Open-source software for building real-time audio and video communication infrastructure on your own servers.
This project is a WebRTC screen sharing server designed to facilitate the streaming of desktop views between multiple participants. It functions as a signaling server to coordinate connection metadata and a relay server to ensure connectivity for users behind restrictive firewalls or symmetric NATs. The server enables real-time screen sharing by establishing direct peer-to-peer connections to reduce latency and server load. It utilizes a relay architecture to maintain stable communication when direct paths are blocked by network firewalls. The system provides coordination for session management, room orchestration, and WebSocket signaling channels to manage participants and media streams.
This is a self-hostable WebRTC server specifically optimized for screen sharing and session orchestration, providing the necessary signaling and relay capabilities for real-time streaming even if it lacks a full multi-party video conferencing suite.
Mirotalksfu is a WebRTC video conferencing platform and AI-integrated meeting suite. It functions as a real-time communication system for hosting high-resolution audio and video meetings, serving as a self-hosted virtual classroom and a collaborative workspace. The platform distinguishes itself by integrating generative AI assistants, speech recognition, and digital avatars into live sessions. It also operates as an RTMP streaming gateway, allowing users to broadcast live meeting content to external audiences and platforms. The system provides a collaboration suite featuring a shared interactive whiteboard, a rich text editor, real-time chat, and project file sharing. Meeting management is supported through screen sharing, local recording, virtual backgrounds, and secure room administration using authentication and passwords. Programmatic control is available via a REST API for automating internal processes and managing core system settings.
Mirotalksfu is a comprehensive, self-hostable WebRTC media server that provides multi-party conferencing, screen sharing, and low-latency streaming through a browser-based interface, fully meeting the requirements for a real-time communication platform.
BigBlueButton is an open-source virtual classroom software and meeting server designed for hosting real-time online teaching sessions. It functions as a WebRTC video conferencing platform and collaboration suite, providing a self-hosted environment for virtual classroom management and online course administration. The system distinguishes itself through specialized educational tools, including an interactive quiz engine, breakout room coordination, and live polling. It provides a comprehensive suite of collaborative assets such as a shared infinite whiteboard, real-time co-authored notes, and presentation annotation tools. The platform covers a broad range of capabilities, including real-time audio and video streaming, live transcription, and session recording. It includes administrative features for participant role moderation, guest access management, and learning analytics to track student engagement. The system is extensible via a plugin model and programmatic APIs for automating classroom management and routing event webhooks. The software supports identity verification through OpenID Connect integration and allows for large-scale deployment via automated server replication.
BigBlueButton is a comprehensive, self-hostable WebRTC media server that provides multi-party conferencing, screen sharing, and low-latency streaming specifically tailored for virtual classrooms and collaborative meetings.
LiveKit is a comprehensive framework for building and orchestrating real-time, multimodal AI agents that interact with users through voice, video, and text. It provides a centralized, event-driven architecture to manage the entire lifecycle of automated participants, from initialization and session state management to graceful shutdown. By utilizing a selective forwarding unit, the platform efficiently routes media streams between participants and agents, ensuring low-latency communication and secure, token-based authentication for all connections. The platform distinguishes itself through its modular pipeline-based media processing, which chains specialized speech-to-text, language, and text-to-speech services into cohesive workflows. It includes advanced capabilities for real-time voice activity detection, enabling natural turn-taking and interruption handling, alongside remote procedure call tooling that allows agents to execute external functions or access local resources during a conversation. Developers can further extend these interactions by integrating photorealistic virtual avatars that synchronize visual expressions with the agent's audio output. Beyond core conversational logic, the system offers extensive support for telephony integration, allowing agents to connect to public networks via SIP for inbound and outbound calling. It provides a robust suite of observability and monitoring tools to track agent performance, connection quality, and session events, ensuring reliability in production environments. The platform also includes specialized utilities for task automation, such as capturing and validating structured user data, and supports multi-step workflow orchestration to handle complex, context-aware interactions. The project provides a command-line interface for scaffolding, deploying, and testing agent applications, with documentation available in machine-readable formats to assist in development.
LiveKit is a robust, self-hostable WebRTC media server that utilizes a selective forwarding unit to provide the low-latency, multi-party audio and video conferencing capabilities required for your communication platform.
