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
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@gmail.com:
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@ng-voice.com wrote:
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@gmail.com wrote:
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Carsten Bock -
suganthi karthick