[asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

Teijo g.aloitus at gmail.com
Tue Jun 27 16:00:31 CDT 2017


Hello,

Yes. When I today understood to set rtcp_mux=yes, at least Chrome (60.0 
beta) worked (quickly tested) as expected.

I'm sure that some day dtls_rekey can be set to the other value than 0 
as well with Chrome.

Best regards,

Teijo

10.4.2017, 16.57, Matt Fredrickson kirjoitti:

> On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkins <dan.jenkins88 at gmail.com> wrote:
> 
>>
>> On Fri, Apr 7, 2017 at 9:44 PM, Teijo <g.aloitus at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
>>> problem until now which remained was that if dtls_rekey was set to the
>>> value other than 0, call hanged up when using chrome after the time where
>>> dtls_rekey was set.
>>>
>>> I suppose that "bad media description" shown in Chrome's window which
>>> causes call to fail, has appeared with Chromes newer versions (currently 58
>>> beta installed) or with Asterisk 13.15.0. Audio codec I'm using is Opus.
>>>
>>> Has somebody else encountered this problem, or more better resolved it?
>>>
>>> Best regards,
>>>
>>> Teijo
>>>
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>>
>>
>> Hi Teijo
>>
>> Take a read of https://nimblea.pe/monkey-business/2017/01/19/webrtc-
>> asterisk-and-chrome-57/ :)
>>
> 
> 13.15.0 should address rtcp-mux issues.
> 
> If there are still issues outstanding, it might be worth reporting a bug on
> issues.asterisk.org.
> 
> Best wishes :-)
> 
> 
> 
> Tämä osa viestin runkoa ladataan pyydettäessä.
> 



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