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....@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 > > _______________________________________________ > sr-dev mailing list > sr-dev@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev > -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:cars...@ng-voice.com Office +49 40 5247593-0 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ _______________________________________________ sr-dev mailing list sr-dev@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev