SpechPhone
Modern SIP/VoIP Softphone written in PHP with Swoole
What is SpechPhone?
SpechPhone is a modern SIP/VoIP softphone written in PHP and optimized with Swoole. It allows establishing real-time voice calls, controlling codecs, and integrating with backend tools without depending on external media libraries.
Key Features
Self-sufficient
No FFmpeg or SoX required - all audio processing built-in
Multiple Codecs
Support for PCMA, PCMU, G.729, Opus, and L16
Audio Processing
Built-in resampling, mixing, noise suppression, and echo cancellation
Asynchronous I/O
Swoole coroutines for hundreds of simultaneous sessions
Full PHP Stack
SIP, RTP, and media control entirely in PHP
Browser Ready
Responsive HTML/JS interface with WebSocket support
Screenshots
Configuration Interface - SIP account setup and codec settings
Call Interface - Active call with real-time controls
Audio Processing - Real-time audio monitoring and codec information
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Mobile Interface - SpechPhone running on Samsung Galaxy S8+
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Mobile Call Interface - Active call on mobile device
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Mobile Configuration - Settings and options on mobile device
Requirements
- Optimized PHP Runtime (pcg729) - Static PHP CLI with native G.729, Opus, and audio processing support
- OpenSSL for TLS/SSL features
- Linux or macOS system (recommended)
Installation
# Install system dependencies
sudo apt update && sudo apt install -y openssl
# Clone the repository
git clone https://github.com/spechshop/spechphone && cd spechphone
git clone https://github.com/spechshop/libspech
# Download pcg729 runtime
curl -L https://github.com/spechshop/pcg729/releases/download/current/php -o php
# Configure runtime
chmod +x ./php
sudo cp php /usr/local/bin/php
# Start services
php middleware.php
php audio.phpArchitecture
Audio Processing
Through Swoole, the application opens a dedicated coroutine instance for each SIP trunk manager. This allows hundreds of calls to occur simultaneously without blocking the main process.
Security Model
- No WebRTC: SpechPhone doesn't use WebRTC (no SRTP / ICE / DTLS)
- Backend Media: Audio is transported via raw RTP (UDP)
- Thin Client: Browser acts as thin client via WebSocket (PCM delivery)
- Production: HTTPS and WSS are mandatory
What Can You Build?
- Enterprise Softphone with dialer and browser integration
- Proxy/SBC Client for SIP servers via UDP
- Audio Streaming via WebSocket or RTP
- Monitoring and Recording Services with native audio utilities