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)" <ma...@aholab.ehu.es>: > El Fri, 04 Apr 2014 08:18:22 +0200 > Rainer Piper <rainer.pi...@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@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
