Releases: bluenviron/mediamtx
v1.0.1
Fixes and improvements
General
- print warning when the write queue is full (#2251)
- limit logging of decode errors (#2253)
- fix maxReaders limit in case of multiple tracks (#2246) (#2264)
Codecs
- h264, h265: raise MaxAccessUnitSize to 8 MiB (bluenviron/mediacommon#59) by @database64128
- Raise maximum NALU count (bluenviron/mediacommon#64) by @milaq
- h264: add limit on maximum number of reordered frames (bluenviron/mediacommon#65)
- mpegts: make 'PTS is missing' a decode error (bluenviron/mediacommon#67)
- h264: support frame_mbs_only_flag = 0 (bluenviron/mediacommon#68)
- av1: fix parsing sequence headers from libsvtav1 (bluenviron/mediacommon#69)
RTSP
- add Speex format
- decode RTP time globally
- emit a decode error in case of packets with wrong SSRC
- allow publishers to set the title of the stream (#979)
- support routing ULPFEC group definitions
- client: support server-sent requests (bluenviron/gortsplib#93) (bluenviron/gortsplib#378)
- client: stop main routine immediately in case of a read error (bluenviron/gortsplib#379)
- re-enable consistency checks on clock rate of tracks (bluenviron/gortsplib#382)
- discard invalid video tracks (bluenviron/gortsplib#381) (bluenviron/gortsplib#383)
- ringbuffer: when buffer is full, preserve queued data (bluenviron/gortsplib#386)
- ringbuffer: discard pending data when buffer is closed (bluenviron/gortsplib#387)
RTMP
- allow RTMP streaming with codecid=av01 or hvc1 (#2232) by @ph0b
- support publishing AV1/H265 with OBS 30 (#2217) (#2234)
- support publishing VP9 tracks with RTMP (#2247)
- fix conversion of AV1/VP9 tracks from HLS/RTMP to RTSP (#2263)
- support ingesting RTMPE streams (#2189)
- add limit on message body size (#2252)
HLS
- embed hls.js into the server (#2202) (#2236)
- bump hls-js to v1.4.10 (#2239)
- fix conversion of AV1/VP9 tracks from HLS/RTMP to RTSP (#2263)
- fix wrong protocol sent to external authentication server (#2213)
- hls source: fix formatting debug log messages (#2243)
- return 404 when requesting hls.min.js.map (#2262)
Dependencies
- build(deps): bump github.com/pion/webrtc/v3 from 3.2.14 to 3.2.15 (#2216)
- build(deps): bump github.com/pion/webrtc/v3 from 3.2.15 to 3.2.16 (#2220)
- build(deps): bump github.com/google/uuid from 1.3.0 to 1.3.1 (#2228)
- build(deps): bump github.com/pion/webrtc/v3 from 3.2.16 to 3.2.17 (#2229)
- build(deps): bump github.com/asticode/go-astits from 1.12.0 to 1.13.0 (bluenviron/mediacommon#56)
v1.0.0
Why 1.0?
This software now supports all the main streaming protocols (SRT / WebRTC / RTSP / RTMP / LL-HLS), a wide range of codecs, a series of innovative protocol-codec combinations (for instance HLS + AV1), and is deployed in production environments. The main objective of the project has been achieved, that is to provide a routing solution for real-time media streams to any user, from householders that want to manage their video feeds to developers that need to route media streams to and from microservices.
There are a couple of secondary features that will be certainly developed in the near future (native recording, native scalability, both can already be achieved by using external integrations) but other than that the focus will be on fixing eventual issues related to the existing features.
