called when the data channel state has changed.
-- (void)channelDidChangeState:(RTCDataChannel*)channel;
-
-// Called when a data buffer was successfully received.
-- (void)channel:(RTCDataChannel*)channel
-    didReceiveMessageWithBuffer:(RTCDataBuffer*)buffer;
-
-@optional
-
-// Called when the buffered amount has changed.
-- (void)channel:(RTCDataChannel*)channel
-    didChangeBufferedAmount:(NSUInteger)amount;
-
-@end
-
-// ObjectiveC wrapper for a DataChannel object.
-// See talk/app/webrtc/datachannelinterface.h
-@interface RTCDataChannel : NSObject
-
-@property(nonatomic, readonly) NSString* label;
-@property(nonatomic, readonly) BOOL isReliable;
-@property(nonatomic, readonly) BOOL isOrdered;
-@property(nonatomic, readonly) NSUInteger maxRetransmitTime;
-@property(nonatomic, readonly) NSUInteger maxRetransmits;
-@property(nonatomic, readonly) NSString* protocol;
-@property(nonatomic, readonly) BOOL isNegotiated;
-@property(nonatomic, readonly) NSInteger streamId;
-@property(nonatomic, readonly) RTCDataChannelState state;
-@property(nonatomic, readonly) NSUInteger bufferedAmount;
-@property(nonatomic, weak) id<RTCDataChannelDelegate> delegate;
-
-- (void)close;
-- (BOOL)sendData:(RTCDataBuffer*)data;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__((
-    unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCEAGLVideoView.h b/src/webrtc-headers/RTCEAGLVideoView.h
deleted file mode 100644
index 32c834ac..00000000
--- a/src/webrtc-headers/RTCEAGLVideoView.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * libjingle
- * Copyright 2014 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-#import <UIKit/UIKit.h>
-
-#import "RTCVideoRenderer.h"
-
-@class RTCEAGLVideoView;
-@protocol RTCEAGLVideoViewDelegate
-
-- (void)videoView:(RTCEAGLVideoView*)videoView didChangeVideoSize:(CGSize)size;
-
-@end
-
-// RTCEAGLVideoView is an RTCVideoRenderer which renders i420 frames in its
-// bounds using OpenGLES 2.0.
-@interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
-
-@property(nonatomic, weak) id<RTCEAGLVideoViewDelegate> delegate;
-
-@end
diff --git a/src/webrtc-headers/RTCFileLogger.h b/src/webrtc-headers/RTCFileLogger.h
deleted file mode 100644
index af43f8ad..00000000
--- a/src/webrtc-headers/RTCFileLogger.h
+++ /dev/null
@@ -1,91 +0,0 @@
-/*
- * libjingle
- * Copyright 2015 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// TODO(tkchin): Move this to a common location.
-#ifndef NS_DESIGNATED_INITIALIZER
-#define NS_DESIGNATED_INITIALIZER
-#endif
-
-typedef NS_ENUM(NSUInteger, RTCFileLoggerSeverity) {
-  kRTCFileLoggerSeverityVerbose,
-  kRTCFileLoggerSeverityInfo,
-  kRTCFileLoggerSeverityWarning,
-  kRTCFileLoggerSeverityError
-};
-
-typedef NS_ENUM(NSUInteger, RTCFileLoggerRotationType) {
-  kRTCFileLoggerTypeCall,
-  kRTCFileLoggerTypeApp,
-};
-
-// This class intercepts WebRTC logs and saves them to a file. The file size
-// will not exceed the given maximum bytesize. When the maximum bytesize is
-// reached, logs are rotated according to the rotationType specified.
-// For kRTCFileLoggerTypeCall, logs from the beginning and the end
-// are preserved while the middle section is overwritten instead.
-// For kRTCFileLoggerTypeApp, the oldest log is overwritten.
-// This class is not threadsafe.
-@interface RTCFileLogger : NSObject
-
-// The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
-@property(nonatomic, assign) RTCFileLoggerSeverity severity;
-
-// The rotation type for this file logger. The default is
-// kRTCFileLoggerTypeCall.
-@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
-
-// Disables buffering disk writes. Should be set before |start|. Buffering
-// is enabled by default for performance.
-@property(nonatomic, assign) BOOL shouldDisableBuffering;
-
-// Default constructor provides default settings for dir path, file size and
-// rotation type.
-- (instancetype)init;
-
-// Create file logger with default rotation type.
-- (instancetype)initWithDirPath:(NSString *)dirPath
-                    maxFileSize:(NSUInteger)maxFileSize;
-
-- (instancetype)initWithDirPath:(NSString *)dirPath
-                    maxFileSize:(NSUInteger)maxFileSize
-                   rotationType:(RTCFileLoggerRotationType)rotationType
-    NS_DESIGNATED_INITIALIZER;
-
-// Starts writing WebRTC logs to disk if not already started. Overwrites any
-// existing file(s).
-- (void)start;
-
-// Stops writing WebRTC logs to disk. This method is also called on dealloc.
-- (void)stop;
-
-// Returns the current contents of the logs, or nil if start has been called
-// without a stop.
-- (NSData *)logData;
-
-@end
diff --git a/src/webrtc-headers/RTCI420Frame.h b/src/webrtc-headers/RTCI420Frame.h
deleted file mode 100644
index 1d145ee7..00000000
--- a/src/webrtc-headers/RTCI420Frame.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// RTCI420Frame is an ObjectiveC version of cricket::VideoFrame.
-// TODO(nisse): It appears it doesn't support any VideoFrame methods,
-// so let it wrap an webrtc::VideoFrameBuffer instead?
-@interface RTCI420Frame : NSObject
-
-@property(nonatomic, readonly) NSUInteger width;
-@property(nonatomic, readonly) NSUInteger height;
-@property(nonatomic, readonly) NSUInteger chromaWidth;
-@property(nonatomic, readonly) NSUInteger chromaHeight;
-// These can return NULL if the object is not backed by a buffer.
-@property(nonatomic, readonly) const uint8_t* yPlane;
-@property(nonatomic, readonly) const uint8_t* uPlane;
-@property(nonatomic, readonly) const uint8_t* vPlane;
-@property(nonatomic, readonly) NSInteger yPitch;
-@property(nonatomic, readonly) NSInteger uPitch;
-@property(nonatomic, readonly) NSInteger vPitch;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__((
-    unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCICECandidate.h b/src/webrtc-headers/RTCICECandidate.h
deleted file mode 100644
index 25e4fe2a..00000000
--- a/src/webrtc-headers/RTCICECandidate.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// RTCICECandidate contains an instance of ICECandidateInterface.
-@interface RTCICECandidate : NSObject
-
-// If present, this contains the identifier of the "media stream
-// identification" as defined in [RFC 3388] for m-line this candidate is
-// associated with.
-@property(nonatomic, copy, readonly) NSString* sdpMid;
-
-// This indicates the index (starting at zero) of m-line in the SDP this
-// candidate is associated with.
-@property(nonatomic, assign, readonly) NSInteger sdpMLineIndex;
-
-// Creates an SDP-ized form of this candidate.
-@property(nonatomic, copy, readonly) NSString* sdp;
-
-// Creates an ICECandidateInterface based on SDP string.
-- (id)initWithMid:(NSString*)sdpMid
-            index:(NSInteger)sdpMLineIndex
-              sdp:(NSString*)sdp;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__((
-    unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCICEServer.h b/src/webrtc-headers/RTCICEServer.h
deleted file mode 100644
index 35acf0db..00000000
--- a/src/webrtc-headers/RTCICEServer.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// RTCICEServer allows for the creation of ICEServer structs.
-@interface RTCICEServer : NSObject
-
-// The server URI, username, and password.
-@property(nonatomic, strong, readonly) NSURL* URI;
-@property(nonatomic, copy, readonly) NSString* username;
-@property(nonatomic, copy, readonly) NSString* password;
-
-// Initializer for RTCICEServer taking uri, username, and password.
-- (id)initWithURI:(NSURL*)URI
-         username:(NSString*)username
-         password:(NSString*)password;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__((
-    unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCLogging.h b/src/webrtc-headers/RTCLogging.h
deleted file mode 100644
index bcd160d9..00000000
--- a/src/webrtc-headers/RTCLogging.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/*
- * libjingle
- * Copyright 2015 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// Subset of rtc::LoggingSeverity.
-typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
-  kRTCLoggingSeverityVerbose,
-  kRTCLoggingSeverityInfo,
-  kRTCLoggingSeverityWarning,
-  kRTCLoggingSeverityError,
-};
-
-#if defined(__cplusplus)
-extern "C" void RTCLogEx(RTCLoggingSeverity severity, NSString* logString);
-extern "C" void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
-extern "C" NSString* RTCFileName(const char* filePath);
-#else
-
-// Wrapper for C++ LOG(sev) macros.
-// Logs the log string to the webrtc logstream for the given severity.
-extern void RTCLogEx(RTCLoggingSeverity severity, NSString* logString);
-
-// Wrapper for rtc::LogMessage::LogToDebug.
-// Sets the minimum severity to be logged to console.
-extern void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
-
-// Returns the filename with the path prefix removed.
-extern NSString* RTCFileName(const char* filePath);
-
-#endif
-
-// Some convenience macros.
-
-#define RTCLogString(format, ...)                    \
-  [NSString stringWithFormat:@"(%@:%d %s): " format, \
-      RTCFileName(__FILE__),                         \
-      __LINE__,                                      \
-      __FUNCTION__,                                  \
-      ##__VA_ARGS__]
-
-#define RTCLogFormat(severity, format, ...)                    \
-  do {                                                         \
-    NSString *logString = RTCLogString(format, ##__VA_ARGS__); \
-    RTCLogEx(severity, logString);                             \
-  } while (false)
-
-#define RTCLogVerbose(format, ...)                                \
-  RTCLogFormat(kRTCLoggingSeverityVerbose, format, ##__VA_ARGS__) \
-
-#define RTCLogInfo(format, ...)                                   \
-  RTCLogFormat(kRTCLoggingSeverityInfo, format, ##__VA_ARGS__)    \
-
-#define RTCLogWarning(format, ...)                                \
-  RTCLogFormat(kRTCLoggingSeverityWarning, format, ##__VA_ARGS__) \
-
-#define RTCLogError(format, ...)                                  \
-  RTCLogFormat(kRTCLoggingSeverityError, format, ##__VA_ARGS__)   \
-
-#if !defined(NDEBUG)
-#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
-#else
-#define RTCLogDebug(format, ...) \
-  do {                           \
-  } while (false)
-#endif
-
-#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
diff --git a/src/webrtc-headers/RTCMediaConstraints.h b/src/webrtc-headers/RTCMediaConstraints.h
deleted file mode 100644
index 425150a8..00000000
--- a/src/webrtc-headers/RTCMediaConstraints.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// RTCMediaConstraints contains the media constraints to be used in
-// RTCPeerConnection and RTCMediaStream.
-@interface RTCMediaConstraints : NSObject
-
-// Initializer for RTCMediaConstraints.  The parameters mandatory and optional
-// contain RTCPair objects with key/value for each constrant.
-- (id)initWithMandatoryConstraints:(NSArray *)mandatory
-               optionalConstraints:(NSArray *)optional;
-
-@end
diff --git a/src/webrtc-headers/RTCMediaSource.h b/src/webrtc-headers/RTCMediaSource.h
deleted file mode 100644
index 164d85de..00000000
--- a/src/webrtc-headers/RTCMediaSource.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-#import "RTCTypes.h"
-
-// RTCMediaSource is an ObjectiveC wrapper for MediaSourceInterface
-@interface RTCMediaSource : NSObject
-
-// The current state of the RTCMediaSource.
-@property(nonatomic, assign, readonly) RTCSourceState state;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__((
-    unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCMediaStream.h b/src/webrtc-headers/RTCMediaStream.h
deleted file mode 100644
index 32271045..00000000
--- a/src/webrtc-headers/RTCMediaStream.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-@class RTCAudioTrack;
-@class RTCVideoTrack;
-@class RTCMediaStream;
-@class RTCMediaStreamTrack;
-
-@protocol RTCMediaStreamDelegate<NSObject>
-
-- (void)OnAddAudioTrack:
-    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
-- (void)OnAddVideoTrack:
-    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
-- (void)OnRemoveAudioTrack:
-    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
-- (void)OnRemoveVideoTrack:
-    (RTCMediaStream*)mediaStream track:(RTCMediaStreamTrack*)track;
-
-@end
-
-// RTCMediaStream is an ObjectiveC wrapper for MediaStreamInterface.
-@interface RTCMediaStream : NSObject
-
-@property(nonatomic, strong, readonly) NSArray *audioTracks;
-@property(nonatomic, strong, readonly) NSArray *videoTracks;
-@property(nonatomic, strong, readonly) NSString *label;
-@property(nonatomic, weak) id<RTCMediaStreamDelegate> delegate;
-
-- (BOOL)addAudioTrack:(RTCAudioTrack *)track;
-- (BOOL)addVideoTrack:(RTCVideoTrack *)track;
-- (BOOL)removeAudioTrack:(RTCAudioTrack *)track;
-- (BOOL)removeVideoTrack:(RTCVideoTrack *)track;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__(
-    (unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCMediaStreamTrack.h b/src/webrtc-headers/RTCMediaStreamTrack.h
deleted file mode 100644
index 052389f1..00000000
--- a/src/webrtc-headers/RTCMediaStreamTrack.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-#import "RTCTypes.h"
-
-@class RTCMediaStreamTrack;
-@protocol RTCMediaStreamTrackDelegate<NSObject>
-
-- (void)mediaStreamTrackDidChange:(RTCMediaStreamTrack*)mediaStreamTrack;
-
-@end
-
-// RTCMediaStreamTrack implements the interface common to RTCAudioTrack and
-// RTCVideoTrack.  Do not create an instance of this class, rather create one
-// of the derived classes.
-@interface RTCMediaStreamTrack : NSObject
-
-@property(nonatomic, readonly) NSString* kind;
-@property(nonatomic, readonly) NSString* label;
-@property(nonatomic, weak) id<RTCMediaStreamTrackDelegate> delegate;
-
-- (BOOL)isEnabled;
-- (BOOL)setEnabled:(BOOL)enabled;
-- (RTCTrackState)state;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__(
-    (unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCNSGLVideoView.h b/src/webrtc-headers/RTCNSGLVideoView.h
deleted file mode 100644
index 2f07239d..00000000
--- a/src/webrtc-headers/RTCNSGLVideoView.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * libjingle
- * Copyright 2014 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#if TARGET_OS_IPHONE
-#error "This file targets OSX."
-#endif
-
-#import <AppKit/NSOpenGLView.h>
-
-#import "RTCVideoRenderer.h"
-
-@class RTCNSGLVideoView;
-@protocol RTCNSGLVideoViewDelegate
-
-- (void)videoView:(RTCNSGLVideoView*)videoView didChangeVideoSize:(CGSize)size;
-
-@end
-
-@interface RTCNSGLVideoView : NSOpenGLView <RTCVideoRenderer>
-
-@property(nonatomic, weak) id<RTCNSGLVideoViewDelegate> delegate;
-
-@end
diff --git a/src/webrtc-headers/RTCOpenGLVideoRenderer.h b/src/webrtc-headers/RTCOpenGLVideoRenderer.h
deleted file mode 100644
index 77893048..00000000
--- a/src/webrtc-headers/RTCOpenGLVideoRenderer.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/*
- * libjingle
- * Copyright 2014 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-#if TARGET_OS_IPHONE
-#import <GLKit/GLKit.h>
-#else
-#import <AppKit/NSOpenGL.h>
-#endif
-
-@class RTCI420Frame;
-
-// RTCOpenGLVideoRenderer issues appropriate OpenGL commands to draw a frame to
-// the currently bound framebuffer. Supports OpenGL 3.2 and OpenGLES 2.0. OpenGL
-// framebuffer creation and management should be handled elsewhere using the
-// same context used to initialize this class.
-@interface RTCOpenGLVideoRenderer : NSObject
-
-// The last successfully drawn frame. Used to avoid drawing frames unnecessarily
-// hence saving battery life by reducing load.
-@property(nonatomic, readonly) RTCI420Frame* lastDrawnFrame;
-
-#if TARGET_OS_IPHONE
-- (instancetype)initWithContext:(EAGLContext*)context;
-#else
-- (instancetype)initWithContext:(NSOpenGLContext*)context;
-#endif
-
-// Draws |frame| onto the currently bound OpenGL framebuffer. |setupGL| must be
-// called before this function will succeed.
-- (BOOL)drawFrame:(RTCI420Frame*)frame;
-
-// The following methods are used to manage OpenGL resources. On iOS
-// applications should release resources when placed in background for use in
-// the foreground application. In fact, attempting to call OpenGLES commands
-// while in background will result in application termination.
-
-// Sets up the OpenGL state needed for rendering.
-- (void)setupGL;
-// Tears down the OpenGL state created by |setupGL|.
-- (void)teardownGL;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__((
-    unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCPair.h b/src/webrtc-headers/RTCPair.h
deleted file mode 100644
index 51d20e72..00000000
--- a/src/webrtc-headers/RTCPair.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// A class to hold a key and value.
-@interface RTCPair : NSObject
-
-@property(nonatomic, strong, readonly) NSString *key;
-@property(nonatomic, strong, readonly) NSString *value;
-
-// Initialize a RTCPair object with a key and value.
-- (id)initWithKey:(NSString *)key value:(NSString *)value;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__(
-    (unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCPeerConnection.h b/src/webrtc-headers/RTCPeerConnection.h
deleted file mode 100644
index 481bbd52..00000000
--- a/src/webrtc-headers/RTCPeerConnection.h
+++ /dev/null
@@ -1,132 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import "RTCPeerConnectionDelegate.h"
-
-@class RTCAudioTrack;
-@class RTCConfiguration;
-@class RTCDataChannel;
-@class RTCDataChannelInit;
-@class RTCDTMFSender;
-@class RTCICECandidate;
-@class RTCICEServers;
-@class RTCMediaConstraints;
-@class RTCMediaStream;
-@class RTCMediaStreamTrack;
-@class RTCSessionDescription;
-@protocol RTCSessionDescriptionDelegate;
-@protocol RTCStatsDelegate;
-
-// RTCPeerConnection is an ObjectiveC friendly wrapper around a PeerConnection
-// object.  See the documentation in talk/app/webrtc/peerconnectioninterface.h.
-// or http://www.webrtc.org/reference/native-apis, which in turn is inspired by
-// the JS APIs: http://dev.w3.org/2011/webrtc/editor/webrtc.html and
-// http://www.w3.org/TR/mediacapture-streams/
-@interface RTCPeerConnection : NSObject
-
-@property(nonatomic, weak) id<RTCPeerConnectionDelegate> delegate;
-
-// Accessor methods to active local streams.
-@property(nonatomic, strong, readonly) NSArray *localStreams;
-
-// The local description.
-@property(nonatomic, assign, readonly) RTCSessionDescription *localDescription;
-
-// The remote description.
-@property(nonatomic, assign, readonly) RTCSessionDescription *remoteDescription;
-
-// The current signaling state.
-@property(nonatomic, assign, readonly) RTCSignalingState signalingState;
-@property(nonatomic, assign, readonly) RTCICEConnectionState iceConnectionState;
-@property(nonatomic, assign, readonly) RTCICEGatheringState iceGatheringState;
-
-// Add a new MediaStream to be sent on this PeerConnection.
-// Note that a SessionDescription negotiation is needed before the
-// remote peer can receive the stream.
-- (BOOL)addStream:(RTCMediaStream *)stream;
-
-// Remove a MediaStream from this PeerConnection.
-// Note that a SessionDescription negotiation is need before the
-// remote peer is notified.
-- (void)removeStream:(RTCMediaStream *)stream;
-
-// Create a data channel.
-- (RTCDataChannel*)createDataChannelWithLabel:(NSString*)label
-                                       config:(RTCDataChannelInit*)config;
-
-// Create a DTMF sender.
-- (RTCDTMFSender*)createDTMFSenderForTrack:(RTCAudioTrack*)audioTrack;
-
-// Create a new offer.
-// Success or failure will be reported via RTCSessionDescriptionDelegate.
-- (void)createOfferWithDelegate:(id<RTCSessionDescriptionDelegate>)delegate
-                    constraints:(RTCMediaConstraints *)constraints;
-
-// Create an answer to an offer.
-// Success or failure will be reported via RTCSessionDescriptionDelegate.
-- (void)createAnswerWithDelegate:(id<RTCSessionDescriptionDelegate>)delegate
-                     constraints:(RTCMediaConstraints *)constraints;
-
-// Sets the local session description.
-// Success or failure will be reported via RTCSessionDescriptionDelegate.
-- (void)
-    setLocalDescriptionWithDelegate:(id<RTCSessionDescriptionDelegate>)delegate
-                 sessionDescription:(RTCSessionDescription *)sdp;
-
-// Sets the remote session description.
-// Success or failure will be reported via RTCSessionDescriptionDelegate.
-- (void)
-    setRemoteDescriptionWithDelegate:(id<RTCSessionDescriptionDelegate>)delegate
-                  sessionDescription:(RTCSessionDescription *)sdp;
-
-// Sets the PeerConnection's global configuration to |configuration|.
-// Any changes to STUN/TURN servers or ICE candidate policy will affect the
-// next gathering phase, and cause the next call to createOffer to generate
-// new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
-// cannot be changed with this method.
-- (BOOL)setConfiguration:(RTCConfiguration *)configuration;
-
-// Provides a remote candidate to the ICE Agent.
-- (BOOL)addICECandidate:(RTCICECandidate *)candidate;
-
-// Terminates all media and closes the transport.
-- (void)close;
-
-// Gets statistics for the media track. If |mediaStreamTrack| is nil statistics
-// are gathered for all tracks.
-// Statistics information will be reported via RTCStatsDelegate.
-- (BOOL)getStatsWithDelegate:(id<RTCStatsDelegate>)delegate
-            mediaStreamTrack:(RTCMediaStreamTrack*)mediaStreamTrack
-            statsOutputLevel:(RTCStatsOutputLevel)statsOutputLevel;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__(
-    (unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCPeerConnectionDelegate.h b/src/webrtc-headers/RTCPeerConnectionDelegate.h
deleted file mode 100644
index bf0c2310..00000000
--- a/src/webrtc-headers/RTCPeerConnectionDelegate.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-#import "RTCTypes.h"
-
-@class RTCDataChannel;
-@class RTCICECandidate;
-@class RTCMediaStream;
-@class RTCPeerConnection;
-
-// RTCPeerConnectionDelegate is a protocol for an object that must be
-// implemented to get messages from PeerConnection.
-@protocol RTCPeerConnectionDelegate<NSObject>
-
-// Triggered when the SignalingState changed.
-- (void)peerConnection:(RTCPeerConnection *)peerConnection
- signalingStateChanged:(RTCSignalingState)stateChanged;
-
-// Triggered when media is received on a new stream from remote peer.
-- (void)peerConnection:(RTCPeerConnection *)peerConnection
-           addedStream:(RTCMediaStream *)stream;
-
-// Triggered when a remote peer close a stream.
-- (void)peerConnection:(RTCPeerConnection *)peerConnection
-        removedStream:(RTCMediaStream *)stream;
-
-// Triggered when renegotiation is needed, for example the ICE has restarted.
-- (void)peerConnectionOnRenegotiationNeeded:(RTCPeerConnection *)peerConnection;
-
-// Called any time the ICEConnectionState changes.
-- (void)peerConnection:(RTCPeerConnection *)peerConnection
-  iceConnectionChanged:(RTCICEConnectionState)newState;
-
-// Called any time the ICEGatheringState changes.
-- (void)peerConnection:(RTCPeerConnection *)peerConnection
-   iceGatheringChanged:(RTCICEGatheringState)newState;
-
-// New Ice candidate have been found.
-- (void)peerConnection:(RTCPeerConnection *)peerConnection
-        gotICECandidate:(RTCICECandidate *)candidate;
-
-// New data channel has been opened.
-- (void)peerConnection:(RTCPeerConnection*)peerConnection
-    didOpenDataChannel:(RTCDataChannel*)dataChannel;
-
-@end
diff --git a/src/webrtc-headers/RTCPeerConnectionFactory.h b/src/webrtc-headers/RTCPeerConnectionFactory.h
deleted file mode 100644
index e1e69b4b..00000000
--- a/src/webrtc-headers/RTCPeerConnectionFactory.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-@class RTCAudioTrack;
-@class RTCConfiguration;
-@class RTCMediaConstraints;
-@class RTCMediaStream;
-@class RTCPeerConnection;
-@class RTCVideoCapturer;
-@class RTCVideoSource;
-@class RTCVideoTrack;
-@protocol RTCPeerConnectionDelegate;
-
-// RTCPeerConnectionFactory is an ObjectiveC wrapper for PeerConnectionFactory.
-// It is the main entry point to the PeerConnection API for clients.
-@interface RTCPeerConnectionFactory : NSObject
-
-// Initialize & de-initialize the SSL subsystem.  Failure is fatal.
-+ (void)initializeSSL;
-+ (void)deinitializeSSL;
-
-// Create an RTCPeerConnection object. RTCPeerConnectionFactory will create
-// required libjingle threads, socket and network manager factory classes for
-// networking.
-- (RTCPeerConnection *)
-    peerConnectionWithICEServers:(NSArray *)servers
-                     constraints:(RTCMediaConstraints *)constraints
-                        delegate:(id<RTCPeerConnectionDelegate>)delegate;
-
-// Creates a peer connection using the default port allocator factory and identity service.
-- (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration
-                                           constraints:(RTCMediaConstraints *)constraints
-                                              delegate:(id<RTCPeerConnectionDelegate>)delegate;
-
-// Create an RTCMediaStream named |label|.
-- (RTCMediaStream *)mediaStreamWithLabel:(NSString *)label;
-
-// Creates a RTCVideoSource. The new source takes ownership of |capturer|.
-// |constraints| decides video resolution and frame rate but can be NULL.
-- (RTCVideoSource *)videoSourceWithCapturer:(RTCVideoCapturer *)capturer
-                                constraints:(RTCMediaConstraints *)constraints;
-
-// Creates a new local VideoTrack. The same |source| can be used in several
-// tracks.
