Looks like here is FreeSWITH fixes it from it's side with rtsp-mux=true.
As i said above - a=rtcp:<port> should present but should use same port with m=audio.
I know that freeswith allows rtcp-mux, but asterisk, for example does not.
Thats why im asking how to handle this issue with rtpengine if it possible.
On Oct 14, 2017 11:02 AM, "Sergey Safarov" s.safarov@gmail.com wrote:
Please look one more example o SDP with rtcp-mux
INVITE sip:3000@pbx.rcsnet.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 91.103.196.12;rport;branch=z9hG4bK32SrX56KHZgya Max-Forwards: 70 From: "" sip:0000000000@91.103.196.12;tag=3Z0mjUyp1matN To: sip:3000@pbx.rcsnet.ru:5060 Call-ID: e3be7793-cd5a-1235-d195-005056be15c6 CSeq: 108480740 INVITE Contact: sip:mod_sofia@91.103.196.12:5060 User-Agent: FreeSWITCH-mod_sofia/1.9.0+git~20170615T144716Z~5f5fb33ea9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 670 X-FS-Support: update_display,send_info Remote-Party-ID: sip:0000000000@91.103.196.12;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1497607971 1497607972 IN IP4 91.103.196.12 s=FreeSWITCH c=IN IP4 91.103.196.12 t=0 0 m=audio 25638 RTP/AVP 102 9 0 8 104 101 a=rtpmap:102 opus/48000/2 a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:104 telephone-event/48000 a=fmtp:104 0-16 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-mux a=rtcp:25638 IN IP4 91.103.196.12 a=ptime:20 m=video 23108 RTP/AVP 103 b=AS:1024 a=rtpmap:103 VP8/90000 a=rtcp-fb:103 ccm fir a=rtcp-fb:103 ccm tmmbr a=rtcp-fb:103 nack a=rtcp-fb:103 nack pli
Log https://freeswitch.org/jira/secure/attachment/26623/originate_log.txt Ticket https://freeswitch.org/jira/browse/FS-10400
Sergey
сб, 14 окт. 2017 г. в 0:13, Yuriy Gorlichenko ovoshlook@gmail.com:
Sorry: small fix webRTC clients accepts a=rtcp:<port> but port suppose should be same with m=audio
2017-10-13 22:58 GMT+03:00 Yuriy Gorlichenko ovoshlook@gmail.com:
Hi all! Some time ago Chromium browser sets rtcpMuxPolicy: required by default (soon on Chrome 58) It means that webRTC based clients not accepts a=rtcp:31757 And uses for RTP and RTCP multiplexing at one port
Main trouble that i found: calls between original SIP client and webRTC client (SIP client is initiator of call)
When sip client sends invite it has a=rtcp:33445 Means it wants 2 different prots for RTCP and RTP
As expected for this case webRTC client says 488 Not accessible here instead of 200 resonse
I suppose rtpengine module should hept to handle it but i can not find any key how to do it
I added form rtpengine_manage() rtcp-mux-offer and rtcp-mux-accept but it only adds "a=rtcp-mux" But not removes a=rtcp and ice cadidate prepeared for it.
Suppose removing a=rtcp:12345 will gives just an issue for RTP session.
Does rtpengine module have some keys for resole this issue?
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