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<body class='hmmessage'><div dir='ltr'>Hey Daniel,<br><br>If you say so, you probably right... I did not try it because on the sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:<br><br><i>"Rtpengine does not (yet) support:</i><BR><i>
</i><ul><li><i>Repacketization or transcoding</i></li><li><i>Playback of pre-recorded streams/announcements</i></li><li><i>Recording of media streams</i></li><li><i>ZRTP"</i></li></ul><br>So I did not test it... I will give it a try. By the way, from where should I download the source code? Also, any "tricky" (common mistake like my last one on WebRTC TLS) I should care about before trying it?<br><br>Thanks,<br>Moacir<br><br><div><hr id="stopSpelling">To: sr-users@lists.sip-router.org<br>From: miconda@gmail.com<br>Date: Wed, 18 May 2016 21:18:01 +0200<br>Subject: Re: [SR-Users] Browser WebRTC transcoder<br><br>
Hello,<BR>
kamailio+rtpengine should do this job quite well.<BR>
Cheers,<br>
Daniel<br>
<BR>
<div class="ecxmoz-cite-prefix">On 18/05/16 19:16, Moacir Ferreira
wrote:<br>
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<div dir="ltr">A question for the community:<br>
<br>
What would be your best advice for a RTP proxy/transcoder to
allow browser WebRTC calls to legacy VoIP?<br>
<br>
Moacir<br>
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<pre class="ecxmoz-signature">--
Daniel-Constantin Mierla
<a class="ecxmoz-txt-link-freetext" href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
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