What codecs are supported by your grandstream? Isn't the g711 in the group?
Cheers, Daniel On 19/05/16 01:51, Moacir Ferreira wrote: > I did not dig into the problem but on my tests I saw that my (old) > Grandstream phone was refusing the call for not having a compatible > codec to talk with the offered ones by the browser (Firefox). Being > this the case, I guess I must include a translator, and all routing > logic, in between the callers. It points to Asterisk that I would like > to avoid for now. But I guess this is not a problem that only affects > me. Someone else must have faced this before. So the question still > open: What solution would be recommended for such case? > > Cheers, > Moacir > > > To: sr-users@lists.sip-router.org > > From: rfu...@sipwise.com > > Date: Wed, 18 May 2016 19:03:10 -0400 > > Subject: Re: [SR-Users] Browser WebRTC transcoder > > > > On 18/05/16 04:57 PM, Moacir Ferreira wrote: > > > Hey Daniel, > > > > > > If you say so, you probably right... I did not try it because on the > > > sipwise GitHub (https://github.com/sipwise/rtpengine) they mention: > > > > > > /"Rtpengine does not (yet) support:/ > > > // > > > > > > * /Repacketization or transcoding/ > > > > This refers to translating one audio codec into another (e.g. opus to > > PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting) > > is supported. > > > > Cheers > > > > _______________________________________________ > > 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 -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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