[sr-dev] Using Kamailio as a webrtc gateway

[suganthi karthick]

Thank you so much.

The conference bridge is an existing working one for SIP clients, and I am
trying to add webrtc support for that.

The webrtc gateway needs to be implemented in a way like a library because
it needs to be integrated into the existing platform.

There are some init functions and config function from the existing module
based on which the gateway can be configured.

Also, when a webrtc call come from a webrtc client, it needs to handle the
signaling and the media has to go to the conference bridge platform.

It would be really helpful if you suggest whether I can use Kamailio for
this purpose and use it as a library and integrate into the exiting
platform?

Your suggestions will be more helpful.

Thanks.


On Mon, Jan 4, 2016 at 3:13 PM, Carsten Bock <carsten at ng-voice.com> wrote:

> Hi,
>
> you can achieve SIP over WebSocket with Kamailio (see
> http://kamailio.org/docs/modules/stable/modules/websocket.html) and
> DTLS-SRTP to "plain" RTP with RTPEngine
> (http://kamailio.org/docs/modules/stable/modules/rtpengine).
>
> An example configuration can be found here:
> https://github.com/caruizdiaz/kamailio-ws
>
> Thanks,
> Carsten
>
> 2016-01-04 10:03 GMT+01:00 suganthi karthick <suganthi.mkk at gmail.com>:
> > Hi,
> >
> > I need to implement a WebRTC gateway for an existing conference bridge.
> The
> > clients application can be a JsSIP client (SIP over websocket or JSON
> over
> > websocket). The WebRTC gateway has to support Signaling and ICE and DTLS.
> >
> > Can I use Kamailio as a base for this development?
> >
> > Thanks
> > Suganthi
> >
> > _______________________________________________
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> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
> >
>
>
>
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>
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