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, VoiceAPI)
- 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 support for optimal speech processing
- Noise Suppression: Real-time audio denoising for crystal-clear conversations
- Direct WebRTC Calls: Native WebRTC support for web-based communications
- STUN/TURN Support: Built-in ICE server management for NAT traversal
- Codec Support: Multiple audio codecs (PCMU, PCMA, G.722, PCM)
- Real-time Media: Low-latency audio streaming and processing
- 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
- LLM Proxy: Built-in proxy for AI language model services
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.
- Rust 1.75 or later
- Cargo package manager
git clone https://github.com/restsend/rustpbx
cd rustpbx
cargo build --releasecp config.toml.example config.toml
# Edit config.toml with your settingscargo run --bin rustpbx -- --conf config.tomlAccess the web interface at http://localhost:8080 to test voice agent features and manage calls.
Build intelligent customer service systems with automated speech recognition, natural language processing, and synthetic voice responses.
Create voice-controlled applications with real-time speech processing and AI-powered conversation handling.
Deploy browser-based contact center solutions with advanced call routing and AI assistance.
Integrate traditional SIP phones and systems with modern AI voice processing capabilities.
- SIP Proxy Server (
proxy/): Full-featured SIP server with modular architecture - User Agent (
useragent/): SIP user agent for outbound calls - Media Engine (
media/): Audio processing pipeline with codec support - Voice Synthesis (
synthesis/): TTS engines for multiple providers - Speech Recognition (
transcription/): ASR engines with streaming support - LLM Integration (
llm/): Language model proxy and integration - Call Recording (
callrecord/): Call recording and storage management - RESTful API (
handler/): HTTP API and WebSocket endpoints
- 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
- Media Proxy Guide - Detailed media proxy configuration
- Architecture Diagrams - System architecture and workflows
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.