A user tutorial guide

[YongzeYao]

I am a programmer trying to code a online web-based audio/video conference room app utilizing WebRTC and found SRS to be mentioned often in blogs/posts related to WebRTC and landed here.

However I have no idea how to use SRS, I read the documents on webrtc.org and those documents claim that I need to provide ICE configuration during creating the RTCPeerConnection object. However, SRS is a SFU server, which I think is not a STUN/TURN server? If that is the case, how do I set up everything and hook things together?

Suppose my front-end is being served at A.com, and I think I need to create a RTCPeerConnection object in my front-end code in order to communicate with other peers using webrtc, and SFU is the other peer in this case. So which TURN server shall I use when creating the RTCPeerConnection object? And where should I deploy my signaling server? How should I handle auth? Is it possible that some other people use my deployed SRS server and use it as their SFU server for free?

[winlinvip]

See https://ossrs.io