Feature request: Add SIP transport

[billsimon]

Describe the feature

SIP is a very widely used signaling transport for voice and video (VoIP) calling. It sets up RTP streams using SDP, much like RTSP or webrtc.

The use case here is SIP video phones and audio/video door bells.

It would be useful to stream these kinds of devices to and from the media server.

[goharahmed]

Hi Bill, Can you explain your use case a bit more. I believe I could help you in adding SIP+SDP interface in this library. WebRTC is already supported so perhaps an OpenSIPS as your SIP<->WebRTC convertor can already achieve this but still.. let me know if still needed. Thanks, Gohar

[billsimon]

Hey Gohar, I moved on from that project though I think this would still be a good feature if anyone were interested in developing it. Thanks for asking!