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Janus WebRTC Server - Flexible Real-time Communication Gateway
Janus is a general purpose WebRTC server designed to provide server-side processing and forwarding of WebRTC streams. Its modular architecture allows for the creation of custom applications via plugins, making it a highly flexible framework for building real-time communication services.
C
Added on 2025年5月11日8,606
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LanguageProject Introduction
Summary
Janus Gateway is an open-source, general-purpose server-side solution implementing the WebRTC server architecture. It acts as a bridge between WebRTC clients and server-side applications, facilitating various real-time communication scenarios through a flexible plugin interface.
Problem Solved
WebRTC is client-side focused. Building complex, multi-party, or server-integrated real-time applications requires significant server-side infrastructure for media handling, signaling, and business logic. Janus provides a robust, flexible server to bridge this gap.
Core Features
Plugin-based Architecture
Highly modular and extensible core with a powerful plugin API to build custom WebRTC applications.
Rich Set of Standard Plugins
Includes official plugins for common use cases like video conferencing, streaming, recording, SIP gateways, and more.
Flexible Transport and Protocol Support
Supports various transport mechanisms (WebSockets, plain HTTP/REST, MQTT) and media protocols (RTP/RTCP, DTLS, ICE, TURN).
Tech Stack
C
WebRTC
RTP/RTCP
DTLS
ICE
libnice
libsrtp
libopus
libvpx
libwebsocket
Use Cases
Due to its plugin-based architecture, Janus can be adapted to a wide variety of real-time communication use cases:
Scalable Video Conferencing Solutions
Details
Facilitating multi-party video and audio calls, enabling features like mixing, recording, and simulcasting through specialized plugins.
User Value
Provides the necessary server infrastructure for handling multiple participants and media streams efficiently, reducing client-side load and complexity.
Live Streaming and Broadcasting
Details
Acting as a server-side component to broadcast live audio/video streams to multiple WebRTC subscribers or ingest streams from broadcasters.
User Value
Enables low-latency live media distribution using standard WebRTC protocols, suitable for events, webinars, or surveillance.
SIP/PSTN Gateway Functionality
Details
Connecting traditional SIP/PSTN endpoints with WebRTC clients, allowing seamless communication between different networks.
User Value
Bridges the gap between legacy communication systems and modern WebRTC applications, expanding reach and interoperability.
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