webrtc_dart - WebRTC for Dart #

webrtc_dart is a pure Dart implementation of WebRTC. No native dependencies, no browser required - just add to your pubspec.yaml and go!

What is webrtc_dart? #

webrtc_dart lets you build WebRTC applications entirely in Dart. Whether you're creating a media server, an SFU (Selective Forwarding Unit), a signaling server, or peer-to-peer data channels - webrtc_dart has you covered.

Unlike browser-based WebRTC, webrtc_dart gives you direct access to RTP packets, making it perfect for building custom media servers, recording solutions, and real-time communication backends.

This is a complete port of werift-webrtc (TypeScript) to Dart.

Quick Start #

Installation #

Add to your pubspec.yaml:

dependencies:
  webrtc_dart: ^0.22.4

Then run:

dart pub get

Your First DataChannel #

Here's a simple example of creating a peer connection and data channel:

import 'package:webrtc_dart/webrtc_dart.dart';

void main() async {
  // Create a new peer connection
  final pc = RtcPeerConnection(RtcConfiguration(
    iceServers: [IceServer(urls: ['stun:stun.l.google.com:19302'])],
  ));

  // Wait for transport initialization (DTLS certificate generation)
  await Future.delayed(Duration(milliseconds: 500));

  // Create a data channel
  final dataChannel = pc.createDataChannel('chat');

  // Handle data channel events
  dataChannel.onStateChange.listen((state) {
    if (state == DataChannelState.open) {
      print('Data channel is open!');
      dataChannel.sendString('Hello from webrtc_dart!');
    }
  });

  dataChannel.onMessage.listen((message) {
    print('Received: $message');
  });

  // Handle ICE candidates
  pc.onIceCandidate.listen((candidate) {
    // Send this candidate to the remote peer via your signaling server
    print('New ICE candidate: ${candidate.toSdp()}');
  });

  // Create an offer
  final offer = await pc.createOffer();
  await pc.setLocalDescription(offer);

  // Send the offer to your remote peer via signaling...
  print('Offer SDP: ${offer.sdp}');
}

Features #

webrtc_dart supports a comprehensive set of WebRTC features:

Browser Compatibility #

webrtc_dart is tested and compatible with:

• Chrome
• Firefox
• Safari
• Other WebRTC implementations (Pion, aiortc, werift)

Examples #

Example 1: Local DataChannel Test #

Connect two peer connections locally and exchange messages:

import 'package:webrtc_dart/webrtc_dart.dart';

void main() async {
  // Create two peer connections
  final pcOffer = RtcPeerConnection();
  final pcAnswer = RtcPeerConnection();

  // Exchange ICE candidates
  pcOffer.onIceCandidate.listen((candidate) async {
    await pcAnswer.addIceCandidate(candidate);
  });

  pcAnswer.onIceCandidate.listen((candidate) async {
    await pcOffer.addIceCandidate(candidate);
  });

  // Handle incoming data channel on answer side
  pcAnswer.onDataChannel.listen((channel) {
    channel.onMessage.listen((message) {
      print('Answer received: $message');
      channel.sendString('Hello back!');
    });
  });

  // Perform offer/answer exchange
  final offer = await pcOffer.createOffer();
  await pcOffer.setLocalDescription(offer);
  await pcAnswer.setRemoteDescription(offer);

  final answer = await pcAnswer.createAnswer();
  await pcAnswer.setLocalDescription(answer);
  await pcOffer.setRemoteDescription(answer);

  // Wait for connection
  await Future.delayed(Duration(seconds: 2));

  // Create and use data channel
  final dc = pcOffer.createDataChannel('chat');
  dc.onStateChange.listen((state) {
    if (state == DataChannelState.open) {
      dc.sendString('Hello from offer side!');
    }
  });

  dc.onMessage.listen((message) {
    print('Offer received: $message');
  });
}

Run this example:

dart run example/quickstart_local.dart

Example 2: Receiving Media (Video/Audio) #

webrtc_dart excels at receiving media streams. Here's how to set up for receiving media:

import 'package:webrtc_dart/webrtc_dart.dart';

void main() async {
  // Create a peer connection
  final pc = RtcPeerConnection(RtcConfiguration(
    iceServers: [IceServer(urls: ['stun:stun.l.google.com:19302'])],
  ));

  // Listen for incoming tracks
  pc.onTrack.listen((transceiver) {
    print('Received ${transceiver.kind} track!');
    print('  mid: ${transceiver.mid}');
    print('  direction: ${transceiver.direction}');
  });

  // Create offer
  final offer = await pc.createOffer();
  await pc.setLocalDescription(offer);

  print('Peer connection ready to receive media');
  print('Offer SDP created (send to remote peer):');
  print(offer.sdp);
}