-- (RTCVideoTrack *)videoTrackWithID:(NSString *)videoId
-                             source:(RTCVideoSource *)source;
-
-// Creates an new AudioTrack.
-- (RTCAudioTrack *)audioTrackWithID:(NSString *)audioId;
-
-@end
diff --git a/src/webrtc-headers/RTCPeerConnectionInterface.h b/src/webrtc-headers/RTCPeerConnectionInterface.h
deleted file mode 100644
index 1bab99ab..00000000
--- a/src/webrtc-headers/RTCPeerConnectionInterface.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * libjingle
- * Copyright 2015 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-// See talk/app/webrtc/peerconnectioninterface.h.
-
-#import <Foundation/Foundation.h>
-
-typedef NS_ENUM(NSInteger, RTCIceTransportsType) {
-  kRTCIceTransportsTypeNone,
-  kRTCIceTransportsTypeRelay,
-  kRTCIceTransportsTypeNoHost,
-  kRTCIceTransportsTypeAll,
-};
-
-// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-08#section-4.1.1
-typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
-  kRTCBundlePolicyBalanced,
-  kRTCBundlePolicyMaxBundle,
-  kRTCBundlePolicyMaxCompat,
-};
-
-// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
-typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) {
-  kRTCRtcpMuxPolicyNegotiate,
-  kRTCRtcpMuxPolicyRequire,
-};
-
-typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
-  kRTCTcpCandidatePolicyEnabled,
-  kRTCTcpCandidatePolicyDisabled,
-};
-
-typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
-  kRTCEncryptionKeyTypeRSA,
-  kRTCEncryptionKeyTypeECDSA,
-};
-
-// Configuration object used for creating a peer connection.
-@interface RTCConfiguration : NSObject
-
-@property(nonatomic, assign) RTCIceTransportsType iceTransportsType;
-@property(nonatomic, copy) NSArray *iceServers;
-@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
-@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
-@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
-@property(nonatomic, assign) int audioJitterBufferMaxPackets;
-@property(nonatomic, assign) int iceConnectionReceivingTimeout;
-@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
-@property(nonatomic, assign) RTCEncryptionKeyType keyType;
-
-- (instancetype)initWithIceTransportsType:(RTCIceTransportsType)iceTransportsType
-                             bundlePolicy:(RTCBundlePolicy)bundlePolicy
-                            rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
-                       tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
-              audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets
-            iceConnectionReceivingTimeout:(int)iceConnectionReceivingTimeout
-       iceBackupCandidatePairPingInterval:(int)iceBackupCandidatePairPingInterval;
-
-@end
diff --git a/src/webrtc-headers/RTCSessionDescription.h b/src/webrtc-headers/RTCSessionDescription.h
deleted file mode 100644
index e3db21cb..00000000
--- a/src/webrtc-headers/RTCSessionDescription.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// Description of an RFC 4566 Session.
-// RTCSessionDescription is an ObjectiveC wrapper for
-// SessionDescriptionInterface.
-@interface RTCSessionDescription : NSObject
-
-// The SDP description.
-@property(nonatomic, copy, readonly) NSString *description;
-
-// The session type.
-@property(nonatomic, copy, readonly) NSString *type;
-
-- (id)initWithType:(NSString *)type sdp:(NSString *)sdp;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__(
-    (unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
-
diff --git a/src/webrtc-headers/RTCSessionDescriptionDelegate.h b/src/webrtc-headers/RTCSessionDescriptionDelegate.h
deleted file mode 100644
index 2fb0af91..00000000
--- a/src/webrtc-headers/RTCSessionDescriptionDelegate.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-@class RTCPeerConnection;
-@class RTCSessionDescription;
-
-extern NSString* const kRTCSessionDescriptionDelegateErrorDomain;
-extern int const kRTCSessionDescriptionDelegateErrorCode;
-
-// RTCSessionDescriptionDelegate is a protocol for listening to callback
-// messages when RTCSessionDescriptions are created or set.
-@protocol RTCSessionDescriptionDelegate<NSObject>
-
-// Called when creating a session.
-- (void)peerConnection:(RTCPeerConnection *)peerConnection
-    didCreateSessionDescription:(RTCSessionDescription *)sdp
-                          error:(NSError *)error;
-
-// Called when setting a local or remote description.
-- (void)peerConnection:(RTCPeerConnection *)peerConnection
-    didSetSessionDescriptionWithError:(NSError *)error;
-
-@end
diff --git a/src/webrtc-headers/RTCStatsDelegate.h b/src/webrtc-headers/RTCStatsDelegate.h
deleted file mode 100644
index 54e8e569..00000000
--- a/src/webrtc-headers/RTCStatsDelegate.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * libjingle
- * Copyright 2014 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-@class RTCPeerConnection;
-
-// RTCSessionDescriptionDelegate is a protocol for receiving statistic
-// reports from RTCPeerConnection.
-@protocol RTCStatsDelegate<NSObject>
-
-- (void)peerConnection:(RTCPeerConnection*)peerConnection
-           didGetStats:(NSArray*)stats;  // NSArray of RTCStatsReport*.
-
-@end
diff --git a/src/webrtc-headers/RTCStatsReport.h b/src/webrtc-headers/RTCStatsReport.h
deleted file mode 100644
index 691c1aa6..00000000
--- a/src/webrtc-headers/RTCStatsReport.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * libjingle
- * Copyright 2014 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// ObjectiveC friendly wrapper around a StatsReport object.
-// See talk/app/webrtc/statsypes.h
-@interface RTCStatsReport : NSObject
-
-@property(nonatomic, readonly) NSString* reportId;
-@property(nonatomic, readonly) NSString* type;
-@property(nonatomic, readonly) CFTimeInterval timestamp;
-@property(nonatomic, readonly) NSArray* values;  // NSArray of RTCPair*.
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__((
-    unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCTypes.h b/src/webrtc-headers/RTCTypes.h
deleted file mode 100644
index 1baa4b50..00000000
--- a/src/webrtc-headers/RTCTypes.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-// Enums that are common to the ObjectiveC version of the PeerConnection API.
-
-// RTCICEConnectionState correspond to the states in webrtc::ICEConnectionState.
-typedef enum {
-  RTCICEConnectionNew,
-  RTCICEConnectionChecking,
-  RTCICEConnectionConnected,
-  RTCICEConnectionCompleted,
-  RTCICEConnectionFailed,
-  RTCICEConnectionDisconnected,
-  RTCICEConnectionClosed,
-  RTCICEConnectionMax,
-} RTCICEConnectionState;
-
-// RTCICEGatheringState the states in webrtc::ICEGatheringState.
-typedef enum {
-  RTCICEGatheringNew,
-  RTCICEGatheringGathering,
-  RTCICEGatheringComplete,
-} RTCICEGatheringState;
-
-// RTCSignalingState correspond to the states in webrtc::SignalingState.
-typedef enum {
-  RTCSignalingStable,
-  RTCSignalingHaveLocalOffer,
-  RTCSignalingHaveLocalPrAnswer,
-  RTCSignalingHaveRemoteOffer,
-  RTCSignalingHaveRemotePrAnswer,
-  RTCSignalingClosed,
-} RTCSignalingState;
-
-// RTCStatsOutputLevel correspond to webrtc::StatsOutputLevel
-typedef enum {
-  RTCStatsOutputLevelStandard,
-  RTCStatsOutputLevelDebug,
-} RTCStatsOutputLevel;
-
-// RTCSourceState corresponds to the states in webrtc::SourceState.
-typedef enum {
-  RTCSourceStateInitializing,
-  RTCSourceStateLive,
-  RTCSourceStateEnded,
-  RTCSourceStateMuted,
-} RTCSourceState;
-
-// RTCTrackState corresponds to the states in webrtc::TrackState.
-typedef enum {
-  RTCTrackStateLive,
-  RTCTrackStateEnded,
-} RTCTrackState;
diff --git a/src/webrtc-headers/RTCVideoCapturer.h b/src/webrtc-headers/RTCVideoCapturer.h
deleted file mode 100644
index d07c091f..00000000
--- a/src/webrtc-headers/RTCVideoCapturer.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-
-// RTCVideoCapturer is an ObjectiveC wrapper for VideoCapturerInterface.
-@interface RTCVideoCapturer : NSObject
-
-// Create a new video capturer using the specified device.
-+ (RTCVideoCapturer *)capturerWithDeviceName:(NSString *)deviceName;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__(
-    (unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCVideoRenderer.h b/src/webrtc-headers/RTCVideoRenderer.h
deleted file mode 100644
index c837b1c5..00000000
--- a/src/webrtc-headers/RTCVideoRenderer.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import <Foundation/Foundation.h>
-#if TARGET_OS_IPHONE
-#import <UIKit/UIKit.h>
-#endif
-
-@class RTCI420Frame;
-
-@protocol RTCVideoRenderer<NSObject>
-
-// The size of the frame.
-- (void)setSize:(CGSize)size;
-
-// The frame to be displayed.
-- (void)renderFrame:(RTCI420Frame*)frame;
-
-@end
diff --git a/src/webrtc-headers/RTCVideoSource.h b/src/webrtc-headers/RTCVideoSource.h
deleted file mode 100644
index 15a9262b..00000000
--- a/src/webrtc-headers/RTCVideoSource.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import "RTCMediaSource.h"
-
-// RTCVideoSource is an ObjectiveC wrapper for VideoSourceInterface.
-@interface RTCVideoSource : RTCMediaSource
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__(
-    (unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/src/webrtc-headers/RTCVideoTrack.h b/src/webrtc-headers/RTCVideoTrack.h
deleted file mode 100644
index 332c3ea8..00000000
--- a/src/webrtc-headers/RTCVideoTrack.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#import "RTCMediaStreamTrack.h"
-
-@protocol RTCVideoRenderer;
-@class RTCPeerConnectionFactory;
-@class RTCVideoSource;
-
-// RTCVideoTrack is an ObjectiveC wrapper for VideoTrackInterface.
-@interface RTCVideoTrack : RTCMediaStreamTrack
-
-@property(nonatomic, readonly) RTCVideoSource* source;
-
-- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
-                         source:(RTCVideoSource*)source
-                        trackId:(NSString*)trackId;
-
-// Register a renderer that will render all frames received on this track.
-- (void)addRenderer:(id<RTCVideoRenderer>)renderer;
-
-// Deregister a renderer.
-- (void)removeRenderer:(id<RTCVideoRenderer>)renderer;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__(
-    (unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
-
-@end
diff --git a/www/cordova-plugin-iosrtc.js b/www/cordova-plugin-iosrtc.js
new file mode 100644
index 00000000..e65b5a2c
--- /dev/null
+++ b/www/cordova-plugin-iosrtc.js
@@ -0,0 +1,5675 @@
+/*
+ * cordova-plugin-iosrtc v8.0.4
+ * Cordova iOS plugin exposing the full WebRTC W3C JavaScript APIs
+ * Copyright 2015-2017 eFace2Face, Inc. (https://eface2face.com)
+ * Copyright 2015-2019 BasqueVoIPMafia (https://github.com/BasqueVoIPMafia)
+ * Copyright 2017-2023 Cordova-RTC (https://github.com/cordova-rtc)
+ * License MIT
+ */
+
+(function(f){if(typeof exports==="object"&&typeof module!=="undefined"){module.exports=f()}else if(typeof define==="function"&&define.amd){define([],f)}else{var g;if(typeof window!=="undefined"){g=window}else if(typeof global!=="undefined"){g=global}else if(typeof self!=="undefined"){g=self}else{g=this}g.iosrtc = f()}})(function(){var define,module,exports;return (function(){function r(e,n,t){function o(i,f){if(!n[i]){if(!e[i]){var c="function"==typeof require&&require;if(!f&&c)return c(i,!0);if(u)return u(i,!0);var a=new Error("Cannot find module '"+i+"'");throw a.code="MODULE_NOT_FOUND",a}var p=n[i]={exports:{}};e[i][0].call(p.exports,function(r){var n=e[i][1][r];return o(n||r)},p,p.exports,r,e,n,t)}return n[i].exports}for(var u="function"==typeof require&&require,i=0;i<t.length;i++)o(t[i]);return o}return r})()({1:[function(_dereq_,module,exports){
+/**
+ * Expose an object with WebRTC Errors.
+ */
+var Errors = (module.exports = {}),
+	/**
+	 * Local variables.
+	 */
+	IntermediateInheritor = function () {};
+
+IntermediateInheritor.prototype = Error.prototype;
+
+/**
+ * Create error classes.
+ */
+addError('InvalidStateError');
+addError('InvalidSessionDescriptionError');
+addError('InternalError');
+addError('MediaStreamError');
+
+function addError(name) {
+	Errors[name] = function () {
+		var tmp = Error.apply(this, arguments);
+
+		this.name = tmp.name = name;
+		this.message = tmp.message;
+
+		Object.defineProperty(this, 'stack', {
+			get: function () {
+				return tmp.stack;
+			}
+		});
+
+		return this;
+	};
+
+	Errors[name].prototype = new IntermediateInheritor();
+}
+
+// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+// TODO remove on 6.0.0
+Errors.detectDeprecatedCallbaksUsage = function detectDeprecatedCallbaksUsage(funcName, arg) {
+	if (typeof arg[1] === 'function' || typeof arg[2] === 'function') {
+		throw new Error(
+			'Callbacks are not supported by "' + funcName + '" anymore, use Promise instead.'
+		);
+	}
+};
+
+},{}],2:[function(_dereq_,module,exports){
+/**
+ * Dependencies.
+ */
+var YaetiEventTarget = _dereq_('yaeti').EventTarget;
+
+var EventTarget = function () {
+	YaetiEventTarget.call(this);
+};
+
+EventTarget.prototype = Object.create(YaetiEventTarget.prototype);
+EventTarget.prototype.constructor = EventTarget;
+
+Object.defineProperties(
+	EventTarget.prototype,
+	Object.getOwnPropertyDescriptors(YaetiEventTarget.prototype)
+);
+
+EventTarget.prototype.dispatchEvent = function (event) {
+	Object.defineProperty(event, 'target', {
+		value: this,
+		writable: false
+	});
+
+	YaetiEventTarget.prototype.dispatchEvent.call(this, event);
+};
+
+/**
+ * Expose the EventTarget class.
+ */
+module.exports = EventTarget;
+
+},{"yaeti":30}],3:[function(_dereq_,module,exports){
+/**
+ * Expose the MediaDeviceInfo class.
+ */
+module.exports = MediaDeviceInfo;
+
+function MediaDeviceInfo(data) {
+	data = data || {};
+
+	Object.defineProperties(this, {
+		// MediaDeviceInfo spec.
+		deviceId: {
+			value: data.deviceId
+		},
+		kind: {
+			value: data.kind
+		},
+		label: {
+			value: data.label
+		},
+		groupId: {
+			value: data.groupId || ''
+		},
+		// SourceInfo old spec.
+		id: {
+			value: data.deviceId
+		},
+		// Deprecated, but useful until there is an alternative
+		facing: {
+			value: ''
+		}
+	});
+}
+
+},{}],4:[function(_dereq_,module,exports){
+/**
+ * Expose the MediaDevices class.
+ */
+module.exports = MediaDevices;
+
+/**
+ * Spec: https://w3c.github.io/mediacapture-main/#dom-mediadevices
+ */
+
+/**
+ * Dependencies.
+ */
+var EventTarget = _dereq_('./EventTarget'),
+	getUserMedia = _dereq_('./getUserMedia'),
+	enumerateDevices = _dereq_('./enumerateDevices');
+
+function MediaDevices(data) {
+	//ondevicechange
+	//enumerateDevices
+	//getDisplayMedia
+	//getSupportedConstraints
+	//getUserMedia
+
+	var self = this;
+
+	// Make this an EventTarget.
+	EventTarget.call(self);
+
+	data = data || {};
+}
+
+MediaDevices.prototype = Object.create(EventTarget.prototype);
+MediaDevices.prototype.constructor = MediaDevices;
+
+MediaDevices.prototype.getUserMedia = function (constraints) {
+	return getUserMedia(constraints);
+};
+
+MediaDevices.prototype.enumerateDevices = function () {
+	return enumerateDevices();
+};
+
+MediaDevices.prototype.getSupportedConstraints = function () {
+	return {
+		// Supported
+		height: true,
+		width: true,
+		deviceId: true,
+		frameRate: true,
+		sampleRate: true,
+		aspectRatio: true,
+		// Not Supported
+		autoGainControl: false,
+		brightness: false,
+		channelCount: false,
+		colorTemperature: false,
+		contrast: false,
+		echoCancellation: false,
+		exposureCompensation: false,
+		exposureMode: false,
+		exposureTime: false,
+		facingMode: true,
+		focusDistance: false,
+		focusMode: false,
+		groupId: false,
+		iso: false,
+		latency: false,
+		noiseSuppression: false,
+		pointsOfInterest: false,
+		resizeMode: false,
+		sampleSize: false,
+		saturation: false,
+		sharpness: false,
+		torch: false,
+		whiteBalanceMode: false,
+		zoom: false
+	};
+};
+
+},{"./EventTarget":2,"./enumerateDevices":19,"./getUserMedia":20}],5:[function(_dereq_,module,exports){
+/**
+ * Expose the MediaStream class.
+ */
+module.exports = MediaStream;
+
+/**
+ * Spec: http://w3c.github.io/mediacapture-main/#mediastream
+ */
+
+/**
+ * Dependencies.
+ */
+var debug = _dereq_('debug')('iosrtc:MediaStream'),
+	exec = _dereq_('cordova/exec'),
+	EventTarget = _dereq_('./EventTarget'),
+	{ MediaStreamTrack } = _dereq_('./MediaStreamTrack'),
+	/**
+	 * Local variables.
+	 */
+
+	// Dictionary of MediaStreams (provided via getMediaStreams() class method).
+	// - key: MediaStream blobId.
+	// - value: MediaStream.
+	mediaStreams;
+
+// TODO longer UUID like native call
+// - "4021904575-2849079001-3048689102-1644344044-4021904575-2849079001-3048689102-1644344044"
+function newMediaStreamId() {
+	return window.crypto.getRandomValues(new Uint32Array(4)).join('-');
+}
+
+// Save original MediaStream
+var originalMediaStream = window.MediaStream || window.Blob;
+//var originalMediaStream = window.Blob;
+var originalMediaStreamTrack = MediaStreamTrack.originalMediaStreamTrack;
+
+/**
+ * Expose the MediaStream class.
+ */
+function MediaStream(arg, id) {
+	debug('new MediaStream(arg) | [arg:%o]', arg);
+
+	// Detect native MediaStream usage
+	// new MediaStream(originalMediaStream) // stream
+	// new MediaStream(originalMediaStreamTrack[]) // tracks
+	if (
+		(!(arg instanceof window.Blob) &&
+			arg instanceof originalMediaStream &&
+			typeof arg.getBlobId === 'undefined') ||
+		(Array.isArray(arg) && arg[0] instanceof originalMediaStreamTrack)
+	) {
+		return new originalMediaStream(arg);
+	}
+
+	// new MediaStream(MediaStream) // stream
+	// new MediaStream(MediaStreamTrack[]) // tracks
+	// new MediaStream() // empty
+
+	id = id || newMediaStreamId();
+	var blobId = 'MediaStream_' + id;
+
+	// Extend returned MediaTream with custom MediaStream
+	var stream;
+	if (originalMediaStream !== window.Blob) {
+		stream = new (Function.prototype.bind.apply(originalMediaStream.bind(this), []))();
+	} else {
+		// Fallback on Blob if originalMediaStream is not a MediaStream and Emulate EventTarget
+		stream = new Blob([blobId], {
+			type: 'stream'
+		});
+
+		var target = document.createTextNode(null);
+		stream.addEventListener = target.addEventListener.bind(target);
+		stream.removeEventListener = target.removeEventListener.bind(target);
+		stream.dispatchEvent = target.dispatchEvent.bind(target);
+	}
+
+	Object.defineProperties(stream, Object.getOwnPropertyDescriptors(MediaStream.prototype));
+
+	// Make it an EventTarget.
+	EventTarget.call(stream);
+
+	// Public atributes.
+	stream._id = id || newMediaStreamId();
+	stream._active = true;
+
+	// Init Stream by Id
+	exec(null, null, 'iosrtcPlugin', 'MediaStream_init', [stream.id]);
+
+	// Public but internal attributes.
+	stream.connected = false;
+
+	stream._addedToConnection = false;
+
+	// Private attributes.
+	stream._audioTracks = {};
+	stream._videoTracks = {};
+
+	// Store the stream into the dictionary.
+	stream._blobId = blobId;
+	mediaStreams[stream._blobId] = stream;
+
+	// Convert arg to array of tracks if possible
+	if (arg instanceof MediaStream || arg instanceof MediaStream.originalMediaStream) {
+		arg = arg.getTracks();
+	}
+
+	if (Array.isArray(arg)) {
+		arg.forEach(function (track) {
+			stream.addTrack(track);
+		});
+	} else if (typeof arg !== 'undefined') {
+		throw new TypeError(
+			"Failed to construct 'MediaStream': No matching constructor signature."
+		);
+	}
+
+	function onResultOK(data) {
+		onEvent.call(stream, data);
+	}
+	exec(onResultOK, null, 'iosrtcPlugin', 'MediaStream_setListener', [stream.id]);
+
+	return stream;
+}
+
+MediaStream.prototype = Object.create(originalMediaStream.prototype, {
+	id: {
+		get: function () {
+			return this._id;
+		}
+	},
+	active: {
+		get: function () {
+			return this._active;
+		}
+	},
+	// Backwards compatibility.
+	label: {
+		get: function () {
+			return this._id;
+		}
+	},
+	addedToConnection: {
+		get: function () {
+			return this._addedToConnection;
+		},
+		set: function (value) {
+			this._addedToConnection = value;
+		}
+	}
+});
+
+Object.defineProperties(
+	MediaStream.prototype,
+	Object.getOwnPropertyDescriptors(EventTarget.prototype)
+);
+
+MediaStream.prototype.constructor = MediaStream;
+
+// Static reference to original MediaStream
+MediaStream.originalMediaStream = originalMediaStream;
+
+/**
+ * Class methods.
+ */
+
+MediaStream.setMediaStreams = function (_mediaStreams) {
+	mediaStreams = _mediaStreams;
+};
+
+MediaStream.getMediaStreams = function () {
+	return mediaStreams;
+};
+
+MediaStream.create = function (dataFromEvent) {
+	debug('create() | [dataFromEvent:%o]', dataFromEvent);
+
+	var trackId,
+		track,
+		stream = new MediaStream([], dataFromEvent.id);
+
+	// We do not use addTrack to prevent false positive "ERROR: video track not added" and "ERROR: audio track not added"
+	// cause the rtcMediaStream already has them internaly.
+
+	for (trackId in dataFromEvent.audioTracks) {
+		if (dataFromEvent.audioTracks.hasOwnProperty(trackId)) {
+			track = new MediaStreamTrack(dataFromEvent.audioTracks[trackId]);
+
+			stream._audioTracks[track.id] = track;
+
+			addListenerForTrackEnded.call(stream, track);
+		}
+	}
+
+	for (trackId in dataFromEvent.videoTracks) {
+		if (dataFromEvent.videoTracks.hasOwnProperty(trackId)) {
+			track = new MediaStreamTrack(dataFromEvent.videoTracks[trackId]);
+
+			stream._videoTracks[track.id] = track;
+
+			addListenerForTrackEnded.call(stream, track);
+		}
+	}
+
+	return stream;
+};
+
+MediaStream.prototype.getBlobId = function () {
+	return this._blobId;
+};
+
+MediaStream.prototype.getAudioTracks = function () {
+	debug('getAudioTracks()');
+
+	var tracks = [],
+		id;
+
+	for (id in this._audioTracks) {
+		if (this._audioTracks.hasOwnProperty(id)) {
+			tracks.push(this._audioTracks[id]);
+		}
+	}
+
+	return tracks;
+};
+
+MediaStream.prototype.getVideoTracks = function () {
+	debug('getVideoTracks()');
+
+	var tracks = [],
+		id;
+
+	for (id in this._videoTracks) {
+		if (this._videoTracks.hasOwnProperty(id)) {
+			tracks.push(this._videoTracks[id]);
+		}
+	}
+
+	return tracks;
+};
+
+MediaStream.prototype.getTracks = function () {
+	debug('getTracks()');
+
+	var tracks = [],
+		id;
+
+	for (id in this._audioTracks) {
+		if (this._audioTracks.hasOwnProperty(id)) {
+			tracks.push(this._audioTracks[id]);
+		}
+	}
+
+	for (id in this._videoTracks) {
+		if (this._videoTracks.hasOwnProperty(id)) {
+			tracks.push(this._videoTracks[id]);
+		}
+	}
+
+	return tracks;
+};
+
+MediaStream.prototype.getTrackById = function (id) {
+	debug('getTrackById()');
+
+	return this._audioTracks[id] || this._videoTracks[id] || null;
+};
+
+MediaStream.prototype.addTrack = function (track) {
+	debug('addTrack() [track:%o]', track);
+
+	if (!(track instanceof MediaStreamTrack)) {
+		throw new Error('argument must be an instance of MediaStreamTrack');
+	}
+
+	if (this._audioTracks[track.id] || this._videoTracks[track.id]) {
+		return;
+	}
+
+	if (track.kind === 'audio') {
+		this._audioTracks[track.id] = track;
+	} else if (track.kind === 'video') {
+		this._videoTracks[track.id] = track;
+	} else {
+		throw new Error('unknown kind attribute: ' + track.kind);
+	}
+
+	addListenerForTrackEnded.call(this, track);
+
+	exec(null, null, 'iosrtcPlugin', 'MediaStream_addTrack', [this.id, track.id]);
+
+	this.dispatchEvent(new Event('update'));
+
+	this.emitConnected();
+};
+
+MediaStream.prototype.removeTrack = function (track) {
+	debug('removeTrack() [track:%o]', track);
+
+	if (!(track instanceof MediaStreamTrack)) {
+		throw new Error('argument must be an instance of MediaStreamTrack');
+	}
+
+	if (!this._audioTracks[track.id] && !this._videoTracks[track.id]) {
+		return;
+	}
+
+	if (track.kind === 'audio') {
+		delete this._audioTracks[track.id];
+	} else if (track.kind === 'video') {
+		delete this._videoTracks[track.id];
+	} else {
+		throw new Error('unknown kind attribute: ' + track.kind);
+	}
+
+	exec(null, null, 'iosrtcPlugin', 'MediaStream_removeTrack', [this.id, track.id]);
+
+	this.dispatchEvent(new Event('update'));
+
+	checkActive.call(this);
+};
+
+MediaStream.prototype.clone = function () {
+	var newStream = MediaStream();
+	this.getTracks().forEach(function (track) {
+		newStream.addTrack(track.clone());
+	});
+
+	return newStream;
+};
+
+// Backwards compatible API.
