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