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Cutting Edge WebRTC Video Conferencing

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Top Related Projects

13,375

Pure Go implementation of the WebRTC API

Jitsi Meet - Secure, Simple and Scalable Video Conferences that you use as a standalone app or embed in your web application.

Janus WebRTC Server

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End-to-end stack for WebRTC. SFU media server and SDKs.

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coturn TURN server project

📡 Simple WebRTC video, voice, and data channels

Quick Overview

Mediasoup is a powerful WebRTC SFU (Selective Forwarding Unit) for Node.js. It enables developers to build scalable video conferencing, broadcasting, and real-time communication applications with advanced features like simulcast and SVC (Scalable Video Coding).

Pros

  • High performance and scalability, capable of handling numerous concurrent connections
  • Supports modern WebRTC features like simulcast and SVC for adaptive video quality
  • Flexible API design allowing integration with various signaling protocols and architectures
  • Active development and community support

Cons

  • Steep learning curve due to its complex architecture and WebRTC concepts
  • Requires significant server resources for large-scale deployments
  • Limited documentation for advanced use cases and troubleshooting
  • May require additional components (like a signaling server) for a complete solution

Code Examples

Creating a WebRTC Router:

const mediasoup = require('mediasoup');

const worker = await mediasoup.createWorker();
const router = await worker.createRouter({
  mediaCodecs: [
    {
      kind: 'audio',
      mimeType: 'audio/opus',
      clockRate: 48000,
      channels: 2
    },
    {
      kind: 'video',
      mimeType: 'video/VP8',
      clockRate: 90000,
      parameters: {
        'x-google-start-bitrate': 1000
      }
    }
  ]
});

Creating a WebRTC Transport:

const transport = await router.createWebRtcTransport({
  listenIps: [
    {
      ip: '0.0.0.0',
      announcedIp: '192.168.1.100'
    }
  ],
  enableUdp: true,
  enableTcp: true,
  preferUdp: true
});

Consuming a remote stream:

const consumer = await transport.consume({
  producerId: remoteProducerId,
  rtpCapabilities: consumerRtpCapabilities
});

// Handle consumer events
consumer.on('transportclose', () => {
  console.log('Transport closed');
});

Getting Started

  1. Install mediasoup:

    npm install mediasoup
    
  2. Create a basic server:

    const mediasoup = require('mediasoup');
    const http = require('http');
    
    (async () => {
      const worker = await mediasoup.createWorker();
      const router = await worker.createRouter({
        mediaCodecs: [
          {
            kind: 'audio',
            mimeType: 'audio/opus',
            clockRate: 48000,
            channels: 2
          },
          {
            kind: 'video',
            mimeType: 'video/VP8',
            clockRate: 90000
          }
        ]
      });
    
      const server = http.createServer();
      server.listen(3000, () => {
        console.log('Server listening on port 3000');
      });
    
      // Implement your signaling logic here
    })();
    
  3. Implement signaling and client-side logic to establish WebRTC connections using the mediasoup API.

Competitor Comparisons

13,375

Pure Go implementation of the WebRTC API

Pros of webrtc

  • Written in Go, offering better performance and easier deployment compared to Node.js-based mediasoup
  • More comprehensive WebRTC implementation, including STUN/TURN functionality
  • Active community with frequent updates and contributions

Cons of webrtc

  • Less focused on SFU (Selective Forwarding Unit) capabilities than mediasoup
  • May require more manual configuration for advanced use cases
  • Potentially steeper learning curve for developers not familiar with Go

Code Comparison

mediasoup (JavaScript):

const mediasoup = require('mediasoup');
const worker = await mediasoup.createWorker();
const router = await worker.createRouter({ mediaCodecs });
const transport = await router.createWebRtcTransport(webRtcTransportOptions);

webrtc (Go):

import "github.com/pion/webrtc/v3"

peerConnection, err := webrtc.NewPeerConnection(webrtc.Configuration{})
dataChannel, err := peerConnection.CreateDataChannel("data", nil)
offer, err := peerConnection.CreateOffer(nil)

Both libraries provide APIs for creating and managing WebRTC connections, but mediasoup focuses on SFU functionality while webrtc offers a more general-purpose WebRTC implementation. The choice between them depends on specific project requirements and the preferred programming language.

Jitsi Meet - Secure, Simple and Scalable Video Conferences that you use as a standalone app or embed in your web application.

Pros of Jitsi Meet

  • Full-featured, ready-to-use video conferencing solution
  • Extensive documentation and community support
  • Includes a complete user interface and additional features like chat and screen sharing

Cons of Jitsi Meet

  • Less flexible for custom implementations
  • Heavier resource usage due to its comprehensive feature set
  • Steeper learning curve for developers wanting to modify core functionality

Code Comparison

Mediasoup (WebRTC transport creation):

const transport = await router.createWebRtcTransport({
  listenIps: [{ ip: '192.168.1.100', announcedIp: null }],
  enableUdp: true,
  enableTcp: true,
  preferUdp: true
});

Jitsi Meet (Conference join):

const options = {
    roomName: 'MyConferenceRoom',
    width: 700,
    height: 700,
    parentNode: document.querySelector('#meet')
};
const api = new JitsiMeetExternalAPI(domain, options);

Both projects are powerful WebRTC solutions, but they serve different purposes. Mediasoup is a flexible WebRTC SFU library for building custom solutions, while Jitsi Meet is a complete video conferencing platform. The choice between them depends on the specific requirements of your project and the level of customization needed.

