[SR-Users] Browser WebRTC transcoder

[Moacir Ferreira]

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?

Cheers,
Moacir

> To: sr-users at lists.sip-router.org
> From: rfuchs at sipwise.com
> Date: Wed, 18 May 2016 19:03:10 -0400
> Subject: Re: [SR-Users] Browser WebRTC transcoder
> 
> On 18/05/16 04:57 PM, Moacir Ferreira wrote:
> > 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:/
> > //
> >
> >   * /Repacketization or transcoding/
> 
> This refers to translating one audio codec into another (e.g. opus to 
> PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting) 
> is supported.
> 
> Cheers
> 
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