you can try to turn of the opus codec support in the browser.
At firefox ... open about:config and search for media.opus.enabled and
set it to false.
At Chrome ... open about:flags and search for opus.
Regards
Rainer
Am 04.04.2014 10:31, schrieb jaflong jaflong:
Hi,
Is it possible to use
Hi,
Is it possible to use pcmu start to end, so I send pcmu instead of opus from
the browser?
Regards
04.04.2014, 12:29, "Jon Bonilla (Manwe)" :
> El Fri, 04 Apr 2014 08:18:22 +0200
> Rainer Piper escribió:
>
>> Hallo,
>> my guess is the audio codec opus
>>
>> asterisk can NOT do transcod
El Fri, 04 Apr 2014 08:18:22 +0200
Rainer Piper escribió:
> Hallo,
> my guess is the audio codec opus
>
> asterisk can NOT do transcoding from opus to pcmu.
>
> The opus codec in asterisk is (just) a path through codec.
>
> your trace right at the end:
> !!! Failed to parse SessionDescription.
upps ... sorry ... *pass th**rough* and not path through :-[
2013-08-23 15:49 + [r397524-397527] Matthew Jordan
* CHANGES: Update CHANGES file to reflect pass through support for
Opus/VP8
Source ->
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog
Hallo,
my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end:
!!! Failed to parse SessionDescription. Failed to parse audio codecs correctly
!!!
Regards
Rainer
Am 03.04.201
My guess would be that it's due to a discrepancy between WebRTC and RFC
5764. WebRTC uses a protocol string of "RTP/SAVPF", while the RFC says
that "UDP/TLS/RTP/SAVPF" shall be used. You can try an SDP rewrite
operation to substitute one for the other. Or teach your non-RTC client
to use a differen
