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rtcp_audio_interval_msec - FreeSWITCH - Confluence
https://freeswitch.org/confluence/display/FREESWITCH/rtcp_audio_interval_msec#:~:text=rtcp_audio_interval_msec%20integer%20Channel%20variable%20to%20set%20the%20interval,RTCP%20%28packets%20not%20sent%2C%20RTCP%20autoadjust%20failing%2C%20etc.%29.
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
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.
Internal Sip Profile — FusionPBX Docs documentation
https://docs.fusionpbx.com/en/latest/advanced/internal_sip_profile.html
rtcp-audio-interval-msec: 5000: False rtcp-video-interval-msec: 5000: False rtp-autofix-timing: false: False rtp-autoflush-during-bridge false: False rtp-hold-timeout-sec: 1800: True rtp-ip $${local_ip_v4} True rtp-rewrite-timestamps: true: False rtp-timeout-sec: 300: True rtp-timer-name: soft: True send-message-query-on-register True: False
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
more than 60 seconds of silence on one side of the call. As my service is. primarily used for recording incoming messages, this means that any message. being recorded longer than 60 seconds gets cut off. My provider says I need. to configure freeswitch to send rtcp keep-alive packets to prevent them from.
Call drop after 25 seconds - Cisco Community
https://community.cisco.com/t5/collaboration-voice-and-video/call-drop-after-25-seconds/ta-p/3121927
This command configures the average interval between successive RTCP report transmissions for a given voice session. For example, if the value argument is set to 25,000 milliseconds, an RTCP report is sent every 25 seconds, on average. If no RTP packets are received during the calculated interval, the call is disconnected.
RTP Control Protocol - Wikipedia
https://en.wikipedia.org/wiki/RTP_Control_Protocol
The RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). Its basic functionality and packet structure is defined in RFC 3550. RTCP provides out-of-band statistics and control information for an RTP session. It partners with RTP in the delivery and packaging of multimedia data, but does not transport any media data itself.
Freeswitch sending empty RTP in time with receiving RTCP
https://stackoverflow.com/questions/62951971/freeswitch-sending-empty-rtp-in-time-with-receiving-rtcp
FreeSWITCH Version: 1.10.3 I was hoping that someone maybe able to help me. Intermittently on inbound calls from the our sip provider we are sending a empty RTP …
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