问题
Based on this article, when implementing a WebRTC solution without a server, I assume it means SFU, the bottleneck is that only 4-6 participants can work.
Is there a solution that can work around this? For example, I just want to use Firebase as the only backend, mainly signaling and no SFU. What is the general implementation strategy to achieve at least 25-50 participants in WebRTC?
Update: This Github project shares a different statement. It states "A full mesh is great for up to ~100 connections"
回答1:
Your real bottleneck with MESH is that each RTCPeerConnection will do its own video encoding in the browser.
The p2p concept naturally includes the requirement that both peers should adjust encoding quality based on network conditions. So, when your browser sends two streams to peers X (good download speed) and Y (bad download speed), the encodings for X and Y will be different - Y will receive lower framerate and bitrate than X.
Sounds reasonable, right? But, unfortunately, mandates separate video encoding for each peer connection.
If multiple peer connections could re-use the same video encoding, then MESH would be much more viable. But Google didn't provide that option in the browser. Simulcast requires SFU, so that's not your case.
So, how many concurrent video encodings can browser perform on a typical machine, for 720p 30 fps video? 5-6, not more. For 640x480 15 fps? Maybe 20 encodings.
In my opinion, the encoding layer and networking layer could be separated in WebRTC design, and even getUserMedia could be extended to getEncodedUserMedia, so that you could send the same encoded content to multiple peers.
So that's the real practical reason people use SFU for multi-peer WebRTC.
回答2:
If you want to make a conference with 25 people all sending their video, then a regular webrtc setup will not work. Except if you massively lower your video quality. The reason for this is that every participant would need to send 24 seperate streams to every other client. So lets say you stream is 128 KB/s then you will need to have 3MB/s in upload speed available. Which isn't always available. Then also downloading that same amount.
The problem is that isn't scalable. That's why you need an SFU. Then you will only send a single stream and receive from others. The other positive thing about SFUs is that you can use simulcast which adapts the quality of your received streams depending on your network speed.
You can use the Janus gateway or mediasoup for example. Here is an already setup mediasoup video conferencing application that is scalable github repository
来源：https://stackoverflow.com/questions/62054484/multi-party-webrtc-without-sfu