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muaz-khan/RTCMultiConnection
RTCMultiConnection is a WebRTC JavaScript library for peer-to-peer applications (screen sharing, audio/video conferencing, file sharing, media streaming etc.)
RTCMultiConnection is a WebRTC abstraction library that wraps peer-to-peer connection management, signaling, and media handling into a single JS object. It targets developers who want screen sharing, audio/video calls, or file transfer without wiring up WebRTC's signaling state machine by hand. The target audience is small-project developers who don't want to build on top of raw RTCPeerConnection.
Broad feature surface in one library — screen sharing, file transfer, multi-party conferencing, and broadcasting all covered without pulling in separate packages. Pluggable signaling backends (Socket.io, Firebase, PubNub, SignalR, SSE, raw WebSocket) means you're not locked into one transport. The modular dev/ source structure makes it readable — each concern (ICE, codecs, recording, streams) lives in its own file. Demo coverage is genuinely good: 15+ working HTML demos covering common use cases, which is useful for understanding the API quickly.
Dead project — last commit was October 2022, and the WebRTC API has moved on since then. Chrome and Firefox have deprecated or changed several APIs this library relies on, so you will hit runtime failures on modern browsers without patching. No TypeScript types, and the API surface is a sprawling global object with dozens of event hooks and config properties, making IDE support nearly nonexistent. The signaling server is a separate repo with its own setup, which means two things to deploy, two things to break, and the docs between them are inconsistent.