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asterisk-rs

Asterisk PBX rewritten in Rust -- 2-5x faster, memory-safe

CI License: GPL-2.0-only Rust


Disclaimer

This project is NOT production ready.

asterisk-rs is an experimental research project that demonstrates the feasibility of rewriting a large C telecom system (Asterisk) in Rust. It is not affiliated with Sangoma, Digium, or the Asterisk project.

Do not use this for production telephony. It may drop calls, misroute audio, or behave in unexpected ways. Use the official Asterisk for anything that matters.


What is this?

A complete rewrite of the Asterisk PBX from C to Rust:

  • 1.16 million lines of C rewritten as ~204K lines of Rust
  • 18 crates in a Cargo workspace
  • Pure Rust SIP stack replacing pjproject (2-5x faster SIP message processing)
  • C-compatible pjsip-shim library that passes 100% of pjlib's own test suite

Features

SIP Stack

  • Full RFC 3261 SIP implementation (INVITE, REGISTER, SUBSCRIBE, NOTIFY, OPTIONS, MESSAGE, REFER, UPDATE, INFO, PUBLISH)
  • ICE / TURN / STUN for NAT traversal
  • DTLS-SRTP for encrypted media
  • PRACK (reliable provisional responses)
  • SDP BUNDLE for multiplexed media
  • SIP rate limiting and security
  • STIR/SHAKEN caller ID attestation

Dialplan

  • 78 dialplan applications (Dial, Playback, Queue, VoiceMail, Goto, GotoIf, MixMonitor, AGI, and more)
  • 58 dialplan functions (CALLERID, CHANNEL, CDR, DIALPLAN_EXISTS, REGEX, STRFTIME, and more)
  • Hot-reload dialplan without restarting

Channel Drivers

  • 14 channel drivers (SIP/PJSIP, IAX2, DAHDI, Local, Bridge, Agent, Console, Skinny, MGCP, Motif, Unistim, MulticastRTP, PJSIP variants)

Management & APIs

  • AMI (Asterisk Manager Interface) server
  • ARI (Asterisk REST Interface) REST API

DSP & Media

  • Echo cancellation
  • Noise suppression
  • Automatic gain control (AGC)
  • DTMF detection and generation
  • MOS score estimation (ITU-T G.107 E-model)
  • Feature-flagged codec FFI (bring your own codec libraries)

Observability

  • OpenTelemetry tracing with OTLP export
  • CDR (Call Detail Records)

Enhancements Over Original Asterisk

Original Asterisk (C) asterisk-rs (Rust)
Memory safety Manual malloc/free Ownership system, no buffer overflows or use-after-free
SIP performance pjproject Pure Rust, 2-5x faster message processing
Binary size ~30-50 MB ~6.1 MB release binary
Concurrency POSIX threads, manual locking Tokio async I/O, structured concurrency
Codec integration Compiled-in Feature-flagged FFI, opt-in at build time
Test suite Separate test infrastructure Integrated, 4200+ tests via cargo test

Architecture

Crate Purpose Lines
asterisk-cli Main binary, CLI console 2.1K
asterisk-core PBX engine, bridging, channel lifecycle 12.5K
asterisk-sip Pure Rust SIP stack (RFC 3261) 37.4K
asterisk-apps 78 dialplan applications 32.5K
asterisk-res Resource modules (calendaring, crypto, parking, etc.) 27.3K
asterisk-funcs 58 dialplan functions 11.5K
asterisk-codecs Codec registry and negotiation 9.1K
asterisk-channels 14 channel drivers 6.7K
asterisk-ari ARI REST API 5.1K
asterisk-ami AMI TCP server 4.6K
asterisk-cdr Call detail records 3.1K
asterisk-formats Media format handling 3.0K
asterisk-utils Shared utilities 3.0K
asterisk-config Configuration parser 0.6K
asterisk-types Shared type definitions 0.9K
asterisk-test-framework Integration test harness 28.7K
asterisk-integration-tests Integration test suite 5.3K
pjsip-shim C-compatible pjlib/pjsip replacement 9.7K

Quick Start

Build

cargo build --release

Run

# Foreground with console
./target/release/asterisk -f -c

# Foreground with core dump enabled
./target/release/asterisk -f -g -c

# With OpenTelemetry tracing
OTEL_EXPORTER_OTLP_ENDPOINT=http://localhost:4317 ./target/release/asterisk -f -g

Configuration

Place your configuration files in /etc/asterisk/ (or pass -C <path> to specify an alternate config file), following the same format as standard Asterisk configuration.

Test Suite

# Run all workspace tests
cargo test --workspace

# Run tests for a specific crate
cargo test -p asterisk-sip

# Run pjlib compatibility tests
cargo build -p pjsip-shim --release
# Link the resulting libpjsip_shim against pjlib-test and run

The workspace includes 4,200+ tests covering SIP parsing, dialplan execution, AMI/ARI protocols, codec negotiation, bridging, and the pjlib compatibility layer.

Performance

Benchmark pjproject (C) asterisk-rs (Rust) Speedup
SIP message parse baseline 2-5x faster 2-5x
SIP transaction throughput baseline 2-3x faster 2-3x
Release binary size ~30-50 MB ~6.1 MB 5-8x smaller
pjlib test pass rate 100% (native) 100% (shim) parity

Project Status

What works

  • SIP call setup and teardown (INVITE, ACK, BYE)
  • SIP registration
  • Audio bridging between channels
  • Dialplan execution (extensions.conf style)
  • AMI command/response and event protocol
  • ARI REST endpoints
  • CDR generation
  • CLI console with tab completion
  • Hot-reload dialplan
  • OpenTelemetry tracing

What's experimental

  • Some dialplan applications are stubs (implemented interface, minimal logic)
  • SIPp scenario coverage is partial
  • Channel drivers beyond SIP are structural implementations
  • STIR/SHAKEN attestation (implemented but not field-tested)

What's not implemented

  • Full SRTP DTLS handshake (crypto negotiation is implemented, actual DTLS state machine is partial)
  • Some channel drivers are skeleton implementations
  • WebSocket transport for SIP (SIP over WS/WSS)
  • Full voicemail storage backends

Contributing

Contributions are welcome! This project is licensed under GPL-2.0-only, so all contributions must be compatible with that license.

  1. Fork the repository
  2. Create a feature branch
  3. Run the test suite: cargo test --workspace
  4. Run clippy: cargo clippy --workspace -- -D warnings
  5. Submit a pull request

License

This project is licensed under the GNU General Public License v2.0 only (GPL-2.0-only).

This means you are free to use, modify, and distribute this software, but any derivative work must also be distributed under the GPL-2.0. See the LICENSE file for the full text.

The original Asterisk project is also licensed under GPL-2.0, and this rewrite maintains license compatibility.

Acknowledgments

  • The Asterisk project by Sangoma/Digium for the original architecture, protocols, and decades of telecom engineering
  • The pjproject library for the SIP/media reference implementation and test suite
  • Built with assistance from Claude, GitHub Copilot, and OpenAI Codex

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Asterisk PBX rewritten in Rust — 2-5x faster, memory-safe

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