[OpenSIPS-Users] SIP to WebRTC Proxy / Gateway

[Rion Carter]

I'm pretty new to SIP, RTP and WebRTC. I am in need of a gateway or proxy
that can let me use an existing SIP Soft-phone to connect to a
WebRTC/SIP-over-websockets server (the  WebRTC/SIP-over-websockets server
does not provide a way for regular SIP softphones to connect).

Would OpenSIPS be able to proxy my requests from my softphone to the WebRTC
endpoint? I have examined the documentation and if I've missed something I
apologize. Most everything I read emphasizes connecting webrtc clients to a
server, and my need is different than that.

Any examples, tutorials or documentation would be appreciated.

Thanks!
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