+MediaStream.prototype.stop = function () {
+	debug('stop()');
+
+	var trackId;
+
+	for (trackId in this._audioTracks) {
+		if (this._audioTracks.hasOwnProperty(trackId)) {
+			this._audioTracks[trackId].stop();
+		}
+	}
+
+	for (trackId in this._videoTracks) {
+		if (this._videoTracks.hasOwnProperty(trackId)) {
+			this._videoTracks[trackId].stop();
+		}
+	}
+};
+
+// TODO: API methods and events.
+
+/**
+ * Private API.
+ */
+
+MediaStream.prototype.emitConnected = function () {
+	debug('emitConnected()');
+
+	var self = this;
+
+	if (this.connected) {
+		return;
+	}
+	this.connected = true;
+
+	setTimeout(
+		function (self) {
+			var event = new Event('connected');
+			Object.defineProperty(event, 'target', { value: self, enumerable: true });
+			self.dispatchEvent(event);
+		},
+		0,
+		self
+	);
+};
+
+function addListenerForTrackEnded(track) {
+	var self = this;
+
+	track.addEventListener('ended', function () {
+		if (track.kind === 'audio' && !self._audioTracks[track.id]) {
+			return;
+		} else if (track.kind === 'video' && !self._videoTracks[track.id]) {
+			return;
+		}
+
+		checkActive.call(self);
+	});
+}
+
+function checkActive() {
+	// A MediaStream object is said to be active when it has at least one MediaStreamTrack
+	// that has not ended. A MediaStream that does not have any tracks or only has tracks
+	// that are ended is inactive.
+
+	var self = this,
+		trackId;
+
+	if (!this.active) {
+		return;
+	}
+	// Fixes Twilio fails to read a local video if the stream is released.
+	if (this._addedToConnection) {
+		return;
+	}
+
+	if (
+		Object.keys(this._audioTracks).length === 0 &&
+		Object.keys(this._videoTracks).length === 0
+	) {
+		debug('no tracks, releasing MediaStream');
+
+		release();
+		return;
+	}
+
+	for (trackId in this._audioTracks) {
+		if (this._audioTracks.hasOwnProperty(trackId)) {
+			if (this._audioTracks[trackId].readyState !== 'ended') {
+				return;
+			}
+		}
+	}
+
+	for (trackId in this._videoTracks) {
+		if (this._videoTracks.hasOwnProperty(trackId)) {
+			if (this._videoTracks[trackId].readyState !== 'ended') {
+				return;
+			}
+		}
+	}
+
+	release();
+
+	function release() {
+		debug('all tracks are ended, releasing MediaStream %s', self.id);
+		self._active = false;
+		self.dispatchEvent(new Event('inactive'));
+
+		// Remove the stream from the dictionary.
+		delete mediaStreams[self._blobId];
+
+		exec(null, null, 'iosrtcPlugin', 'MediaStream_release', [self.id]);
+	}
+}
+
+function onEvent(data) {
+	var type = data.type,
+		event,
+		track;
+
+	debug('onEvent() | [type:%s, data:%o]', type, data);
+
+	switch (type) {
+		case 'addtrack':
+			// check if a track already exists before initializing a new
+			// track and calling setListener again.
+			if (data.track.kind === 'audio') {
+				track = this._audioTracks[data.track.id];
+			} else if (data.track.kind === 'video') {
+				track = this._videoTracks[data.track.id];
+			}
+			if (!track) {
+				track = new MediaStreamTrack(data.track);
+				if (track.kind === 'audio') {
+					this._audioTracks[track.id] = track;
+				} else if (track.kind === 'video') {
+					this._videoTracks[track.id] = track;
+				}
+				addListenerForTrackEnded.call(this, track);
+			}
+
+			event = new Event('addtrack');
+			event.track = track;
+
+			this.dispatchEvent(event);
+
+			// Also emit 'update' for the MediaStreamRenderer.
+			this.dispatchEvent(new Event('update'));
+			break;
+
+		case 'removetrack':
+			if (data.track.kind === 'audio') {
+				track = this._audioTracks[data.track.id];
+				delete this._audioTracks[data.track.id];
+			} else if (data.track.kind === 'video') {
+				track = this._videoTracks[data.track.id];
+				delete this._videoTracks[data.track.id];
+			}
+
+			if (!track) {
+				throw new Error(
+					'"removetrack" event fired on MediaStream for a non existing MediaStreamTrack'
+				);
+			}
+
+			event = new Event('removetrack');
+			event.track = track;
+
+			this.dispatchEvent(event);
+
+			// Also emit 'update' for the MediaStreamRenderer.
+			this.dispatchEvent(new Event('update'));
+
+			// Check whether the MediaStream still is active.
+			checkActive.call(this);
+			break;
+	}
+}
+
+},{"./EventTarget":2,"./MediaStreamTrack":7,"cordova/exec":undefined,"debug":23}],6:[function(_dereq_,module,exports){
+/**
+ * Expose the MediaStreamRenderer class.
+ */
+module.exports = MediaStreamRenderer;
+
+/**
+ * Dependencies.
+ */
+var debug = _dereq_('debug')('iosrtc:MediaStreamRenderer'),
+	exec = _dereq_('cordova/exec'),
+	randomNumber = _dereq_('random-number').generator({ min: 10000, max: 99999, integer: true }),
+	EventTarget = _dereq_('./EventTarget'),
+	MediaStream = _dereq_('./MediaStream');
+
+function MediaStreamRenderer(element) {
+	debug('new() | [element:"%s"]', element);
+
+	var self = this;
+
+	// Make this an EventTarget.
+	EventTarget.call(this);
+
+	if (!(element instanceof HTMLElement)) {
+		throw new Error('a valid HTMLElement is required');
+	}
+
+	// Public atributes.
+	this.element = element;
+	this.stream = undefined;
+	this.videoWidth = undefined;
+	this.videoHeight = undefined;
+
+	// Private attributes.
+	this.id = randomNumber();
+
+	function onResultOK(data) {
+		onEvent.call(self, data);
+	}
+
+	exec(onResultOK, null, 'iosrtcPlugin', 'new_MediaStreamRenderer', [this.id]);
+
+	this.refresh();
+
+	// TODO cause video resizing jiggling add semaphore
+	//this.refreshInterval = setInterval(function () {
+	//	self.refresh(self);
+	//}, 500);
+
+	element.render = this;
+}
+
+MediaStreamRenderer.prototype = Object.create(EventTarget.prototype);
+MediaStreamRenderer.prototype.constructor = MediaStreamRenderer;
+
+MediaStreamRenderer.prototype.render = function (stream) {
+	debug('render() [stream:%o]', stream);
+
+	var self = this;
+
+	if (!(stream instanceof MediaStream.originalMediaStream)) {
+		throw new Error('render() requires a MediaStream instance as argument');
+	}
+
+	self.stream = stream;
+
+	exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_render', [self.id, stream.id]);
+
+	// Subscribe to 'update' event so we call native mediaStreamChanged() on it.
+	stream.addEventListener('update', function () {
+		if (self.stream !== stream) {
+			return;
+		}
+
+		debug('MediaStream emits "update", calling native mediaStreamChanged()');
+
+		exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_mediaStreamChanged', [self.id]);
+	});
+
+	// Subscribe to 'inactive' event and emit "close" so the video element can react.
+	stream.addEventListener('inactive', function () {
+		if (self.stream !== stream) {
+			return;
+		}
+
+		debug('MediaStream emits "inactive", emiting "close" and closing this MediaStreamRenderer');
+
+		self.dispatchEvent(new Event('close'));
+		self.close();
+	});
+
+	if (stream.connected) {
+		connected();
+	} else {
+		// Otherwise subscribe to 'connected' event to emulate video elements events.
+		stream.addEventListener('connected', function () {
+			if (self.stream !== stream) {
+				return;
+			}
+
+			connected();
+		});
+	}
+
+	function connected() {
+		// Emit video events.
+		self.element.dispatchEvent(new Event('loadedmetadata'));
+		self.element.dispatchEvent(new Event('loadeddata'));
+		self.element.dispatchEvent(new Event('canplay'));
+		self.element.dispatchEvent(new Event('canplaythrough'));
+	}
+};
+
+MediaStreamRenderer.prototype.save = function (callback) {
+	debug('save()');
+
+	if (!this.stream) {
+		callback(null);
+		return;
+	}
+
+	function onResultOK(data) {
+		callback(data);
+	}
+
+	function onResultError() {
+		callback(null);
+	}
+
+	exec(onResultOK, onResultError, 'iosrtcPlugin', 'MediaStreamRenderer_save', [this.id]);
+};
+
+MediaStreamRenderer.prototype.refresh = function () {
+	debug('refresh()');
+
+	var elementPositionAndSize = getElementPositionAndSize.call(this),
+		computedStyle,
+		videoRatio,
+		elementRatio,
+		elementLeft = elementPositionAndSize.left,
+		elementTop = elementPositionAndSize.top,
+		elementWidth = elementPositionAndSize.width,
+		elementHeight = elementPositionAndSize.height,
+		videoViewWidth,
+		videoViewHeight,
+		visible,
+		opacity,
+		zIndex,
+		mirrored,
+		objectFit,
+		clip,
+		borderRadius,
+		paddingTop,
+		paddingBottom,
+		paddingLeft,
+		paddingRight,
+		backgroundColorRgba,
+		self = this;
+
+	computedStyle = window.getComputedStyle(this.element);
+
+	// get background color
+	backgroundColorRgba = computedStyle.backgroundColor
+		.replace(/rgba?\((.*)\)/, '$1')
+		.split(',')
+		.map(function (x) {
+			return x.trim();
+		});
+	backgroundColorRgba[3] = '0';
+	this.element.style.backgroundColor = 'rgba(' + backgroundColorRgba.join(',') + ')';
+	backgroundColorRgba.length = 3;
+
+	// get padding values
+	paddingTop = parseInt(computedStyle.paddingTop) | 0;
+	paddingBottom = parseInt(computedStyle.paddingBottom) | 0;
+	paddingLeft = parseInt(computedStyle.paddingLeft) | 0;
+	paddingRight = parseInt(computedStyle.paddingRight) | 0;
+
+	// fix position according to padding
+	elementLeft += paddingLeft;
+	elementTop += paddingTop;
+
+	// fix width and height according to padding
+	elementWidth -= paddingLeft + paddingRight;
+	elementHeight -= paddingTop + paddingBottom;
+
+	videoViewWidth = elementWidth;
+	videoViewHeight = elementHeight;
+
+	// visible
+	if (computedStyle.visibility === 'hidden') {
+		visible = false;
+	} else {
+		visible = !!this.element.offsetHeight; // Returns 0 if element or any parent is hidden.
+	}
+
+	// opacity
+	opacity = parseFloat(computedStyle.opacity);
+
+	// zIndex
+	zIndex = parseFloat(computedStyle.zIndex) || parseFloat(this.element.style.zIndex) || 0;
+
+	// mirrored (detect "-webkit-transform: scaleX(-1);" or equivalent)
+	if (
+		computedStyle.transform === 'matrix(-1, 0, 0, 1, 0, 0)' ||
+		computedStyle['-webkit-transform'] === 'matrix(-1, 0, 0, 1, 0, 0)'
+	) {
+		mirrored = true;
+	} else {
+		mirrored = false;
+	}
+
+	// objectFit ('contain' is set as default value)
+	objectFit = computedStyle.objectFit || 'contain';
+
+	// clip
+	if (objectFit === 'none') {
+		clip = false;
+	} else {
+		clip = true;
+	}
+
+	// borderRadius
+	borderRadius = parseFloat(computedStyle.borderRadius);
+	if (/%$/.test(borderRadius)) {
+		borderRadius = Math.min(elementHeight, elementWidth) * borderRadius;
+	}
+
+	/**
+	 * No video yet, so just update the UIView with the element settings.
+	 */
+
+	if (!this.videoWidth || !this.videoHeight) {
+		debug('refresh() | no video track yet');
+
+		nativeRefresh.call(this);
+		return;
+	}
+
+	videoRatio = this.videoWidth / this.videoHeight;
+
+	/**
+	 * Element has no width and/or no height.
+	 */
+
+	if (!elementWidth || !elementHeight) {
+		debug('refresh() | video element has 0 width and/or 0 height');
+
+		nativeRefresh.call(this);
+		return;
+	}
+
+	/**
+	 * Set video view position and size.
+	 */
+
+	elementRatio = elementWidth / elementHeight;
+
+	switch (objectFit) {
+		case 'cover':
+			// The element has higher or equal width/height ratio than the video.
+			if (elementRatio >= videoRatio) {
+				videoViewWidth = elementWidth;
+				videoViewHeight = videoViewWidth / videoRatio;
+			} else if (elementRatio < videoRatio) {
+				// The element has lower width/height ratio than the video.
+				videoViewHeight = elementHeight;
+				videoViewWidth = videoViewHeight * videoRatio;
+			}
+			break;
+
+		case 'fill':
+			videoViewHeight = elementHeight;
+			videoViewWidth = elementWidth;
+			break;
+
+		case 'none':
+			videoViewHeight = this.videoHeight;
+			videoViewWidth = this.videoWidth;
+			break;
+
+		case 'scale-down':
+			// Same as 'none'.
+			if (this.videoWidth <= elementWidth && this.videoHeight <= elementHeight) {
+				videoViewHeight = this.videoHeight;
+				videoViewWidth = this.videoWidth;
+			} else {
+				// Same as 'contain'.
+				if (elementRatio >= videoRatio) {
+					// The element has higher or equal width/height ratio than the video.
+					videoViewHeight = elementHeight;
+					videoViewWidth = videoViewHeight * videoRatio;
+				} else if (elementRatio < videoRatio) {
+					// The element has lower width/height ratio than the video.
+					videoViewWidth = elementWidth;
+					videoViewHeight = videoViewWidth / videoRatio;
+				}
+			}
+			break;
+
+		default:
+			// 'contain'.
+			objectFit = 'contain';
+			if (elementRatio >= videoRatio) {
+				// The element has higher or equal width/height ratio than the video.
+				videoViewHeight = elementHeight;
+				videoViewWidth = videoViewHeight * videoRatio;
+			} else if (elementRatio < videoRatio) {
+				// The element has lower width/height ratio than the video.
+				videoViewWidth = elementWidth;
+				videoViewHeight = videoViewWidth / videoRatio;
+			}
+			break;
+	}
+
+	nativeRefresh.call(this);
+
+	function hash(str) {
+		var hash = 5381,
+			i = str.length;
+
+		while (i) {
+			hash = (hash * 33) ^ str.charCodeAt(--i);
+		}
+
+		return hash >>> 0;
+	}
+
+	function nativeRefresh() {
+		var data = {
+				elementLeft: Math.round(elementLeft),
+				elementTop: Math.round(elementTop),
+				elementWidth: Math.round(elementWidth),
+				elementHeight: Math.round(elementHeight),
+				videoViewWidth: Math.round(videoViewWidth),
+				videoViewHeight: Math.round(videoViewHeight),
+				visible: visible,
+				backgroundColor: backgroundColorRgba.join(','),
+				opacity: opacity,
+				zIndex: zIndex,
+				mirrored: mirrored,
+				objectFit: objectFit,
+				clip: clip,
+				borderRadius: borderRadius
+			},
+			newRefreshCached = hash(JSON.stringify(data));
+
+		if (newRefreshCached === self.refreshCached) {
+			return;
+		}
+
+		self.refreshCached = newRefreshCached;
+
+		debug('refresh() | [data:%o]', data);
+
+		exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_refresh', [this.id, data]);
+	}
+};
+
+MediaStreamRenderer.prototype.close = function () {
+	debug('close()');
+
+	if (!this.stream) {
+		return;
+	}
+	this.stream = undefined;
+
+	exec(null, null, 'iosrtcPlugin', 'MediaStreamRenderer_close', [this.id]);
+	if (this.refreshInterval) {
+		clearInterval(this.refreshInterval);
+		delete this.refreshInterval;
+	}
+};
+
+/**
+ * Private API.
+ */
+
+function onEvent(data) {
+	var type = data.type,
+		event;
+
+	debug('onEvent() | [type:%s, data:%o]', type, data);
+
+	switch (type) {
+		case 'videoresize':
+			this.videoWidth = data.size.width;
+			this.videoHeight = data.size.height;
+			this.refresh();
+
+			event = new Event(type);
+			event.videoWidth = data.size.width;
+			event.videoHeight = data.size.height;
+			this.dispatchEvent(event);
+
+			break;
+	}
+}
+
+function getElementPositionAndSize() {
+	var rect = this.element.getBoundingClientRect();
+
+	return {
+		left: rect.left + this.element.clientLeft,
+		top: rect.top + this.element.clientTop,
+		width: this.element.clientWidth,
+		height: this.element.clientHeight
+	};
+}
+
+},{"./EventTarget":2,"./MediaStream":5,"cordova/exec":undefined,"debug":23,"random-number":28}],7:[function(_dereq_,module,exports){
+/**
+ * Expose the MediaStreamTrack class.
+ */
+module.exports.MediaStreamTrack = MediaStreamTrack;
+module.exports.newMediaStreamTrackId = newMediaStreamTrackId;
+
+/**
+ * Spec: http://w3c.github.io/mediacapture-main/#mediastreamtrack
+ */
+
+/**
+ * Dependencies.
+ */
+var debug = _dereq_('debug')('iosrtc:MediaStreamTrack'),
+	exec = _dereq_('cordova/exec'),
+	enumerateDevices = _dereq_('./enumerateDevices'),
+	MediaTrackCapabilities = _dereq_('./MediaTrackCapabilities'),
+	MediaTrackSettings = _dereq_('./MediaTrackSettings'),
+	EventTarget = _dereq_('./EventTarget');
+
+// Save original MediaStreamTrack
+var originalMediaStreamTrack = window.MediaStreamTrack || function dummyMediaStreamTrack() {};
+
+// All "active" tracks, to ensure RTCRtpSender gets the same object
+var activeMediaStreamTracks = new Map();
+
+function newMediaStreamTrackId() {
+	return window.crypto.getRandomValues(new Uint32Array(4)).join('-');
+}
+
+function MediaStreamTrack(dataFromEvent) {
+	if (!dataFromEvent) {
+		throw new Error('Illegal constructor');
+	}
+
+	debug('new() | [dataFromEvent:%o]', dataFromEvent);
+
+	var self = this;
+
+	// Make this an EventTarget.
+	EventTarget.call(this);
+
+	// Public atributes.
+	this.id = dataFromEvent.id; // NOTE: It's a string.
+	this.kind = dataFromEvent.kind;
+	this.label = dataFromEvent.label;
+	this.muted = false; // TODO: No "muted" property in ObjC API.
+	this.capabilities = dataFromEvent.capabilities;
+	this.readyState = dataFromEvent.readyState;
+
+	// Private attributes.
+	this._enabled = dataFromEvent.enabled;
+	this._ended = false;
+
+	this.dataFromEvent = dataFromEvent;
+
+	activeMediaStreamTracks.set(this.id, this);
+
+	function onResultOK(data) {
+		onEvent.call(self, data);
+	}
+
+	exec(onResultOK, null, 'iosrtcPlugin', 'MediaStreamTrack_setListener', [this.id]);
+}
+
+MediaStreamTrack.prototype = Object.create(EventTarget.prototype);
+MediaStreamTrack.prototype.constructor = MediaStreamTrack;
+
+// Static reference to original MediaStreamTrack
+MediaStreamTrack.originalMediaStreamTrack = originalMediaStreamTrack;
+
+// Setters.
+Object.defineProperty(MediaStreamTrack.prototype, 'enabled', {
+	get: function () {
+		return this._enabled;
+	},
+	set: function (value) {
+		debug('enabled = %s', !!value);
+
+		this._enabled = !!value;
+		exec(null, null, 'iosrtcPlugin', 'MediaStreamTrack_setEnabled', [this.id, this._enabled]);
+	}
+});
+
+MediaStreamTrack.prototype.getConstraints = function () {
+	debug('MediaStreamTrack.prototype.getConstraints  is not implemented.');
+	return {};
+};
+
+MediaStreamTrack.prototype.applyConstraints = function (constraints) {
+	debug('MediaStreamTrack.prototype.applyConstraints  is not implemented.', constraints);
+	return Promise.reject(new Error('applyConstraints is not implemented.'));
+};
+
+MediaStreamTrack.prototype.clone = function () {
+	var newTrackId = newMediaStreamTrackId();
+
+	exec(null, null, 'iosrtcPlugin', 'MediaStreamTrack_clone', [this.id, newTrackId]);
+
+	return new MediaStreamTrack({
+		id: newTrackId,
+		kind: this.kind,
+		label: this.label,
+		readyState: this.readyState,
+		enabled: this.enabled,
+		trackId: this.dataFromEvent.trackId
+	});
+};
+
+MediaStreamTrack.prototype.getCapabilities = function () {
+	return new MediaTrackCapabilities(this.capabilities);
+};
+
+MediaStreamTrack.prototype.getSettings = function () {
+	//throw new Error('Not implemented.');
+	// SHAM
+	return new MediaTrackSettings();
+};
+
+MediaStreamTrack.prototype.stop = function () {
+	debug('stop()');
+
+	if (this._ended) {
+		return;
+	}
+
+	exec(null, null, 'iosrtcPlugin', 'MediaStreamTrack_stop', [this.id]);
+};
+
+// TODO: API methods and events.
+
+/**
+ * Class methods.
+ */
+
+MediaStreamTrack.getSources = function () {
+	debug('getSources()');
+
+	return enumerateDevices.apply(this, arguments);
+};
+
+MediaStreamTrack.findOrCreate = function (trackData) {
+	var id = trackData.id;
+	debug('findOrCreate()', id);
+
+	return activeMediaStreamTracks.get(id) || new MediaStreamTrack(trackData);
+};
+
+/**
+ * Private API.
+ */
+
+function onEvent(data) {
+	var type = data.type;
+
+	debug('onEvent() | [type:%s, data:%o]', type, data);
+
+	switch (type) {
+		case 'statechange':
+			this.readyState = data.readyState;
+			this._enabled = data.enabled;
+
+			switch (data.readyState) {
+				case 'initializing':
+					break;
+				case 'live':
+					break;
+				case 'ended':
+					activeMediaStreamTracks.delete(this.id);
+					this._ended = true;
+					this.dispatchEvent(new Event('ended'));
+					break;
+				case 'failed':
+					break;
+			}
+			break;
+	}
+}
+
+},{"./EventTarget":2,"./MediaTrackCapabilities":8,"./MediaTrackSettings":9,"./enumerateDevices":19,"cordova/exec":undefined,"debug":23}],8:[function(_dereq_,module,exports){
+/**
+ * Expose the MediaTrackSettings class.
+ */
+module.exports = MediaTrackCapabilities;
+
+// Ref https://www.w3.org/TR/mediacapture-streams/#dom-mediatrackcapabilities
+function MediaTrackCapabilities(data) {
+	data = data || {};
+
+	this.deviceId = data.deviceId;
+}
+
+},{}],9:[function(_dereq_,module,exports){
+/**
+ * Expose the MediaTrackSettings class.
+ */
+module.exports = MediaTrackSettings;
+
+// Ref https://www.w3.org/TR/mediacapture-streams/#dom-mediatracksettings
+function MediaTrackSettings(data) {
+	data = data || {};
+}
+
+},{}],10:[function(_dereq_,module,exports){
+/**
+ * Expose the RTCDTMFSender class.
+ */
+module.exports = RTCDTMFSender;
+
+/**
+ * Dependencies.
+ */
+var debug = _dereq_('debug')('iosrtc:RTCDTMFSender'),
+	debugerror = _dereq_('debug')('iosrtc:ERROR:RTCDTMFSender'),
+	exec = _dereq_('cordova/exec'),
+	randomNumber = _dereq_('random-number').generator({ min: 10000, max: 99999, integer: true }),
+	EventTarget = _dereq_('./EventTarget');
+
+debugerror.log = console.warn.bind(console);
+
+function RTCDTMFSender(peerConnection, track) {
+	var self = this;
+
+	// Make this an EventTarget.
+	EventTarget.call(this);
+
+	debug('new() | [track:%o]', track);
+
+	// Public atributes (accessed as read-only properties)
+	this._track = track;
+	// TODO: read these from the properties exposed in Swift?
+	this._duration = 100;
+	this._interToneGap = 70;
+	this._toneBuffer = '';
+
+	// Private attributes.
+	this.peerConnection = peerConnection;
+	this.dsId = randomNumber();
+
+	function onResultOK(data) {
+		onEvent.call(self, data);
+	}
+
+	exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_createDTMFSender', [
+		this.peerConnection.pcId,
+		this.dsId,
+		this._track.id
+	]);
+}
+
+RTCDTMFSender.prototype = Object.create(EventTarget.prototype);
+RTCDTMFSender.prototype.constructor = RTCDTMFSender;
+
+Object.defineProperty(RTCDTMFSender.prototype, 'canInsertDTMF', {
+	get: function () {
+		// TODO: check if it's muted or stopped?
