<div dir="ltr">Please look one more example o SDP with rtcp-mux<div><br></div><div><pre style="word-wrap:break-word;white-space:pre-wrap"> ------------------------------------------------------------------------
INVITE <a href="http://sip:3000@pbx.rcsnet.ru:5060">sip:3000@pbx.rcsnet.ru:5060</a> SIP/2.0
Via: SIP/2.0/UDP 91.103.196.12;rport;branch=z9hG4bK32SrX56KHZgya
Max-Forwards: 70
From: "" <<a href="mailto:sip%3A0000000000@91.103.196.12">sip:0000000000@91.103.196.12</a>>;tag=3Z0mjUyp1matN
To: <<a href="http://sip:3000@pbx.rcsnet.ru:5060">sip:3000@pbx.rcsnet.ru:5060</a>>
Call-ID: e3be7793-cd5a-1235-d195-005056be15c6
CSeq: 108480740 INVITE
Contact: <<a href="http://sip:mod_sofia@91.103.196.12:5060">sip:mod_sofia@91.103.196.12:5060</a>>
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: <<a href="mailto:sip%3A0000000000@91.103.196.12">sip:0000000000@91.103.196.12</a>>;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
<font color="#ff0000" size="4">a=rtcp-mux</font>
<font size="4" color="#ff0000">a=rtcp:25638 IN IP4 91.103.196.12</font>
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
------------------------------------------------------------------------</pre></div><div><br></div><div>Log <a href="https://freeswitch.org/jira/secure/attachment/26623/originate_log.txt">https://freeswitch.org/jira/secure/attachment/26623/originate_log.txt</a><br></div><div>Ticket <a href="https://freeswitch.org/jira/browse/FS-10400">https://freeswitch.org/jira/browse/FS-10400</a></div><div><br></div><div>Sergey</div><div><br></div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr">сб, 14 окт. 2017 г. в 0:13, Yuriy Gorlichenko <<a href="mailto:ovoshlook@gmail.com">ovoshlook@gmail.com</a>>:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Sorry:<br>small fix<br>webRTC clients accepts <br>a=rtcp:<port><br>but port suppose should be same with <br>m=audio</div><div class="gmail_extra"><br><div class="gmail_quote">2017-10-13 22:58 GMT+03:00 Yuriy Gorlichenko <span dir="ltr"><<a href="mailto:ovoshlook@gmail.com" target="_blank">ovoshlook@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi all!<br>Some time ago Chromium browser sets rtcpMuxPolicy: required by default (soon on Chrome 58)<br>It means that webRTC based clients not accepts <br>a=rtcp:31757<br>And uses for RTP and RTCP multiplexing at one port<br><br>Main trouble that i found: calls between original SIP client and webRTC client (SIP client is initiator of call)<br><br>When sip client sends invite it has<br>a=rtcp:33445<br>Means it wants 2 different prots for RTCP and RTP<br><br>As expected for this case webRTC client says 488 Not accessible here instead of 200 resonse<br><br>I suppose rtpengine module should hept to handle it but i can not find any key how to do it<br><br>I added form rtpengine_manage()<br>rtcp-mux-offer and rtcp-mux-accept but it only adds "a=rtcp-mux"<br>But not removes a=rtcp and ice cadidate prepeared for it.<br><br>Suppose removing a=rtcp:12345 will gives just an issue for RTP session.<br><br>Does rtpengine module have some keys for resole this issue?<br><br><br> <br><br></div>
</blockquote></div><br></div>
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