Sip.js: Multiplexing RTCP required in Chrome Canary
Per this announcement:
As of the most recent Chrome Canary build, the default RTCP multiplexing policy is "require", instead of "negotiate". This will affect the next Chrome release, M57.
For any application that doesn't yet support RTCP multiplexing, you can get the old behavior by explicitly setting the RTCRtpMuxPolicy to "negotiate" in the RTCConfiguration.
Some endpoints (Asterisk, for example) do not yet support rtcp-mux and will therefore stop working after the next Chrome release.
seanbright
All 3 comments
This is not something we can do anything about in SIP.js. If you would like to discuss this further, I would recommend reaching out to Asterisk or you can discuss with other SIP.js users on our mailing list. Thanks.
egreenmachine
on 18 Jan 2017
I'm sorry. You are correct. I did not read the announcement before replying and just used your summary as a basis for my reply. Yes, we should add yet another hack. Re-opening this issue.
egreenmachine
on 18 Jan 2017
It should be noted that I tested this with FreeSWITCH 1.6.14 and RTPEngine and neither had issues with multiplexing RTCP with RTP. It appears that Asterisk is the lone issue here.
egreenmachine
on 24 Jan 2017
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Most helpful comment
I'm sorry. You are correct. I did not read the announcement before replying and just used your summary as a basis for my reply. Yes, we should add yet another hack. Re-opening this issue.