+		return this._track && this._track.kind === 'audio' && this._track.enabled;
+	}
+});
+
+Object.defineProperty(RTCDTMFSender.prototype, 'track', {
+	get: function () {
+		return this._track;
+	}
+});
+
+Object.defineProperty(RTCDTMFSender.prototype, 'duration', {
+	get: function () {
+		return this._duration;
+	}
+});
+
+Object.defineProperty(RTCDTMFSender.prototype, 'interToneGap', {
+	get: function () {
+		return this._interToneGap;
+	}
+});
+
+Object.defineProperty(RTCDTMFSender.prototype, 'toneBuffer', {
+	get: function () {
+		return this._toneBuffer;
+	}
+});
+
+RTCDTMFSender.prototype.insertDTMF = function (tones, duration, interToneGap) {
+	if (isClosed.call(this)) {
+		return;
+	}
+
+	debug('insertDTMF() | [tones:%o, duration:%o, interToneGap:%o]', tones, duration, interToneGap);
+
+	if (!tones) {
+		return;
+	}
+
+	this._duration = duration || 100;
+	this._interToneGap = interToneGap || 70;
+
+	var self = this;
+
+	function onResultOK(data) {
+		onEvent.call(self, data);
+	}
+
+	exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDTMFSender_insertDTMF', [
+		this.peerConnection.pcId,
+		this.dsId,
+		tones,
+		this._duration,
+		this._interToneGap
+	]);
+};
+
+/**
+ * Private API.
+ */
+
+function isClosed() {
+	return this.peerConnection.signalingState === 'closed';
+}
+
+function onEvent(data) {
+	var type = data.type,
+		event;
+
+	debug('onEvent() | [type:%s, data:%o]', type, data);
+
+	if (type === 'tonechange') {
+		event = new Event('tonechange');
+		event.tone = data.tone;
+		this.dispatchEvent(event);
+	}
+}
+
+},{"./EventTarget":2,"cordova/exec":undefined,"debug":23,"random-number":28}],11:[function(_dereq_,module,exports){
+(function (setImmediate){
+/**
+ * Expose the RTCDataChannel class.
+ */
+module.exports = RTCDataChannel;
+
+/**
+ * Dependencies.
+ */
+var debug = _dereq_('debug')('iosrtc:RTCDataChannel'),
+	debugerror = _dereq_('debug')('iosrtc:ERROR:RTCDataChannel'),
+	exec = _dereq_('cordova/exec'),
+	randomNumber = _dereq_('random-number').generator({ min: 10000, max: 99999, integer: true }),
+	EventTarget = _dereq_('./EventTarget');
+
+debugerror.log = console.warn.bind(console);
+
+function RTCDataChannel(peerConnection, label, options, dataFromEvent) {
+	var self = this;
+
+	// Make this an EventTarget.
+	EventTarget.call(this);
+
+	// Created via pc.createDataChannel().
+	if (!dataFromEvent) {
+		debug('new() | [label:%o, options:%o]', label, options);
+
+		if (typeof label !== 'string') {
+			label = '';
+		}
+
+		options = options || {};
+
+		if (
+			options.hasOwnProperty('maxPacketLifeTime') &&
+			options.hasOwnProperty('maxRetransmits')
+		) {
+			throw new SyntaxError('both maxPacketLifeTime and maxRetransmits can not be present');
+		}
+
+		if (options.hasOwnProperty('id')) {
+			if (typeof options.id !== 'number' || isNaN(options.id) || options.id < 0) {
+				throw new SyntaxError('id must be a number');
+			}
+			// TODO:
+			//   https://code.google.com/p/webrtc/issues/detail?id=4618
+			if (options.id > 1023) {
+				throw new SyntaxError(
+					'id cannot be greater than 1023 (https://code.google.com/p/webrtc/issues/detail?id=4614)'
+				);
+			}
+		}
+
+		// Public atributes.
+		this.label = label;
+		this.ordered = options.hasOwnProperty('ordered') ? !!options.ordered : true;
+		this.maxPacketLifeTime = options.hasOwnProperty('maxPacketLifeTime')
+			? Number(options.maxPacketLifeTime)
+			: null;
+		this.maxRetransmits = options.hasOwnProperty('maxRetransmits')
+			? Number(options.maxRetransmits)
+			: null;
+		this.protocol = options.hasOwnProperty('protocol') ? String(options.protocol) : '';
+		this.negotiated = options.hasOwnProperty('negotiated') ? !!options.negotiated : false;
+		this.id = options.hasOwnProperty('id') ? Number(options.id) : undefined;
+		this.readyState = 'connecting';
+		this.bufferedAmount = 0;
+		this.bufferedAmountLowThreshold = 0;
+
+		// Private attributes.
+		this.peerConnection = peerConnection;
+		this.dcId = randomNumber();
+
+		exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_createDataChannel', [
+			this.peerConnection.pcId,
+			this.dcId,
+			label,
+			options
+		]);
+	} else {
+		// Created via pc.ondatachannel.
+		debug('new() | [dataFromEvent:%o]', dataFromEvent);
+
+		// Public atributes.
+		this.label = dataFromEvent.label;
+		this.ordered = dataFromEvent.ordered;
+		this.maxPacketLifeTime = dataFromEvent.maxPacketLifeTime;
+		this.maxRetransmits = dataFromEvent.maxRetransmits;
+		this.protocol = dataFromEvent.protocol;
+		this.negotiated = dataFromEvent.negotiated;
+		this.id = dataFromEvent.id;
+		this.readyState = dataFromEvent.readyState;
+		this.bufferedAmount = dataFromEvent.bufferedAmount;
+		this.bufferedAmountLowThreshold = dataFromEvent.bufferedAmountLowThreshold;
+
+		// Private attributes.
+		this.peerConnection = peerConnection;
+		this.dcId = dataFromEvent.dcId;
+
+		exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_setListener', [
+			this.peerConnection.pcId,
+			this.dcId
+		]);
+	}
+
+	function onResultOK(data) {
+		if (data.type) {
+			onEvent.call(self, data);
+		} else {
+			// Special handler for received binary message.
+			onEvent.call(self, {
+				type: 'message',
+				message: data
+			});
+		}
+	}
+}
+
+RTCDataChannel.prototype = Object.create(EventTarget.prototype);
+RTCDataChannel.prototype.constructor = RTCDataChannel;
+
+// Just 'arraybuffer' binaryType is implemented in Chromium.
+Object.defineProperty(RTCDataChannel.prototype, 'binaryType', {
+	get: function () {
+		return 'arraybuffer';
+	},
+	set: function (type) {
+		if (type !== 'arraybuffer') {
+			throw new Error('just "arraybuffer" is implemented for binaryType');
+		}
+	}
+});
+
+RTCDataChannel.prototype.send = function (data) {
+	if (isClosed.call(this) || this.readyState !== 'open') {
+		return;
+	}
+
+	debug('send() | [data:%o]', data);
+
+	if (!data) {
+		return;
+	}
+
+	setImmediate(() => {
+		if (typeof data === 'string' || data instanceof String) {
+			exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendString', [
+				this.peerConnection.pcId,
+				this.dcId,
+				data
+			]);
+		} else if (window.ArrayBuffer && data instanceof window.ArrayBuffer) {
+			exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendBinary', [
+				this.peerConnection.pcId,
+				this.dcId,
+				data
+			]);
+		} else if (
+			(window.Int8Array && data instanceof window.Int8Array) ||
+			(window.Uint8Array && data instanceof window.Uint8Array) ||
+			(window.Uint8ClampedArray && data instanceof window.Uint8ClampedArray) ||
+			(window.Int16Array && data instanceof window.Int16Array) ||
+			(window.Uint16Array && data instanceof window.Uint16Array) ||
+			(window.Int32Array && data instanceof window.Int32Array) ||
+			(window.Uint32Array && data instanceof window.Uint32Array) ||
+			(window.Float32Array && data instanceof window.Float32Array) ||
+			(window.Float64Array && data instanceof window.Float64Array) ||
+			(window.DataView && data instanceof window.DataView)
+		) {
+			exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_sendBinary', [
+				this.peerConnection.pcId,
+				this.dcId,
+				data.buffer
+			]);
+		} else {
+			throw new Error('invalid data type');
+		}
+	});
+};
+
+RTCDataChannel.prototype.close = function () {
+	if (isClosed.call(this)) {
+		return;
+	}
+
+	debug('close()');
+
+	this.readyState = 'closing';
+
+	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCDataChannel_close', [
+		this.peerConnection.pcId,
+		this.dcId
+	]);
+};
+
+/**
+ * Private API.
+ */
+
+function isClosed() {
+	return (
+		this.readyState === 'closed' ||
+		this.readyState === 'closing' ||
+		this.peerConnection.signalingState === 'closed'
+	);
+}
+
+function onEvent(data) {
+	var type = data.type,
+		event;
+
+	debug('onEvent() | [type:%s, data:%o]', type, data);
+
+	switch (type) {
+		case 'new':
+			// Update properties and exit without firing the event.
+			this.ordered = data.channel.ordered;
+			this.maxPacketLifeTime = data.channel.maxPacketLifeTime;
+			this.maxRetransmits = data.channel.maxRetransmits;
+			this.protocol = data.channel.protocol;
+			this.negotiated = data.channel.negotiated;
+			this.id = data.channel.id;
+			this.readyState = data.channel.readyState;
+			this.bufferedAmount = data.channel.bufferedAmount;
+			break;
+
+		case 'statechange':
+			this.readyState = data.readyState;
+
+			switch (data.readyState) {
+				case 'connecting':
+					break;
+				case 'open':
+					this.dispatchEvent(new Event('open'));
+					break;
+				case 'closing':
+					break;
+				case 'closed':
+					this.dispatchEvent(new Event('close'));
+					break;
+			}
+			break;
+
+		case 'message':
+			event = new Event('message');
+			event.data = data.message;
+			this.dispatchEvent(event);
+			break;
+
+		case 'bufferedamount':
+			this.bufferedAmount = data.bufferedAmount;
+
+			if (
+				this.bufferedAmountLowThreshold > 0 &&
+				this.bufferedAmountLowThreshold > this.bufferedAmount
+			) {
+				event = new Event('bufferedamountlow');
+				event.bufferedAmount = this.bufferedAmount;
+				this.dispatchEvent(event);
+			}
+
+			break;
+	}
+}
+
+}).call(this,_dereq_("timers").setImmediate)
+},{"./EventTarget":2,"cordova/exec":undefined,"debug":23,"random-number":28,"timers":29}],12:[function(_dereq_,module,exports){
+/**
+ * Expose the RTCIceCandidate class.
+ */
+module.exports = RTCIceCandidate;
+
+/**
+* RFC-5245: http://tools.ietf.org/html/rfc5245#section-15.1
+*
+* candidate-attribute   = "candidate" ":" foundation SP component-id SP
+                           transport SP
+                           priority SP
+                           connection-address SP     ;from RFC 4566
+                           port         ;port from RFC 4566
+                           SP cand-type
+                           [SP rel-addr]
+                           [SP rel-port]
+                           *(SP extension-att-name SP
+                                extension-att-value)
+*
+* foundation            = 1*32ice-char
+* component-id          = 1*5DIGIT
+* transport             = "UDP" / transport-extension
+* transport-extension   = token              ; from RFC 3261
+* priority              = 1*10DIGIT
+* cand-type             = "typ" SP candidate-types
+* candidate-types       = "host" / "srflx" / "prflx" / "relay" / token
+* rel-addr              = "raddr" SP connection-address
+* rel-port              = "rport" SP port
+* extension-att-name    = byte-string    ;from RFC 4566
+* extension-att-value   = byte-string
+* ice-char              = ALPHA / DIGIT / "+" / "/"
+*/
+
+/**
+* RFC-3261: https://tools.ietf.org/html/rfc3261#section-25.1
+*
+* token          =  1*(alphanum / "-" / "." / "!" / "%" / "*"
+                     / "_" / "+" / "`" / "'" / "~" )
+*/
+
+/*
+* RFC-4566: https://tools.ietf.org/html/rfc4566#section-9
+*
+* port =                1*DIGIT
+* IP4-address =         b1 3("." decimal-uchar)
+* b1 =                  decimal-uchar
+                         ; less than "224"
+* ; The following is consistent with RFC 2373 [30], Appendix B.
+* IP6-address =         hexpart [ ":" IP4-address ]
+* hexpart =             hexseq / hexseq "::" [ hexseq ] /
+                         "::" [ hexseq ]
+* hexseq  =             hex4 *( ":" hex4)
+* hex4    =             1*4HEXDIG
+* decimal-uchar =       DIGIT
+                         / POS-DIGIT DIGIT
+                         / ("1" 2*(DIGIT))
+                         / ("2" ("0"/"1"/"2"/"3"/"4") DIGIT)
+                         / ("2" "5" ("0"/"1"/"2"/"3"/"4"/"5"))
+*/
+
+var candidateToJson = (function () {
+	var candidateFieldName = {
+		FOUNDATION: 'foundation',
+		COMPONENT_ID: 'componentId',
+		TRANSPORT: 'transport',
+		PRIORITY: 'priority',
+		CONNECTION_ADDRESS: 'connectionAddress',
+		PORT: 'port',
+		CANDIDATE_TYPE: 'candidateType',
+		REMOTE_CANDIDATE_ADDRESS: 'remoteConnectionAddress',
+		REMOTE_CANDIDATE_PORT: 'remotePort'
+	};
+
+	var candidateType = {
+		HOST: 'host',
+		SRFLX: 'srflx',
+		PRFLX: 'prflx',
+		RELAY: 'relay'
+	};
+
+	var transport = {
+		TCP: 'TCP',
+		UDP: 'UDP'
+	};
+
+	var IPV4SEG = '(?:25[0-5]|(?:2[0-4]|1{0,1}[0-9]){0,1}[0-9])';
+	var IPV4ADDR = `(?:${IPV4SEG}\\.){3}${IPV4SEG}`;
+	var IPV6SEG = '[0-9a-fA-F]{1,4}';
+	var IPV6ADDR =
+		`(?:${IPV6SEG}:){7,7}${IPV6SEG}|` + // 1:2:3:4:5:6:7:8
+		`(?:${IPV6SEG}:){1,7}:|` + // 1::                              1:2:3:4:5:6:7::
+		`(?:${IPV6SEG}:){1,6}:${IPV6SEG}|` + // 1::8             1:2:3:4:5:6::8  1:2:3:4:5:6::8
+		`(?:${IPV6SEG}:){1,5}(?::${IPV6SEG}){1,2}|` + // 1::7:8           1:2:3:4:5::7:8  1:2:3:4:5::8
+		`(?:${IPV6SEG}:){1,4}(?::${IPV6SEG}){1,3}|` + // 1::6:7:8         1:2:3:4::6:7:8  1:2:3:4::8
+		`(?:${IPV6SEG}:){1,3}(?::${IPV6SEG}){1,4}|` + // 1::5:6:7:8       1:2:3::5:6:7:8  1:2:3::8
+		`(?:${IPV6SEG}:){1,2}(?::${IPV6SEG}){1,5}|` + // 1::4:5:6:7:8     1:2::4:5:6:7:8  1:2::8
+		`${IPV6SEG}:(?:(?::${IPV6SEG}){1,6})|` + // 1::3:4:5:6:7:8   1::3:4:5:6:7:8  1::8
+		`:(?:(?::${IPV6SEG}){1,7}|:)|` + // ::2:3:4:5:6:7:8  ::2:3:4:5:6:7:8 ::8       ::
+		`fe80:(?::${IPV6SEG}){0,4}%[0-9a-zA-Z]{1,}|` + // fe80::7:8%eth0   fe80::7:8%1     (link-local IPv6 addresses with zone index)
+		`::(?:ffff(?::0{1,4}){0,1}:){0,1}${IPV4ADDR}|` + // ::255.255.255.255   ::ffff:255.255.255.255  ::ffff:0:255.255.255.255 (IPv4-mapped IPv6 addresses and IPv4-translated addresses)
+		`(?:${IPV6SEG}:){1,4}:${IPV4ADDR}`; // 2001:db8:3:4::192.0.2.33  64:ff9b::192.0.2.33 (IPv4-Embedded IPv6 Address)
+
+	var TOKEN = "[0-9a-zA-Z\\-\\.!\\%\\*_\\+\\`\\'\\~]+";
+
+	var CANDIDATE_TYPE = '';
+	Object.keys(candidateType).forEach(function (key) {
+		CANDIDATE_TYPE += candidateType[key] + '|';
+	});
+	CANDIDATE_TYPE += TOKEN;
+
+	var pattern = {
+		COMPONENT_ID: '[0-9]{1,5}',
+		FOUNDATION: '[a-zA-Z0-9\\+\\/\\-]+',
+		PRIORITY: '[0-9]{1,10}',
+		TRANSPORT: transport.UDP + '|' + TOKEN,
+		CONNECTION_ADDRESS: IPV4ADDR + '|' + IPV6ADDR,
+		PORT: '[0-9]{1,5}',
+		CANDIDATE_TYPE: CANDIDATE_TYPE
+	};
+
+	return function candidateToJson(iceCandidate) {
+		var iceCandidateJson = null;
+
+		if (iceCandidate && typeof iceCandidate === 'string') {
+			var ICE_CANDIDATE_PATTERN = new RegExp(
+				`candidate:(${pattern.FOUNDATION})` + // 10
+					`\\s(${pattern.COMPONENT_ID})` + // 1
+					`\\s(${pattern.TRANSPORT})` + // UDP
+					`\\s(${pattern.PRIORITY})` + // 1845494271
+					`\\s(${pattern.CONNECTION_ADDRESS})` + // 13.93.107.159
+					`\\s(${pattern.PORT})` + // 53705
+					'\\s' +
+					'typ' +
+					`\\s(${pattern.CANDIDATE_TYPE})` + // typ prflx
+					'(?:\\s' +
+					'raddr' +
+					`\\s(${pattern.CONNECTION_ADDRESS})` + // raddr 10.1.221.7
+					'\\s' +
+					'rport' +
+					`\\s(${pattern.PORT}))?` // rport 54805
+			);
+
+			var iceCandidateFields = iceCandidate.match(ICE_CANDIDATE_PATTERN);
+			if (iceCandidateFields) {
+				iceCandidateJson = {};
+				Object.keys(candidateFieldName).forEach(function (key, i) {
+					// i+1 because match returns the entire match result
+					// and the parentheses-captured matched results.
+					if (iceCandidateFields.length > i + 1 && iceCandidateFields[i + 1]) {
+						iceCandidateJson[candidateFieldName[key]] = iceCandidateFields[i + 1];
+					}
+				});
+			}
+		}
+
+		return iceCandidateJson;
+	};
+})();
+
+// See https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidate/RTCIceCandidate
+function RTCIceCandidate(data) {
+	data = data || {};
+
+	// Public atributes.
+	this.sdpMid = data.sdpMid;
+	this.sdpMLineIndex = data.sdpMLineIndex;
+	this.candidate = data.candidate;
+
+	// Parse candidate SDP:
+	// Example: candidate:1829696681 1 udp 2122262783 2a01:cb05:8d3e:a300:e1ad:79c1:7096:8ba0 49778 typ host generation 0 ufrag c9L6 network-id 2 network-cost 10
+	var iceCandidateFields = candidateToJson(this.candidate);
+	if (iceCandidateFields) {
+		this.foundation = iceCandidateFields.foundation;
+		this.component = iceCandidateFields.componentId;
+		this.priority = iceCandidateFields.priority;
+		this.type = iceCandidateFields.candidateType;
+
+		this.address = iceCandidateFields.connectionAddress;
+		this.ip = iceCandidateFields.connectionAddress;
+		this.protocol = iceCandidateFields.transport;
+		this.port = iceCandidateFields.port;
+
+		this.relatedAddress = iceCandidateFields.remoteConnectionAddress || null;
+		this.relatedPort = iceCandidateFields.remotePort || null;
+	}
+}
+
+},{}],13:[function(_dereq_,module,exports){
+(function (global){
+/**
+ * Expose the RTCPeerConnection class.
+ */
+module.exports = RTCPeerConnection;
+
+/**
+ * Dependencies.
+ */
+var debug = _dereq_('debug')('iosrtc:RTCPeerConnection'),
+	debugerror = _dereq_('debug')('iosrtc:ERROR:RTCPeerConnection'),
+	exec = _dereq_('cordova/exec'),
+	randomNumber = _dereq_('random-number').generator({ min: 10000, max: 99999, integer: true }),
+	EventTarget = _dereq_('./EventTarget'),
+	RTCSessionDescription = _dereq_('./RTCSessionDescription'),
+	RTCIceCandidate = _dereq_('./RTCIceCandidate'),
+	RTCDataChannel = _dereq_('./RTCDataChannel'),
+	RTCDTMFSender = _dereq_('./RTCDTMFSender'),
+	RTCRtpReceiver = _dereq_('./RTCRtpReceiver'),
+	RTCRtpSender = _dereq_('./RTCRtpSender'),
+	{ RTCRtpTransceiver, addTransceiverToPeerConnection } = _dereq_('./RTCRtpTransceiver'),
+	RTCStatsReport = _dereq_('./RTCStatsReport'),
+	MediaStream = _dereq_('./MediaStream'),
+	{ MediaStreamTrack, newMediaStreamTrackId } = _dereq_('./MediaStreamTrack'),
+	Errors = _dereq_('./Errors');
+
+debugerror.log = console.warn.bind(console);
+
+function deprecateWarning(method, newMethod) {
+	if (!newMethod) {
+		console.warn(method + ' is deprecated.');
+	} else {
+		console.warn(method + ' method is deprecated, use ' + newMethod + ' instead.');
+	}
+}
+
+function RTCPeerConnection(pcConfig, pcConstraints) {
+	debug('new() | [pcConfig:%o, pcConstraints:%o]', pcConfig, pcConstraints);
+
+	var self = this;
+
+	// Make this an EventTarget.
+	EventTarget.call(this);
+
+	// Restore corrupted RTCPeerConnection.prototype
+	// TODO find why webrtc-adapter prevent events onnegotiationneeded to be trigger.
+	// Object.defineProperties(this, RTCPeerConnection.prototype_descriptor);
+
+	// Fix webrtc-adapter bad SHIM on addTrack causing error when original does support multiple streams.
+	// NotSupportedError: The adapter.js addTrack, addStream polyfill only supports a single stream which is associated with the specified track.
+	Object.defineProperty(this, 'addTrack', RTCPeerConnection.prototype_descriptor.addTrack);
+	Object.defineProperty(this, 'addStream', RTCPeerConnection.prototype_descriptor.addStream);
+	Object.defineProperty(
+		this,
+		'getLocalStreams',
+		RTCPeerConnection.prototype_descriptor.getLocalStreams
+	);
+	Object.defineProperty(
+		this,
+		'addTransceiver',
+		RTCPeerConnection.prototype_descriptor.addTransceiver
+	);
+	Object.defineProperty(
+		this,
+		'getOrCreateTrack',
+		RTCPeerConnection.prototype_descriptor.getOrCreateTrack
+	);
+
+	// Public atributes.
+	this._localDescription = null;
+	this.remoteDescription = null;
+	this.signalingState = 'stable';
+	this.iceGatheringState = 'new';
+	this.iceConnectionState = 'new';
+	this.pcConfig = fixPcConfig(pcConfig);
+
+	// Private attributes.
+	this.pcId = randomNumber();
+	this.localStreams = {};
+	this.remoteStreams = {};
+
+	this.tracks = {};
+	this.transceivers = [];
+
+	function onResultOK(data) {
+		onEvent.call(self, data);
+	}
+
+	exec(onResultOK, null, 'iosrtcPlugin', 'new_RTCPeerConnection', [
+		this.pcId,
+		this.pcConfig,
+		pcConstraints
+	]);
+}
+
+RTCPeerConnection.prototype = Object.create(EventTarget.prototype);
+RTCPeerConnection.prototype.constructor = RTCPeerConnection;
+
+Object.defineProperties(RTCPeerConnection.prototype, {
+	localDescription: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		get: function () {
+			return this._localDescription;
+		}
+	},
+	connectionState: {
+		get: function () {
+			return this.iceConnectionState;
+		}
+	},
+	onicecandidate: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		set: function (callback) {
+			return this.addEventListener('icecandidate', callback);
+		}
+	},
+	onaddstream: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		set: function (callback) {
+			return this.addEventListener('addstream', callback);
+		}
+	},
+	ontrack: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		set: function (callback) {
+			return this.addEventListener('track', callback);
+		}
+	},
+	oniceconnectionstatechange: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		set: function (callback) {
+			return this.addEventListener('iceconnectionstatechange', callback);
+		}
+	},
+	onnegotiationneeded: {
+		// Fix webrtc-adapter TypeError: Attempting to change the getter of an unconfigurable property.
+		configurable: true,
+		set: function (callback) {
+			return this.addEventListener('negotiationneeded', callback);
+		}
+	}
+});
+
+RTCPeerConnection.prototype.createOffer = function (options) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('RTCPeerConnection.prototype.createOffer', arguments);
+
+	var self = this;
+
+	if (isClosed.call(this)) {
+		return;
+	}
+
+	debug('createOffer() [options:%o]', options);
+
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			if (data.transceivers) {
+				self.updateTransceiversState(data.transceivers);
+			}
+
+			var desc = new RTCSessionDescription(data);
+
+			debug('createOffer() | success [desc:%o]', desc);
+			resolve(desc);
+		}
+
+		function onResultError(error) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			debugerror('createOffer() | failure: %s', error);
+			reject(new global.DOMException(error));
+		}
+
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createOffer', [
+			self.pcId,
+			options
+		]);
+	});
+};
+
+RTCPeerConnection.prototype.createAnswer = function (options) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('RTCPeerConnection.prototype.createAnswer', arguments);
+
+	var self = this;
+
+	if (isClosed.call(this)) {
+		return;
+	}
+
+	debug('createAnswer() [options:%o]', options);
+
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			if (data.transceivers) {
+				self.updateTransceiversState(data.transceivers);
+			}
+
+			var desc = new RTCSessionDescription(data);
+
+			debug('createAnswer() | success [desc:%o]', desc);
+			resolve(desc);
+		}
+
+		function onResultError(error) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			debugerror('createAnswer() | failure: %s', error);
+			reject(new global.DOMException(error));
+		}
+
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_createAnswer', [
+			self.pcId,
+			options
+		]);
+	});
+};
+
+RTCPeerConnection.prototype.setLocalDescription = function (desc) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage(
+		'RTCPeerConnection.prototype.setLocalDescription',
+		arguments
+	);
+
+	var self = this;
+
+	if (isClosed.call(this)) {
+		return new Promise(function (resolve, reject) {
+			reject(new Errors.InvalidStateError('peerconnection is closed'));
+		});
+	}
+
+	// "This is no longer necessary, however; RTCPeerConnection.setLocalDescription() and other
+	// methods which take SDP as input now directly accept an object conforming to the RTCSessionDescriptionInit dictionary,
+	// so you don't have to instantiate an RTCSessionDescription yourself.""
