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

Matt Fredrickson creslin at digium.com
Thu Jan 19 16:31:31 CST 2017


On Thu, Jan 19, 2017 at 3:23 AM, Dan Jenkins <dan.jenkins88 at gmail.com> wrote:
>
>
> On Thu, Jan 19, 2017 at 9:09 AM, Dan Jenkins <dan.jenkins88 at gmail.com>
> wrote:
>>
>> Hi Sebastian,
>>
>> I haven't opened an issue in Asterisk yet - ran out of time yesterday -
>> first thing on my list to do today.
>>
>> I have now tested the flag in jssip (jssip makes it extremely easy to test
>> as it gives you direct access to the parameters you pass into
>> RTCPeerConnection) and it does in fact work. Once I've made the issue - I'll
>> be writing a blog post about the issue to further warn others about it.

Thanks for letting us know about this problem Dan.  This is
concerning, but it is good to know that they have a compatibility flag
for the time being.  I think someone will still need to go through and
evaluate how challenging it will be to add rtcp-mux support to
Asterisk.

>From the recollection I have of the last time I poked at that code, it
seems like it wouldn't be too hard to add rtcp-mux support.  My
prediction is that we'll probably look into this sooner rather than
later, given the change in climate with Chrome's support.

Matthew Fredrickson

>>
>> Dan
>>
>> On Thu, Jan 19, 2017 at 12:33 AM, Sebastian Gutierrez <scgm11 at gmail.com>
>> wrote:
>>>
>>> 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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>>>
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>>
>>
>
> Sorry - I top posted...
>
> Anyway - here is the issue
>
> https://issues.asterisk.org/jira/browse/ASTERISK-26732
>
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-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA



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