webrtc-mediasoup

Build a WebRTC SFU using a mediasoup-based Node.js server to route audio and video streams. It enforces a modular Worker → Router → Transport topology, enabling scalable, configurable per-room routing for multi-party conferences.

Routing media in a WebRTC SFU with mediasoup involves managing the worker and router lifecycle. You configure WebRtcTransport for ICE/DTLS/SRTP, PlainTransport for RTP bridges, and PipeTransport for router-to-router sharding.

Yes, mediasoup requires a Bring Your Own signaling pattern for session management. It uses protoo as a reference signaling implementation, allowing flexible integration with your existing authentication and session layers.

Yes, you can bridge WebRTC streams with ffmpeg or GStreamer using mediasoup. The server provides PlainTransport for RTP bridges and DirectTransport for in-process control, enabling seamless integration into streaming pipelines.

Mediasoup handles router-to-router scaling in WebRTC sessions by using PipeTransport. This allows you to shard routers across multiple workers, ensuring scalable and configurable media routing for high-volume streaming backends.

A limitation of using a mediasoup SFU for room-based meetings is that it does not include a built-in signaling layer. You must implement and manage your own signaling protocol using a reference like protoo to handle client integration.