[SR-Users] Browser WebRTC transcoder

[Daniel-Constantin Mierla]

What codecs are supported by your grandstream? Isn't the g711 in the group?

Cheers,
Daniel


On 19/05/16 01:51, Moacir Ferreira wrote:
> 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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>
>
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-- 
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

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