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!