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ConfigsMay 30, 2026·3 min de lectura

Mediasoup — WebRTC SFU for Real-Time Video and Audio

Build scalable video conferencing and live streaming servers with a high-performance Selective Forwarding Unit in Node.js and Rust.

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Single
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Mediasoup Overview
Comando de instalación directa
npx -y tokrepo@latest install cef41d7c-5c02-11f1-9bc6-00163e2b0d79 --target codex

Ejecutar después de confirmar el plan con dry-run.

Introduction

Mediasoup is a WebRTC Selective Forwarding Unit (SFU) library for Node.js with a high-performance C++ media worker, recently rewritten in Rust. It routes media streams between participants without mixing or transcoding, enabling low-latency group video calls, live streaming, and broadcasting at scale.

What Mediasoup Does

  • Routes audio and video streams between WebRTC peers via selective forwarding (no transcoding)
  • Manages multiple workers, routers, and transports for horizontal scaling
  • Supports simulcast and SVC so clients receive the quality layer matching their bandwidth
  • Provides producer-consumer architecture for flexible media routing topologies
  • Handles DTLS, ICE, RTP/RTCP, and SCTP protocols internally

Architecture Overview

Mediasoup runs a Node.js control plane that spawns native worker processes (C++ or Rust) for media handling. Each worker manages one or more routers, which define media routing scopes. Transports (WebRTC, plain RTP, or pipe) connect external endpoints to routers. Producers publish media into a router, and consumers subscribe to it. The worker handles all packet forwarding at near-kernel speed while the Node.js layer manages signaling logic and room orchestration. Inter-worker pipe transports enable multi-worker scaling on multi-core machines.

Self-Hosting & Configuration

  • Install via npm; the native worker compiles automatically during installation
  • Configure worker settings including RTC port ranges, log levels, and DTLS certificates
  • Set media codecs per router to control which audio and video formats are negotiated
  • Use plain RTP transports for integration with recording servers or external media processors
  • Deploy behind a TURN server for NAT traversal in restrictive network environments

Key Features

  • Near-zero latency forwarding without CPU-intensive transcoding or mixing
  • Simulcast and SVC layer selection for adaptive bitrate delivery
  • Multi-worker architecture scales across all CPU cores on a single machine
  • Protocol-level flexibility with WebRTC, plain RTP, and pipe transport types
  • Active protocol compliance tracking with regular updates for new WebRTC standards

Comparison with Similar Tools

  • Janus — a C-based gateway with plugin architecture; Mediasoup is a library you embed into your Node.js application
  • Pion — Go WebRTC building blocks; Mediasoup provides a higher-level SFU abstraction
  • LiveKit — a turnkey SFU server; Mediasoup gives more control but requires building your own signaling
  • Kurento — MCU-style media server with transcoding; Mediasoup is SFU-only for lower latency
  • Ion SFU — a Go-based SFU built on Pion; Mediasoup's C++/Rust worker offers different performance characteristics

FAQ

Q: Does Mediasoup transcode video? A: No. It is a pure SFU that forwards packets without decoding. For transcoding, pipe media to an external processor like FFmpeg.

Q: How many participants can a single server handle? A: Depends on bandwidth and media settings. A typical server handles hundreds of participants in SFU mode since CPU load is minimal.

Q: Is there a client-side library? A: Yes. mediasoup-client for browsers and libmediasoupclient for native C++ applications handle the client-side WebRTC interaction.

Q: Can I record streams? A: Yes, by creating a plain RTP consumer and piping the stream to a recording tool like GStreamer or FFmpeg.

Sources

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