Re: [SR-Users] What going on this SDP

2014-04-04 Thread Rainer Piper
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

Re: [SR-Users] What going on this SDP

2014-04-04 Thread jaflong jaflong
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

Re: [SR-Users] What going on this SDP

2014-04-04 Thread 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 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.

Re: [SR-Users] What going on this SDP

2014-04-04 Thread Rainer Piper
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

Re: [SR-Users] What going on this SDP

2014-04-03 Thread Rainer Piper
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

Re: [SR-Users] What going on this SDP

2014-04-03 Thread Richard Fuchs
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

[SR-Users] What going on this SDP

2014-04-03 Thread jaflong jaflong
Hi List, Can anyone help me understand why this is getting rejected Please note the specific message further dow the log. "Failed to parse SessionDescription.  Failed to parse audio codecs correctly" This is on Chrome. On Firefox  There is a further message in the console "Could not negotiate