[SR-Users] Browser WebRTC transcoder

Moacir Ferreira moacirferreira at hotmail.com
Fri May 20 09:49:29 CEST 2016


Yes, now it is working both ways. I was missing the rtpengine and right configuration.
I will do some further testing to see if everything is functional.
 
Thanks,
 
> To: sr-users at lists.sip-router.org
> From: rfuchs at sipwise.com
> Date: Thu, 19 May 2016 08:22:26 -0400
> Subject: Re: [SR-Users] Browser WebRTC transcoder
> 
> On 05/19/2016 04:52 AM, Moacir Ferreira wrote:
> ...
> > So the Grandstream offers a lot of codecs but will get a "Not Found"
> > from Kamailio. Look in the other way:
> 
> That would be a SIP signalling (e.g. Kamailio config) problem. Perhaps a
> missing registration.
> 
> > Here the Grandstream says "Media type not available". As I am not a real
> > SIP guy, I got no clue why does not work!
> 
> This you can solve with rtpengine. The required codecs (PCM) are there,
> you just need to break the encryption (RTP <> SRTP) and some other
> features of WebRTC (ICE, BUNDLE, rtcp-mux, ...), all of which rtpengine
> can do.
> 
> Cheers
> 
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