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.
- RFC 3261 Compliant: Full compliance with SIP specification
- Multiple Transport Support: UDP, TCP, TLS, WebSocket (TLS/WebSocket require the
rustls
andwebsocket
features, enabled by default) - 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
- Transport support
- UDP
- TCP
- TLS
- WebSocket
- Digest Authentication
- Transaction Layer
- Dialog Layer
- WASM target
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)
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.
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
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
use rsipstack::transport::{udp::UdpConnection, SipAddr};
use tokio_util::sync::CancellationToken;
// Create UDP connection bound to an ephemeral local port
let cancel_token = CancellationToken::new();
let connection = UdpConnection::create_connection(
"0.0.0.0:0".parse()?,
None,
Some(cancel_token.child_token()),
)
.await?;
// Prepare the remote target
let target_addr = SipAddr::new(
rsip::transport::Transport::Udp,
rsip::HostWithPort::try_from("127.0.0.1:5060")?,
);
// 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?;
use rsipstack::transport::{
SipAddr, TcpListenerConnection, TransportEvent, TransportLayer,
};
use tokio_util::sync::CancellationToken;
// Build a transport layer and register listeners
let cancel_token = CancellationToken::new();
let transport_layer = TransportLayer::new(cancel_token.clone());
let tcp_listener = TcpListenerConnection::new(
SipAddr::new(
rsip::transport::Transport::Tcp,
rsip::HostWithPort::try_from("0.0.0.0:5060")?,
),
None,
)
.await?;
transport_layer.add_transport(tcp_listener.into());
// Access the transport event stream
let mut events = transport_layer
.inner
.transport_rx
.lock()
.unwrap()
.take()
.expect("transport receiver");
tokio::spawn(async move {
while let Some(event) = events.recv().await {
match event {
TransportEvent::New(connection) => println!("New connection: {}", connection),
TransportEvent::Incoming(msg, connection, source) => {
println!("Received message from {}: {}", source, msg);
// Use `connection` to reply if needed
}
TransportEvent::Closed(connection) => {
println!("Connection closed: {}", connection);
}
}
}
});
// Start accepting connections (this is normally driven by `Endpoint::serve`)
transport_layer
.serve_listens()
.await
.expect("failed to start listeners");
To add TLS or WebSocket listeners, construct a TlsListenerConnection
or
WebSocketListenerConnection
and register it with transport_layer.add_transport(...)
.
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.clone())
.build();
// Start endpoint background task
let endpoint_inner = endpoint.inner.clone();
tokio::spawn(async move {
if let Err(err) = endpoint_inner.serve().await {
eprintln!("endpoint stopped: {err}");
}
});
// Handle incoming transactions
let mut incoming = endpoint
.incoming_transactions()
.expect("transaction receiver available");
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
}
}
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?;
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
}
}
cargo test
# 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
============================
We welcome contributions! Please see our Contributing Guide for details.
This project is licensed under the MIT License - see the LICENSE file for details.