<h1 align="center">
<a href="https://webrtc.rs"><img src="https://raw.githubusercontent.com/webrtc-rs/webrtc-rs.github.io/master/res/rtc.png" alt="WebRTC.rs"></a>
<br>
</h1>
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<p align="center">
<strong>Sans-I/O WebRTC implementation in Rust</strong>
</p>
<p align="center">
<strong>Sponsored with 💖 by</strong><br>
</p>
<p align="center">
<strong>Gold Sponsors:</strong><br>
<a href="https://www.recall.ai" target="_blank">
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/recall.svg"
alt="Recall.ai">
</a><br>
<p align="center">
<strong>Silver Sponsors:</strong><br>
<a href="https://getstream.io/video/voice-calling/?utm_source=https://github.com/webrtc-rs/webrtc&utm_medium=sponsorship&utm_content=&utm_campaign=webrtcRepo_July2023_video_klmh22" target="_blank">
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<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/ChannelTalk_logo.png" alt="ChannelTalk">
</a><br>
<strong>Bronze Sponsors:</strong><br>
<a href="https://github.com/AdrianEddy" target="_blank">AdrianEddy</a><br>
</p>
<details>
<summary><b>Table of Content</b></summary>
- [Overview](#overview)
- [Features](#features)
- [Building](#building)
- [Open Source License](#open-source-license)
- [Contributing](#contributing)
</details>
## Overview
**RTC** is a pure Rust implementation of [WebRTC](https://www.w3.org/TR/webrtc/) using a **sans-I/O architecture**.
Unlike traditional WebRTC libraries, RTC separates protocol logic from I/O operations, giving you complete control over
networking, threading, and async runtime integration.
### What is Sans-I/O?
Sans-I/O (without I/O) is a design pattern where the library handles protocol logic but **you** control all I/O
operations. Instead of the library performing network reads and writes directly, you feed it network data and it tells
you what to send.
**Benefits:**
- 🚀 **Runtime Independent** - Works with tokio, async-std, smol, or blocking I/O
- 🎯 **Full Control** - You control threading, scheduling, and I/O multiplexing
- 🧪 **Testable** - Protocol logic can be tested without real network I/O
- 🔌 **Flexible** - Easy integration with existing networking code
## Sans-I/O Event Loop Pattern
The sans-I/O architecture uses a simple event loop with six core methods:
### Core API Methods
1. **`poll_write()`** - Get outgoing network packets to send via UDP
2. **`poll_event()`** - Process connection state changes and notifications
3. **`poll_read()`** - Get incoming application messages (RTP, RTCP, data)
4. **`poll_timeout()`** - Get next timer deadline for retransmissions/keepalives
5. **`handle_read()`** - Feed incoming network packets into the connection
6. **`handle_timeout()`** - Notify about timer expiration
Additional methods for external control:
* handle_write() - Queue application messages (RTP/RTCP/data) for sending
* handle_event() - Inject external events into the connection
### Event Loop Example
```rust
use rtc::peer_connection::RTCPeerConnection;
use rtc::peer_connection::configuration::RTCConfigurationBuilder;
use rtc::peer_connection::event::{RTCPeerConnectionEvent, RTCTrackEvent};
use rtc::peer_connection::state::RTCPeerConnectionState;
use rtc::peer_connection::message::RTCMessage;
use rtc::peer_connection::sdp::RTCSessionDescription;
use rtc::shared::{TaggedBytesMut, TransportContext, TransportProtocol};
use rtc::sansio::Protocol;
use std::time::{Duration, Instant};
use tokio::net::UdpSocket;
use bytes::BytesMut;
#[tokio::main]
async fn main() -> Result<(), Box<dyn std::error::Error>> {
// Setup peer connection
let config = RTCConfigurationBuilder::new().build();
let mut pc = RTCPeerConnection::new(config)?;
// Signaling: Create offer and set local description
let offer = pc.create_offer(None)?;
pc.set_local_description(offer.clone())?;
// TODO: Send offer.sdp to remote peer via your signaling channel
// signaling_channel.send_offer(&offer.sdp).await?;
// TODO: Receive answer from remote peer via your signaling channel
// let answer_sdp = signaling_channel.receive_answer().await?;
// let answer = RTCSessionDescription::answer(answer_sdp)?;
// pc.set_remote_description(answer)?;
// Bind UDP socket
let socket = UdpSocket::bind("0.0.0.0:0").await?;
let local_addr = socket.local_addr()?;
let mut buf = vec![0u8; 2000];
'EventLoop: loop {
// 1. Send outgoing packets
while let Some(msg) = pc.poll_write() {
socket.send_to(&msg.message, msg.transport.peer_addr).await?;
}
// 2. Handle events
while let Some(event) = pc.poll_event() {
match event {
RTCPeerConnectionEvent::OnConnectionStateChangeEvent(state) => {
println!("Connection state: {state}");
if state == RTCPeerConnectionState::Failed {
