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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.
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 14.4.0.
https://blogs.asterisk.org/2018/05/16/receiver-estimated-maximum-bitrate-support/
To support remb an existing frame type called AST_FRAME_RTCP was leveraged for both the reception and sending of remb. When the res_rtp_asterisk module receives remb information it creates an Asterisk frame (just like as if it had received DTMF or audio/video), populates it with the information from the RTCP message, and raises it.
https://reviewboard.asterisk.org/r/3439/
the A=rtcp attribute in SDP points out a different port than the mediaport+1 to receive RTCP on. This patch adds a new api to rtpengine and res_rtp_asterisk and updates chan_sip to use it. This patch needs to be modified to handle the IP address argument too.
http://forums.asterisk.org/viewtopic.php?t=18886
Sep 01, 2011 · I'm Using Asterisk-1.4.13 in Intel(R) Core(TM)2 CPU 6600 @ 2.40GHz. When My call terminate to an SIP gateway I got rtp.c:891 ast_rtcp_read: RTCP Read too short WARNING. And im Ny side I can hear other party's voice but other party couldn't hear me. Using Codec G723.1 or G723. Can Anybody give me any solution of it? Thanks. Suberna
http://downloads.asterisk.org/pub/security/AST-2017-005.html
Resolution. The RTP stack will now only learn a new source address if it has been told to expect the address to change. The RTCP support has now also been updated to drop RTCP reports that are not regarding the RTP session currently in progress.
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://github.com/asterisk/asterisk/blob/master/configs/samples/rtp.conf.sample
Jun 13, 2019 · asterisk / configs / samples / rtp.conf.sample Find file Copy path jcolp res_rtp_asterisk: Add support for DTLS packet fragmentation. a8e5cf5 Jun 13, 2019
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.
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