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<body class='hmmessage'><div dir='ltr'>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?<br><br>Cheers,<br>Moacir<br><br><div>> To: sr-users@lists.sip-router.org<br>> From: rfuchs@sipwise.com<br>> Date: Wed, 18 May 2016 19:03:10 -0400<br>> Subject: Re: [SR-Users] Browser WebRTC transcoder<br>> <br>> On 18/05/16 04:57 PM, Moacir Ferreira wrote:<br>> > Hey Daniel,<br>> ><br>> > If you say so, you probably right... I did not try it because on the<br>> > sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:<br>> ><br>> > /"Rtpengine does not (yet) support:/<br>> > //<br>> ><br>> > * /Repacketization or transcoding/<br>> <br>> This refers to translating one audio codec into another (e.g. opus to <br>> PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting) <br>> is supported.<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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