e: [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
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
eone else must have faced this before. So the question still
open: What solution would be recommended for such case?
Cheers,
Moacir
> To: sr-users@lists.sip-router.org
> From: rfu...@sipwise.com
> Date: Wed, 18 May 2016
meone else must have faced this before. So the question still
> open: What solution would be recommended for such case?
>
> Cheers,
> Moacir
>
> > To: sr-users@lists.sip-router.org
> > From: rfu...@sipwise.com
> > Date: Wed, 18 May 2016 19:03:10 -0400
> > Subject
org
> From: rfu...@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
>
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 trans
il.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 questi
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
>
>
> ___
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
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mail
