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
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