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Janus Gateway
One Janus installation can host a video meeting in one session, stream a camera feed in another, and bridge a browser call into a telephone network in a third, because Janus is a general-purpose WebRTC server whose work is done by interchangeable plugins. The VideoRoom plugin publishes each participant's streams to everyone else in the room with simulcast and viewport selection, so a class or team call adapts quality per viewer instead of forcing one bitrate on all. Streaming mounts live or prerecorded sources, including RTSP cameras and RTP feeds, and hands them to any number of WebRTC viewers, while AudioBridge mixes voices into conference rooms with recording and signed-token access. SIP and NoSIP plugins register with existing VoIP infrastructure and place or receive calls, translating between WebRTC media and telephone audio so a web page can ring a desk phone. Record&Play captures publisher streams to files you can replay through the same server, TextRoom adds data-channel chat rooms, and Lua or JavaScript plugins let you script custom media logic without recompiling the core. Applications talk to Janus over HTTP or WebSocket APIs, with RabbitMQ, MQTT, and NATS transports for bus-driven deployments, and event handlers report session activity to your monitoring stack. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. GPL-3.0 licensed.
Benefits
- Many Media Apps, One Server
- Interchangeable plugins provide video rooms, streaming, audio conferencing, SIP telephony, text rooms, and recording on one WebRTC core, so separate products stop needing separate media servers and operations knowledge.
- Rooms That Adapt Per Viewer
- VideoRoom forwards publisher streams with simulcast layers and viewport-based selection, allowing large calls to give each viewer an appropriate quality instead of mixing everyone into a single heavy composite.
- Telephony Bridges Included
- SIP and NoSIP plugins register with VoIP providers, handle incoming and outgoing calls, and translate media, so browser interfaces can reach desk phones and PSTN services without a second gateway.
- Scriptable Media Logic
- Lua and JavaScript plugin runtimes let developers prototype custom media behavior quickly, while the C core keeps forwarding, DTLS, and congestion handling on the performance-critical path.
Features
- VideoRoom SFU Plugin
- VideoRoom publishes participant audio and video to room subscribers with simulcast, SVC, and viewport selection, plus RTP forwarding to external services such as streaming platforms.
- Streaming Plugin Mountpoints
- Streaming serves live RTP, RTSP, and prerecorded sources to WebRTC viewers through mountpoints, with helpers for sources such as cameras and software encoders.
- SIP And NoSIP Plugins
- SIP plugins register with external VoIP servers, manage calls with hold and transfer, and support late offers, while NoSIP handles calls without registration.
- Record And Play
- Record&Play records publisher media into Janus recording files that can be listed, converted, and replayed to WebRTC viewers as on-demand streams.
- Multiple API Transports
- Applications control sessions over HTTP and WebSocket APIs, with RabbitMQ, MQTT, NATS, and Unix socket transports plus pluggable event handlers for monitoring.