simplesip 0.1.3

Simple SIP client library with RTP audio streaming capabilities

SimpleSIP - Simple SIP Client Library

A lightweight, easy-to-use Python library for SIP (Session Initiation Protocol) communication with RTP audio streaming capabilities. Perfect for building VoIP applications, automated calling systems, and SIP-based integrations.

Features

• Simple API - Easy-to-use interface for SIP operations
• Full SIP Support - Registration, calls, and session management
• Real-time Audio - RTP audio streaming with μ-law (PCMU) encoding
• Audio Capture - Built-in microphone support with PyAudio integration
• Authentication - Digest authentication support
• Async Operations - Non-blocking operations with threading
• Call States - Comprehensive call state management
• Extensible - Easy to extend and customize for your needs

Quick Start

Installation

pip install simplesip

For audio support, install with optional dependencies:

pip install simplesip[audio]
# or
pip install simplesip pyaudio

Basic Usage

from simplesip import SimpleSIPClient
import time

# Create a SIP client
client = SimpleSIPClient(
    username="your_username",
    password="your_password", 
    server="your_sip_server.com"
)

# Connect to the SIP server
client.connect()

# Make a call
client.call("1234567890")

# Wait for the call to be established
while client.call_state.value != 'connected':
    time.sleep(0.1)

# Keep the call active for 10 seconds
time.sleep(10)

# Hang up
client.hangup()

# Disconnect from server
client.disconnect()

API Reference

SimpleSIPClient

The main class for SIP operations.

Constructor

SimpleSIPClient(username, password, server, port=5060, local_port=None, timeout=5)

Parameters:

• username (str): SIP username
• password (str): SIP password
• server (str): SIP server hostname or IP
• port (int): SIP server port (default: 5060)
• local_port (int): Local port for SIP (default: random)
• timeout (int): Connection timeout in seconds (default: 5)

Methods

connect()

Connect to the SIP server and register.

client.connect()

call(number)

Initiate a call to the specified number.

client.call("1234567890")

hangup()

End the current call.

client.hangup()

disconnect()

Disconnect from the SIP server.

client.disconnect()

send_audio(audio_data)

Send audio data during an active call.

client.send_audio(ulaw_audio_data)

set_audio_callback(callback, format='pcmu')

Set a callback function to handle incoming audio.

def audio_handler(audio_data, format):
    # Process incoming audio
    pass

client.set_audio_callback(audio_handler, format='pcmu')

get_call_status()

Get the current call status.

status = client.get_call_status()
print(f"State: {status['state']}")
print(f"Duration: {status['duration']}")

Call States

The library uses an enum for call states:

• CallState.IDLE - No active call
• CallState.INVITING - Outgoing call in progress
• CallState.RINGING - Call is ringing
• CallState.CONNECTED - Call connected but no media
• CallState.STREAMING - Call with active audio streaming

Audio Support

SimpleSIP supports real-time audio streaming using RTP protocol with μ-law (PCMU) encoding.

Audio Formats

• PCMU (μ-law): Primary format for SIP/RTP
• PCM: Linear 16-bit audio for processing

Audio Callback

Set up an audio callback to handle incoming audio:

def handle_audio(audio_data, format):
    if format == 'pcmu':
        # Handle μ-law encoded audio
        pass
    elif format == 'pcm':
        # Handle linear PCM audio
        pass

client.set_audio_callback(handle_audio, format='pcmu')

Microphone Integration

import pyaudio
import numpy as np

# Audio configuration
CHUNK = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 8000

# Initialize PyAudio
p = pyaudio.PyAudio()
stream = p.open(
    format=FORMAT,
    channels=CHANNELS,
    rate=RATE,
    input=True,
    frames_per_buffer=CHUNK
)

# Capture and send audio
while client.call_state.value in ['connected', 'streaming']:
    data = stream.read(CHUNK)
    # Convert PCM to μ-law (implement conversion)
    ulaw_data = pcm_to_ulaw(data)
    client.send_audio(ulaw_data)

Examples

Complete Call Example

from simplesip import SimpleSIPClient
import time
import threading

def audio_callback(audio_data, format):
    """Handle incoming audio data"""
    print(f"Received {len(audio_data)} bytes of {format} audio")

# Create client
client = SimpleSIPClient(
    username="1001",
    password="secret123",
    server="sip.example.com"
)

# Set up audio handling
client.set_audio_callback(audio_callback, format='pcmu')

try:
    # Connect to server
    print("Connecting to SIP server...")
    client.connect()
    print("Connected and registered!")
    
    # Make a call
    print("Making call to 1002...")
    client.call("1002")
    
    # Wait for call to be answered
    timeout = 30
    start_time = time.time()
    
    while client.call_state.value not in ['connected', 'streaming']:
        if time.time() - start_time > timeout:
            print("Call timeout!")
            break
        
        status = client.get_call_status()
        print(f"Call status: {status['state']}")
        time.sleep(1)
    
    if client.call_state.value in ['connected', 'streaming']:
        print("Call connected! Audio streaming active.")
        
        # Keep call active for 30 seconds
        time.sleep(30)
        
        # Hang up
        print("Hanging up...")
        client.hangup()
    
finally:
    # Clean disconnect
    client.disconnect()
    print("Disconnected from server")

Configuration Options

# Create client with custom settings
client = SimpleSIPClient(
    username="myuser",
    password="mypass",
    server="192.168.1.100",
    port=5060,           # Custom SIP port
    local_port=5061,     # Custom local port
    timeout=10           # Custom timeout
)

# Set custom timeout
client.timeout = 30

# Enable debug logging
import logging
logging.basicConfig(level=logging.DEBUG)

Development

Requirements

• Python 3.8+
• numpy (for audio processing)
• pyaudio (optional, for microphone support)

Running Tests

# Install development dependencies
pip install -e .[dev]

# Run tests
pytest

# Run tests with coverage
pytest --cov=simplesip

Building Documentation

# Install docs dependencies
pip install sphinx sphinx-rtd-theme

# Build docs
cd docs
make html

Contributing

Contributions are welcome! Please follow these steps:

1. Fork the repository
2. Create a feature branch: git checkout -b feature-name
3. Make your changes
4. Add tests for new functionality
5. Run tests: pytest
6. Commit changes: git commit -am 'Add feature'
7. Push to branch: git push origin feature-name
8. Submit a Pull Request

Known Limitations

• Currently supports PCMU (μ-law) audio encoding only
• IPv4 only (IPv6 support planned)
• Basic SIP features (advanced features in development)
• No built-in STUN/TURN support yet

Roadmap

• Additional audio codecs (G.711 A-law, G.722)
• IPv6 support
• STUN/TURN support for NAT traversal
• SIP over TLS (SIPS)
• Conference calling support
• Call transfer and hold functionality
• Advanced SDP negotiation

License

This project is licensed under the MIT License - see the LICENSE file for details.

Author

Awais Khan

• Email: [e-mail]
• GitHub: @Awaiskhan404

Acknowledgments

• Built with Python's socket and threading libraries
• Audio processing powered by NumPy
• Inspired by the SIP protocol specification (RFC 3261)

Support

• Bug Reports: GitHub Issues
• Discussions: GitHub Discussions
• Email: [e-mail]

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Made with ❤️ for the Python and VoIP communities