New features
SRT
- support publishing, reading, proxying with SRT (#2068)
WebRTC
- support proxying WebRTC streams with WHEP (#2142)
HLS
UDP
- support reading MPEG-1 tracks (#2147)
General
Fixes and improvements
RTSP
- support G726 format (bluenviron/gortsplib#330)
- fix race condition in WritePacketRTP() (bluenviron/gortsplib#334)
- fix SDP unmarshaling with Vurix NVR (#2128)
- add VP8/VP9 limits
HLS
- show IP in logs in case of failed authentication (#2099)
- prevent brute-force attacks by waiting before sending responses (#2100)
- reply status code 204 to OPTIONS requests (#2141)
- prefer Opus tracks to MPEG-4 tracks (#2158)
- fix parsing decimal EXT-X-TARGETDURATION (bluenviron/gohlslib#55)
- fix parsing EXT-X-STREAM-INF with spaces (bluenviron/gohlslib#56)
- fix parsing playlists without trailing newline (bluenviron/gohlslib#58)
- add Cache-Control header to all responses
- prepend prefix to segments. . This is needed to prevent usage of cached segments from previous muxing sessions
WebRTC
- show both IP and port during session creation and in API (#2096)
- send session ID to external auth server (#1981) (#2098)
- show IP in logs in case of failed authentication (#2099)
- prevent brute-force attacks by waiting before sending responses (#2100)
- speed up track detection (#2105)
- fix race condition when broadcasting RTP packets (#2117)
- reply status code 204 to OPTIONS requests (#2141)
UDP
- support using domain names instead of IPs (#2150)
API
- fix crash when calling /v1/webrtcsessions/list just after session creation (#2097)
- add transport to RTSP sessions (#2151)
- remove sourceReady from docs (#2163)
General
- return an error in case the random number generator fails (#2120)
- remove warning when decoding VP8 or VP9 (#2159). . avoid printing 'received a non-starting fragment without any previous starting fragment'
- disable check for missing key frames (#1904) (#2161)
- rename disablePublisherOverride into overridePublisher (#2164)
- remove 'disable' from names of configuration parameters (#2101)
- fix crash in case of specially-crafted HTTP requests (#2166) (#2169)
- Add video player options via query string (#2145)
- mpegts: fix panic with specially-crafted strings; add fuzzing (bluenviron/mediacommon#29)
- h264, h265: raise MaxNALUSize (bluenviron/mediacommon#30)
- h264, h265: rename MaxNALUSize to MaxAccessUnitSize and apply to entire access unit (bluenviron/mediacommon#36)
- h264: fix 'invalid POC' error (bluenviron/mediacommon#55)
Dependencies
- build(deps): bump github.com/pion/rtp from 1.7.13 to 1.8.0 (#2091)
- build(deps): bump github.com/pion/webrtc/v3 from 3.2.12 to 3.2.13 (#2092)
- build(deps): bump github.com/gookit/color from 1.5.3 to 1.5.4 (#2089)
- build(deps): bump github.com/abema/go-mp4 from 0.10.1 to 0.11.0 (#2112)
- build(deps): bump github.com/pion/rtp from 1.8.0 to 1.8.1 (#2129)
- build(deps): bump golang.org/x/net from 0.12.0 to 0.13.0 (#2139)
- build(deps): bump github.com/pion/webrtc/v3 from 3.2.13 to 3.2.14 (#2140)
- build(deps): bump golang.org/x/term from 0.10.0 to 0.11.0 (#2148)
- build(deps): bump golang.org/x/net from 0.13.0 to 0.14.0 (#2170)
- build(deps): bump github.com/pion/ice/v2 from 2.3.9 to 2.3.10 (#2171)
- build(deps): bump github.com/asticode/go-astits
v0.23.8
Fixes and improvements
General
- hls, webrtc: add Authorization to Access-Control-Allow-Headers (#2018) (#2020)
- stop execution in case of panics when handling HTTP requests (#2021)
- disable colored log lines when output is not a terminal (#1477) (#2050)
- make sure components are closed in a specific order (#2065)
- update list of supported codecs inside error messages (#2058) (#2073)
WebRTC
- allow removing default WebRTC ICE server with environment variables (#2064)
- fix race condition that caused random crashes during handshake (#2072)
- fix memory leak during shutdown or session kick (#2079)
- display publish-related errors in web page (#1836) (#2080)
Raspberry Pi Camera
- fix a compile error with recent libcamera (#2081) by @hideaki-t
- rpi camera: add rpiCameraHDR parameter (#1876) (#2083)
API
- add path to RTMP connections, RTSP sessions, WebRTC sessions (#1962) (#2022)
- apidocs: fix source/reader types (#2027)
- fix error in case of nested paths (#2040) by @Jordy84
- return 404 when a path configuration is not found (#2067) (#2074)
- allow to edit properties of path config "all" (#2067) (#2075)
- add 'readyTime' to paths (#2049) (#2082)
Dependencies
v0.23.7
Fixes and improvements
General
- set Access-Control-Allow-Headers to a static string (#1973)
WebRTC
- do not pass preflight requests to external auth (#1941) (#1972)
- in the web page, pass query parameters to inner requests (#1976)
- fix memory leak when publishing or reading (#1884) (#1983)
- fix bitrate not being applied (#1984)
- forbid publishing zero tracks (#1991)
- allow setting Opus bitrate (#1908) (#1985)
- add option to disable audio effects (#1908) (#1989)
- move codec and bitrate settings on client side (#1990)
- support publishing with OBS and WebRTC (#1998)
- allow using special characters in ICE server credentials (#1953) (#2000)
RTSP
- generate RTCP receiver reports even before receiving RTCP sender reports (bluenviron/gortsplib#318). . (#1739)
HLS
- in the web page, pass query parameters to inner requests (#1976)
RPI Camera
API
Dependencies
- bump github.com/pion/webrtc/v3 from 3.2.10 to 3.2.11 (#2002)
v0.23.6
Fixes and improvements
General
- fix 'runOnDemandRestart: yes' (#1947)
- rename environment variable RTSP_PATH into MTX_PATH (#1967)
- add Arch Linux package to the README (#1957) (#1969)
WebRTC
- make preflight OPTIONS requests work with external auth (#1941) (#1964)
- fix using inline credentials in URLs (#1919) (#1966)
RTSP
- return an error in case of invalid packet (bluenviron/gortsplib#305). . when reading with TCP and packet has an unknown format.
