rsipstack - A SIP Stack written in Rust

A RFC 3261 compliant SIP stack written in Rust. The goal of this project is to provide a high-performance, reliable, and easy-to-use SIP stack that can be used in various scenarios.

Features

• RFC 3261 Compliant: Full compliance with SIP specification
• Multiple Transport Support: UDP, TCP, TLS, WebSocket
• Transaction Layer: Complete SIP transaction state machine
• Dialog Layer: SIP dialog management
• Digest Authentication: Built-in authentication support
• High Performance: Built with Rust for maximum performance
• Easy to Use: Simple and intuitive API design

TODO

• Transport support
  • UDP
  • TCP
  • TLS
  • WebSocket
• Digest Authentication
• Transaction Layer
• Dialog Layer
• WASM target

Use Cases

This SIP stack can be used in various scenarios, including but not limited to:

• Integration with WebRTC for browser-based communication, such as WebRTC SBC.
• Building custom SIP proxies or registrars
• Building custom SIP user agents (SIP.js alternative)

Why Rust?

We are a group of developers who are passionate about SIP and Rust. We believe that Rust is a great language for building high-performance network applications, and we want to bring the power of Rust to the SIP/WebRTC/SFU world.

Quick Start Examples

SIP Proxy Server

A stateful SIP proxy that routes calls between registered users:

# Run proxy server
cargo run --example proxy -- --port 25060 --addr 127.0.0.1

# Run with external IP
cargo run --example proxy -- --port 25060 --external-ip 1.2.3.4

This example demonstrates:

• SIP user registration and location service
• Call routing between registered users
• Transaction forwarding and response handling
• Session management for active calls
• Handling INVITE, BYE, REGISTER, and ACK methods

SIP User Agent Client

A complete SIP client with registration, calling, and media support:

# Local demo proxy
cargo run --example client -- --port 25061 --sip-server 127.0.0.1:25060 --auto-answer

# Register with a SIP server
cargo run --example client -- --sip-server sip.example.com --user alice --password secret --auto-answer

This example demonstrates:

• SIP user registration with digest authentication
• Making and receiving SIP calls (INVITE/BYE)
• Dialog management for call sessions
• RTP media streaming with file playback
• STUN support for NAT traversal

API Usage Guide

1. Simple SIP Connection

use rsipstack::transport::{udp::UdpConnection, SipAddr};

// Create UDP connection
let connection = UdpConnection::create_connection("127.0.0.1:5060".parse()?, None).await?;

// Send raw SIP message
let sip_message = "OPTIONS sip:test@example.com SIP/2.0\r\n...";
connection.send_raw(sip_message.as_bytes(), &target_addr).await?;

2. Using New Listener APIs

use rsipstack::transport::{
    TcpListenerConnection, WebSocketListenerConnection, TlsListenerConnection,
    TlsConfig, SipAddr
};
use tokio_util::sync::CancellationToken;
use tokio::sync::mpsc::unbounded_channel;

// Create TCP listener
let socket_addr: std::net::SocketAddr = "127.0.0.1:5060".parse()?;
let local_addr = SipAddr::new(rsip::transport::Transport::Tcp, socket_addr.into());
let tcp_listener = TcpListenerConnection::new(local_addr, None).await?;

let cancel_token = CancellationToken::new();
let (sender, mut receiver) = unbounded_channel();

// Start TCP listener
tcp_listener.serve_listener(cancel_token.clone(), sender.clone()).await?;

// Create WebSocket listener
let ws_local_addr = SipAddr::new(rsip::transport::Transport::Ws, socket_addr.into());
let ws_listener = WebSocketListenerConnection::new(ws_local_addr, None, false).await?;
ws_listener.serve_listener(cancel_token.clone(), sender.clone()).await?;

// Create TLS listener with configuration
let tls_config = TlsConfig {
    cert: Some(cert_pem_bytes),
    key: Some(key_pem_bytes),
    ..Default::default()
};
let tls_local_addr = SipAddr::new(rsip::transport::Transport::Tls, socket_addr.into());
let tls_listener = TlsListenerConnection::new(tls_local_addr, None, tls_config).await?;
tls_listener.serve_listener(cancel_token.clone(), sender.clone()).await?;

// Handle incoming connections
while let Some(event) = receiver.recv().await {
    match event {
        TransportEvent::New(connection) => {
            println!("New connection: {}", connection);
        }
        TransportEvent::Incoming(msg, connection, source) => {
            println!("Received message from {}: {}", source, msg);
        }
        TransportEvent::Closed(connection) => {
            println!("Connection closed: {}", connection);
        }
    }
}