+	// Source: https://developer.mozilla.org/en-US/docs/Web/API/RTCSessionDescription/RTCSessionDescription#Example
+	// Still we do instnanciate RTCSessionDescription, so internal object is used properly.
+
+	if (!(desc instanceof RTCSessionDescription)) {
+		desc = new RTCSessionDescription(desc);
+	}
+
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			if (data.transceivers) {
+				self.updateTransceiversState(data.transceivers);
+			}
+
+			debug('setLocalDescription() | success');
+			// Update localDescription.
+			self._localDescription = data.type === '' ? null : new RTCSessionDescription(data);
+			resolve();
+		}
+
+		function onResultError(error) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			debugerror('setLocalDescription() | failure: %s', error);
+			reject(
+				new Errors.InvalidSessionDescriptionError('setLocalDescription() failed: ' + error)
+			);
+		}
+
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setLocalDescription', [
+			self.pcId,
+			desc
+		]);
+	});
+};
+
+RTCPeerConnection.prototype.setRemoteDescription = function (desc) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage(
+		'RTCPeerConnection.prototype.setRemoteDescription',
+		arguments
+	);
+
+	var self = this;
+
+	if (isClosed.call(this)) {
+		return new Promise(function (resolve, reject) {
+			reject(new Errors.InvalidStateError('peerconnection is closed'));
+		});
+	}
+
+	debug('setRemoteDescription() [desc:%o]', desc);
+
+	// Remove extmap-allow-mixed sdp header
+	if (desc && desc.sdp && desc.sdp.indexOf('\na=extmap-allow-mixed') !== -1) {
+		desc = new RTCSessionDescription({
+			type: desc.type,
+			sdp: desc.sdp
+				.split('\n')
+				.filter((line) => {
+					return line.trim() !== 'a=extmap-allow-mixed';
+				})
+				.join('\n')
+		});
+	}
+
+	// "This is no longer necessary, however; RTCPeerConnection.setLocalDescription() and other
+	// methods which take SDP as input now directly accept an object conforming to the RTCSessionDescriptionInit dictionary,
+	// so you don't have to instantiate an RTCSessionDescription yourself.""
+	// Source: https://developer.mozilla.org/en-US/docs/Web/API/RTCSessionDescription/RTCSessionDescription#Example
+	// Still we do instnanciate RTCSessionDescription so internal object is used properly.
+
+	if (!(desc instanceof RTCSessionDescription)) {
+		desc = new RTCSessionDescription(desc);
+	}
+
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			if (data.transceivers) {
+				self.updateTransceiversState(data.transceivers);
+			}
+
+			debug('setRemoteDescription() | success');
+			// Update remoteDescription.
+			self.remoteDescription = new RTCSessionDescription(data);
+			resolve();
+		}
+
+		function onResultError(error) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			debugerror('setRemoteDescription() | failure: %s', error);
+			reject(
+				new Errors.InvalidSessionDescriptionError('setRemoteDescription() failed: ' + error)
+			);
+		}
+
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_setRemoteDescription', [
+			self.pcId,
+			desc
+		]);
+	});
+};
+
+RTCPeerConnection.prototype.addIceCandidate = function (candidate) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('RTCPeerConnection.prototype.addIceCandidate', arguments);
+
+	var self = this;
+
+	if (isClosed.call(this)) {
+		return new Promise(function (resolve, reject) {
+			reject(new Errors.InvalidStateError('peerconnection is closed'));
+		});
+	}
+
+	debug('addIceCandidate() | [candidate:%o]', candidate);
+
+	if (typeof candidate !== 'object') {
+		return new Promise(function (resolve, reject) {
+			reject(
+				new global.DOMException(
+					'addIceCandidate() must be called with a RTCIceCandidate instance or RTCIceCandidateInit object as argument'
+				)
+			);
+		});
+	}
+
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			debug('addIceCandidate() | success');
+			// Update remoteDescription.
+			if (self.remoteDescription && data.remoteDescription) {
+				self.remoteDescription.type = data.remoteDescription.type;
+				self.remoteDescription.sdp = data.remoteDescription.sdp;
+			} else if (data.remoteDescription) {
+				self.remoteDescription = new RTCSessionDescription(data.remoteDescription);
+			}
+			resolve();
+		}
+
+		function onResultError() {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			debugerror('addIceCandidate() | failure');
+			reject(new global.DOMException('addIceCandidate() failed'));
+		}
+
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_addIceCandidate', [
+			self.pcId,
+			candidate
+		]);
+	});
+};
+
+RTCPeerConnection.prototype.getConfiguration = function () {
+	debug('getConfiguration()');
+
+	return this.pcConfig;
+};
+
+RTCPeerConnection.prototype.getLocalStreams = function () {
+	debug('getLocalStreams()');
+	deprecateWarning('getLocalStreams', 'getSenders');
+
+	var streams = [],
+		id;
+
+	for (id in this.localStreams) {
+		if (this.localStreams.hasOwnProperty(id)) {
+			streams.push(this.localStreams[id]);
+		}
+	}
+
+	return streams;
+};
+
+RTCPeerConnection.prototype.getRemoteStreams = function () {
+	debug('getRemoteStreams()');
+	deprecateWarning('getRemoteStreams', 'getReceivers');
+
+	var streams = [],
+		id;
+
+	for (id in this.remoteStreams) {
+		if (this.remoteStreams.hasOwnProperty(id)) {
+			streams.push(this.remoteStreams[id]);
+		}
+	}
+
+	return streams;
+};
+
+RTCPeerConnection.prototype.getReceivers = function () {
+	return this.getTransceivers()
+		.filter((transceiver) => !transceiver.stopped)
+		.map((transceiver) => transceiver.receiver);
+};
+
+RTCPeerConnection.prototype.getSenders = function () {
+	return this.getTransceivers()
+		.filter((transceiver) => !transceiver.stopped)
+		.map((transceiver) => transceiver.sender);
+};
+
+RTCPeerConnection.prototype.getTransceivers = function () {
+	return this.transceivers;
+};
+
+RTCPeerConnection.prototype.addTrack = function (track, ...streams) {
+	var stream = streams[0];
+
+	if (isClosed.call(this)) {
+		throw new Errors.InvalidStateError('peerconnection is closed');
+	}
+
+	// Add localStreams if missing
+	// Disable to match browser behavior
+	//stream = stream || Object.values(this.localStreams)[0] || new MediaStream();
+
+	// Fix webrtc-adapter bad SHIM on addStream
+	if (stream) {
+		this.addStream(stream);
+	}
+
+	var transceiver = new RTCRtpTransceiver(this, {
+		sender: new RTCRtpSender(this, {
+			track: track
+		}),
+		receiver: new RTCRtpReceiver(this),
+		direction: 'sendrecv',
+		currentDirection: null,
+		mid: null
+	});
+
+	for (var streamId in this.localStreams) {
+		if (this.localStreams.hasOwnProperty(streamId)) {
+			// Target provided stream argument or first added stream to group track
+			if (!stream || (stream && stream.id === streamId)) {
+				stream = this.localStreams[streamId];
+				stream.addTrack(track);
+				exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_addTrack', [
+					this.pcId,
+					track.id,
+					transceiver._id,
+					transceiver.receiver._id,
+					transceiver.sender._id,
+					streamId
+				]);
+				break;
+			}
+		}
+	}
+
+	// No Stream matched add track without stream
+	if (!stream) {
+		exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_addTrack', [
+			this.pcId,
+			track.id,
+			transceiver._id,
+			transceiver.receiver._id,
+			transceiver.sender._id,
+			null
+		]);
+	}
+
+	this.getOrCreateTrack(track);
+
+	this.transceivers.push(transceiver);
+
+	return transceiver.sender;
+};
+
+RTCPeerConnection.prototype.removeTrack = function (sender) {
+	var id, track, stream, hasTrack;
+
+	if (!(sender instanceof RTCRtpSender)) {
+		throw new Error('removeTrack() must be called with a RTCRtpSender instance as argument');
+	}
+
+	track = sender.track;
+
+	// No sender track found
+	if (!track) {
+		return;
+	}
+
+	function matchLocalTrack(localTrack) {
+		return localTrack.id === track.id;
+	}
+
+	for (id in this.localStreams) {
+		if (this.localStreams.hasOwnProperty(id)) {
+			// Check if track is belong to stream
+			hasTrack = this.localStreams[id].getTracks().filter(matchLocalTrack).length > 0;
+
+			if (hasTrack) {
+				stream = this.localStreams[id];
+				stream.removeTrack(track);
+				exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_removeTrack', [
+					this.pcId,
+					sender._id
+				]);
+				delete this.tracks[track.id];
+				break;
+			}
+		}
+	}
+
+	// No Stream matched remove track without stream
+	if (!stream) {
+		for (id in this.tracks) {
+			if (this.tracks.hasOwnProperty(id)) {
+				if (track.id === id) {
+					exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_removeTrack', [
+						this.pcId,
+						sender._id
+					]);
+					delete this.tracks[track.id];
+				}
+			}
+		}
+	}
+};
+
+RTCPeerConnection.prototype.getOrCreateTrack = function (trackInput) {
+	var { id } = trackInput,
+		existingTrack = this.tracks[id];
+
+	if (existingTrack) {
+		return existingTrack;
+	}
+
+	var track;
+	if (trackInput instanceof MediaStreamTrack) {
+		track = trackInput;
+	} else {
+		track = MediaStreamTrack.findOrCreate(trackInput);
+	}
+
+	this.tracks[id] = track;
+	track.addEventListener('ended', () => {
+		delete this.tracks[id];
+	});
+
+	return track;
+};
+
+RTCPeerConnection.prototype.addTransceiver = function (trackOrKind, init) {
+	if (isClosed.call(this)) {
+		throw new Errors.InvalidStateError('peerconnection is closed');
+	}
+
+	var kind,
+		track = null,
+		self = this,
+		trackIdOrKind;
+	if (trackOrKind instanceof MediaStreamTrack) {
+		kind = trackOrKind.kind;
+		track = trackOrKind;
+		trackIdOrKind = trackOrKind.id;
+	} else {
+		if (!(trackOrKind === 'audio' || trackOrKind === 'video')) {
+			throw new TypeError(
+				'An invalid string was specified as trackOrKind. The string must be either "audio" or "video".'
+			);
+		}
+		kind = trackOrKind;
+		trackIdOrKind = trackOrKind;
+	}
+
+	var receiverTrackID = newMediaStreamTrackId();
+
+	var receiver = new RTCRtpReceiver(this, {
+		track: new MediaStreamTrack({
+			id: receiverTrackID,
+			kind,
+			enabled: true,
+			readyState: 'live',
+			trackId: 'unknown'
+		})
+	});
+
+	var sender = new RTCRtpSender(this, {
+		track
+	});
+
+	var data = init || {};
+	data.sender = sender;
+	data.receiver = receiver;
+
+	var transceiver = new RTCRtpTransceiver(this, data);
+
+	this.transceivers.push(transceiver);
+
+	var initJson = {};
+	if (init && init.direction) {
+		initJson.direction = init.direction;
+	} else {
+		initJson.direction = 'sendrecv';
+	}
+
+	if (init && init.sendEncodings) {
+		initJson.sendEncodings = init.sendEncodings;
+	}
+
+	if (init && init.streams) {
+		initJson.streams = init.streams.map((stream) => stream.id);
+	} else {
+		initJson.streams = [];
+	}
+
+	debug(
+		'addTransceiver() [trackIdOrKind:%s, init:%o, initJson: %o]',
+		trackIdOrKind,
+		init,
+		initJson
+	);
+
+	addTransceiverToPeerConnection(this, trackIdOrKind, initJson, transceiver)
+		.then((update) => {
+			self.updateTransceiversState(update.transceivers);
+		})
+		.catch((error) => {
+			debugerror('addTransceiver() | failure: %s', error);
+		});
+
+	return transceiver;
+};
+
+RTCPeerConnection.prototype.updateTransceiversState = function (transceivers) {
+	debug('updateTransceiversState()');
+	this.transceivers = transceivers.map((transceiver) => {
+		const existingTransceiver = this.transceivers.find(
+			(localTransceiver) => localTransceiver._id === transceiver.id
+		);
+
+		if (existingTransceiver) {
+			existingTransceiver.update(transceiver);
+			return existingTransceiver;
+		}
+
+		return new RTCRtpTransceiver(this, transceiver);
+	});
+};
+
+RTCPeerConnection.prototype.getStreamById = function (id) {
+	debug('getStreamById()');
+
+	return this.localStreams[id] || this.remoteStreams[id] || null;
+};
+
+RTCPeerConnection.prototype.addStream = function (stream) {
+	var self = this;
+
+	if (isClosed.call(this)) {
+		throw new Errors.InvalidStateError('peerconnection is closed');
+	}
+
+	debug('addStream()');
+
+	if (!(stream instanceof MediaStream.originalMediaStream)) {
+		throw new Error('addStream() must be called with a MediaStream instance as argument');
+	}
+
+	if (this.localStreams[stream.id]) {
+		debugerror('addStream() | given stream already in present in local streams');
+		return;
+	}
+
+	this.localStreams[stream.id] = stream;
+
+	stream.addedToConnection = true;
+
+	stream.getTracks().forEach(function (track) {
+		self.getOrCreateTrack(track);
+	});
+
+	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_addStream', [this.pcId, stream.id]);
+};
+
+RTCPeerConnection.prototype.removeStream = function (stream) {
+	var self = this;
+
+	if (isClosed.call(this)) {
+		throw new Errors.InvalidStateError('peerconnection is closed');
+	}
+
+	debug('removeStream()');
+
+	if (!(stream instanceof MediaStream.originalMediaStream)) {
+		throw new Error('removeStream() must be called with a MediaStream instance as argument');
+	}
+
+	if (!this.localStreams[stream.id]) {
+		debugerror('removeStream() | given stream not present in local streams');
+		return;
+	}
+
+	delete this.localStreams[stream.id];
+
+	stream.getTracks().forEach(function (track) {
+		delete self.tracks[track.id];
+	});
+
+	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_removeStream', [this.pcId, stream.id]);
+};
+
+RTCPeerConnection.prototype.createDataChannel = function (label, options) {
+	if (isClosed.call(this)) {
+		throw new Errors.InvalidStateError('peerconnection is closed');
+	}
+
+	debug('createDataChannel() [label:%s, options:%o]', label, options);
+
+	return new RTCDataChannel(this, label, options);
+};
+
+RTCPeerConnection.prototype.createDTMFSender = function (track) {
+	if (isClosed.call(this)) {
+		throw new Errors.InvalidStateError('peerconnection is closed');
+	}
+
+	debug('createDTMFSender() [track:%o]', track);
+
+	return new RTCDTMFSender(this, track);
+};
+
+RTCPeerConnection.prototype.getStats = function (selector) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('RTCPeerConnection.prototype.getStats', arguments);
+
+	var self = this;
+
+	if (selector && !(selector instanceof MediaStreamTrack)) {
+		throw new Error(
+			'getStats() must be called with null or a valid MediaStreamTrack instance as argument'
+		);
+	}
+
+	if (isClosed.call(this)) {
+		throw new Errors.InvalidStateError('peerconnection is closed');
+	}
+
+	// debug('getStats() [selector:%o]', selector);
+
+	return new Promise(function (resolve, reject) {
+		function onResultOK(array) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			resolve(new RTCStatsReport(array));
+		}
+
+		function onResultError(error) {
+			if (isClosed.call(self)) {
+				return;
+			}
+
+			debugerror('getStats() | failure: %s', error);
+			reject(new global.DOMException(error));
+		}
+
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'RTCPeerConnection_getStats', [
+			self.pcId,
+			selector ? selector.id : null
+		]);
+	});
+};
+
+RTCPeerConnection.prototype.close = function () {
+	if (isClosed.call(this)) {
+		return;
+	}
+
+	debug('close()');
+
+	exec(null, null, 'iosrtcPlugin', 'RTCPeerConnection_close', [this.pcId]);
+};
+
+// Save current RTCPeerConnection.prototype
+RTCPeerConnection.prototype_descriptor = Object.getOwnPropertyDescriptors(
+	RTCPeerConnection.prototype
+);
+
+/**
+ * Private API.
+ */
+function fixPcConfig(pcConfig) {
+	if (!pcConfig) {
+		return {
+			iceServers: []
+		};
+	}
+
+	var iceServers = pcConfig.iceServers,
+		i,
+		len,
+		iceServer;
+
+	if (!Array.isArray(iceServers)) {
+		pcConfig.iceServers = [];
+		return pcConfig;
+	}
+
+	for (i = 0, len = iceServers.length; i < len; i++) {
+		iceServer = iceServers[i];
+
+		// THe Objective-C wrapper of WebRTC is old and does not implement .urls.
+		if (iceServer.url) {
+			continue;
+		} else if (Array.isArray(iceServer.urls)) {
+			iceServer.url = iceServer.urls[0];
+		} else if (typeof iceServer.urls === 'string') {
+			iceServer.url = iceServer.urls;
+		}
+	}
+
+	return pcConfig;
+}
+
+function isClosed() {
+	return this.signalingState === 'closed';
+}
+
+function onEvent(data) {
+	var type = data.type,
+		self = this,
+		event = new Event(type),
+		dataChannel,
+		transceiver,
+		id;
+
+	Object.defineProperty(event, 'target', { value: self, enumerable: true });
+
+	debug('onEvent() | [type:%s, data:%o]', type, data);
+
+	if (data.transceivers) {
+		this.updateTransceiversState(data.transceivers);
+	}
+
+	switch (type) {
+		case 'signalingstatechange':
+			this.signalingState = data.signalingState;
+			break;
+
+		case 'icegatheringstatechange':
+			this.iceGatheringState = data.iceGatheringState;
+			break;
+
+		case 'iceconnectionstatechange':
+			this.iceConnectionState = data.iceConnectionState;
+
+			// Emit "connected" on remote streams if ICE connected.
+			if (data.iceConnectionState === 'connected') {
+				for (id in this.remoteStreams) {
+					if (this.remoteStreams.hasOwnProperty(id)) {
+						this.remoteStreams[id].emitConnected();
+					}
+				}
+			}
+			break;
+
+		case 'icecandidate':
+			if (data.candidate) {
+				event.candidate = new RTCIceCandidate(data.candidate);
+			} else {
+				event.candidate = null;
+			}
+			// Update _localDescription.
+			if (this._localDescription && data.localDescription) {
+				this._localDescription.type = data.localDescription.type;
+				this._localDescription.sdp = data.localDescription.sdp;
+			} else if (data.localDescription) {
+				this._localDescription = new RTCSessionDescription(data.localDescription);
+			}
+			break;
+
+		case 'negotiationneeded':
+			break;
+
+		case 'track':
+			var track = (event.track = new MediaStreamTrack(data.track));
+			event.receiver = new RTCRtpReceiver(self, { track: track });
+
+			transceiver = this.transceivers.find((t) => t.receiver.track.id === track.id);
+			event.transceiver = transceiver;
+			event.streams = [];
+
+			// Add stream only if available in case of Unified-Plan of track event without stream
+			if (data.stream && data.streamId) {
+				var stream = this.remoteStreams[data.streamId] || MediaStream.create(data.stream);
+				event.streams.push(stream);
+			}
+
+			// Store remote track
+			this.getOrCreateTrack(track);
+
+			break;
+
+		case 'addstream':
+			// Append to the remote streams.
+			this.remoteStreams[data.streamId] =
+				this.remoteStreams[data.streamId] || MediaStream.create(data.stream);
+
+			event.stream = this.remoteStreams[data.streamId];
+
+			// Emit "connected" on the stream if ICE connected.
+			if (
+				this.iceConnectionState === 'connected' ||
+				this.iceConnectionState === 'completed'
+			) {
+				event.stream.emitConnected();
+			}
+			break;
+
+		case 'removestream':
+			event.stream = this.remoteStreams[data.streamId];
+
+			// Remove from the remote streams.
+			delete this.remoteStreams[data.streamId];
+			break;
+
+		case 'datachannel':
+			dataChannel = new RTCDataChannel(this, null, null, data.channel);
+			event.channel = dataChannel;
+			break;
+	}
+
+	this.dispatchEvent(event);
+}
+
+}).call(this,typeof global !== "undefined" ? global : typeof self !== "undefined" ? self : typeof window !== "undefined" ? window : {})
+},{"./Errors":1,"./EventTarget":2,"./MediaStream":5,"./MediaStreamTrack":7,"./RTCDTMFSender":10,"./RTCDataChannel":11,"./RTCIceCandidate":12,"./RTCRtpReceiver":14,"./RTCRtpSender":15,"./RTCRtpTransceiver":16,"./RTCSessionDescription":17,"./RTCStatsReport":18,"cordova/exec":undefined,"debug":23,"random-number":28}],14:[function(_dereq_,module,exports){
+/**
+ * Expose the RTCRtpReceiver class.
+ */
+module.exports = RTCRtpReceiver;
+
+var randomNumber = _dereq_('random-number').generator({ min: 10000, max: 99999, integer: true });
+
+function RTCRtpReceiver(pc, data) {
+	data = data || {};
+	this._id = data.id || randomNumber();
+
+	this._pc = pc;
+	this.track = data.track ? pc.getOrCreateTrack(data.track) : null;
+	this.params = data.params || {};
+}
+
+RTCRtpReceiver.prototype.getParameters = function () {
+	return this.params;
+};
+
+RTCRtpReceiver.prototype.getStats = function () {
+	return this._pc.getStats();
+};
+
+RTCRtpReceiver.prototype.update = function ({ track, params }) {
+	if (track) {
+		this.track = this._pc.getOrCreateTrack(track);
+	} else {
+		this.track = null;
+	}
+
+	this.params = params;
+};
+
+},{"random-number":28}],15:[function(_dereq_,module,exports){
+/**
+ * Expose the RTCRtpSender class.
+ */
+module.exports = RTCRtpSender;
+
+/**
+ * Dependencies.
+ */
+var exec = _dereq_('cordova/exec'),
+	{ MediaStreamTrack } = _dereq_('./MediaStreamTrack'),
+	randomNumber = _dereq_('random-number').generator({ min: 10000, max: 99999, integer: true });
+
+function RTCRtpSender(pc, data) {
+	data = data || {};
+	this._id = data.id || randomNumber();
+
+	this._pc = pc;
+	this.track = data.track ? pc.getOrCreateTrack(data.track) : null;
+	this.params = data.params || {};
+	if (this.track) {
+		this.dtmf = pc.createDTMFSender(this.track);
+	}
+}
+
+RTCRtpSender.prototype.getParameters = function () {
+	return this.params;
+};
+
+RTCRtpSender.prototype.getStats = function () {
+	return this._pc.getStats();
+};
+
+RTCRtpSender.prototype.setParameters = function (params) {
+	Object.assign(this.params, params);
+	return new Promise((resolve, reject) => {
+		function onResultOK(result) {
+			resolve(result);
+		}
+
+		function onResultError(error) {
+			reject(error);
+		}
+
+		exec(
+			onResultOK,
+			onResultError,
+			'iosrtcPlugin',
+			'RTCPeerConnection_RTCRtpSender_setParameters',
+			[this._pc.pcId, this._id, params]
+		);
+	});
+};
+
+RTCRtpSender.prototype.replaceTrack = function (withTrack) {
+	var self = this,
+		pc = self._pc;
+
+	return new Promise((resolve, reject) => {
+		function onResultOK(result) {
+			self.track = result.track ? MediaStreamTrack.findOrCreate(result.track) : null;
+			resolve();
+		}
+
+		function onResultError(error) {
+			reject(error);
+		}
+
+		var trackId = withTrack ? withTrack.id : null;
+
+		exec(
+			onResultOK,
+			onResultError,
+			'iosrtcPlugin',
+			'RTCPeerConnection_RTCRtpSender_replaceTrack',
+			[this._pc.pcId, this._id, trackId]
+		);
+
+		// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/negotiationneeded_event
+		var event = new Event('negotiationneeded');
+		pc.dispatchEvent(event);
+
+		pc.addEventListener('signalingstatechange', function listener() {
+			if (pc.signalingState === 'closed') {
+				pc.removeEventListener('signalingstatechange', listener);
+				reject();
+			} else if (pc.signalingState === 'stable') {
+				pc.removeEventListener('signalingstatechange', listener);
+				resolve();
+			}
+		});
+	});
+};
+
+RTCRtpSender.prototype.update = function ({ track, params }) {
+	if (track) {
+		this.track = this._pc.getOrCreateTrack(track);
+	} else {
+		this.track = null;
+	}
+
+	this.params = params;
+};
+
+},{"./MediaStreamTrack":7,"cordova/exec":undefined,"random-number":28}],16:[function(_dereq_,module,exports){
+const RTCRtpSender = _dereq_('./RTCRtpSender');
+const RTCRtpReceiver = _dereq_('./RTCRtpReceiver');
+
+/**
+ * Expose the RTCRtpTransceiver class.
+ */
+module.exports = { RTCRtpTransceiver, addTransceiverToPeerConnection };
+
+/**
+ * Dependencies.