- allow using odd interleaved IDs (bluenviron/gortsplib#304) (#1762)
- client: support URLs with IPv6 and no port (bluenviron/gortsplib#313) (bluenviron/gortsplib#316)
HLS
- support ISO8601 dates (bluenviron/gohlslib#50) (#1958)
Dependencies
v0.23.5
Fixes and improvements
-
Add Docker images with FFmpeg included and change docker repository name (#1771) (#1909) (#1923). Available images now are:
name FFmpeg included RPI Camera support bluenviron/mediamtx:latest ❌ ❌ bluenviron/mediamtx:latest-ffmpeg ✔️ ❌ bluenviron/mediamtx:latest-rpi ❌ ✔️ bluenviron/mediamtx:latest-ffmpeg-rpi ✔️ ✔️ -
return an error in case default configuration file can't be opened (#1920)
-
bump github.com/pion/webrtc/v3 from 3.2.8 to 3.2.9 (#1906)
v0.23.4
Fixes and improvements
General
- allow using special characters in external commands (#1652) (#1868), on Windows, when using cmd.exe or a bat file as external command.
- print error that caused an external command to exit (#1869)
- fix sending session ID to external authentication (#1871). this fixes a regression introduced in v0.23.0
- replace math/rand with crypto/rand, increasing security (#1872)
WebRTC
- Add WebRTC stream id to whep response headers (#1879) by @vvitkovsky
RTSP
- support otherDataPresent and crcCheckPresent inside StreamMuxConfig of MP4A-LATM format (bluenviron/mediacommon#18) (#1881)
- fix using multicast when a single interface doesn't support it (bluenviron/gortsplib#293) (https://github.com/bluenviron/mediamtx/discussions/1744)
RTMP
- fix timestamp conversion from RTSP/HLS to RTMP (#1899). this was causing moments of silence and timing errors when reading with RTMP a stream originally published with RTSP or HLS.
- support reading MP4A-LATM-encoded AAC with RTMP and HLS (#1694) (#1898)
HLS
UDP
- fix using multicast when a single interface doesn't support it (#1874)
RPI Camera
API
- fix setting default parameters when creating a path (#1853) (#1905). . this fixes a regression introduced in v0.23.0.
Dependencies
v0.23.3
Fixes and improvements
WebRTC
- fix WHIP/WHEP implementation (#1857) (#1861). Offers and answers are now encoded in SDP in place of JSON; Location header is sent to clients. This fixes compatibility with GStreamer and whipsink.
HLS
- fix parsing multivariant playlists with windows-style newlines (bluenviron/gohlslib#43)
Dependencies
v0.23.2
RTSP
- fix sending keepalives to sources (#1812) (bluenviron/gortsplib#287). This fixes a regression introduced in v0.23.0
WebRTC
- Expose E-Tag, Accept-Patch and Link headers for cross-origin WHIP/WHEP requests (#1841)
- add POST and PATCH methods to Access-Control-Allow-Methods (#1848)
API
- allow using paths/list endpoint when a path is being deleted (#1849)
v0.23.1
Note
This release is mostly dedicated to fix regressions introduced in v0.23.0. Major releases allow to introduce new features, but every addition is linked the risk of breaking something. Thanks to all the people that immediately tested the new version, reported issues and allowed to quickly develop the fixes included in this release: @tbnguyen1407, @alpe12, @felder, @nigelsim, @blu006, @Tokolosh, @0ip
Fixes and improvements
WebRTC
- fix exception in browser when webrtcICEServers is empty (#1817) (#1821). this fixes a regression introduced in v0.23.0.
RTMP
- fix crash when publishing video-only tracks (#1819) (#1822). this fixes a regression introduced in v0.23.0.
HLS
API
- bump API prefix from /v1 to /v2 (#1815)
- add /get endpoints (#1577) (#1823). this allows to get entities by ID or name after /list endpoints were changed in v0.23.0.
- fix wrong pageCount in /list endpoints (#1813) (#1824)
- add item count to /list endpoints (#1813) (#1829)
- sort results of /list endpoints (#1828)
General
- fix missing H264/H265 keyframe warning message (#1825)