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

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

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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
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6 changes: 6 additions & 0 deletions conf/janus.plugin.sip.jcfg.sample
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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
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