+ */
+var debugerror = _dereq_('debug')('iosrtc:ERROR:RTCRtpTransceiver'),
+	exec = _dereq_('cordova/exec'),
+	randomNumber = _dereq_('random-number').generator({ min: 10000, max: 99999, integer: true }),
+	EventTarget = _dereq_('./EventTarget');
+
+debugerror.log = console.warn.bind(console);
+
+function addTransceiverToPeerConnection(peerConnection, trackIdOrKind, init, transceiver) {
+	return new Promise((resolve, reject) => {
+		exec(onResultOK, reject, 'iosrtcPlugin', 'RTCPeerConnection_addTransceiver', [
+			peerConnection.pcId,
+			trackIdOrKind,
+			init,
+			transceiver._id,
+			transceiver.sender ? transceiver.sender._id : 0,
+			transceiver.receiver ? transceiver.receiver._id : 0,
+			transceiver.receiver && transceiver.receiver.track
+				? transceiver.receiver.track.id
+				: null
+		]);
+
+		function onResultOK(data) {
+			resolve(data);
+		}
+	});
+}
+
+function RTCRtpTransceiver(peerConnection, data) {
+	data = data || {};
+
+	this.peerConnection = peerConnection;
+
+	this._id = data.id || randomNumber();
+	this._receiver =
+		data.receiver instanceof RTCRtpReceiver
+			? data.receiver
+			: new RTCRtpReceiver(peerConnection, data.receiver);
+	this._sender =
+		data.sender instanceof RTCRtpSender
+			? data.sender
+			: new RTCRtpSender(peerConnection, data.sender);
+	this._stopped = false;
+
+	if (data.mid) {
+		this._mid = data.mid;
+	} else {
+		this._mid = null;
+	}
+
+	if (data.direction) {
+		this._direction = data.direction;
+		this._currentDirection = data.direction;
+	} else {
+		this._direction = 'sendrecv';
+		this._currentDirection = 'sendrecv';
+	}
+}
+
+RTCRtpTransceiver.prototype = Object.create(EventTarget.prototype);
+RTCRtpTransceiver.prototype.constructor = RTCRtpTransceiver;
+
+Object.defineProperties(RTCRtpTransceiver.prototype, {
+	currentDirection: {
+		get: function () {
+			return this._currentDirection;
+		}
+	},
+	direction: {
+		get: function () {
+			return this._direction;
+		},
+		set: function (direction) {
+			var self = this;
+			this._direction = direction;
+
+			exec(
+				onResultOK,
+				null,
+				'iosrtcPlugin',
+				'RTCPeerConnection_RTCRtpTransceiver_setDirection',
+				[this.peerConnection.pcId, this._id, direction]
+			);
+
+			function onResultOK(data) {
+				self.peerConnection.updateTransceiversState(data.transceivers);
+			}
+		}
+	},
+	mid: {
+		get: function () {
+			return this._mid;
+		}
+	},
+	receiver: {
+		get: function () {
+			return this._receiver;
+		}
+	},
+	sender: {
+		get: function () {
+			return this._sender;
+		}
+	},
+	stopped: {
+		get: function () {
+			return this._stopped;
+		}
+	},
+	receiverId: {
+		get: function () {
+			return this._receiver.track.id;
+		}
+	}
+});
+
+RTCRtpTransceiver.prototype.stop = function () {
+	var self = this;
+
+	exec(onResultOK, null, 'iosrtcPlugin', 'RTCPeerConnection_RTCRtpTransceiver_stop', [
+		this.peerConnection.pcId,
+		this._id
+	]);
+
+	function onResultOK(data) {
+		self.peerConnection.updateTransceiversState(data.transceivers);
+	}
+};
+
+RTCRtpTransceiver.prototype.update = function (data) {
+	if (data.direction) {
+		this._direction = data.direction;
+	}
+	if (data.currentDirection) {
+		this._currentDirection = data.currentDirection;
+	}
+	if (data.mid) {
+		this._mid = data.mid;
+	}
+
+	this._stopped = data.stopped;
+
+	this.receiver.update(data.receiver);
+	this.sender.update(data.sender);
+};
+
+},{"./EventTarget":2,"./RTCRtpReceiver":14,"./RTCRtpSender":15,"cordova/exec":undefined,"debug":23,"random-number":28}],17:[function(_dereq_,module,exports){
+/**
+ * Expose the RTCSessionDescription class.
+ */
+module.exports = RTCSessionDescription;
+
+function RTCSessionDescription(data) {
+	data = data || {};
+
+	// Public atributes.
+	this.type = data.type;
+	this.sdp = data.sdp;
+}
+
+},{}],18:[function(_dereq_,module,exports){
+class RTCStatsReport {
+	constructor(data) {
+		this.data = new Map((data || []).map((el) => [el.reportId, el]));
+	}
+
+	get size() {
+		return this.data.size;
+	}
+
+	has(key) {
+		return this.data.has(key);
+	}
+	get(key) {
+		return this.data.get(key);
+	}
+	forEach(callbackfn, thisArg) {
+		return this.data.forEach(callbackfn, thisArg);
+	}
+
+	keys() {
+		return this.data.keys();
+	}
+	values() {
+		return this.data.values();
+	}
+	entries() {
+		return this.data.entries();
+	}
+	*[Symbol.iterator]() {
+		for (const value of this.data) {
+			yield value;
+		}
+	}
+}
+
+/**
+ * Expose the RTCStatsReport class.
+ */
+module.exports = RTCStatsReport;
+
+},{}],19:[function(_dereq_,module,exports){
+/**
+ * Expose the enumerateDevices function.
+ */
+module.exports = enumerateDevices;
+
+/**
+ * Dependencies.
+ */
+var debug = _dereq_('debug')('iosrtc:enumerateDevices'),
+	exec = _dereq_('cordova/exec'),
+	MediaDeviceInfo = _dereq_('./MediaDeviceInfo'),
+	Errors = _dereq_('./Errors');
+
+function enumerateDevices() {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('cordova.plugins.iosrtc.enumerateDevices', arguments);
+
+	return new Promise(function (resolve) {
+		function onResultOK(data) {
+			debug('enumerateDevices() | success');
+			resolve(getMediaDeviceInfos(data.devices));
+		}
+
+		exec(onResultOK, null, 'iosrtcPlugin', 'enumerateDevices', []);
+	});
+}
+
+/**
+ * Private API.
+ */
+
+function getMediaDeviceInfos(devices) {
+	debug('getMediaDeviceInfos() | [devices:%o]', devices);
+
+	var id,
+		mediaDeviceInfos = [];
+
+	for (id in devices) {
+		if (devices.hasOwnProperty(id)) {
+			mediaDeviceInfos.push(new MediaDeviceInfo(devices[id]));
+		}
+	}
+
+	return mediaDeviceInfos;
+}
+
+},{"./Errors":1,"./MediaDeviceInfo":3,"cordova/exec":undefined,"debug":23}],20:[function(_dereq_,module,exports){
+/**
+ * Expose the getUserMedia function.
+ */
+module.exports = getUserMedia;
+
+/**
+ * Dependencies.
+ */
+var debug = _dereq_('debug')('iosrtc:getUserMedia'),
+	debugerror = _dereq_('debug')('iosrtc:ERROR:getUserMedia'),
+	exec = _dereq_('cordova/exec'),
+	MediaStream = _dereq_('./MediaStream'),
+	Errors = _dereq_('./Errors');
+
+function isPositiveInteger(number) {
+	return typeof number === 'number' && number >= 0 && number % 1 === 0;
+}
+
+function isPositiveFloat(number) {
+	return typeof number === 'number' && number >= 0;
+}
+
+function getUserMedia(constraints) {
+	// Detect callback usage to assist 5.0.1 to 5.0.2 migration
+	// TODO remove on 6.0.0
+	Errors.detectDeprecatedCallbaksUsage('cordova.plugins.iosrtc.getUserMedia', arguments);
+
+	debug('[original constraints:%o]', constraints);
+
+	var audioRequested = false,
+		videoRequested = false,
+		newConstraints = {};
+
+	if (
+		typeof constraints !== 'object' ||
+		(!constraints.hasOwnProperty('audio') && !constraints.hasOwnProperty('video'))
+	) {
+		return new Promise(function (resolve, reject) {
+			reject(
+				new Errors.MediaStreamError(
+					'constraints must be an object with at least "audio" or "video" keys'
+				)
+			);
+		});
+	}
+
+	if (constraints.audio) {
+		audioRequested = true;
+	}
+
+	if (constraints.video) {
+		videoRequested = true;
+	}
+
+	// https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints
+	// Example:
+	//
+	// getUserMedia({
+	//  audio: {
+	//      deviceId: 'azer-asdf-zxcv',
+	//  },
+	//  video: {
+	//      deviceId: 'qwer-asdf-zxcv',
+	//      aspectRatio: 1.777.
+	//      facingMode: 'user',
+	//      width: {
+	//          min: 400,
+	//          max: 600
+	//      },
+	//      frameRate: {
+	//          min: 1.0,
+	//          max: 60.0
+	//      }
+	//  }
+	// });
+
+	/*
+	// See: https://www.w3.org/TR/mediacapture-streams/#media-track-constraints
+	dictionary MediaTrackConstraintSet {
+	 ConstrainULong     width;
+	 ConstrainULong     height;
+	 ConstrainDouble    aspectRatio;
+	 ConstrainDouble    frameRate;
+	 ConstrainDOMString facingMode;
+	 ConstrainDOMString resizeMode;
+	 ConstrainULong     sampleRate;
+	 ConstrainULong     sampleSize;
+	 ConstrainBoolean   echoCancellation;
+	 ConstrainBoolean   autoGainControl;
+	 ConstrainBoolean   noiseSuppression;
+	 ConstrainDouble    latency;
+	 ConstrainULong     channelCount;
+	 ConstrainDOMString deviceId;
+	 ConstrainDOMString groupId;
+	};
+
+	 // typedef ([Clamp] unsigned long or ConstrainULongRange) ConstrainULong;
+	 // We convert unsigned long to ConstrainULongRange.exact
+
+	 dictionary ULongRange {
+		[Clamp] unsigned long max;
+		[Clamp] unsigned long min;
+	 };
+
+	 dictionary ConstrainULongRange : ULongRange {
+		  [Clamp] unsigned long exact;
+		  [Clamp] unsigned long ideal;
+	 };
+
+	 // See: https://www.w3.org/TR/mediacapture-streams/#dom-doublerange
+	 // typedef (double or ConstrainDoubleRange) ConstrainDouble;
+	 // We convert double to ConstrainDoubleRange.exact
+	 dictionary ConstrainDouble {
+		double max;
+		double min;
+	 };
+
+	 dictionary ConstrainDoubleRange : DoubleRange {
+		double exact;
+		double ideal;
+	 };
+
+	 // typedef (boolean or ConstrainBooleanParameters) ConstrainBoolean;
+	 dictionary ConstrainBooleanParameters {
+		boolean exact;
+		boolean ideal;
+	 };
+
+	 // typedef (DOMString or sequence<DOMString> or ConstrainDOMStringParameters) ConstrainDOMString;
+	 // We convert DOMString to ConstrainDOMStringParameters.exact
+	 dictionary ConstrainDOMStringParameters {
+		(DOMString or sequence<DOMString>) exact;
+		(DOMString or sequence<DOMString>) ideal;
+	 };
+	*/
+
+	// Get video constraints
+	if (videoRequested) {
+		// Handle object video constraints
+		newConstraints.video = {};
+
+		// Handle Stupid not up-to-date webrtc-adapter
+		// Note: Firefox [38+] does support a subset of constraints with getUserMedia(), but not the outdated syntax that Chrome and Opera are using.
+		// The mandatory / optional syntax was deprecated a in 2014, and minWidth and minHeight the year before that.
+		if (
+			typeof constraints.video === 'object' &&
+			(typeof constraints.video.optional === 'object' ||
+				typeof constraints.video.mandatory === 'object')
+		) {
+			var videoConstraints = {};
+
+			if (Array.isArray(constraints.video.optional)) {
+				/*
+				Example of constraints.video.optional:
+					{
+						"optional": [
+							{
+								"minWidth": 640
+							},
+							{
+								"maxWidth": 640
+							},
+							{
+								"minHeight": 480
+							},
+							{
+								"maxHeight": 480
+							},
+							{
+								"sourceId": "com.apple.avfoundation.avcapturedevice.built-in_video:0"
+							}
+						]
+					}
+				*/
+
+				// Convert optional array to object
+				Object.values(constraints.video.optional).forEach(function (optional) {
+					var optionalConstraintName = Object.keys(optional)[0],
+						optionalConstraintValue = optional[optionalConstraintName];
+					videoConstraints[optionalConstraintName] = Array.isArray(
+						optionalConstraintValue
+					)
+						? optionalConstraintValue[0]
+						: optionalConstraintValue;
+				});
+			} else if (typeof constraints.video.mandatory === 'object') {
+				videoConstraints = constraints.video.mandatory;
+			}
+
+			if (typeof videoConstraints.sourceId === 'string') {
+				newConstraints.video.deviceId = {
+					ideal: videoConstraints.sourceId
+				};
+			}
+
+			if (isPositiveInteger(videoConstraints.minWidth)) {
+				newConstraints.video.width = {
+					min: videoConstraints.minWidth
+				};
+			}
+
+			if (isPositiveInteger(videoConstraints.maxWidth)) {
+				newConstraints.video.width = newConstraints.video.width || {};
+				newConstraints.video.width.max = videoConstraints.maxWidth;
+			}
+
+			if (isPositiveInteger(videoConstraints.minHeight)) {
+				newConstraints.video.height = {
+					min: videoConstraints.minHeight
+				};
+			}
+
+			if (isPositiveInteger(videoConstraints.maxHeight)) {
+				newConstraints.video.height = newConstraints.video.height || {};
+				newConstraints.video.height.max = videoConstraints.maxHeight;
+			}
+
+			if (isPositiveFloat(videoConstraints.minFrameRate)) {
+				newConstraints.video.frameRate = {
+					min: parseFloat(videoConstraints.minFrameRate, 10)
+				};
+			}
+
+			if (isPositiveFloat(videoConstraints.maxFrameRate)) {
+				newConstraints.video.frameRate = newConstraints.video.frameRate || {};
+				newConstraints.video.frameRate.max = parseFloat(videoConstraints.maxFrameRate, 10);
+			}
+		}
+
+		// Get requested video deviceId.
+		if (typeof constraints.video.deviceId === 'string') {
+			newConstraints.video.deviceId = {
+				exact: constraints.video.deviceId
+			};
+		} else if (typeof constraints.video.sourceId === 'string') {
+			// Also check video sourceId (mangled by adapter.js).
+			newConstraints.video.deviceId = {
+				exact: constraints.video.sourceId
+			};
+		} else if (typeof constraints.video.deviceId === 'object') {
+			// Also check deviceId.(exact|ideal)
+			if (!!constraints.video.deviceId.exact) {
+				newConstraints.video.deviceId = {
+					exact: Array.isArray(constraints.video.deviceId.exact)
+						? constraints.video.deviceId.exact[0]
+						: constraints.video.deviceId.exact
+				};
+			} else if (!!constraints.video.deviceId.ideal) {
+				newConstraints.video.deviceId = {
+					ideal: Array.isArray(constraints.video.deviceId.ideal)
+						? constraints.video.deviceId.ideal[0]
+						: constraints.video.deviceId.ideal
+				};
+			}
+		}
+
+		// Get requested width min/max, exact.
+		if (typeof constraints.video.width === 'object') {
+			newConstraints.video.width = {};
+			if (isPositiveInteger(constraints.video.width.min)) {
+				newConstraints.video.width.min = constraints.video.width.min;
+			}
+			if (isPositiveInteger(constraints.video.width.max)) {
+				newConstraints.video.width.max = constraints.video.width.max;
+			}
+			if (isPositiveInteger(constraints.video.width.exact)) {
+				newConstraints.video.width.exact = constraints.video.width.exact;
+			}
+			if (isPositiveInteger(constraints.video.width.ideal)) {
+				newConstraints.video.width.ideal = constraints.video.width.ideal;
+			}
+		} else if (isPositiveInteger(constraints.video.width)) {
+			// Get requested width long as exact
+			newConstraints.video.width = {
+				exact: constraints.video.width
+			};
+		}
+
+		// Get requested height min/max, exact.
+		if (typeof constraints.video.height === 'object') {
+			newConstraints.video.height = {};
+			if (isPositiveInteger(constraints.video.height.min)) {
+				newConstraints.video.height.min = constraints.video.height.min;
+			}
+			if (isPositiveInteger(constraints.video.height.max)) {
+				newConstraints.video.height.max = constraints.video.height.max;
+			}
+			if (isPositiveInteger(constraints.video.height.exact)) {
+				newConstraints.video.height.exact = constraints.video.height.exact;
+			}
+			if (isPositiveInteger(constraints.video.height.ideal)) {
+				newConstraints.video.height.ideal = constraints.video.height.ideal;
+			}
+		} else if (isPositiveInteger(constraints.video.height)) {
+			// Get requested height long as exact
+			newConstraints.video.height = {
+				exact: constraints.video.height
+			};
+		}
+
+		// Get requested frameRate min/max.
+		if (typeof constraints.video.frameRate === 'object') {
+			newConstraints.video.frameRate = {};
+			if (isPositiveFloat(constraints.video.frameRate.min)) {
+				newConstraints.video.frameRate.min = parseFloat(
+					constraints.video.frameRate.min,
+					10
+				);
+			}
+			if (isPositiveFloat(constraints.video.frameRate.max)) {
+				newConstraints.video.frameRate.max = parseFloat(
+					constraints.video.frameRate.max,
+					10
+				);
+			}
+			if (isPositiveInteger(constraints.video.frameRate.exact)) {
+				newConstraints.video.frameRate.exact = constraints.video.frameRate.exact;
+			}
+			if (isPositiveInteger(constraints.video.frameRate.ideal)) {
+				newConstraints.video.frameRate.ideal = constraints.video.frameRate.ideal;
+			}
+		} else if (isPositiveFloat(constraints.video.frameRate)) {
+			// Get requested frameRate double as exact
+			newConstraints.video.frameRate = {
+				exact: parseFloat(constraints.video.frameRate, 10)
+			};
+		}
+
+		// get aspectRatio (e.g 1.7777777777777777)
+		// TODO ConstrainDouble min, max
+		if (typeof constraints.video.aspectRatio === 'object') {
+			newConstraints.video.aspectRatio = {};
+			if (isPositiveFloat(constraints.video.aspectRatio.min)) {
+				newConstraints.video.aspectRatio.min = parseFloat(
+					constraints.video.aspectRatio.min,
+					10
+				);
+			}
+			if (isPositiveFloat(constraints.video.aspectRatio.max)) {
+				newConstraints.video.aspectRatio.max = parseFloat(
+					constraints.video.aspectRatio.max,
+					10
+				);
+			}
+			if (isPositiveInteger(constraints.video.aspectRatio.exact)) {
+				newConstraints.video.aspectRatio.exact = constraints.video.aspectRatio.exact;
+			}
+			if (isPositiveInteger(constraints.video.aspectRatio.ideal)) {
+				newConstraints.video.aspectRatio.ideal = constraints.video.aspectRatio.ideal;
+			}
+		} else if (isPositiveFloat(constraints.video.aspectRatio)) {
+			newConstraints.video.aspectRatio = {
+				exact: parseFloat(constraints.video.aspectRatio, 10)
+			};
+		}
+
+		// get facingMode (e.g environment, user)
+		// TODO ConstrainDOMStringParameters ideal, exact
+		if (typeof constraints.video.facingMode === 'string') {
+			newConstraints.video.facingMode = {
+				exact: constraints.video.facingMode
+			};
+		} else if (typeof constraints.video.facingMode === 'object') {
+			if (typeof constraints.video.facingMode.exact === 'string') {
+				newConstraints.video.facingMode = {
+					exact: constraints.video.facingMode.exact
+				};
+			} else if (typeof constraints.video.facingMode.ideal === 'string') {
+				newConstraints.video.facingMode = {
+					ideal: constraints.video.facingMode.ideal
+				};
+			}
+		}
+	}
+
+	// Get audio constraints
+	if (audioRequested) {
+		// Handle object audio constraints
+		newConstraints.audio = {};
+
+		// Handle Stupid not up-to-date webrtc-adapter
+		// Note: Firefox [38+] does support a subset of constraints with getUserMedia(), but not the outdated syntax that Chrome and Opera are using.
+		// The mandatory / optional syntax was deprecated a in 2014, and minWidth and minHeight the year before that.
+		if (
+			typeof constraints.audio === 'object' &&
+			(typeof constraints.audio.optional === 'object' ||
+				typeof constraints.audio.mandatory === 'object')
+		) {
+			if (typeof constraints.audio.optional === 'object') {
+				if (typeof constraints.audio.optional.sourceId === 'string') {
+					newConstraints.audio.deviceId = {
+						ideal: constraints.audio.optional.sourceId
+					};
+				} else if (
+					Array.isArray(constraints.audio.optional) &&
+					typeof constraints.audio.optional[0] === 'object' &&
+					typeof constraints.audio.optional[0].sourceId === 'string'
+				) {
+					newConstraints.audio.deviceId = {
+						ideal: constraints.audio.optional[0].sourceId
+					};
+				}
+			} else if (
+				constraints.audio.mandatory &&
+				typeof constraints.audio.mandatory.sourceId === 'string'
+			) {
+				newConstraints.audio.deviceId = {
+					exact: constraints.audio.mandatory.sourceId
+				};
+			}
+		}
+
+		// Get requested audio deviceId.
+		if (typeof constraints.audio.deviceId === 'string') {
+			newConstraints.audio.deviceId = {
+				exact: constraints.audio.deviceId
+			};
+		} else if (typeof constraints.audio.sourceId === 'string') {
+			// Also check audio sourceId (mangled by adapter.js).
+			newConstraints.audio.deviceId = {
+				exact: constraints.audio.sourceId
+			};
+		} else if (typeof constraints.audio.deviceId === 'object') {
+			// Also check deviceId.(exact|ideal)
+			if (!!constraints.audio.deviceId.exact) {
+				newConstraints.audio.deviceId = {
+					exact: Array.isArray(constraints.audio.deviceId.exact)
+						? constraints.audio.deviceId.exact[0]
+						: constraints.audio.deviceId.exact
+				};
+			} else if (!!constraints.audio.deviceId.ideal) {
+				newConstraints.audio.deviceId = {
+					ideal: Array.isArray(constraints.audio.deviceId.ideal)
+						? constraints.audio.deviceId.ideal[0]
+						: constraints.audio.deviceId.ideal
+				};
+			}
+		}
+	}
+
+	debug('[computed constraints:%o]', newConstraints);
+
+	return new Promise(function (resolve, reject) {
+		function onResultOK(data) {
+			debug('getUserMedia() | success');
+			var stream = MediaStream.create(data.stream);
+			resolve(stream);
+			// Emit "connected" on the stream.
+			stream.emitConnected();
+		}
+
+		function onResultError(error) {
+			debugerror('getUserMedia() | failure: %s', error);
+			reject(new Errors.MediaStreamError('getUserMedia() failed: ' + error));
+		}
+
+		exec(onResultOK, onResultError, 'iosrtcPlugin', 'getUserMedia', [newConstraints]);
+	});
+}
+
+},{"./Errors":1,"./MediaStream":5,"cordova/exec":undefined,"debug":23}],21:[function(_dereq_,module,exports){
+(function (global){
+/**
+ * Variables.
+ */
+
+var // Dictionary of MediaStreamRenderers.
+	// - key: MediaStreamRenderer id.
+	// - value: MediaStreamRenderer.
+	mediaStreamRenderers = {},
+	// Dictionary of MediaStreams.
+	// - key: MediaStream blobId.
+	// - value: MediaStream.
+	mediaStreams = {},
+	/**
+	 * Dependencies.
+	 */
+	debug = _dereq_('debug')('iosrtc'),
+	exec = _dereq_('cordova/exec'),
+	domready = _dereq_('domready'),
+	getUserMedia = _dereq_('./getUserMedia'),
+	enumerateDevices = _dereq_('./enumerateDevices'),
+	RTCPeerConnection = _dereq_('./RTCPeerConnection'),
+	RTCSessionDescription = _dereq_('./RTCSessionDescription'),
+	RTCIceCandidate = _dereq_('./RTCIceCandidate'),
+	MediaDevices = _dereq_('./MediaDevices'),
+	MediaStream = _dereq_('./MediaStream'),
+	{ MediaStreamTrack } = _dereq_('./MediaStreamTrack'),
+	videoElementsHandler = _dereq_('./videoElementsHandler'),
+	{ RTCRtpTransceiver } = _dereq_('./RTCRtpTransceiver');
+
+/**
+ * Expose the iosrtc object.
+ */
+module.exports = {
+	// Expose WebRTC classes and functions.
+	getUserMedia: getUserMedia,
+	enumerateDevices: enumerateDevices,
+	getMediaDevices: enumerateDevices, // TMP
+	RTCPeerConnection: RTCPeerConnection,
+	RTCSessionDescription: RTCSessionDescription,
+	RTCIceCandidate: RTCIceCandidate,
+	MediaDevices: MediaDevices,
+	MediaStream: MediaStream,
+	MediaStreamTrack: MediaStreamTrack,
+
+	// Expose a function to refresh current videos rendering a MediaStream.
+	refreshVideos: videoElementsHandler.refreshVideos,
+
+	// Expose a function to handle a video not yet inserted in the DOM.
+	observeVideo: videoElementsHandler.observeVideo,
+
+	// Select audio output (earpiece or speaker).
+	selectAudioOutput: selectAudioOutput,
+
+	// Set default audio output (earpiece or speaker).
+	setDefaultAudioOutput: setDefaultAudioOutput,
+
+	// turnOnSpeaker with options
+	turnOnSpeaker: turnOnSpeaker,
+
+	// Checking permission (audio and camera)
+	requestPermission: requestPermission,
+
+	// Expose a function to initAudioDevices if needed, sets the audio session active
+	initAudioDevices: initAudioDevices,
+
+	// Expose a function to pollute window and naigator namespaces.
+	registerGlobals: registerGlobals,
+
+	// Expose the debug module.
+	debug: _dereq_('debug'),
+
+	// Debug function to see what happens internally.
+	dump: dump,
+
+	// Debug Stores to see what happens internally.
+	mediaStreamRenderers: mediaStreamRenderers,
+	mediaStreams: mediaStreams
+};
+
+domready(function () {
+	// Let the MediaStream class and the videoElementsHandler share same MediaStreams container.
