Embed real-time video conferencing and audio communication features directly into your custom web or mobile applications.
Jitsi Meet is an open-source platform for real-time audio and video communication. It provides a complete infrastructure for hosting secure video conferences, supporting features such as screen sharing, messaging, and participant polling. The platform is designed for both standalone use and integration into external web or mobile applications. The system utilizes a selective forwarding unit architecture to route media streams between participants, ensuring efficient communication across multiple users. It relies on standardized real-time transport protocols to manage data transmission and includes mechanisms for network path negotiation to bypass firewalls and network address translation. Security is maintained through the implementation of end-to-end encryption and standard protocols to protect the privacy of communication sessions. The platform offers extensive configuration and deployment options, allowing for self-hosted installations on private servers or scalable deployments within cloud environments. It supports infrastructure management through containerized microservices and load balancing to maintain performance during high usage. Developers can extend the platform's functionality through programmatic interfaces, including software development kits and sandboxed interface injection, to align the communication experience with specific organizational requirements.
Jitsi Meet is a comprehensive, self-hostable WebRTC platform that provides both a standalone application and the necessary SDKs to integrate multi-party video, audio, screen sharing, and chat into your own custom software.
PeerJS is a real-time communication library and framework designed to establish direct peer-to-peer data and media connections between browsers. It provides a simplified wrapper for managing the lifecycles of WebRTC connectivity, enabling the exchange of audio, video, and data without a central relay. The project includes a coordination server that handles signaling management by mapping unique user identities to network connection details. This server facilitates the discovery of remote peers and the exchange of metadata required to perform the initial connection handshake. The framework supports several communication patterns, including real-time video calling, decentralized data messaging, and direct binary file transfers. It utilizes an event-driven system to manage connection states and employs heartbeat monitoring to track the health of active peer sessions.
PeerJS provides a robust WebRTC wrapper and signaling server that enables the integration of real-time audio and video communication into custom applications, though it focuses on peer-to-peer connectivity rather than providing a pre-built multi-party conferencing suite.
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 provides the necessary infrastructure for real-time audio and video conferencing, though it functions primarily as a server-side platform rather than a client-side SDK.
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 project is a specialized screen-sharing and signaling server rather than a comprehensive SDK for building full-featured video and audio conferencing applications.
Deskreen is a wireless screen mirroring and virtual display tool that streams a computer screen or specific application windows to any device with a web browser. It functions as a virtual display streamer and a web-based secondary monitor, allowing users to extend their desktop workspace to remote devices over a local network. The system supports end-to-end encrypted screen sharing to protect display data and utilizes virtual display adapters to treat remote browsers as extended screens. It includes capabilities for multi-device broadcasting, enabling a single video source to be mirrored across several connected devices simultaneously. The tool provides security and session management through connection password protection, trusted device restrictions, and active connection monitoring. Additional utility features include real-time video quality optimization and screen orientation flipping to support teleprompter hardware.
This tool is a specialized application for screen mirroring and virtual display extension rather than a general-purpose SDK or library for building custom video conferencing platforms.
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 remote desktop and browser isolation platform rather than a general-purpose video conferencing SDK, though it utilizes WebRTC to stream shared sessions and input.
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 powerful WebRTC-based media routing platform that provides the necessary APIs and self-hostable infrastructure to integrate professional-grade, low-latency video and audio conferencing into custom production workflows.
Coturn is a network server that facilitates peer-to-peer media traffic for real-time communication applications. It functions as a relay platform for voice, video, and data transmission, enabling direct connections between clients located behind restrictive firewalls and network address translators. The server implements standard network traversal protocols to manage media packet exchange and client authentication. It utilizes a multi-threaded architecture and event-driven polling to handle high-throughput traffic, while employing hash-based message authentication codes to verify client identity and secure access to relay services. The platform includes a modular interface for persistent storage of credentials and server state across various database backends. It also provides integrated monitoring capabilities to track traffic volume, connection status, and operational health metrics, allowing for the identification of performance bottlenecks in distributed communication environments.
This is a STUN/TURN relay server that provides the necessary network traversal infrastructure for WebRTC, but it is a low-level networking component rather than a full-featured SDK for building conferencing applications.