Configs2026年7月23日·1 分钟阅读

Janus Gateway — General Purpose WebRTC Server

A modular, open-source WebRTC gateway that handles media negotiation, recording, streaming, and SIP bridging through a plugin-based architecture.

Agent 就绪

Agent 可直接安装

这个资产可安装;Agent 先选择当前运行时、检查安装计划,再运行匹配命令。

Native · 98/100策略:允许
Agent 入口
任意 MCP/CLI Agent
类型
Skill
安装
Single
信任
信任等级:Established
入口
Janus WebRTC Gateway
直接安装命令
npx -y tokrepo@latest install baa90456-8672-11f1-9bc6-00163e2b0d79 --target codex

先 dry-run 确认安装计划,再运行此命令。

Introduction

Janus is a general-purpose WebRTC server developed by Meetecho. Unlike monolithic video platforms, Janus provides a lightweight core that delegates media logic to plugins, making it suitable for building custom real-time communication applications from video rooms to SIP gateways.

What Janus Gateway Does

  • Negotiates WebRTC sessions (ICE, DTLS, SRTP) between browsers and server-side plugins
  • Provides built-in plugins for video rooms, SIP bridging, streaming, recording, and screen sharing
  • Handles media relay, mixing, and forwarding with low latency overhead
  • Exposes a JSON-based API over HTTP, WebSocket, RabbitMQ, or MQTT transports
  • Supports simulcast, SVC, and bandwidth estimation for adaptive quality

Architecture Overview

Janus runs as a single C process with a thin core that handles WebRTC session setup (ICE candidates, DTLS handshakes, SRTP encryption). All application logic lives in dynamically loaded plugins. Each plugin manages its own sessions and media flows. The core routes media packets between browser peers and plugins based on session handles. Transport plugins handle the signaling layer, decoupling the API protocol from the media path.

Self-Hosting & Configuration

  • Build from source on Linux; requires libnice, libsrtp2, and OpenSSL development headers
  • Configure transport plugins (HTTP, WebSocket, MQTT) in /etc/janus/ config files
  • Enable or disable media plugins by toggling their config files in the plugins directory
  • Set up a TURN server (like Coturn) for NAT traversal in production deployments
  • Use the built-in admin API for runtime monitoring and session management

Key Features

  • Plugin architecture that separates core WebRTC handling from application logic
  • Built-in video room plugin supporting up to dozens of participants per room
  • SIP plugin for bridging WebRTC calls to traditional telephony networks
  • Streaming plugin for one-to-many live video distribution via WebRTC
  • Recording plugin that saves sessions to MJR files for later replay or transcoding

Comparison with Similar Tools

  • LiveKit — modern Go-based SFU; easier to deploy for video rooms but less flexible plugin model than Janus
  • mediasoup — Node.js SFU library; embeddable in JS apps but requires building your own signaling layer
  • Kurento — Java-based media server; stronger media processing pipeline but heavier resource footprint
  • Pion — Go WebRTC library; lower-level building blocks, not a ready-to-use server like Janus

FAQ

Q: How many participants can a Janus video room support? A: Depends on server resources and media quality settings. A well-configured server handles 30-50 participants in a single room with selective forwarding.

Q: Does Janus do transcoding? A: By default it operates as an SFU (selective forwarding unit) and does not transcode. Plugins can be extended for MCU-style mixing if needed.

Q: What browsers are supported? A: All modern browsers with WebRTC support: Chrome, Firefox, Safari, and Edge. Mobile browsers on iOS and Android are also supported.

Q: Can I use Janus for production live streaming? A: Yes. The streaming plugin distributes live feeds to thousands of WebRTC viewers. For very large audiences, combine it with an HLS/DASH CDN fallback.

Sources

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