+	MediaStream.setMediaStreams(mediaStreams);
+	videoElementsHandler(mediaStreams, mediaStreamRenderers);
+
+	// refreshVideos on device orientation change to resize peers video
+	// while local video will resize du orientation change
+	window.addEventListener('resize', function () {
+		videoElementsHandler.refreshVideos();
+	});
+});
+
+function selectAudioOutput(output) {
+	debug('selectAudioOutput() | [output:"%s"]', output);
+
+	switch (output) {
+		case 'earpiece':
+			exec(null, null, 'iosrtcPlugin', 'selectAudioOutputEarpiece', []);
+			break;
+		case 'speaker':
+			exec(null, null, 'iosrtcPlugin', 'selectAudioOutputSpeaker', []);
+			break;
+		default:
+			throw new Error('output must be "earpiece" or "speaker"');
+	}
+}
+
+function setDefaultAudioOutput(output) {
+	debug('setDefaultAudioOutput() | [output:"%s"]', output);
+	if (!['earpiece', 'speaker'].includes(output)) {
+		throw new Error('output must be "earpiece" or "speaker"');
+	}
+	exec(null, null, 'iosrtcPlugin', 'setDefaultAudioOutput', [output === 'speaker']);
+}
+
+function turnOnSpeaker(isTurnOn) {
+	debug('turnOnSpeaker() | [isTurnOn:"%s"]', isTurnOn);
+
+	exec(null, null, 'iosrtcPlugin', 'RTCTurnOnSpeaker', [isTurnOn]);
+}
+
+function requestPermission(needMic, needCamera, callback) {
+	debug('requestPermission() | [needMic:"%s", needCamera:"%s"]', needMic, needCamera);
+
+	function ok() {
+		callback(true);
+	}
+
+	function error() {
+		callback(false);
+	}
+	exec(ok, error, 'iosrtcPlugin', 'RTCRequestPermission', [needMic, needCamera]);
+}
+
+function initAudioDevices() {
+	debug('initAudioDevices()');
+
+	exec(null, null, 'iosrtcPlugin', 'initAudioDevices', []);
+}
+
+function callbackifyMethod(originalMethod) {
+	return function () {
+		var success,
+			failure,
+			originalArgs = Array.prototype.slice.call(arguments);
+
+		var callbackArgs = [];
+		originalArgs.forEach(function (arg) {
+			if (typeof arg === 'function') {
+				if (!success) {
+					success = arg;
+				} else {
+					failure = arg;
+				}
+			} else {
+				callbackArgs.push(arg);
+			}
+		});
+
+		var promiseResult = originalMethod.apply(this, callbackArgs);
+
+		// Only apply then if callback success available
+		if (typeof success === 'function') {
+			promiseResult = promiseResult.then(success);
+		}
+
+		// Only apply catch if callback failure available
+		if (typeof failure === 'function') {
+			promiseResult = promiseResult.catch(failure);
+		}
+
+		return promiseResult;
+	};
+}
+
+function callbackifyPrototype(proto, method) {
+	var originalMethod = proto[method];
+	proto[method] = callbackifyMethod(originalMethod);
+}
+
+function restoreCallbacksSupport() {
+	debug('restoreCallbacksSupport()');
+	getUserMedia = callbackifyMethod(getUserMedia);
+	enumerateDevices = callbackifyMethod(enumerateDevices);
+	callbackifyPrototype(RTCPeerConnection.prototype, 'createAnswer');
+	callbackifyPrototype(RTCPeerConnection.prototype, 'createOffer');
+	callbackifyPrototype(RTCPeerConnection.prototype, 'setRemoteDescription');
+	callbackifyPrototype(RTCPeerConnection.prototype, 'setLocalDescription');
+	callbackifyPrototype(RTCPeerConnection.prototype, 'addIceCandidate');
+	callbackifyPrototype(RTCPeerConnection.prototype, 'getStats');
+}
+
+function registerGlobals(doNotRestoreCallbacksSupport) {
+	debug('registerGlobals()');
+
+	if (!global.navigator) {
+		global.navigator = {};
+	}
+
+	// Restore Callback support
+	if (!doNotRestoreCallbacksSupport) {
+		restoreCallbacksSupport();
+	}
+
+	navigator.getUserMedia = getUserMedia;
+	navigator.webkitGetUserMedia = getUserMedia;
+
+	// Prevent WebRTC-adapter to overide navigator.mediaDevices after shim is applied since ios 14.3
+	if (!(navigator.mediaDevices instanceof MediaDevices)) {
+		Object.defineProperty(
+			navigator,
+			'mediaDevices',
+			{
+				value: new MediaDevices(),
+				writable: false
+			},
+			{
+				enumerable: false,
+				configurable: false,
+				writable: false,
+				value: 'static'
+			}
+		);
+	}
+
+	window.RTCPeerConnection = RTCPeerConnection;
+	window.webkitRTCPeerConnection = RTCPeerConnection;
+	window.RTCSessionDescription = RTCSessionDescription;
+	window.RTCIceCandidate = RTCIceCandidate;
+	window.MediaStream = MediaStream;
+	window.webkitMediaStream = MediaStream;
+	window.MediaStreamTrack = MediaStreamTrack;
+	window.RTCRtpTransceiver = RTCRtpTransceiver;
+	window.MediaStreamTrackEvent = window.Event;
+
+	// Apply CanvasRenderingContext2D.drawImage monkey patch
+	var drawImage = CanvasRenderingContext2D.prototype.drawImage;
+	CanvasRenderingContext2D.prototype.drawImage = (function () {
+		// Methods to address the memory leaks problems in Safari
+		let temporaryImage, imageElement;
+		const BASE64_MARKER = ';base64,';
+		const objectURL = window.URL || window.webkitURL;
+
+		function convertDataURIToBlob(dataURI) {
+			// Validate input data
+			if (!dataURI) {
+				return;
+			}
+
+			// Convert image (in base64) to binary data
+			const base64Index = dataURI.indexOf(BASE64_MARKER) + BASE64_MARKER.length;
+			const base64 = dataURI.substring(base64Index);
+			const raw = window.atob(base64);
+			const rawLength = raw.length;
+			let array = new Uint8Array(new ArrayBuffer(rawLength));
+
+			for (let i = 0; i < rawLength; i++) {
+				array[i] = raw.charCodeAt(i);
+			}
+
+			// Create and return a new blob object using binary data
+			return new Blob([array], { type: 'image/jpeg' });
+		}
+
+		return function (arg) {
+			const context = this;
+			let args = Array.prototype.slice.call(arguments);
+			if (arg instanceof HTMLVideoElement && arg.render) {
+				arg.render.save(function (base64Image) {
+					// Destroy old image
+					if (temporaryImage) {
+						objectURL.revokeObjectURL(temporaryImage);
+					}
+
+					// Create a new image from binary data
+					const imageDataBlob = convertDataURIToBlob(
+						'data:image/jpg;base64,' + base64Image
+					);
+
+					// Create a new object URL object
+					imageElement = imageElement || new Image();
+					temporaryImage = objectURL.createObjectURL(imageDataBlob);
+
+					imageElement.addEventListener('load', function () {
+						args.splice(0, 1, imageElement);
+						drawImage.apply(context, args);
+					});
+
+					// Set the new image
+					imageElement.src = temporaryImage;
+				});
+			} else {
+				return drawImage.apply(context, args);
+			}
+		};
+	})();
+}
+
+function dump() {
+	exec(null, null, 'iosrtcPlugin', 'dump', []);
+}
+
+}).call(this,typeof global !== "undefined" ? global : typeof self !== "undefined" ? self : typeof window !== "undefined" ? window : {})
+},{"./MediaDevices":4,"./MediaStream":5,"./MediaStreamTrack":7,"./RTCIceCandidate":12,"./RTCPeerConnection":13,"./RTCRtpTransceiver":16,"./RTCSessionDescription":17,"./enumerateDevices":19,"./getUserMedia":20,"./videoElementsHandler":22,"cordova/exec":undefined,"debug":23,"domready":25}],22:[function(_dereq_,module,exports){
+/**
+ * Expose a function that must be called when the library is loaded.
+ * And also a helper function.
+ */
+module.exports = videoElementsHandler;
+module.exports.observeVideo = observeVideo;
+module.exports.refreshVideos = refreshVideos;
+
+/**
+ * Dependencies.
+ */
+var debug = _dereq_('debug')('iosrtc:videoElementsHandler'),
+	MediaStreamRenderer = _dereq_('./MediaStreamRenderer'),
+	/**
+	 * Local variables.
+	 */
+
+	// RegExp for Blob URI.
+	BLOB_INTERNAL_URI_REGEX = new RegExp(/^blob:/),
+	// Dictionary of MediaStreamRenderers (provided via module argument).
+	// - key: MediaStreamRenderer id.
+	// - value: MediaStreamRenderer.
+	mediaStreamRenderers,
+	// Dictionary of MediaStreams (provided via module argument).
+	// - key: MediaStream blobId.
+	// - value: MediaStream.
+	mediaStreams,
+	// Video element mutation observer.
+	videoObserver = new MutationObserver(function (mutations) {
+		var i, numMutations, mutation, video;
+
+		for (i = 0, numMutations = mutations.length; i < numMutations; i++) {
+			mutation = mutations[i];
+
+			// HTML video element.
+			video = mutation.target;
+
+			// .srcObject removed.
+			if (!video.srcObject && !video.src) {
+				// If this video element was previously handling a MediaStreamRenderer, release it.
+				releaseMediaStreamRenderer(video);
+				continue;
+			}
+
+			handleVideo(video);
+		}
+	}),
+	// DOM mutation observer.
+	domObserver = new MutationObserver(function (mutations) {
+		var i, numMutations, mutation, j, numNodes, node;
+
+		for (i = 0, numMutations = mutations.length; i < numMutations; i++) {
+			mutation = mutations[i];
+
+			// Check if there has been addition or deletion of nodes.
+			if (mutation.type !== 'childList') {
+				continue;
+			}
+
+			// Check added nodes.
+			for (j = 0, numNodes = mutation.addedNodes.length; j < numNodes; j++) {
+				node = mutation.addedNodes[j];
+
+				checkNewNode(node);
+			}
+
+			// Check removed nodes.
+			for (j = 0, numNodes = mutation.removedNodes.length; j < numNodes; j++) {
+				node = mutation.removedNodes[j];
+
+				checkRemovedNode(node);
+			}
+		}
+
+		function checkNewNode(node) {
+			var j, childNode;
+
+			if (node.nodeName === 'VIDEO') {
+				debug('new video element added');
+
+				// Avoid same node firing more than once (really, may happen in some cases).
+				if (node._iosrtcVideoHandled) {
+					return;
+				}
+				node._iosrtcVideoHandled = true;
+
+				// Observe changes in the video element.
+				observeVideo(node);
+			} else {
+				for (j = 0; j < node.childNodes.length; j++) {
+					childNode = node.childNodes.item(j);
+
+					checkNewNode(childNode);
+				}
+			}
+		}
+
+		function checkRemovedNode(node) {
+			var j, childNode;
+
+			if (node.nodeName === 'VIDEO') {
+				debug('video element removed');
+
+				// If this video element was previously handling a MediaStreamRenderer, release it.
+				releaseMediaStreamRenderer(node);
+				delete node._iosrtcVideoHandled;
+			} else {
+				for (j = 0; j < node.childNodes.length; j++) {
+					childNode = node.childNodes.item(j);
+
+					checkRemovedNode(childNode);
+				}
+			}
+		}
+	});
+
+function refreshVideos() {
+	debug('refreshVideos()');
+
+	var id;
+
+	for (id in mediaStreamRenderers) {
+		if (mediaStreamRenderers.hasOwnProperty(id)) {
+			mediaStreamRenderers[id].refresh();
+		}
+	}
+}
+
+function videoElementsHandler(_mediaStreams, _mediaStreamRenderers) {
+	var existingVideos = document.querySelectorAll('video'),
+		i,
+		len,
+		video;
+
+	mediaStreams = _mediaStreams;
+	mediaStreamRenderers = _mediaStreamRenderers;
+
+	// Search the whole document for already existing HTML video elements and observe them.
+	for (i = 0, len = existingVideos.length; i < len; i++) {
+		video = existingVideos.item(i);
+
+		debug('video element found');
+
+		observeVideo(video);
+	}
+
+	// Observe the whole document for additions of new HTML video elements and observe them.
+	domObserver.observe(document, {
+		// Set to true if additions and removals of the target node's child elements (including text nodes) are to
+		// be observed.
+		childList: true,
+		// Set to true if mutations to target's attributes are to be observed.
+		attributes: false,
+		// Set to true if mutations to target's data are to be observed.
+		characterData: false,
+		// Set to true if mutations to not just target, but also target's descendants are to be observed.
+		subtree: true,
+		// Set to true if attributes is set to true and target's attribute value before the mutation needs to be
+		// recorded.
+		attributeOldValue: false,
+		// Set to true if characterData is set to true and target's data before the mutation needs to be recorded.
+		characterDataOldValue: false
+		// Set to an array of attribute local names (without namespace) if not all attribute mutations need to be
+		// observed.
+		// attributeFilter:
+	});
+}
+
+function observeVideo(video) {
+	debug('observeVideo()');
+
+	// If the video already has a srcObject property but is not yet handled by the plugin
+	// then handle it now.
+	var hasStream = video.srcObject || video.src;
+	if (hasStream && !video._iosrtcMediaStreamRendererId) {
+		handleVideo(video);
+	}
+
+	// Add .srcObject observer to the video element.
+	videoObserver.observe(video, {
+		// Set to true if additions and removals of the target node's child elements (including text
+		// nodes) are to be observed.
+		childList: false,
+		// Set to true if mutations to target's attributes are to be observed.
+		attributes: true,
+		// Set to true if mutations to target's data are to be observed.
+		characterData: false,
+		// Set to true if mutations to not just target, but also target's descendants are to be observed.
+		subtree: false,
+		// Set to true if attributes is set to true and target's attribute value before the mutation
+		// needs to be recorded.
+		attributeOldValue: false,
+		// Set to true if characterData is set to true and target's data before the mutation needs to be
+		// recorded.
+		characterDataOldValue: false,
+		// Set to an array of attribute local names (without namespace) if not all attribute mutations
+		// need to be observed.
+		// srcObject DO not trigger MutationObserver
+		attributeFilter: ['srcObject', 'src']
+	});
+
+	// MutationObserver fail to trigger when using srcObject on ony tested browser.
+	// But video.srcObject = new MediaStream() will trigger onloadstart and
+	// video.srcObject = null will trigger onemptied events.
+
+	video.addEventListener('loadstart', function () {
+		var hasStream = video.srcObject || video.src;
+
+		if (hasStream && !video._iosrtcMediaStreamRendererId) {
+			// If this video element was NOT previously handling a MediaStreamRenderer, release it.
+			handleVideo(video);
+		} else if (hasStream && video._iosrtcMediaStreamRendererId) {
+			// The video element has received a new srcObject.
+			var stream = video.srcObject;
+			if (stream && typeof stream.getBlobId === 'function') {
+				// Release previous renderer
+				releaseMediaStreamRenderer(video);
+				// Install new renderer
+				provideMediaStreamRenderer(video, stream.getBlobId());
+			}
+		}
+	});
+
+	video.addEventListener('emptied', function () {
+		var hasStream = video.srcObject || video.src;
+		if (!hasStream && video._iosrtcMediaStreamRendererId) {
+			// If this video element was previously handling a MediaStreamRenderer, release it.
+			releaseMediaStreamRenderer(video);
+		}
+	});
+
+	// Intercept video 'error' events if it's due to the attached MediaStream.
+	video.addEventListener('error', function (event) {
+		if (
+			video.error.code === window.MediaError.MEDIA_ERR_SRC_NOT_SUPPORTED &&
+			BLOB_INTERNAL_URI_REGEX.test(video.src)
+		) {
+			debug('stopping "error" event propagation for video element');
+
+			event.stopImmediatePropagation();
+		}
+	});
+}
+
+/**
+ * Private API.
+ */
+
+function handleVideo(video) {
+	var stream;
+
+	// The app has set video.srcObject.
+	if (video.srcObject) {
+		stream = video.srcObject;
+		if (stream && typeof stream.getBlobId === 'function') {
+			if (!stream.getBlobId()) {
+				// If this video element was previously handling a MediaStreamRenderer, release it.
+				releaseMediaStreamRenderer(video);
+
+				return;
+			}
+
+			provideMediaStreamRenderer(video, stream.getBlobId());
+		}
+	} else if (video.src) {
+		var xhr = new XMLHttpRequest();
+		xhr.open('GET', video.src, true);
+		xhr.responseType = 'blob';
+		xhr.onload = function () {
+			if (xhr.status !== 200) {
+				// If this video element was previously handling a MediaStreamRenderer, release it.
+				releaseMediaStreamRenderer(video);
+
+				return;
+			}
+
+			var reader = new FileReader();
+
+			// Some versions of Safari fail to set onloadend property, some others do not react
+			// on 'loadend' event. Try everything here.
+			try {
+				reader.onloadend = onloadend;
+			} catch (error) {
+				reader.addEventListener('loadend', onloadend);
+			}
+			reader.readAsText(xhr.response);
+
+			function onloadend() {
+				var mediaStreamBlobId = reader.result;
+
+				// The retrieved URL does not point to a MediaStream.
+				if (!mediaStreamBlobId || typeof mediaStreamBlobId !== 'string') {
+					// If this video element was previously handling a MediaStreamRenderer, release it.
+					releaseMediaStreamRenderer(video);
+
+					return;
+				}
+
+				provideMediaStreamRenderer(video, mediaStreamBlobId);
+			}
+		};
+		xhr.send();
+	}
+}
+
+function provideMediaStreamRenderer(video, mediaStreamBlobId) {
+	var mediaStream = mediaStreams[mediaStreamBlobId],
+		mediaStreamRenderer = mediaStreamRenderers[video._iosrtcMediaStreamRendererId];
+
+	if (!mediaStream) {
+		releaseMediaStreamRenderer(video);
+
+		return;
+	}
+
+	if (mediaStreamRenderer) {
+		mediaStreamRenderer.render(mediaStream);
+	} else {
+		mediaStreamRenderer = new MediaStreamRenderer(video);
+		mediaStreamRenderer.render(mediaStream);
+
+		mediaStreamRenderers[mediaStreamRenderer.id] = mediaStreamRenderer;
+		video._iosrtcMediaStreamRendererId = mediaStreamRenderer.id;
+	}
+
+	// Close the MediaStreamRenderer of this video if it emits "close" event.
+	mediaStreamRenderer.addEventListener('close', function () {
+		if (mediaStreamRenderers[video._iosrtcMediaStreamRendererId] !== mediaStreamRenderer) {
+			return;
+		}
+
+		releaseMediaStreamRenderer(video);
+	});
+
+	// Override some <video> properties.
+	// NOTE: This is a terrible hack but it works.
+	Object.defineProperties(video, {
+		videoWidth: {
+			configurable: true,
+			get: function () {
+				return mediaStreamRenderer.videoWidth || 0;
+			}
+		},
+		videoHeight: {
+			configurable: true,
+			get: function () {
+				return mediaStreamRenderer.videoHeight || 0;
+			}
+		},
+		readyState: {
+			configurable: true,
+			get: function () {
+				if (
+					mediaStreamRenderer &&
+					mediaStreamRenderer.stream &&
+					mediaStreamRenderer.stream.connected
+				) {
+					return video.HAVE_ENOUGH_DATA;
+				} else {
+					return video.HAVE_NOTHING;
+				}
+			}
+		}
+	});
+}
+
+function releaseMediaStreamRenderer(video) {
+	if (!video._iosrtcMediaStreamRendererId) {
+		return;
+	}
+
+	var mediaStreamRenderer = mediaStreamRenderers[video._iosrtcMediaStreamRendererId];
+
+	if (mediaStreamRenderer) {
+		delete mediaStreamRenderers[video._iosrtcMediaStreamRendererId];
+		mediaStreamRenderer.close();
+	}
+
+	delete video._iosrtcMediaStreamRendererId;
+
+	// Remove overrided <video> properties.
+	delete video.videoWidth;
+	delete video.videoHeight;
+	delete video.readyState;
+}
+
+},{"./MediaStreamRenderer":6,"debug":23}],23:[function(_dereq_,module,exports){
+(function (process){
+/* eslint-env browser */
+
+/**
+ * This is the web browser implementation of `debug()`.
+ */
+
+exports.log = log;
+exports.formatArgs = formatArgs;
+exports.save = save;
+exports.load = load;
+exports.useColors = useColors;
+exports.storage = localstorage();
+
+/**
+ * Colors.
+ */
+
+exports.colors = [
+	'#0000CC',
+	'#0000FF',
+	'#0033CC',
+	'#0033FF',
+	'#0066CC',
+	'#0066FF',
+	'#0099CC',
+	'#0099FF',
+	'#00CC00',
+	'#00CC33',
+	'#00CC66',
+	'#00CC99',
+	'#00CCCC',
+	'#00CCFF',
+	'#3300CC',
+	'#3300FF',
+	'#3333CC',
+	'#3333FF',
+	'#3366CC',
+	'#3366FF',
+	'#3399CC',
+	'#3399FF',
+	'#33CC00',
+	'#33CC33',
+	'#33CC66',
+	'#33CC99',
+	'#33CCCC',
+	'#33CCFF',
+	'#6600CC',
+	'#6600FF',
+	'#6633CC',
+	'#6633FF',
+	'#66CC00',
+	'#66CC33',
+	'#9900CC',
+	'#9900FF',
+	'#9933CC',
+	'#9933FF',
+	'#99CC00',
+	'#99CC33',
+	'#CC0000',
+	'#CC0033',
+	'#CC0066',
+	'#CC0099',
+	'#CC00CC',
+	'#CC00FF',
+	'#CC3300',
+	'#CC3333',
+	'#CC3366',
+	'#CC3399',
+	'#CC33CC',
+	'#CC33FF',
+	'#CC6600',
+	'#CC6633',
+	'#CC9900',
+	'#CC9933',
+	'#CCCC00',
+	'#CCCC33',
+	'#FF0000',
+	'#FF0033',
+	'#FF0066',
+	'#FF0099',
+	'#FF00CC',
+	'#FF00FF',
+	'#FF3300',
+	'#FF3333',
+	'#FF3366',
+	'#FF3399',
+	'#FF33CC',
+	'#FF33FF',
+	'#FF6600',
+	'#FF6633',
+	'#FF9900',
+	'#FF9933',
+	'#FFCC00',
+	'#FFCC33'
+];
+
+/**
+ * Currently only WebKit-based Web Inspectors, Firefox >= v31,
+ * and the Firebug extension (any Firefox version) are known
+ * to support "%c" CSS customizations.
+ *
+ * TODO: add a `localStorage` variable to explicitly enable/disable colors
+ */
+
+// eslint-disable-next-line complexity
+function useColors() {
+	// NB: In an Electron preload script, document will be defined but not fully
+	// initialized. Since we know we're in Chrome, we'll just detect this case
+	// explicitly
+	if (typeof window !== 'undefined' && window.process && (window.process.type === 'renderer' || window.process.__nwjs)) {
+		return true;
+	}
+
+	// Internet Explorer and Edge do not support colors.
+	if (typeof navigator !== 'undefined' && navigator.userAgent && navigator.userAgent.toLowerCase().match(/(edge|trident)\/(\d+)/)) {
+		return false;
+	}
+
+	// Is webkit? http://stackoverflow.com/a/16459606/376773
+	// document is undefined in react-native: https://github.com/facebook/react-native/pull/1632
+	return (typeof document !== 'undefined' && document.documentElement && document.documentElement.style && document.documentElement.style.WebkitAppearance) ||
+		// Is firebug? http://stackoverflow.com/a/398120/376773
+		(typeof window !== 'undefined' && window.console && (window.console.firebug || (window.console.exception && window.console.table))) ||
+		// Is firefox >= v31?
+		// https://developer.mozilla.org/en-US/docs/Tools/Web_Console#Styling_messages
+		(typeof navigator !== 'undefined' && navigator.userAgent && navigator.userAgent.toLowerCase().match(/firefox\/(\d+)/) && parseInt(RegExp.$1, 10) >= 31) ||
+		// Double check webkit in userAgent just in case we are in a worker
+		(typeof navigator !== 'undefined' && navigator.userAgent && navigator.userAgent.toLowerCase().match(/applewebkit\/(\d+)/));
+}
+
+/**
+ * Colorize log arguments if enabled.
+ *
+ * @api public
+ */
+
+function formatArgs(args) {
+	args[0] = (this.useColors ? '%c' : '') +
+		this.namespace +
+		(this.useColors ? ' %c' : ' ') +
+		args[0] +
+		(this.useColors ? '%c ' : ' ') +
+		'+' + module.exports.humanize(this.diff);
+
+	if (!this.useColors) {
+		return;
+	}
+
+	const c = 'color: ' + this.color;
+	args.splice(1, 0, c, 'color: inherit');
+
+	// The final "%c" is somewhat tricky, because there could be other
+	// arguments passed either before or after the %c, so we need to
+	// figure out the correct index to insert the CSS into
+	let index = 0;
+	let lastC = 0;
+	args[0].replace(/%[a-zA-Z%]/g, match => {
+		if (match === '%%') {
+			return;
+		}
+		index++;
+		if (match === '%c') {
+			// We only are interested in the *last* %c
+			// (the user may have provided their own)
+			lastC = index;
+		}
+	});
+
+	args.splice(lastC, 0, c);
+}
+
+/**
+ * Invokes `console.log()` when available.
+ * No-op when `console.log` is not a "function".
+ *
+ * @api public
+ */
+function log(...args) {
+	// This hackery is required for IE8/9, where
+	// the `console.log` function doesn't have 'apply'
+	return typeof console === 'object' &&
+		console.log &&
+		console.log(...args);
+}
+
+/**
+ * Save `namespaces`.
+ *
+ * @param {String} namespaces
+ * @api private
+ */
+function save(namespaces) {
+	try {
+		if (namespaces) {
+			exports.storage.setItem('debug', namespaces);
+		} else {
+			exports.storage.removeItem('debug');
+		}
+	} catch (error) {
+		// Swallow
+		// XXX (@Qix-) should we be logging these?
+	}
+}
+
+/**
+ * Load `namespaces`.
