[asterisk-dev] Issues with WebRTC going forward - rtcp-mux now required

[Sebastian Gutierrez]

Dan,

have you open an issue on asterisk jira?
did you test the flag in jssip?

thanks

best regards


On Jan 18, 2017, 10:52 -0600, Dan Jenkins <dan.jenkins88 at gmail.com>, wrote:
> Hi All,
>
> I've been working with a company who utilise WebRTC using Asterisk behind Kamailio to connect browser users and their SIP infrastructure and just came across an issue making/receiving calls in Chrome Canary and Chrome Dev.
>
> Long story short; the issue is that rtcp-mux has now been set as required in Chrome's WebRTC stack - https://groups.google.com/forum/#!topic/discuss-webrtc/eM57DEy89MY
>
> For now; there is a workaround of being able to pass in a flag to the RTCPeerConnection call to get the old "negotiate" behaviour and I'm talking to the Chrome team to find out how long this flag will be around for.
>
> I've been told that Asterisk doesn't support rtcp-mux as of today and so I'm raising the issue here. It seems - if Asterisk wants to support WebRTC long term; it will need to support rtcp-mux - I quote Sean Bright from a conversation we had in IRC where he said it was "non-trivial" to support.
>
> I don't know more than this and I don't mean to say something is difficult to fix when I honestly don't know the effort levels in order to fix this. So I'm raising this here for a conversation.
>
> I will be testing the client side flag fix in jssip in a moment and then writing up a blog post about it if it does indeed fix the issue, at least temporarily while the flag is available.
>
> Dan
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