Freeswitch Rtcp-Audio-Interval-Msec

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rtcp_audio_interval_msec - FreeSWITCH - Confluence

https://freeswitch.org/confluence/display/FREESWITCH/rtcp_audio_interval_msec

rtcp_audio_interval_msec integer Channel variable to set the interval in msec between each RTCP SR packet.. setting the rtp_timer_name to none will create all sorts of undesired side effects with RTCP (packets not sent, RTCP autoadjust failing, etc.).

RTCP - FreeSWITCH - Confluence

https://freeswitch.org/confluence/display/FREESWITCH/RTCP

Only the Sender Report (SR) messages are sent by FreeSWITCH. Discussion. RTCP can be configured on a per-session basis or for the entire SIP profile. Its transmission intervals can be set from between 100 ms to 5000 ms. Per Session. Use the rtcp_audio_interval_msec and rtcp_video_internal_msec channel variables.

Freeswitch inserting silence in RTP stream when …

https://github.com/signalwire/freeswitch/issues/735

After some further investigation, I discovered that rtcp-audio-interval-msec was set on FreeSWITCH 1 in my scenario. Removing this seems to have resolved the problem for me (as we are not using RTCP). However this may still be a bug for users who that are using this feature.

[Freeswitch-users] RTCP Keep Alive issue - hangup after 60 ...

https://freeswitch-users.freeswitch.narkive.com/9sXMijpp/rtcp-keep-alive-issue-hangup-after-60-seconds-of-silence

Sent: 14 January 2011 15:00. To: FreeSWITCH Users Help. Subject: Re: [Freeswitch-users] RTCP Keep Alive issue - hangup after 60. seconds of silence. While this will turn on RTCP your provider needs to be beaten for requiring. such a resource wasting process. /b. On Jan 14, 2011, at 6:49 AM, Christopher Rienzo wrote:

Freeswitch sending empty RTP in time with receiving RTCP

https://stackoverflow.com/questions/62951971/freeswitch-sending-empty-rtp-in-time-with-receiving-rtcp

FreeSWITCH 2 sends all the RTP packets to FreeSWITCH 1 but FreeSWITCH 1 is sending a RTP packet with an empty payload to the SIP provider, on either side of this empty RTP packet there is RTP with the MARKER bit set. Adjusting the below RTCP timers on FreeSWITCH 2 affects the frequency of the problem on FreeSWITCH 1.

Freeswitch透传RTCP消息 bug 的修复._smallbutstrong的专栏 …

https://blog.csdn.net/smallbutstrong/article/details/8451322

在有些视频协议中,采用了payload specific 的feedback RTCP消息来请求关键帧. 而Freeswitch 透传RTCP消息有问题. 首先文档中没有说明有passthru的透传设置. 其设置应该如下. 但是设置完以后, 还是发现很多问题, 首先音频的RTCP包被丢弃了. 其bug如下.将switch_rtp.c 的 …

mod_sofia - FreeSWITCH - Confluence

https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia

parse-all-invite-headers. Type: Boolean. When true, mod_sofia will parse all inbound invite headers and set variables with the values of them. Some custom variables are prepended with "X-" to differentiate them from standard SIP headers. For more information search sofia.c for the string " un_name " to see how variables are built.

mod sofia - FreeSWITCH

https://freeswitch.org/confluence/spaces/flyingpdf/pdfpageexport.action?pageId=1048707

[Freeswitch-users] RTCP FIR on floor change/new member join

https://freeswitch-users.freeswitch.narkive.com/6XvrYmIA/rtcp-fir-on-floor-change-new-member-join

BTW, I am not even sure if I have configured the RTCP in Freeswitch correctly. What I did was to uncommented "rtcp-video-interval-msec" and "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see in Chrome's log: webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No …

sip_local_sdp_str - FreeSWITCH - Confluence

https://freeswitch.org/confluence/display/FREESWITCH/sip_local_sdp_str

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