Skip to content
New issue

Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.

Sign up for GitHub

By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.

Already on GitHub? Sign in to your account

T.140 support in SIP plugin, and WebRTC gateway (again! see #1898) #3231

Open
wants to merge 4 commits into
base: master
Choose a base branch
from
Open

T.140 support in SIP plugin, and WebRTC gateway (again! see #1898) #3231

Show file tree
Hide file tree
Changes from all commits
Commits
Show all changes
4 commits
Select commit Hold shift + click to select a range
6558e8d
T.140 support in SIP plugin, and WebRTC gateway (again! see #1898)
lminiero Jun 6, 2023
15bfe53
Aligned with latest changes
lminiero Jul 13, 2023
58e2b4e
Aligned to latest changes
lminiero Nov 16, 2023
0aa8784
Fixed conflict and aligned with latest changes
lminiero Feb 6, 2024
File filter

Filter by extension

Filter by extension
Conversations
Failed to load comments.
Loading
Jump to
Jump to file
Failed to load files.
Loading
Diff view
Diff view
6 changes: 6 additions & 0 deletions conf/janus.plugin.sip.jcfg.sample
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -40,6 +40,12 @@ general: {
#dscp_audio_rtp = 46
#dscp_video_rtp = 26

# If you want to allow the plugin to also negotiate Real-Time Text (RTT)
# via T.140 and datachannels on the WebRTC side, you need to set the
# 'allow_t140' property to true: by default, the property is disabled,
# meaning that attempts to negotiate RTT on either side will be rejected.
#allow_t140 = true

# In case you want to use SIPS for some sessions, Sofia may need to
# have access to a certificate to use: this is especially true for
# Sofia >= 1.13, which will fail to create the agent if no certificate
Expand Down
Loading
Loading