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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