A SIP Stack written in Rust

WIP This is a work in progress and is not yet ready for production use.

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.

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.

How to run

# the sip phone will serve at: YOUR_NETWORK_IP:25060
cargo run --example client

Make a call to sip:YOUR_NETWORK_IP:25060 from another sip client.(e.g. linphone)

Benchmark tools

# 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
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