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.

With over 20,000 GitHub stars and adoption by companies building everything from telehealth platforms to AI voice agents, LiveKit is the most widely deployed open-source real-time communication server available. The Go-based Selective Forwarding Unit handles hundreds of concurrent participants per node with adaptive bitrate streaming, simulcast layers, SVC codec support for VP9 and AV1, and end-to-end encryption. Client SDKs span JavaScript, Swift, Kotlin, Flutter, React Native, Rust, Python, Unity, and ESP32 embedded devices, while server-side APIs cover Node.js, Go, Ruby, Java, Python, Rust, PHP, and .NET. The Agents framework enables building AI-powered voice and video applications — real-time speech-to-text, LLM-driven conversations, and computer vision pipelines — running as server-side participants in any room. Egress records sessions to S3-compatible storage or streams to RTMP endpoints, while Ingress pulls external feeds from OBS via RTMP, WHIP, or SRT into LiveKit rooms. The SIP bridge connects traditional telephony to WebRTC rooms for hybrid conferencing. JWT-based authentication, webhook notifications, room-level moderation APIs, and selective subscription give operators granular control. Deploy as a single binary for development, Docker Compose for production single-node, or Kubernetes with the official Helm chart for distributed multi-region clusters using Redis for state coordination. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. Apache 2.0 licensed.