Hello,

Kamailio can be used to gateway signaling from webrtc/websocket to other transports like UDP/TCP/TLS. See websocket module for an example to start with and search of github for more advanced example by Carlos Ruiz Diaz. Some talks were given at Kamailio World Conference in the past that shows further examples.

If you need to do gatewaying of the SRTP to RTP, then you have to add RTPEngine as well.

Cheers,
Daniel

On 28/02/15 22:13, Rion Carter wrote:

I'm pretty new to SIP, RTP/SRTP, WebRTC and Websockets, so I hope this question is coherent. I have a group of SIP Softphones that need to connect to a WebRTC/SIP-over-Websockets server. Can Kamailio be configured to let me do this?

Any examples, tutorials or documentation would be appreciated. I'm trying to determine how feasible this task is. :)

Thanks!
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Daniel-Constantin Mierla
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