You can build a standalone webrtc gateway using kamailio and rtpengine. The forward sip traffic to your existing application.
Daniel 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. > > Thanks. > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu
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