You can build a standalone webrtc gateway using kamailio and rtpengine. The forward sip traffic to your existing application.


On 04/01/16 13:56, suganthi karthick wrote:
Hi all,

I need to implement a WebRTC gateway for an existing conference bridge. The WebRTC gateway has to support Signaling, ICE and DTLS. The webrtc clients can be JsSIP or any webrtc client.

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 conference platform, based on which the webrtc gateway has to  be configured. 

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

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

Your suggestions will be more helpful.



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