[SR-Users] Kamailio as a SIP softphone to WebRTC/SIP-over-websockets proxy?

[Rion Carter]

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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