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:

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?

Thanks,
Moacir

To: sr-users@lists.sip-router.org
From: miconda@gmail.com
Date: Wed, 18 May 2016 21:18:01 +0200
Subject: Re: [SR-Users] Browser WebRTC transcoder

Hello,
kamailio+rtpengine should do this job quite well.
Cheers,
Daniel

On 18/05/16 19:16, Moacir Ferreira wrote:
A question for the community:
 
What would be your best advice for a RTP proxy/transcoder to allow browser WebRTC calls to legacy VoIP?
 
Moacir


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