Neko is a virtual desktop infrastructure platform that provides containerized browser isolation and remote desktop environments. It enables users to host secure, ephemeral browser instances that can be accessed and managed through a standard web browser, ensuring consistent execution across different host systems. The platform distinguishes itself through its collaborative capabilities, allowing multiple users to view and interact with a single shared browser session in real time. It synchronizes keyboard, mouse, and gamepad inputs from multiple participants while providing integrated tools for real-time chat and file exchange. To maintain performance, the system utilizes hardware-accelerated rendering and adaptive bitrate control, which dynamically adjusts media quality based on real-time network throughput. The project covers a broad range of administrative and operational requirements, including identity management, session persistence, and granular access control. It supports complex network environments through configurable STUN and TURN integration, reverse proxy support, and customizable firewall traversal settings. Users can further extend the platform by customizing browser environments, applying administrative policies, and offloading graphics processing to dedicated hardware. The software is distributed as container images with multi-architecture support, and its configuration is managed through a comprehensive framework that includes URL-based parameters and persistent storage mounting for user data.
Neko is a self-hosted platform that uses WebRTC to stream a containerized browser session to multiple users, providing the required real-time media streaming, conferencing, and TURN/STUN capabilities for collaborative remote desktop access.
Openfire is an XMPP communication server and enterprise messaging platform designed for real-time collaboration. It serves as a communication hub providing instant messaging, presence tracking, and multi-user chat capabilities for organizational use. The server supports federated network routing via an XMPP federation gateway, allowing users across different domains to exchange messages. It is designed for high availability through server node clustering and multi-node synchronization to balance client traffic and ensure continuous uptime. The platform integrates with external directory services and custom identity providers for automated user and group synchronization. Its capability surface includes audio and video conferencing, STUN and TURN services for network traversal, and a plugin-based architecture for adding custom functionality. Administrative control is provided through APIs for user account management and presence monitoring, while network security is handled via TLS encryption for socket connections.
Openfire is a robust XMPP-based communication server that includes built-in support for STUN/TURN services and audio/video conferencing, making it a viable, albeit XMPP-centric, solution for your real-time media streaming needs.
Mediasoup is a selective forwarding unit used for real-time media routing. It enables the development of low-latency audio and video communication systems by routing streams between participants without transcoding. The project provides embedded media routing logic that can be integrated directly into an application. It supports simulcast and quality layering, allowing the system to adapt resolution and bitrate based on real-time bandwidth estimations to maintain connection stability. The capability surface includes media track management for audio, video, and screen capture, as well as bidirectional data exchange via SCTP channels. It handles connection negotiation and secure session establishment across diverse platforms and browsers using WebRTC and RTP transport. The system also includes observability tools for tracking real-time stream statistics and active speaker data.
Mediasoup is a powerful WebRTC selective forwarding unit that provides the core media routing and transport logic required to build a multi-party conferencing system, though it functions as a library to be integrated into an application rather than a pre-built, ready-to-deploy server.
VDO.Ninja is a low-latency peer-to-peer media routing service and video streaming platform designed to integrate remote audio and video feeds into professional production workflows. It functions as a WebRTC broadcast integration tool and studio controller, allowing for the direct transmission of high-definition media between publishers and viewers with minimal delay. The platform distinguishes itself through extensive protocol bridging, converting between WebRTC, WHIP, WHEP, SRT, and RTMP to ensure compatibility across diverse network environments and professional studio software. It includes a director-led guest management system that organizes participants via virtual lobbies, waiting rooms, and access controls to coordinate remote guests. The system covers a broad range of capabilities, including professional audio routing to virtual cables and digital audio workstations, hardware control for PTZ cameras via MIDI and WebHID, and real-time visual effects such as machine-learning background removal. It also provides comprehensive recording tools for multitrack audio and headless cloud capture, alongside programmatic APIs for managing session orchestration and media routing. The application can be self-hosted on private HTTPS servers and supports specialized deployments on embedded Linux devices and Nvidia Jetson hardware.
VDO.Ninja is a self-hostable WebRTC-based platform that provides low-latency media routing and multi-party conferencing features, making it a highly capable tool for real-time video streaming and production.