rustrtc

A pure Rust implementation of WebRTC.

Features

PeerConnection: The main entry point for WebRTC connections.
Data Channels: Support for reliable and unreliable data channels.
Media Support: RTP/SRTP handling for audio and video.
ICE/STUN: Interactive Connectivity Establishment and STUN protocol support.
DTLS: Datagram Transport Layer Security for secure communication.
SDP: Session Description Protocol parsing and generation.

Performance (vs webrtc-rs & pion)

From Apple M4 machine result

mpi@mpis-MacBook-Air rustrtc % cargo build --release --example benchmark && ./target/release/examples/benchmark

Comparison (Baseline: webrtc)
Metric               | webrtc     | rustrtc    | pion      
--------------------------------------------------------------------------------
Duration (s)         | 10.02      | 10.02      | 11.02     
Setup Latency (ms)   | 1.14       | 0.69       | 1.80      
Throughput (MB/s)    | 135.45     | 213.38     | 177.92    
Msg Rate (msg/s)     | 138696.71  | 218497.60  | 182190.56 
CPU Usage (%)        | 820.38     | 829.33     | 596.12    
Memory (MB)          | 28.00      | 10.00      | 41.00     
--------------------------------------------------------------------------------

Key Findings:

Throughput: rustrtc is ~57% faster than webrtc-rs and ~20% faster than pion.
Memory: rustrtc uses ~64% less memory than webrtc-rs and ~75% less than pion.
Setup Latency: Significantly faster connection setup (0.69ms vs 1.14ms/1.80ms).

Usage

Here is a simple example of how to create a PeerConnection and handle an offer:

use rustrtc::{PeerConnection, RtcConfiguration, SessionDescription, SdpType};

#[tokio::main]
async fn main() {
    let config = RtcConfiguration::default();
    let pc = PeerConnection::new(config);

    // Create a Data Channel
    let dc = pc.create_data_channel("data", None).unwrap();

    // Handle received messages
    let dc_clone = dc.clone();
    tokio::spawn(async move {
        while let Some(event) = dc_clone.recv().await {
            if let rustrtc::DataChannelEvent::Message(data) = event {
                println!("Received: {:?}", String::from_utf8_lossy(&data));
            }
        }
    });

    // Create an offer
    let offer = pc.create_offer().await.unwrap();
    pc.set_local_description(offer).unwrap();

    // Wait for ICE gathering to complete
    pc.wait_for_gathering_complete().await;

    // Get the complete SDP with candidates
    let complete_offer = pc.local_description().unwrap();
    println!("Offer SDP: {}", complete_offer.to_sdp_string());
}

Configuration

rustrtc allows customizing the WebRTC session via RtcConfiguration:

ice_servers: Configure STUN/TURN servers.
ice_transport_policy: Control ICE candidate gathering (e.g., All, Relay).
ssrc_start: Set the starting SSRC value for local tracks.
media_capabilities: Configure supported codecs (payload types, names) and SCTP ports.

use rustrtc::{PeerConnection, RtcConfiguration, IceServer, IceTransportPolicy, config::MediaCapabilities};

let mut config = RtcConfiguration::default();

// Configure ICE servers
config.ice_servers.push(IceServer::new(vec!["stun:stun.l.google.com:19302"]));

// Set ICE transport policy (optional)
config.ice_transport_policy = IceTransportPolicy::All;

config.ssrc_start = 10000;

// Customize media capabilities
let mut caps = MediaCapabilities::default();
// ... configure audio/video/application caps ...
config.media_capabilities = Some(caps);

let pc = PeerConnection::new(config);

Examples

You can run the examples provided in the repository.

Echo Server

The echo server example demonstrates how to accept a WebRTC connection, receive data on a data channel, and echo it back. It also supports video playback if an IVF file is provided.

Run the server:
```
cargo run --example echo_server
```
Open your browser and navigate to http://127.0.0.1:3000.

DataChannel Chat

A multi-user chat room using WebRTC DataChannels.

Run the server:
```
cargo run --example datachannel_chat
```
Open your browser and navigate to http://127.0.0.1:3000. Open multiple tabs to chat between them.

Audio Saver

Records audio from the browser's microphone and saves it to a file (output.ulaw) on the server.

Run the server:
```
cargo run --example audio_saver
```
Open your browser and navigate to http://127.0.0.1:3000. Click "Start" to begin recording.

RTP Play (FFmpeg)

Streams a video file (examples/static/output.ivf) via RTP to a UDP port, which can be played back using ffplay.

Run the server:
```
cargo run --example rtp_play
```

In a separate terminal, run ffplay (requires ffmpeg installed):

ffplay -protocol_whitelist file,udp,rtp -i examples/rtp_play.sdp

License

This project is licensed under the MIT License.

rustrtc 0.2.0