asterisk users - Apr 2017 - Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

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

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
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>      https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

Hi Teijo

Take a read of
https://nimblea.pe/monkey-business/2017/01/19/webrtc-asterisk-and-chrome-57/
:)

Dan
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Thank you Dan for this information.

Best regards,

Teijo

8.4.2017, 15:23, Dan Jenkins kirjoitti:
> 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
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> 
> 
> Hi Teijo
> 
> Take a read of
>
https://nimblea.pe/monkey-business/2017/01/19/webrtc-asterisk-and-chrome-57/
> :)
> 
> Dan
> 
> 
>

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
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>      https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> 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 :-)

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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