return Ok(());
}
}
RTCPeerConnectionEvent::OnTrack(RTCTrackEvent::OnOpen(init)) => {
println!("New track: {}", init.track_id);
}
_ => {}
}
}
// 3. Handle incoming messages
while let Some(message) = pc.poll_read() {
match message {
RTCMessage::RtpPacket(track_id, packet) => {
println!("RTP packet on track {track_id}");
}
RTCMessage::DataChannelMessage(channel_id, msg) => {
println!("Data channel message");
}
_ => {}
}
}
// 4. Handle timeouts
let timeout = pc.poll_timeout()
.unwrap_or(Instant::now() + Duration::from_secs(86400));
let delay = timeout.saturating_duration_since(Instant::now());
if delay.is_zero() {
pc.handle_timeout(Instant::now())?;
continue;
}
// 5. Multiplex I/O
tokio::select! {
_ = stop_rx.recv() => {
break 'EventLoop,
}
_ = tokio::time::sleep(delay) => {
pc.handle_timeout(Instant::now())?;
}
Ok(message) = message_rx.recv() => {
pc.handle_write(message)?;
}
Ok(event) = event_rx.recv() => {
pc.handle_event(event)?;
}
Ok((n, peer_addr)) = socket.recv_from(&mut buf) => {
pc.handle_read(TaggedBytesMut {
now: Instant::now(),
transport: TransportContext {
local_addr,
peer_addr,
ecn: None,
transport_protocol: TransportProtocol::UDP,
},
message: BytesMut::from(&buf[..n]),
})?;
}
}
}
pc.close()?;
Ok(())
}
```
## Features
- ✅ **ICE** (Interactive Connectivity Establishment) - NAT traversal with STUN/TURN
- ✅ **DTLS** (Datagram Transport Layer Security) - Encryption for media and data
- ✅ **SCTP** (Stream Control Transmission Protocol) - Reliable data channels
- ✅ **RTP/RTCP** - Real-time media transport and control
- ✅ **SDP** (Session Description Protocol) - Offer/answer negotiation
- ✅ **Data Channels** - Bidirectional peer-to-peer data transfer
- ✅ **Media Tracks** - Audio/video transmission
- ✅ **Trickle ICE** - Progressive candidate gathering
- ✅ **Simulcast & SVC** - Scalable video coding
## Examples
The repository includes comprehensive examples demonstrating various use cases:
- [data-channels-offer-answer](https://github.com/webrtc-rs/rtc/tree/master/examples/examples/data-channels-offer-answer/) -
Complete data channel setup with
signaling
- [reflect](https://github.com/webrtc-rs/rtc/tree/master/examples/examples/reflect/) - Echo server that reflects
media back to sender
- [save-to-disk-vpx](https://github.com/webrtc-rs/rtc/tree/master/examples/examples/save-to-disk-vpx/) - Receive and
save VP8/VP9 video
- [play-from-disk-vpx](https://github.com/webrtc-rs/rtc/tree/master/examples/examples/play-from-disk-vpx/) - Send
VP8/VP9 video from disk
- [simulcast](https://github.com/webrtc-rs/rtc/tree/master/examples/examples/simulcast) - Receive 3
simulcast encodings in one track
- [broadcast](https://github.com/webrtc-rs/rtc/tree/master/examples/examples/broadcast) - Broadcast a video to multiple
peers
Run an example:
```bash
cargo run --example data-channels-offer-answer --features examples
```
## Architecture
RTC is built from composable crates, each implementing a specific protocol:
## RTC Crates
<p align="center">
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">RTC<a href="https://crates.io/crates/rtc"><img src="https://img.shields.io/crates/v/rtc.svg"></a>
<br>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">Media<a href="https://crates.io/crates/rtc-media"><img src="https://img.shields.io/crates/v/rtc-media.svg"></a>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">Interceptor<a href="https://crates.io/crates/rtc-interceptor"><img src="https://img.shields.io/crates/v/rtc-interceptor.svg"></a>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">DataChannel<a href="https://crates.io/crates/rtc-datachannel"><img src="https://img.shields.io/crates/v/rtc-datachannel.svg"></a>
<br>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">RTP<a href="https://crates.io/crates/rtc-rtp"><img src="https://img.shields.io/crates/v/rtc-rtp.svg"></a>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">RTCP<a href="https://crates.io/crates/rtc-rtcp"><img src="https://img.shields.io/crates/v/rtc-rtcp.svg"></a>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">SRTP<a href="https://crates.io/crates/rtc-srtp"><img src="https://img.shields.io/crates/v/rtc-srtp.svg"></a>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">SCTP<a href="https://crates.io/crates/rtc-sctp"><img src="https://img.shields.io/crates/v/rtc-sctp.svg"></a>
<br>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">DTLS<a href="https://crates.io/crates/rtc-dtls"><img src="https://img.shields.io/crates/v/rtc-dtls.svg"></a>
