[sr-dev] Using Kamailio as a webrtc gateway
Carsten Bock
carsten at ng-voice.com
Mon Jan 4 10:43:54 CET 2016
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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