rustrtc

A high-performance implementation of WebRTC.

Features

• 🚀High performance: ~2.8x 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

nice@miuda.ai rustrtc % cargo run -r --example benchmark

Comparison (Baseline: webrtc)
Metric               | webrtc     | rustrtc    | pion      
--------------------------------------------------------------------------------
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)
webrtc     | ██████████████                           254.55
rustrtc    | ████████████████████████████████████████ 713.66
pion       | █████████████████                        309.11

Message Rate (msg/s) (Higher is better)
webrtc     | ██████████████                           260659.38
rustrtc    | ████████████████████████████████████████ 730788.92
pion       | █████████████████                        316533.37

Setup Latency (ms) (Lower is better)
webrtc     | ████████████████████████████████████████ 1.36
rustrtc    | ██████                                   0.22
pion       | ██████████████████████████               0.90

CPU Usage (%) (Lower is better)
webrtc     | ███████████████████████████████████████  1480.45
rustrtc    | ████████████████████████████████████████ 1497.50
pion       | █████████████████████████████            1121.20

Memory (MB) (Lower is better)
webrtc     | ██████████████████████████               29.00
rustrtc    | █████████████                            15.00
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:

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.

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:

   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:

   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:

   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:

   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:

   cargo run --example rtp_play
   

2. 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.