+ *
+ * @return {String} returns the previously persisted debug modes
+ * @api private
+ */
+function load() {
+	let r;
+	try {
+		r = exports.storage.getItem('debug');
+	} catch (error) {
+		// Swallow
+		// XXX (@Qix-) should we be logging these?
+	}
+
+	// If debug isn't set in LS, and we're in Electron, try to load $DEBUG
+	if (!r && typeof process !== 'undefined' && 'env' in process) {
+		r = process.env.DEBUG;
+	}
+
+	return r;
+}
+
+/**
+ * Localstorage attempts to return the localstorage.
+ *
+ * This is necessary because safari throws
+ * when a user disables cookies/localstorage
+ * and you attempt to access it.
+ *
+ * @return {LocalStorage}
+ * @api private
+ */
+
+function localstorage() {
+	try {
+		// TVMLKit (Apple TV JS Runtime) does not have a window object, just localStorage in the global context
+		// The Browser also has localStorage in the global context.
+		return localStorage;
+	} catch (error) {
+		// Swallow
+		// XXX (@Qix-) should we be logging these?
+	}
+}
+
+module.exports = _dereq_('./common')(exports);
+
+const {formatters} = module.exports;
+
+/**
+ * Map %j to `JSON.stringify()`, since no Web Inspectors do that by default.
+ */
+
+formatters.j = function (v) {
+	try {
+		return JSON.stringify(v);
+	} catch (error) {
+		return '[UnexpectedJSONParseError]: ' + error.message;
+	}
+};
+
+}).call(this,_dereq_('_process'))
+},{"./common":24,"_process":27}],24:[function(_dereq_,module,exports){
+
+/**
+ * This is the common logic for both the Node.js and web browser
+ * implementations of `debug()`.
+ */
+
+function setup(env) {
+	createDebug.debug = createDebug;
+	createDebug.default = createDebug;
+	createDebug.coerce = coerce;
+	createDebug.disable = disable;
+	createDebug.enable = enable;
+	createDebug.enabled = enabled;
+	createDebug.humanize = _dereq_('ms');
+
+	Object.keys(env).forEach(key => {
+		createDebug[key] = env[key];
+	});
+
+	/**
+	* Active `debug` instances.
+	*/
+	createDebug.instances = [];
+
+	/**
+	* The currently active debug mode names, and names to skip.
+	*/
+
+	createDebug.names = [];
+	createDebug.skips = [];
+
+	/**
+	* Map of special "%n" handling functions, for the debug "format" argument.
+	*
+	* Valid key names are a single, lower or upper-case letter, i.e. "n" and "N".
+	*/
+	createDebug.formatters = {};
+
+	/**
+	* Selects a color for a debug namespace
+	* @param {String} namespace The namespace string for the for the debug instance to be colored
+	* @return {Number|String} An ANSI color code for the given namespace
+	* @api private
+	*/
+	function selectColor(namespace) {
+		let hash = 0;
+
+		for (let i = 0; i < namespace.length; i++) {
+			hash = ((hash << 5) - hash) + namespace.charCodeAt(i);
+			hash |= 0; // Convert to 32bit integer
+		}
+
+		return createDebug.colors[Math.abs(hash) % createDebug.colors.length];
+	}
+	createDebug.selectColor = selectColor;
+
+	/**
+	* Create a debugger with the given `namespace`.
+	*
+	* @param {String} namespace
+	* @return {Function}
+	* @api public
+	*/
+	function createDebug(namespace) {
+		let prevTime;
+
+		function debug(...args) {
+			// Disabled?
+			if (!debug.enabled) {
+				return;
+			}
+
+			const self = debug;
+
+			// Set `diff` timestamp
+			const curr = Number(new Date());
+			const ms = curr - (prevTime || curr);
+			self.diff = ms;
+			self.prev = prevTime;
+			self.curr = curr;
+			prevTime = curr;
+
+			args[0] = createDebug.coerce(args[0]);
+
+			if (typeof args[0] !== 'string') {
+				// Anything else let's inspect with %O
+				args.unshift('%O');
+			}
+
+			// Apply any `formatters` transformations
+			let index = 0;
+			args[0] = args[0].replace(/%([a-zA-Z%])/g, (match, format) => {
+				// If we encounter an escaped % then don't increase the array index
+				if (match === '%%') {
+					return match;
+				}
+				index++;
+				const formatter = createDebug.formatters[format];
+				if (typeof formatter === 'function') {
+					const val = args[index];
+					match = formatter.call(self, val);
+
+					// Now we need to remove `args[index]` since it's inlined in the `format`
+					args.splice(index, 1);
+					index--;
+				}
+				return match;
+			});
+
+			// Apply env-specific formatting (colors, etc.)
+			createDebug.formatArgs.call(self, args);
+
+			const logFn = self.log || createDebug.log;
+			logFn.apply(self, args);
+		}
+
+		debug.namespace = namespace;
+		debug.enabled = createDebug.enabled(namespace);
+		debug.useColors = createDebug.useColors();
+		debug.color = selectColor(namespace);
+		debug.destroy = destroy;
+		debug.extend = extend;
+		// Debug.formatArgs = formatArgs;
+		// debug.rawLog = rawLog;
+
+		// env-specific initialization logic for debug instances
+		if (typeof createDebug.init === 'function') {
+			createDebug.init(debug);
+		}
+
+		createDebug.instances.push(debug);
+
+		return debug;
+	}
+
+	function destroy() {
+		const index = createDebug.instances.indexOf(this);
+		if (index !== -1) {
+			createDebug.instances.splice(index, 1);
+			return true;
+		}
+		return false;
+	}
+
+	function extend(namespace, delimiter) {
+		const newDebug = createDebug(this.namespace + (typeof delimiter === 'undefined' ? ':' : delimiter) + namespace);
+		newDebug.log = this.log;
+		return newDebug;
+	}
+
+	/**
+	* Enables a debug mode by namespaces. This can include modes
+	* separated by a colon and wildcards.
+	*
+	* @param {String} namespaces
+	* @api public
+	*/
+	function enable(namespaces) {
+		createDebug.save(namespaces);
+
+		createDebug.names = [];
+		createDebug.skips = [];
+
+		let i;
+		const split = (typeof namespaces === 'string' ? namespaces : '').split(/[\s,]+/);
+		const len = split.length;
+
+		for (i = 0; i < len; i++) {
+			if (!split[i]) {
+				// ignore empty strings
+				continue;
+			}
+
+			namespaces = split[i].replace(/\*/g, '.*?');
+
+			if (namespaces[0] === '-') {
+				createDebug.skips.push(new RegExp('^' + namespaces.substr(1) + '$'));
+			} else {
+				createDebug.names.push(new RegExp('^' + namespaces + '$'));
+			}
+		}
+
+		for (i = 0; i < createDebug.instances.length; i++) {
+			const instance = createDebug.instances[i];
+			instance.enabled = createDebug.enabled(instance.namespace);
+		}
+	}
+
+	/**
+	* Disable debug output.
+	*
+	* @return {String} namespaces
+	* @api public
+	*/
+	function disable() {
+		const namespaces = [
+			...createDebug.names.map(toNamespace),
+			...createDebug.skips.map(toNamespace).map(namespace => '-' + namespace)
+		].join(',');
+		createDebug.enable('');
+		return namespaces;
+	}
+
+	/**
+	* Returns true if the given mode name is enabled, false otherwise.
+	*
+	* @param {String} name
+	* @return {Boolean}
+	* @api public
+	*/
+	function enabled(name) {
+		if (name[name.length - 1] === '*') {
+			return true;
+		}
+
+		let i;
+		let len;
+
+		for (i = 0, len = createDebug.skips.length; i < len; i++) {
+			if (createDebug.skips[i].test(name)) {
+				return false;
+			}
+		}
+
+		for (i = 0, len = createDebug.names.length; i < len; i++) {
+			if (createDebug.names[i].test(name)) {
+				return true;
+			}
+		}
+
+		return false;
+	}
+
+	/**
+	* Convert regexp to namespace
+	*
+	* @param {RegExp} regxep
+	* @return {String} namespace
+	* @api private
+	*/
+	function toNamespace(regexp) {
+		return regexp.toString()
+			.substring(2, regexp.toString().length - 2)
+			.replace(/\.\*\?$/, '*');
+	}
+
+	/**
+	* Coerce `val`.
+	*
+	* @param {Mixed} val
+	* @return {Mixed}
+	* @api private
+	*/
+	function coerce(val) {
+		if (val instanceof Error) {
+			return val.stack || val.message;
+		}
+		return val;
+	}
+
+	createDebug.enable(createDebug.load());
+
+	return createDebug;
+}
+
+module.exports = setup;
+
+},{"ms":26}],25:[function(_dereq_,module,exports){
+/*!
+  * domready (c) Dustin Diaz 2014 - License MIT
+  */
+!function (name, definition) {
+
+  if (typeof module != 'undefined') module.exports = definition()
+  else if (typeof define == 'function' && typeof define.amd == 'object') define(definition)
+  else this[name] = definition()
+
+}('domready', function () {
+
+  var fns = [], listener
+    , doc = document
+    , hack = doc.documentElement.doScroll
+    , domContentLoaded = 'DOMContentLoaded'
+    , loaded = (hack ? /^loaded|^c/ : /^loaded|^i|^c/).test(doc.readyState)
+
+
+  if (!loaded)
+  doc.addEventListener(domContentLoaded, listener = function () {
+    doc.removeEventListener(domContentLoaded, listener)
+    loaded = 1
+    while (listener = fns.shift()) listener()
+  })
+
+  return function (fn) {
+    loaded ? setTimeout(fn, 0) : fns.push(fn)
+  }
+
+});
+
+},{}],26:[function(_dereq_,module,exports){
+/**
+ * Helpers.
+ */
+
+var s = 1000;
+var m = s * 60;
+var h = m * 60;
+var d = h * 24;
+var w = d * 7;
+var y = d * 365.25;
+
+/**
+ * Parse or format the given `val`.
+ *
+ * Options:
+ *
+ *  - `long` verbose formatting [false]
+ *
+ * @param {String|Number} val
+ * @param {Object} [options]
+ * @throws {Error} throw an error if val is not a non-empty string or a number
+ * @return {String|Number}
+ * @api public
+ */
+
+module.exports = function(val, options) {
+  options = options || {};
+  var type = typeof val;
+  if (type === 'string' && val.length > 0) {
+    return parse(val);
+  } else if (type === 'number' && isFinite(val)) {
+    return options.long ? fmtLong(val) : fmtShort(val);
+  }
+  throw new Error(
+    'val is not a non-empty string or a valid number. val=' +
+      JSON.stringify(val)
+  );
+};
+
+/**
+ * Parse the given `str` and return milliseconds.
+ *
+ * @param {String} str
+ * @return {Number}
+ * @api private
+ */
+
+function parse(str) {
+  str = String(str);
+  if (str.length > 100) {
+    return;
+  }
+  var match = /^(-?(?:\d+)?\.?\d+) *(milliseconds?|msecs?|ms|seconds?|secs?|s|minutes?|mins?|m|hours?|hrs?|h|days?|d|weeks?|w|years?|yrs?|y)?$/i.exec(
+    str
+  );
+  if (!match) {
+    return;
+  }
+  var n = parseFloat(match[1]);
+  var type = (match[2] || 'ms').toLowerCase();
+  switch (type) {
+    case 'years':
+    case 'year':
+    case 'yrs':
+    case 'yr':
+    case 'y':
+      return n * y;
+    case 'weeks':
+    case 'week':
+    case 'w':
+      return n * w;
+    case 'days':
+    case 'day':
+    case 'd':
+      return n * d;
+    case 'hours':
+    case 'hour':
+    case 'hrs':
+    case 'hr':
+    case 'h':
+      return n * h;
+    case 'minutes':
+    case 'minute':
+    case 'mins':
+    case 'min':
+    case 'm':
+      return n * m;
+    case 'seconds':
+    case 'second':
+    case 'secs':
+    case 'sec':
+    case 's':
+      return n * s;
+    case 'milliseconds':
+    case 'millisecond':
+    case 'msecs':
+    case 'msec':
+    case 'ms':
+      return n;
+    default:
+      return undefined;
+  }
+}
+
+/**
+ * Short format for `ms`.
+ *
+ * @param {Number} ms
+ * @return {String}
+ * @api private
+ */
+
+function fmtShort(ms) {
+  var msAbs = Math.abs(ms);
+  if (msAbs >= d) {
+    return Math.round(ms / d) + 'd';
+  }
+  if (msAbs >= h) {
+    return Math.round(ms / h) + 'h';
+  }
+  if (msAbs >= m) {
+    return Math.round(ms / m) + 'm';
+  }
+  if (msAbs >= s) {
+    return Math.round(ms / s) + 's';
+  }
+  return ms + 'ms';
+}
+
+/**
+ * Long format for `ms`.
+ *
+ * @param {Number} ms
+ * @return {String}
+ * @api private
+ */
+
+function fmtLong(ms) {
+  var msAbs = Math.abs(ms);
+  if (msAbs >= d) {
+    return plural(ms, msAbs, d, 'day');
+  }
+  if (msAbs >= h) {
+    return plural(ms, msAbs, h, 'hour');
+  }
+  if (msAbs >= m) {
+    return plural(ms, msAbs, m, 'minute');
+  }
+  if (msAbs >= s) {
+    return plural(ms, msAbs, s, 'second');
+  }
+  return ms + ' ms';
+}
+
+/**
+ * Pluralization helper.
+ */
+
+function plural(ms, msAbs, n, name) {
+  var isPlural = msAbs >= n * 1.5;
+  return Math.round(ms / n) + ' ' + name + (isPlural ? 's' : '');
+}
+
+},{}],27:[function(_dereq_,module,exports){
+// shim for using process in browser
+var process = module.exports = {};
+
+// cached from whatever global is present so that test runners that stub it
+// don't break things.  But we need to wrap it in a try catch in case it is
+// wrapped in strict mode code which doesn't define any globals.  It's inside a
+// function because try/catches deoptimize in certain engines.
+
+var cachedSetTimeout;
+var cachedClearTimeout;
+
+function defaultSetTimout() {
+    throw new Error('setTimeout has not been defined');
+}
+function defaultClearTimeout () {
+    throw new Error('clearTimeout has not been defined');
+}
+(function () {
+    try {
+        if (typeof setTimeout === 'function') {
+            cachedSetTimeout = setTimeout;
+        } else {
+            cachedSetTimeout = defaultSetTimout;
+        }
+    } catch (e) {
+        cachedSetTimeout = defaultSetTimout;
+    }
+    try {
+        if (typeof clearTimeout === 'function') {
+            cachedClearTimeout = clearTimeout;
+        } else {
+            cachedClearTimeout = defaultClearTimeout;
+        }
+    } catch (e) {
+        cachedClearTimeout = defaultClearTimeout;
+    }
+} ())
+function runTimeout(fun) {
+    if (cachedSetTimeout === setTimeout) {
+        //normal enviroments in sane situations
+        return setTimeout(fun, 0);
+    }
+    // if setTimeout wasn't available but was latter defined
+    if ((cachedSetTimeout === defaultSetTimout || !cachedSetTimeout) && setTimeout) {
+        cachedSetTimeout = setTimeout;
+        return setTimeout(fun, 0);
+    }
+    try {
+        // when when somebody has screwed with setTimeout but no I.E. maddness
+        return cachedSetTimeout(fun, 0);
+    } catch(e){
+        try {
+            // When we are in I.E. but the script has been evaled so I.E. doesn't trust the global object when called normally
+            return cachedSetTimeout.call(null, fun, 0);
+        } catch(e){
+            // same as above but when it's a version of I.E. that must have the global object for 'this', hopfully our context correct otherwise it will throw a global error
+            return cachedSetTimeout.call(this, fun, 0);
+        }
+    }
+
+
+}
+function runClearTimeout(marker) {
+    if (cachedClearTimeout === clearTimeout) {
+        //normal enviroments in sane situations
+        return clearTimeout(marker);
+    }
+    // if clearTimeout wasn't available but was latter defined
+    if ((cachedClearTimeout === defaultClearTimeout || !cachedClearTimeout) && clearTimeout) {
+        cachedClearTimeout = clearTimeout;
+        return clearTimeout(marker);
+    }
+    try {
+        // when when somebody has screwed with setTimeout but no I.E. maddness
+        return cachedClearTimeout(marker);
+    } catch (e){
+        try {
+            // When we are in I.E. but the script has been evaled so I.E. doesn't  trust the global object when called normally
+            return cachedClearTimeout.call(null, marker);
+        } catch (e){
+            // same as above but when it's a version of I.E. that must have the global object for 'this', hopfully our context correct otherwise it will throw a global error.
+            // Some versions of I.E. have different rules for clearTimeout vs setTimeout
+            return cachedClearTimeout.call(this, marker);
+        }
+    }
+
+
+
+}
+var queue = [];
+var draining = false;
+var currentQueue;
+var queueIndex = -1;
+
+function cleanUpNextTick() {
+    if (!draining || !currentQueue) {
+        return;
+    }
+    draining = false;
+    if (currentQueue.length) {
+        queue = currentQueue.concat(queue);
+    } else {
+        queueIndex = -1;
+    }
+    if (queue.length) {
+        drainQueue();
+    }
+}
+
+function drainQueue() {
+    if (draining) {
+        return;
+    }
+    var timeout = runTimeout(cleanUpNextTick);
+    draining = true;
+
+    var len = queue.length;
+    while(len) {
+        currentQueue = queue;
+        queue = [];
+        while (++queueIndex < len) {
+            if (currentQueue) {
+                currentQueue[queueIndex].run();
+            }
+        }
+        queueIndex = -1;
+        len = queue.length;
+    }
+    currentQueue = null;
+    draining = false;
+    runClearTimeout(timeout);
+}
+
+process.nextTick = function (fun) {
+    var args = new Array(arguments.length - 1);
+    if (arguments.length > 1) {
+        for (var i = 1; i < arguments.length; i++) {
+            args[i - 1] = arguments[i];
+        }
+    }
+    queue.push(new Item(fun, args));
+    if (queue.length === 1 && !draining) {
+        runTimeout(drainQueue);
+    }
+};
+
+// v8 likes predictible objects
+function Item(fun, array) {
+    this.fun = fun;
+    this.array = array;
+}
+Item.prototype.run = function () {
+    this.fun.apply(null, this.array);
+};
+process.title = 'browser';
+process.browser = true;
+process.env = {};
+process.argv = [];
+process.version = ''; // empty string to avoid regexp issues
+process.versions = {};
+
+function noop() {}
+
+process.on = noop;
+process.addListener = noop;
+process.once = noop;
+process.off = noop;
+process.removeListener = noop;
+process.removeAllListeners = noop;
+process.emit = noop;
+process.prependListener = noop;
+process.prependOnceListener = noop;
+
+process.listeners = function (name) { return [] }
+
+process.binding = function (name) {
+    throw new Error('process.binding is not supported');
+};
+
+process.cwd = function () { return '/' };
+process.chdir = function (dir) {
+    throw new Error('process.chdir is not supported');
+};
+process.umask = function() { return 0; };
+
+},{}],28:[function(_dereq_,module,exports){
+void function(root){
+
+  function defaults(options){
+    var options = options || {}
+    var min = options.min
+    var max = options.max
+    var integer = options.integer || false
+    if ( min == null && max == null ) {
+      min = 0
+      max = 1
+    } else if ( min == null ) {
+      min = max - 1
+    } else if ( max == null ) {
+      max = min + 1
+    }
+    if ( max < min ) throw new Error('invalid options, max must be >= min')
+    return {
+      min:     min
+    , max:     max
+    , integer: integer
+    }
+  }
+
+  function random(options){
+    options = defaults(options)
+    if ( options.max === options.min ) return options.min
+    var r = Math.random() * (options.max - options.min + Number(!!options.integer)) + options.min
+    return options.integer ? Math.floor(r) : r
+  }
+
+  function generator(options){
+    options = defaults(options)
+    return function(min, max, integer){
+      options.min     = min != null ? min : options.min
+      options.max     = max != null ? max : options.max
+      options.integer = integer != null ? integer : options.integer
+      return random(options)
+    }
+  }
+
+  module.exports =  random
+  module.exports.generator = generator
+  module.exports.defaults = defaults
+}(this)
+
+},{}],29:[function(_dereq_,module,exports){
+(function (setImmediate,clearImmediate){
+var nextTick = _dereq_('process/browser.js').nextTick;
+var apply = Function.prototype.apply;
+var slice = Array.prototype.slice;
+var immediateIds = {};
+var nextImmediateId = 0;
+
+// DOM APIs, for completeness
+
+exports.setTimeout = function() {
+  return new Timeout(apply.call(setTimeout, window, arguments), clearTimeout);
+};
+exports.setInterval = function() {
+  return new Timeout(apply.call(setInterval, window, arguments), clearInterval);
+};
+exports.clearTimeout =
+exports.clearInterval = function(timeout) { timeout.close(); };
+
+function Timeout(id, clearFn) {
+  this._id = id;
+  this._clearFn = clearFn;
+}
+Timeout.prototype.unref = Timeout.prototype.ref = function() {};
+Timeout.prototype.close = function() {
+  this._clearFn.call(window, this._id);
+};
+
+// Does not start the time, just sets up the members needed.
+exports.enroll = function(item, msecs) {
+  clearTimeout(item._idleTimeoutId);
+  item._idleTimeout = msecs;
+};
+
+exports.unenroll = function(item) {
+  clearTimeout(item._idleTimeoutId);
+  item._idleTimeout = -1;
+};
+
+exports._unrefActive = exports.active = function(item) {
+  clearTimeout(item._idleTimeoutId);
+
+  var msecs = item._idleTimeout;
+  if (msecs >= 0) {
+    item._idleTimeoutId = setTimeout(function onTimeout() {
+      if (item._onTimeout)
+        item._onTimeout();
+    }, msecs);
+  }
+};
+
+// That's not how node.js implements it but the exposed api is the same.
+exports.setImmediate = typeof setImmediate === "function" ? setImmediate : function(fn) {
+  var id = nextImmediateId++;
+  var args = arguments.length < 2 ? false : slice.call(arguments, 1);
+
+  immediateIds[id] = true;
+
+  nextTick(function onNextTick() {
+    if (immediateIds[id]) {
+      // fn.call() is faster so we optimize for the common use-case
+      // @see http://jsperf.com/call-apply-segu
+      if (args) {
+        fn.apply(null, args);
+      } else {
+        fn.call(null);
+      }
+      // Prevent ids from leaking
+      exports.clearImmediate(id);
+    }
+  });
+
+  return id;
+};
+
+exports.clearImmediate = typeof clearImmediate === "function" ? clearImmediate : function(id) {
+  delete immediateIds[id];
+};
+}).call(this,_dereq_("timers").setImmediate,_dereq_("timers").clearImmediate)
+},{"process/browser.js":27,"timers":29}],30:[function(_dereq_,module,exports){
+module.exports =
+{
+	EventTarget : _dereq_('./lib/EventTarget'),
+	Event       : _dereq_('./lib/Event')
+};
+
+},{"./lib/Event":31,"./lib/EventTarget":32}],31:[function(_dereq_,module,exports){
+(function (global){
+/**
+ * In browsers export the native Event interface.
+ */
+
+module.exports = global.Event;
+
+}).call(this,typeof global !== "undefined" ? global : typeof self !== "undefined" ? self : typeof window !== "undefined" ? window : {})
+},{}],32:[function(_dereq_,module,exports){
+function yaetiEventTarget()
+{
+	this._listeners = {};
+}
+
+Object.defineProperties(yaetiEventTarget.prototype,
+	{
+		listeners:
+		{
+			get: function()
+			{
+				return this._listeners;
+			}
+		}
+	});
+
+yaetiEventTarget.prototype.addEventListener = function(type, newListener)
+{
+	var listenersType;
+	var i;
+	var listener;
+
+	if (!type || !newListener)
+		return;
+
+	listenersType = this._listeners[type];
+
+	if (listenersType === undefined)
+		this._listeners[type] = listenersType = [];
+
+	for (i = 0; !!(listener = listenersType[i]); i++)
+	{
+		if (listener === newListener)
+			return;
+	}
+
+	listenersType.push(newListener);
+};
+
+yaetiEventTarget.prototype.removeEventListener = function(type, oldListener)
+{
+	var listenersType;
+	var i;
+	var listener;
+
+	if (!type || !oldListener)
+		return;
+
+	listenersType = this._listeners[type];
+
+	if (listenersType === undefined)
+		return;
+
+	for (i = 0; !!(listener = listenersType[i]); i++)
+	{
+		if (listener === oldListener)
+		{
+			listenersType.splice(i, 1);
+			break;
+		}
+	}
+
+	if (listenersType.length === 0)
+		delete this._listeners[type];
+};
+
+yaetiEventTarget.prototype.dispatchEvent = function(event)
+{
+	var type;
+	var listenersType;
+	var dummyListener;
+	var stopImmediatePropagation = false;
+	var i;
+	var listener;
+
+	if (!event || typeof event.type !== 'string')
+		throw new Error('`event` must have a valid `type` property');
+
+	// Do some stuff to emulate DOM Event behavior (just if this is not a
+	// DOM Event object).
+	if (event._yaeti)
+	{
+		event.target = this;
+		event.cancelable = true;
+	}
+
+	// Attempt to override the stopImmediatePropagation() method.
+	try
+	{
+		event.stopImmediatePropagation = function()
+		{
+			stopImmediatePropagation = true;
+		};
+	}
+	catch (error)
+	{}
+
+	type = event.type;
+	listenersType = (this._listeners[type] || []);
+
+	dummyListener = this['on' + type];
+
+	if (typeof dummyListener === 'function')
+	{
+		try
+		{
+			dummyListener.call(this, event);
+		}
+		catch (error)
+		{
+			console.error(error);
+		}
+	}
+
+	for (i = 0; !!(listener = listenersType[i]); i++)
+	{
+		if (stopImmediatePropagation)
+			break;
+
+		try
+		{
+			listener.call(this, event);
+		}
+		catch (error)
+		{
+			console.error(error);
+		}
+	}
+
+	return !event.defaultPrevented;
+};
+
+module.exports = yaetiEventTarget;
+
+},{}]},{},[21])(21)
+});