Janus WebRTC Server

Pros of Janus-Gateway

  • More mature and widely adopted, with a larger community and ecosystem
  • Supports a broader range of protocols and use cases out-of-the-box
  • Offers a plugin architecture for easy extensibility

Cons of Janus-Gateway

  • Generally considered more complex to set up and configure
  • Can be less performant in high-scale scenarios compared to Mediasoup
  • C-based core may be less accessible for developers primarily working with web technologies

Code Comparison

Janus-Gateway (C):

janus_plugin *create(void) {
    janus_plugin *plugin = (janus_plugin *)calloc(1, sizeof(janus_plugin));
    plugin->init = janus_echotest_init;
    plugin->destroy = janus_echotest_destroy;
    plugin->handle_message = janus_echotest_handle_message;
    return plugin;
}

Mediasoup (JavaScript):

const mediasoup = require('mediasoup');

const worker = await mediasoup.createWorker({
  logLevel: 'warn',
  rtcMinPort: 10000,
  rtcMaxPort: 10100
});

Both projects are powerful WebRTC solutions, but they cater to different needs. Janus-Gateway offers a more comprehensive feature set and flexibility, while Mediasoup provides a more modern, JavaScript-centric approach with potentially better performance in certain scenarios.

9,473

End-to-end stack for WebRTC. SFU media server and SDKs.

Pros of LiveKit

  • More comprehensive solution with server and client SDKs for multiple languages
  • Built-in support for recording and selective forwarding unit (SFU)
  • Active development with frequent updates and releases

Cons of LiveKit

  • Less flexible for custom implementations compared to mediasoup
  • Steeper learning curve due to more complex architecture
  • Potentially higher resource usage for smaller-scale applications

Code Comparison

mediasoup (JavaScript):

const worker = await mediasoup.createWorker();
const router = await worker.createRouter({ mediaCodecs });
const transport = await router.createWebRtcTransport(webRtcTransportOptions);

LiveKit (Go):

room, err := server.CreateRoom(&livekit.CreateRoomRequest{
    Name: "my-room",
})
participant, err := room.CreateParticipant("user123", &livekit.ParticipantInfo{
    Identity: "John Doe",
})

Summary

mediasoup offers a more flexible and lightweight approach, ideal for developers who need fine-grained control over their WebRTC infrastructure. LiveKit provides a more comprehensive, out-of-the-box solution with additional features like recording and SFU support. The choice between the two depends on project requirements, scalability needs, and development preferences.

11,072

coturn TURN server project

Pros of coturn

  • Widely adopted and battle-tested TURN server implementation
  • Supports multiple protocols (TURN, STUN, ICE) and transport methods
  • Lightweight and can handle a large number of concurrent connections

Cons of coturn

  • Focused solely on TURN/STUN functionality, not a complete WebRTC solution
  • Requires additional components for a full WebRTC stack
  • Less active development compared to mediasoup

Code comparison

coturn (simple TURN server configuration):

listening-port=3478
listening-ip=0.0.0.0
realm=example.com
user=username:password

mediasoup (creating a WebRTC transport):

const transport = await router.createWebRtcTransport({
  listenIps: [{ ip: '0.0.0.0', announcedIp: publicIp }],
  enableUdp: true,
  enableTcp: true
});

Summary

coturn is a specialized TURN/STUN server, while mediasoup is a full-featured WebRTC SFU (Selective Forwarding Unit). coturn excels in providing NAT traversal solutions, whereas mediasoup offers a more comprehensive WebRTC infrastructure. The choice between them depends on the specific requirements of your project and whether you need a complete WebRTC solution or just NAT traversal capabilities.

📡 Simple WebRTC video, voice, and data channels

Pros of simple-peer

  • Easier to set up and use, with a simpler API
  • Lightweight and focused on basic WebRTC peer-to-peer connections
  • Better suited for small-scale or hobby projects

Cons of simple-peer

  • Limited scalability for large-scale applications
  • Fewer advanced features and customization options
  • Less suitable for complex, multi-party video conferencing scenarios

Code Comparison

simple-peer example:

const peer = new SimplePeer({ initiator: true })
peer.on('signal', data => {
  // send signal data to peer
})
peer.on('connect', () => {
  peer.send('hello')
})

mediasoup example:

const router = await mediasoupWorker.createRouter({ mediaCodecs })
const transport = await router.createWebRtcTransport(webRtcTransportOptions)
const producer = await transport.produce({ kind: 'video', rtpParameters })
const consumer = await transport.consume({ producerId: producer.id })

Summary

simple-peer is more straightforward and easier to use for basic WebRTC connections, making it ideal for small projects or quick prototypes. However, it lacks the advanced features and scalability of mediasoup, which is better suited for complex, large-scale video conferencing applications that require more control and customization.

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README

mediasoup v3

Website and Documentation

Support Forum

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).

Authors

Social

Sponsor

You can support mediasoup by sponsoring it. Thanks!

License

ISC

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