Yes, now it is working both ways. I was missing the rtpengine and right configuration. I will do some further testing to see if everything is functional. Thanks, > To: sr-users@lists.sip-router.org > From: rfu...@sipwise.com > Date: Thu, 19 May 2016 08:22:26 -0400 > Subject: Re: [SR-Users] Browser WebRTC transcoder > > On 05/19/2016 04:52 AM, Moacir Ferreira wrote: > ... > > So the Grandstream offers a lot of codecs but will get a "Not Found" > > from Kamailio. Look in the other way: > > That would be a SIP signalling (e.g. Kamailio config) problem. Perhaps a > missing registration. > > > Here the Grandstream says "Media type not available". As I am not a real > > SIP guy, I got no clue why does not work! > > This you can solve with rtpengine. The required codecs (PCM) are there, > you just need to break the encryption (RTP <> SRTP) and some other > features of WebRTC (ICE, BUNDLE, rtcp-mux, ...), all of which rtpengine > can do. > > 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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