# rustrtc
A high-performance implementation of WebRTC.
## Features
- **🚀High performance:** ~64% faster than `pion` (go version).
- **🍡WebRTC Compliant**: Full compliance with webrtc/chrome.
- **📺Media Support**: RTP/SRTP handling for audio and video.
- **👌ICE/STUN**: Interactive Connectivity Establishment and STUN protocol support.
## Benchmark game (rustrtc vs webrtc-rs & pion) in 0.2.28
**CPU:** `AMD Ryzen 7 5700X 8-Core Processor`
**OS** `5.15.0-118-generic #128-Ubuntu`
**Compiler** `rustc 1.91.0 (f8297e351 2025-10-28)`, `go version go1.23.0 linux/amd64`
```shell
nice@miuda.ai rustrtc % cargo run -r --example benchmark
Comparison (Baseline: webrtc)
Duration (s) | 10.07 | 10.02 | 10.13
Setup Latency (ms) | 1.36 | 0.22 | 0.90
Throughput (MB/s) | 254.55 | 713.66 | 309.11
Msg Rate (msg/s) | 260659.38 | 730788.92 | 316533.37
CPU Usage (%) | 1480.45 | 1497.50 | 1121.20
Memory (MB) | 29.00 | 15.00 | 44.00
--------------------------------------------------------------------------------
Performance Charts
==================
Throughput (MB/s) (Higher is better)
pion | █████████████████ 309.11
Message Rate (msg/s) (Higher is better)
pion | █████████████████ 316533.37
Setup Latency (ms) (Lower is better)
pion | ██████████████████████████ 0.90
CPU Usage (%) (Lower is better)
pion | █████████████████████████████ 1121.20
Memory (MB) (Lower is better)
pion | ████████████████████████████████████████ 44.00
```
**Key Findings:**
- **Throughput**: `rustrtc` is ~2.8x faster than `webrtc-rs` and ~2.3x faster than `pion`.
- **Memory**: `rustrtc` uses ~48% less memory than `webrtc-rs` and ~66% less than `pion`.
- **Setup Latency**: Significantly faster connection setup (0.22ms vs 1.36ms/0.90ms).
## Usage
Here is a simple example of how to create a `PeerConnection` and handle an offer:
```rust
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().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.
```rust
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.
### SFU (Selective Forwarding Unit)
A multi-user video conferencing server. It receives media from each participant and forwards it to others.
1. Run the server:
```bash
cargo run --example rustrtc_sfu
```
2. Open your browser and navigate to `http://127.0.0.1:8081`. Open multiple tabs/windows to simulate multiple users.
### 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.
1. Run the server:
```bash
cargo run --example echo_server
```
2. Open your browser and navigate to `http://127.0.0.1:3000`.
### DataChannel Chat
A multi-user chat room using WebRTC DataChannels.
1. Run the server:
```bash
cargo run --example datachannel_chat
```
2. 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.
1. Run the server:
```bash
cargo run --example audio_saver
```
2. 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`.
1. Run the server:
```bash
cargo run --example rtp_play
```
2. In a separate terminal, run `ffplay` (requires ffmpeg installed):
```bash
ffplay -protocol_whitelist file,udp,rtp -i examples/rtp_play.sdp
```
## License
This project is licensed under the MIT License.