3. Using Endpoint and Transactions

use rsipstack::{EndpointBuilder, transport::TransportLayer};
use tokio_util::sync::CancellationToken;

// Build endpoint with transport layer
let cancel_token = CancellationToken::new();
let transport_layer = TransportLayer::new(cancel_token.clone());
let endpoint = EndpointBuilder::new()
    .with_transport_layer(transport_layer)
    .with_cancel_token(cancel_token)
    .build();

// Handle incoming transactions
let mut incoming = endpoint.incoming_transactions();
while let Some(transaction) = incoming.recv().await {
    // Process transaction based on method
    match transaction.original.method {
        rsip::Method::Register => {
            transaction.reply(rsip::StatusCode::OK).await?;
        }
        rsip::Method::Options => {
            transaction.reply(rsip::StatusCode::OK).await?;
        }
        // ... handle other methods
    }
}

4. Creating a User Agent Client

use rsipstack::dialog::{DialogLayer, registration::Registration};
use rsipstack::dialog::authenticate::Credential;
use rsipstack::dialog::invitation::InviteOption;
use std::sync::Arc;
use tokio::sync::mpsc::unbounded_channel;

// Create dialog layer
let dialog_layer = Arc::new(DialogLayer::new(endpoint.inner.clone()));

// Register with server
let credential = Credential {
    username: "alice".to_string(),
    password: "secret".to_string(),
    realm: None,
};

let mut registration = Registration::new(endpoint.inner.clone(), Some(credential.clone()));
let response = registration.register("sip:registrar.example.com".parse()?, None).await?;

// Make outgoing call
let invite_option = InviteOption {
    callee: "sip:bob@example.com".parse()?,
    caller: "sip:alice@example.com".parse()?,
    content_type: None,
    offer: None,
    contact: "sip:alice@192.168.1.100:5060".parse()?,
    credential: Some(credential),
    headers: None,
};

let (state_sender, _state_receiver) = unbounded_channel();
let (invite_dialog, response) = dialog_layer.do_invite(invite_option, state_sender).await?;

5. Implementing a Proxy

use rsipstack::transaction::{Transaction, key::{TransactionKey, TransactionRole}};
use rsipstack::rsip_ext::RsipHeadersExt;
use rsip::prelude::HeadersExt;
use std::collections::HashMap;

// Handle incoming requests
while let Some(mut transaction) = incoming.recv().await {
    match transaction.original.method {
        rsip::Method::Register => {
            // Store user registration
            let user = User::try_from(&transaction.original)?;
            users.insert(user.username.clone(), user);
            transaction.reply(rsip::StatusCode::OK).await?;
        }
        rsip::Method::Invite => {
            // Route call to registered user  
            let callee = transaction.original.to_header()?.uri()?.auth
                .map(|a| a.user)
                .unwrap_or_default();
            if let Some(target) = users.get(&callee) {
                // Create new client transaction for forwarding
                let mut forwarded_req = transaction.original.clone();
                let via = transaction.endpoint_inner.get_via(None, None)?;
                forwarded_req.headers.push_front(via.into());
                
                let key = TransactionKey::from_request(&forwarded_req, TransactionRole::Client)?;
                let mut forwarded_tx = Transaction::new_client(
                    key, 
                    forwarded_req, 
                    transaction.endpoint_inner.clone(), 
                    None
                );
                forwarded_tx.destination = Some(target.destination.clone());
                forwarded_tx.send().await?;
            } else {
                transaction.reply(rsip::StatusCode::NotFound).await?;
            }
        }
        // ... handle other methods
    }
}

Running Tests

Unit Tests

cargo test

Benchmark Tests

# Run server
cargo run -r --bin bench_ua  -- -m server -p 5060

# Run client with 1000 calls
cargo run -r  --bin bench_ua  -- -m client -p 5061 -s 127.0.0.1:5060 -c 1000

The test monitor:

=== SIP Benchmark UA Stats ===
Dialogs: 9992
Active Calls: 9983
Rejected Calls: 0
Failed Calls: 0
Total Calls: 250276
Calls/Second: 1501
============================

Documentation

• API Documentation
• Examples

Contributing

We welcome contributions! Please see our Contributing Guide for details.

License

This project is licensed under the MIT License - see the LICENSE file for details.