[SR-Users] What going on this SDP

jaflong jaflong jaflong at yandex.com
Fri Apr 4 10:31:54 CEST 2014


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)" <manwe at aholab.ehu.es>:
> El Fri, 04 Apr 2014 08:18:22 +0200
> Rainer Piper <rainer.piper at soho-piper.de> 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.  Failed to parse audio codecs
>>  correctly !!!
>
> Just in case you don't know the patch:
>
> https://github.com/meetecho/asterisk-opus
>
> cheers,
>
> Jon
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users



More information about the sr-users mailing list