如何将传入的 WebRTC 音频流混合到服务器上的单个 stream 中？

Question

I'm working on a project which involves building an audio-conferencing app for the web. Currently my working system uses a WebSocket server to negotiate connections between peers, which can then stream audio directly to one another. However, I wish to implement the server as its own client/peer, which will receive all incoming audio streams, "mix" them into a single source/stream, and then stream it to all peers individually. The goal is to avoid direct peer-to-peer connections between user connections.

Perhaps a more simple question would be how I can accomplish the concept of the given figure, the green squares being RTCPeerConnections, and the server "forwarding" the incoming streams to the recipient? Figure

How can I accomplish this, and is the concept feasible in regards to system resources of the server?

Thanks.

Answer 1

You can use kurento . Its based on webRTC and its Media Server features include group communications, transcoding, recording, mixing, broadcasting and routing of audiovisual flows.

Answer 2

The concept you are looking for is called Multi Conferencing Unit (MCU). MCU is not part of standard WebRTC. WebRTC is peer-to-peer only.

There are several media server solution that offer the MCU functionality. kurento, as suggested by Milad, is one options. Others examples are Jitsi Videobridge or Janus .

A more recent approach you might want to consider is SFU (Selective Forwarding Unit).