pure_dart_webrtc 1.0.0

A pure Dart implementation of WebRTC, including support for ICE, STUN, TURN, DTLS, TLS, SRTP, RTP, RTCP, and SDP.

dart_webrtc

dart_webrtc (WebrTC Implementation for The Dart language)

A pure Dart implementation of WebRTC, including support for ICE, STUN, TURN, DTLS, TLS, SRTP, RTP, RTCP, and SDP.

Inspired by webrtc-nuts-and-bolts

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📦 Installation

clone the repo

Then run:

dart pub get

Requires Dart 3.x or later.

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📚 Documentation (WIP)

• API Reference
• Example usage in the /bin directory

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💡 Examples

GitHub: examples

DTLS server

dart bin/srtp_webrtc.dart
dart bin/dart_webrtc.dart
dart lib/src/dtls/examples/server/psk_ccm8.dart
dart lib/src/dtls/examples/server/psk_ccm.dart

SRTP/DTLS server

dart bin/srtp_webrtc.dart

DTLS client

dart lib/src/dtls3/handshaker/client/dtls_client.dart

STUN Server

dart lib/src/stun3/stun_server7.dart

STUN client

dart lib/src/stun3/stun_client.dart

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STUN/DTLS/SRTP/RTCP multiplexing

This example recieves stun, dtls, rtp packets. It decrypts, encrypts and sends the packets back to the sender. copy the sdp offer from this file:

lib\src\sdp8\sdp_test.dart

Adjust the the fingerprint and ice candidate

cd WebRTC-Simple-SDP-Handshake-Demo
php -S localhost:3000

Navigate to the link in your browser=> localhost:3000 Paste the sdp in the sdp offer box

Run this command

dart bin\srtp_webrtc.dart

In your browser, click create answer

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🎯 Roadmap

✅ Version 1.0 Goals

Signaling & NAT traversal

• ✅ SIP
• ✅ STUN
• ❌ TURN (UDP)

ICE

• ✅ Vanilla ICE
• ✅ Trickle ICE
• ❌ ICE Restart
• ✅ ICE-Lite (Client-side)
• ❌ ICE-Lite (Server-side)

Security

• ✅ DTLS
  • ✅ DTLS-SRTP
  • ✅ Curve25519
  • ✅ P-256
• ❌ TLS 1.2

Channels

• ❌ DataChannel
• ❌ MediaChannel
  • ❌ sendonly
  • ❌ recvonly
  • ❌ sendrecv
  • ❌ Multi-track
  • ❌ RTX
  • ❌ RED

RTP/RTCP

• ❌ RFC 3550 (RTP base)
• ❌ RTP Payload Formats:
  • ❌ VP8
  • ❌ VP9
  • ❌ H264
  • ❌ AV1
  • ❌ RED (RFC 2198)
• ❌ RTCP:
  • ❌ SR/RR
  • ❌ PLI
  • ❌ REMB
  • ❌ NACK
  • ❌ TransportWideCC

SDP / PeerConnection

• ✅ SDP parsing and generation
  • ✅ Reuse inactive m-line
• ❌ PeerConnection API
• ❌ Simulcast (recv only)
• ❌ Bandwidth Estimation (sender-side)

Media Recorder

• ❌ OPUS
• ❌ VP8
• ❌ VP9
• ❌ H264
• ❌ AV1

Compatibility & Interop

• ✅ Chrome / Edge / Firefox
• ❌ Pion
• ❌ aiortc
• ❌ sipsorcery
• ✅ webrtc-rs
• ❌ Interop E2E testing

Testing

• ❌ Unit tests
• ❌ Web Platform Tests

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🔜 Roadmap for 2.0

• ❌ API compatible with browser RTCPeerConnection
• ❌ Simulcast (send support)
• ❌ TURN over TCP
• ❌ getStats() support
• ❌ Support for more cipher suites

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🔗 References

• aiortc (Python)
• pion/webrtc (Go)
• sipsorcery (C#)
• webrtc-rs (Rust)