[sr-dev] Using Kamailio as a webrtc gateway

[suganthi karthick]

We have to interface the conference bridge platform and the webrtc gateway.
Shall we use Kamailio for this and work on top of it?

Thanks.

On Mon, Jan 4, 2016 at 3:25 PM, suganthi karthick <suganthi.mkk at gmail.com>
wrote:

> 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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>> >
>>
>>
>>
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>>
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>>
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