Feature request: Add SIP transport

bluenviron / mediamtx

Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.

MIT License

11.63k stars 1.47k forks source link

Feature request: Add SIP transport #2312

Open billsimon opened 1 year ago

billsimon commented 1 year ago

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 commented 3 months ago

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 commented 3 months ago

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!