[SR-Users] rtcp-mux policy for RTPengine module

[Sergey Safarov]

Please look one more example o SDP with rtcp-mux

   ------------------------------------------------------------------------
   INVITE sip:3000 at 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 at 91.103.196.12>;tag=3Z0mjUyp1matN
   To: <sip:3000 at pbx.rcsnet.ru:5060>
   Call-ID: e3be7793-cd5a-1235-d195-005056be15c6
   CSeq: 108480740 INVITE
   Contact: <sip:mod_sofia at 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 at 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 at 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 at 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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