RustPBX is a high-performance, secure software-defined PBX (Private Branch Exchange) system implemented in Rust, designed to support AI-powered communication pipelines and modern voice applications.

• Full SIP Stack: Complete SIP proxy server with registration, authentication, and call routing
• Media Proxy: Advanced RTP/RTCP media proxying with NAT traversal support
• Multi-Transport: UDP, TCP, and WebSocket transport support
• Call Recording: Built-in call recording with multiple storage backends
• User Management: Flexible user authentication and registration system

• Speech-to-Text (ASR): Real-time speech recognition with multiple providers (Tencent Cloud, Aliyun, Deepgram)
• Text-to-Speech (TTS): High-quality speech synthesis with emotion and speaker control
• LLM Integration: OpenAI-compatible LLM proxy for intelligent conversation handling
• Voice Activity Detection: WebRTC and Silero VAD and Ten VAD support for optimal speech processing
• Noise Suppression: Real-time audio denoising(rnnoise) for crystal-clear conversations

• RESTful Endpoints: Complete REST API for call management and control
• WebSocket Commands: Real-time call control via WebSocket connections
• Call Management: List, monitor, and control active calls

• Rust 1.75 or later
• Cargo package manager
• opus, alsa
• pkg-config

Linux:

apt-get install -y libasound2-dev libopus-dev

Mac:

brew install opus

git clone https://github.com/restsend/rustpbx
cd rustpbx
cargo build --release

cp config.toml.example config.toml
# Edit config.toml with your settings

cargo run --bin rustpbx -- --conf config.toml

Access the web interface at http://localhost:8080 to test voice agent features and manage calls.

1. Pull the Docker image:

docker pull ghcr.io/restsend/rustpbx:latest

2. Create config.toml:

Proxy and UserAgent both worked.

# Create config.toml
cat > config.toml << EOF
http_addr = "0.0.0.0:8080"
log_level = "info"
recorder_path = "/tmp/recorders"
media_cache_path = "/tmp/mediacache"
database_url = "sqlite://./db/rustpbx.sqlite3"

[ua]
addr="0.0.0.0"
udp_port=13050

[console]
#session_secret = "please_change_me_to_a_random_secret"
base_path = "/console"

[proxy]
modules = ["acl", "auth", "registrar", "call"]
addr = "0.0.0.0"
udp_port = 15060
registrar_expires = 60
ws_handler= "/ws"
media_proxy = "auto"
trunks_source = "config" # or "database"
routes_source = "config" # or "database"

# ACL rules
acl_rules = [
    "allow 10.0.0.0/8",
    "allow all",
]

[[proxy.user_backends]]
type = "memory"
users = [
    { username = "bob", password = "123456" },
    { username = "alice", password = "123456" },
]

[[proxy.user_backends]]
type = "extension"

[callrecord]
type = "local"
root = "/tmp/cdr"
EOF

4. Run with Docker:

docker run -d \
  --name rustpbx \
  -p 8080:8080 \
  -p 15060:15060/udp \
  -p 13050:13050/udp \
  -p 20000-30000:20000-30000/udp \
  --env-file .env \
  -v $(pwd)/db:/app/db \
  -v $(pwd)/config.toml:/app/config.toml \
  -v $(pwd)/recordings:/tmp/recorders \
  ghcr.io/restsend/rustpbx:latest \
  --conf /app/config.toml

• Create super user via cli(optional)

docker exec rustpbx /app/rustpbx --conf /app/config.toml --super-user=YOUR --super-password=PASS

1. Access the service:

• Web Interface: http://localhost:8080/console/
  • Login via YOUR + PASS
• SIP Proxy: localhost:15060
• User Agent: localhost:13050

See https://github.com/restsend/rustpbxgo

The SIP workflow demonstrates how external applications can initiate calls through RustPBX, leveraging the full SIP protocol stack for reliable voice communications.

The WebRTC workflow shows how web applications can establish direct peer-to-peer connections via RustPBX, enabling modern browser-based voice applications.

For detailed API documentation, see API Documentation.

• Modular proxy architecture with pluggable modules
• User authentication and registration
• Call routing and forwarding
• CDR (Call Detail Records) generation

• Automatic NAT detection and media proxying
• Configurable RTP port ranges
• Support for multiple codecs
• Real-time media relay

• Multiple ASR/TTS provider support
• Configurable LLM endpoints
• Voice activity detection
• Audio preprocessing and enhancement

• API Reference - Complete REST API documentation
• Architecture Diagrams - System architecture and workflows
• VoiceAgent Integration with Telephony Networks
• VoiceAgent 与电话网络互通的技术实现

This project is currently in active development. We welcome contributions and feedback from the community.

MIT License - see LICENSE file for details.

Work in Progress - Core functionality is implemented and being actively refined. The system is suitable for development and testing environments.