<br>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">mDNS<a href="https://crates.io/crates/rtc-mdns"><img src="https://img.shields.io/crates/v/rtc-mdns.svg"></a>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">STUN<a href="https://crates.io/crates/rtc-stun"><img src="https://img.shields.io/crates/v/rtc-stun.svg"></a>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">TURN<a href="https://crates.io/crates/rtc-turn"><img src="https://img.shields.io/crates/v/rtc-turn.svg"></a>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">ICE<a href="https://crates.io/crates/rtc-ice"><img src="https://img.shields.io/crates/v/rtc-ice.svg"></a>
<br>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">SDP<a href="https://crates.io/crates/rtc-sdp"><img src="https://img.shields.io/crates/v/rtc-sdp.svg"></a>
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc/master/doc/check.png">Shared<a href="https://crates.io/crates/rtc-shared"><img src="https://img.shields.io/crates/v/rtc-shared.svg"></a>
</p>
### Dependency Graph
<p align="center">
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc-rs.github.io/master/res/rtc_crates_dep_graph.png" alt="RTC Crates Dependency Graph">
</p>
### Protocol Stack
<p align="center">
<img src="https://raw.githubusercontent.com/webrtc-rs/webrtc-rs.github.io/master/res/rtc_stack.png" alt="RTC Protocols Stack">
</p>
## Common Use Cases
### Data Channels
```rust
use rtc::data_channel::RTCDataChannelInit;
fn example(mut pc: RTCPeerConnection) -> Result<(), Box<dyn std::error::Error>> {
// Create a data channel
let init = RTCDataChannelInit {
ordered: true,
max_retransmits: None,
..Default::default()
};
let mut dc = pc.create_data_channel("my-channel", Some(init))?;
// Send data
dc.send_text("Hello, WebRTC!")?;
Ok(())
}
```
### Media Tracks
```rust
use rtc::media_stream::MediaStreamTrack;
use rtc::rtp_transceiver::rtp_sender::{RTCRtpCodec, RtpCodecKind};
fn example(mut pc: RTCPeerConnection) -> Result<(), Box<dyn std::error::Error>> {
// Create a video track
let track = MediaStreamTrack::new(
"stream-id".to_string(),
"track-id".to_string(),
"Camera".to_string(),
RtpCodecKind::Video,
vec![RTCRtpEncodingParameters {
rtp_coding_parameters: RTCRtpCodingParameters {
ssrc: Some(ssrc),
..Default::default()
},
codec: RTCRtpCodec::default(),
..Default::default()
}],
);
// Add to peer connection
let sender_id = pc.add_track(track)?;
Ok(())
}
```
### Signaling
WebRTC requires an external signaling channel (e.g., WebSocket, HTTP) to exchange offers and answers:
```rust
fn example(mut pc: RTCPeerConnection) -> Result<(), Box<dyn std::error::Error>> {
// Create and send offer
let offer = pc.create_offer(None)?;
pc.set_local_description(offer.clone())?;
// Send offer.sdp via your signaling channel
// Receive and apply answer
// let answer = receive_answer_from_signaling()?;
// pc.set_remote_description(answer)?;
Ok(())
}
```
## Specification Compliance
This implementation follows these specifications:
- [W3C WebRTC 1.0](https://www.w3.org/TR/webrtc/) - Main WebRTC API specification
- [RFC 8829](https://datatracker.ietf.org/doc/html/rfc8829) - JSEP: JavaScript Session Establishment Protocol
- [RFC 8866](https://datatracker.ietf.org/doc/html/rfc8866) - SDP: Session Description Protocol
- [RFC 8445](https://datatracker.ietf.org/doc/html/rfc8445) - ICE: Interactive Connectivity Establishment
- [RFC 6347](https://datatracker.ietf.org/doc/html/rfc6347) - DTLS: Datagram Transport Layer Security
- [RFC 9260](https://datatracker.ietf.org/doc/html/rfc9260) - SCTP: Stream Control Transmission Protocol
- [RFC 8831](https://datatracker.ietf.org/doc/html/rfc8831) - WebRTC Data Channels
- [RFC 3550](https://datatracker.ietf.org/doc/html/rfc3550) - RTP: Real-time Transport Protocol
- [RFC 3711](https://datatracker.ietf.org/doc/html/rfc3711) - SRTP: Secure Real-time Transport Protocol
## Documentation
- [API Documentation](https://docs.rs/rtc) - Complete API reference
- [Examples](https://github.com/webrtc-rs/rtc/tree/master/examples/examples) - Working code examples
- [Sans-I/O Pattern](https://sans-io.readthedocs.io/) - Detailed explanation of the sans-I/O design
- [WebRTC for the Curious](https://webrtcforthecurious.com/) - Comprehensive WebRTC guide
## Building and Testing
```bash
# Build the library
cargo build
# Run tests
cargo test
# Build documentation
cargo doc --open
# Run examples
cargo run --example data-channels-offer --features examples
```
## Contributing
Contributions are welcome! Please feel free to submit a Pull Request.
## License
This project is licensed under either of:
- MIT License ([LICENSE-MIT](LICENSE-MIT) or http://opensource.org/licenses/MIT)
- Apache License, Version 2.0 ([LICENSE-APACHE](LICENSE-APACHE) or http://www.apache.org/licenses/LICENSE-2.0)
at your option.
## Acknowledgments
Special thanks to all contributors and the WebRTC-rs community for making this project possible.