Run this example:

dart run example/quickstart_media.dart

Example 3: Using STUN/TURN Servers #

Configure ICE servers for NAT traversal:

import 'package:webrtc_dart/webrtc_dart.dart';

void main() async {
  final pc = RtcPeerConnection(RtcConfiguration(
    iceServers: [
      // STUN server
      IceServer(urls: ['stun:stun.l.google.com:19302']),

      // TURN server (UDP)
      IceServer(
        urls: ['turn:your-turn-server.com:3478'],
        username: 'user',
        credential: 'password',
      ),
    ],
  ));

  print('Peer connection created with STUN/TURN configuration');
  print('ICE servers configured:');
  for (final server in pc.configuration.iceServers) {
    print('  - ${server.urls}');
  }
}

Run this example:

dart run example/quickstart_turn.dart

Example 4: ICE Restart #

Reconnect without creating a new peer connection:

import 'package:webrtc_dart/webrtc_dart.dart';

void main() async {
  final pc = RtcPeerConnection();

  // ... after connection is established and network changes ...

  // Trigger ICE restart
  pc.restartIce();

  // Create a new offer with ICE restart flag
  final offer = await pc.createOffer();
  await pc.setLocalDescription(offer);

  // Send the new offer to the remote peer for renegotiation
  print('ICE restart triggered');
  print('New offer SDP for renegotiation:');
  print(offer.sdp);
}

Run this example:

dart run example/quickstart_ice_restart.dart

More Examples #

Architecture #

webrtc_dart is designed with a modular architecture:

webrtc_dart
├── ICE      - Interactive Connectivity Establishment
├── DTLS     - Datagram Transport Layer Security
├── SCTP     - Stream Control Transmission Protocol (DataChannels)
├── RTP      - Real-time Transport Protocol (Media)
├── RTCP     - RTP Control Protocol
├── SRTP     - Secure RTP
└── SDP      - Session Description Protocol

Code Structure #

lib/src/
├── stun/          # STUN/TURN protocol
├── ice/           # ICE agent and candidates
├── dtls/          # DTLS handshake and encryption
├── srtp/          # SRTP/SRTCP encryption
├── rtp/           # RTP/RTCP media transport
├── sctp/          # SCTP reliable transport
├── datachannel/   # DataChannel API
├── media/         # Media tracks and transceivers
├── codec/         # Codec implementations
├── sdp/           # SDP parsing and generation
├── stats/         # Statistics collection
├── container/     # WebM/MP4 recording
└── peer_connection.dart  # Main PeerConnection API

Design Philosophy #

• Pure Dart: No native bindings or browser dependencies
• Direct RTP Access: Full control over media packets for custom processing
• Browser-Compatible API: Familiar RTCPeerConnection-style interface
• Modular: Use only what you need

Test Coverage #

2171 tests passing with 80% code coverage.

# Run all tests
dart test

# Run specific test suite
dart test test/ice/
dart test test/dtls/
dart test test/peer_connection_test.dart

Browser Interop Testing #

Automated tests verify compatibility with Chrome, Firefox, and Safari:

cd interop
npm install
npm test              # Test all browsers
npm run test:chrome   # Test Chrome only
npm run test:firefox  # Test Firefox only
npm run test:safari   # Test Safari/WebKit only

Configuration Options #

import 'package:webrtc_dart/webrtc_dart.dart';

final pc = RtcPeerConnection(RtcConfiguration(
  // ICE configuration
  iceServers: [
    IceServer(urls: ['stun:stun.l.google.com:19302']),
  ],

  // Supported codecs
  codecs: RtcCodecConfiguration(
    video: [
      RtcRtpCodecParameters(
        mimeType: 'video/VP8',
        clockRate: 90000,
      ),
    ],
    audio: [
      RtcRtpCodecParameters(
        mimeType: 'audio/opus',
        clockRate: 48000,
        channels: 2,
      ),
    ],
  ),

  // RTP header extensions
  headerExtensions: RtcHeaderExtensions(
    video: [/* ... */],
    audio: [/* ... */],
  ),
));

Related Projects #

• werift-webrtc - The original TypeScript implementation this project is ported from
• node-sfu - A complete SFU built with werift

Contributing #

Contributions are welcome! Check out the examples directory for usage patterns.

# Format code
dart format .

# Analyze code
dart analyze

# Run tests
dart test

License #

MIT License - see LICENSE for details.

Acknowledgments #

This project is a Dart port of werift-webrtc by Yuki Shindo.

webrtc_dart is inspired by and references:

• werift-webrtc - TypeScript WebRTC implementation
• aiortc - Python WebRTC implementation
• pion/webrtc - Go WebRTC implementation