Backed by nearly 30,000 GitHub stars and powering millions of meetings through meet.jit.si, Jitsi Meet is the open-source video conferencing platform that runs entirely in the browser with no plugins or downloads required. The WebRTC-based architecture delivers HD audio and video through the Jitsi Videobridge selective forwarding unit, while Prosody handles XMPP signaling and Jicofo manages conference focus and orechestration. End-to-end encryption uses AES-GCM 128-bit keys exchanged via the Olm protocol, protecting audio, video, and screen sharing through the Insertable Streams API in Chromium-based browsers. Moderators can split participants into breakout rooms with individual password protection, enable lobby waiting rooms for admission control, set meeting passwords, and track speaker statistics with per-participant talk time metrics. Virtual backgrounds leverage TensorFlow.js body segmentation with WebGPU and WebGL acceleration, automatically adapting quality through device tier detection. Server-side recording and YouTube live streaming run through Jibri, while Jigasi provides a SIP gateway for PSTN dial-in and speech-to-text transcription. The platform includes in-meeting chat with private conversations, polls, raise-hand functionality, emoji reactions, and tile and speaker view layouts. Mobile applications for iOS and Android complement the web client, and both web and native SDKs enable embedding video conferences into custom applications. 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.

OpenIM gives your own applications one-to-one and group chat, powered by an instant messaging server you host and client SDKs for mobile, web, and desktop. Conversations carry unread counts, drafts, pinned messages, and typing indicators, and each chat keeps paging history, keyword search, read receipts, and revoke for mistakes. Custom message payloads carry cards, orders, or any application data your product renders inside a conversation. Your backend drives the server over a REST API that provisions users, issues login tokens, creates groups with owners and members, and sends messages from trusted services. Webhooks call your business servers before and after events, so moderation, billing, and CRM workflows react as messages happen. Groups scale to hundreds of thousands of members with invitations, join requests, member muting, and group-wide mute controls. Offline messages wait on the server and sync down on reconnect, merged by message ID so nothing arrives twice. Deployment runs as Docker Compose on one host or as gateway and RPC microservices on Kubernetes. 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.

Viewers arrive on browsers, phones, and desktop players, and SRS meets each of them with the right protocol from a single published stream. A broadcaster pushing RTMP can reach browser viewers through WebRTC with sub-second latency, phone apps through HLS, and desktop players through HTTP-FLV, all from the same live source without running parallel encoders. WHIP endpoints accept WebRTC ingest from browser studios and tools like OBS, while WHEP serves WebRTC playback, so an entire publish-to-viewer path can stay on modern web protocols. Operators shape each channel with virtual hosts, authenticate publishers and players through HTTP hooks that consult your own backend, and record live streams to disk as DVR archives for replay. Forwarding republishes a stream to YouTube, Facebook, or another SRS instance, and origin clustering lets edge servers pull from a central origin so one ingest feeds viewers in several regions. The built-in HTTP API reports streams, clients, and server health for monitoring, and GB28181 support ingests surveillance camera networks alongside creator broadcasts. Configuration lives in a single reloadable file, so changing a vhost or enabling HLS rarely means restarting the listeners your audience is attached to. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. MIT 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.

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.

A single ejabberd node carries chat, device telemetry, voice calls, and federated messaging at once, because the same Erlang server speaks XMPP, MQTT, SIP, and Matrix over one authentication, storage, and clustering layer. Users get standards-based instant messaging with group chats, message archives, publish-subscribe, message carbons, push notifications, and OMEMO encryption, and your server federates with the global XMPP network out of the box. Devices publish and subscribe through the built-in MQTT broker on TCP or WebSocket transports, sharing accounts and access rules with your chat users instead of needing a second broker. Voice and video calls route through the integrated SIP gateway and STUN/TURN support, so WebRTC clients get signalling and media relay from the same deployment. Matrix support bridges federated rooms into the same infrastructure, letting two decentralised ecosystems interoperate without a separate bridge stack to babysit. Clustering with consistent hashing scales a deployment across nodes, and a single node is rated at two million concurrent sessions for very large communities. Operators manage everything from the web admin and command line, with SQL or Mnesia storage, LDAP authentication, and virtual hosting for multiple domains on one server. Running on a dedicated VPS on RepoCloud with guaranteed CPU, RAM, and SSD, full root SSH access, and a browser serial console. GPL-2.0 licensed.