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Cutting Edge WebRTC Video Conferencing

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mediasoup v3

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Design Goals

mediasoup and its client side libraries are designed to accomplish with the following goals:

  • Be a SFU (Selective Forwarding Unit).
  • Support both WebRTC and plain RTP input and output.
  • Be a Node.js module or Rust crate in server side.
  • Be a tiny TypeScript and C++ libraries in client side.
  • Be minimalist: just handle the media layer.
  • Be signaling agnostic: do not mandate any signaling protocol.
  • Be super low level API.
  • Support all existing WebRTC endpoints.
  • Enable integration with well known multimedia libraries/tools.

Architecture

Use Cases

mediasoup and its client side libraries provide a super low level API. They are intended to enable different use cases and scenarios, without any constraint or assumption. Some of these use cases are:

  • Group video chat applications.
  • One-to-many (or few-to-many) broadcasting applications in real-time.
  • RTP streaming.

Features

  • ECMAScript 6/Idiomatic Rust low level API.
  • Multi-stream: multiple audio/video streams over a single ICE + DTLS transport.
  • IPv6 ready.
  • ICE / DTLS / RTP / RTCP over UDP and TCP.
  • Simulcast and SVC support.
  • Congestion control.
  • Sender and receiver bandwidth estimation with spatial/temporal layers distribution algorithm.
  • Data message exchange (via WebRTC DataChannels, SCTP over plain UDP, and direct termination in Node.js/Rust).
  • Extremely powerful (media worker thread/subprocess coded in C++ on top of libuv).

Demo Online

Try it at v3demo.mediasoup.org (source code).

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You can support mediasoup by sponsoring it. Thanks!

License

ISC

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