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https://support.digium.com/community/s/article/How-does-asterisk-and-Switchvox-choose-RTP-ports
Sep 06, 2016 · Switchvox or asterisk has no obligation or requirement to ask the network which ports are open. The PBX is always going to look at the range that is setup on its configuration and choose a port(s) within that range, If the port happens to be blocked at the firewall level, the customer will experience a call without any audio (or video).
https://blogs.asterisk.org/2017/04/26/rtcp-mux-webrtc/
rtcp-mux in Asterisk. To get around this problem, the Asterisk team decided to add support for rtcp-mux into Asterisk before it became too late. I added support for rtcp-mux for chan_pjsip, and Sean Bright added rtcp-mux for chan_sip. The feature is available starting in Asterisk 13.15.0 and Asterisk …
https://wiki.asterisk.org/wiki/display/AST/RTP+task+list
Mar 05, 2015 · There is nothing that attempts to modify the RTCP transmission interval, and there is no code to parse the new RTCP packe types defined by RFC 4585. Any work done in this section will be breaking new ground in Asterisk's support of RTP/AVPF. Create RTP/AVPF RTCP packet decoders.
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Jan 31, 2018 · The WebRTC support in Asterisk has evolved and improved over time (in particular with Asterisk 15) but has not yet fully ventured into the user experience area. The two most important areas of this are the handling of lost or out of order packets and bandwidth management.
http://forums.asterisk.org/viewtopic.php?t=83682
I have used asterisk for years now and every since 1.6, I have used the "rtcp set stats on" in order to watch for issues and monitor quality with calls. I have recently had a couple builds with 1.8.8 and even tried 1.8.13 and the command seems to execute, but nothing is displayed on the CLI at the completion of a call like it normally does.
https://tools.ietf.org/html/rfc3605
RFC 3605 RTCP attribute in SDP October 2003 states that "other ports used by the media application (such as the RTCP port) should be derived algorithmically from the base media port." RTCP port numbers were necessarily derived from the base media port in older versions of RTP (such as []), but now that this restriction has been lifted, there is a need to specify RTCP ports explicitly in SDP.Cited by: 117
https://www.netlab.tkk.fi/opetus/s38130/k99/presentations/4.pdf
accompanying RTP Control Protocol (RTCP) provides feedback of the quality of the data delivery and information about session participants. A RTP session usually is composed of a RTP port number (UDP port), a RTCP port number (consecutive UDP port) and the participant's IP address.
https://community.freepbx.org/t/logging-qos-statistics/18686
rtcp events can be logged to Asterisk full then parsed out. That would be much easier than real time. If desiring real time I would use media proxy or something designed to do what you want. My comments are based on Asterisk 1.8 support. Asterisk 11 may have improved RTCP-XR handling.
https://www.3cx.com/PBX/RTCP/
RTCP stands for Real-time Transport Control Protocol and is defined in RFC 3550.RTCP works hand in hand with RTP.RTP does the delivery of the actual data, whereas RTCP is used to send control packets to participants in a call.
https://stackoverflow.com/questions/42688499/sipml5-negotiate-rtcpmuxpolicy
Mar 10, 2017 · In previous versions of chrome rtcp multiplxing set to 'negotiate' Starting from version 57 chrome changed rtcp multiplxing to 'require' Asterisk, as I understand it does not support rtcp multiplexing. webrtc allow RTCRtpMuxPolicy flag options "negotiate" and "require" In Sipml5 API 2.0.3 , as I understood there is no option to set RTCRtpMuxPolicy.
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