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!