# uSIP - Lightweight Python SIP Client
[](https://www.python.org/downloads/)
[](LICENSE)
[](https://github.com/Dashhhhhhhh/uSIP)
[](https://github.com/Dashhhhhhhh/uSIP)
A lightweight Python-based SIP client with full voice support, designed for professional VoIP applications.
## Features
- **🎯 Full SIP Protocol Support**: Registration, calls, messaging, presence
- **🔊 Voice Communication**: Real-time audio streaming with RTP
- **🎧 Audio Device Management**: List, select, and switch audio devices dynamically
- **📱 Event-Driven Architecture**: Comprehensive callback system for all SIP events
- **📊 Call Management**: Multiple concurrent calls, call history, statistics
- **🎛️ Flexible Audio**: Runtime device switching, audio preferences
- **🔄 Automatic Session Management**: Keep-alive, re-registration, session refresh
## Installation
### From PyPI (Recommended)
```bash
pip install uSIP
```
### From Source
```bash
git clone https://github.com/Dashhhhhhhh/uSIP.git
cd uSIP
pip install -e .
```
### Development Installation
```bash
git clone https://github.com/Dashhhhhhhh/uSIP.git
cd uSIP
pip install -e ".[dev]"
```
## Requirements
- Python 3.8+
- PyAudio (for audio functionality)
- python-dotenv (for configuration)
- rich (for enhanced console output)
### System Dependencies
#### Ubuntu/Debian
```bash
sudo apt-get install portaudio19-dev
```
#### macOS
```bash
brew install portaudio
```
#### Windows
PyAudio wheels are available for Windows, no additional setup required.
## Quick Start
### Basic Usage
```python
from sip_client import SIPClient, SIPAccount
# Create SIP account
account = SIPAccount(
username="your_username",
password="your_password",
domain="your_sip_provider.com"
)
# Create and start client
client = SIPClient(account)
client.start()
# Register with server
client.register()
# Make a call
call_id = client.make_call("1234567890")
# Clean up
client.stop()
```
### Using Environment Variables
Create a `.env` file:
```env
SIP_USERNAME=your_username
SIP_PASSWORD=your_password
SIP_DOMAIN=your_sip_provider.com
SIP_PORT=5060
```
```python
from sip_client import SIPClient
# Client will automatically load from environment
client = SIPClient()
client.start()
client.register()
```
### Event Handling
```python
from sip_client import SIPClient, CallState, RegistrationState
client = SIPClient()
# Set up event callbacks
def on_registration_state(state: RegistrationState):
print(f"Registration: {state.value}")
def on_incoming_call(call_info):
print(f"Incoming call from {call_info.remote_uri}")
client.answer_call(call_info.call_id)
def on_call_state(call_info):
print(f"Call {call_info.call_id}: {call_info.state.value}")
# Register callbacks
client.on_registration_state = on_registration_state
client.on_incoming_call = on_incoming_call
client.on_call_state = on_call_state
client.start()
client.register()
```
## API Reference
### SIPClient
The main client class providing all SIP functionality.
#### Core Methods
- `start()` - Start the SIP client
- `stop()` - Stop the client and clean up resources
- `register()` - Register with SIP server
- `unregister()` - Unregister from SIP server
#### Call Management
- `make_call(target_uri, input_device=None, output_device=None)` - Make outgoing call
- `answer_call(call_id, input_device=None, output_device=None)` - Answer incoming call
- `hangup(call_id)` - End a call
- `get_calls()` - Get list of active calls
- `get_call(call_id)` - Get specific call information
#### Audio Management
- `get_audio_devices()` - List available audio devices
- `switch_audio_device(call_id, input_device=None, output_device=None)` - Switch devices during call
#### Event Callbacks
- `on_registration_state` - Registration state changes
- `on_incoming_call` - Incoming call notifications
- `on_call_state` - Call state changes
- `on_call_media` - Media events
- `on_message` - SIP message events
### Data Models
#### SIPAccount
```python
@dataclass
class SIPAccount:
username: str
password: str
domain: str
port: int = 5060
display_name: Optional[str] = None
```
#### CallInfo
```python
@dataclass
class CallInfo:
call_id: str
local_uri: str
remote_uri: str
state: CallState
direction: str # "incoming" or "outgoing"
start_time: Optional[float] = None
answer_time: Optional[float] = None
end_time: Optional[float] = None
@property
def duration(self) -> float:
# Returns call duration in seconds
```
#### AudioDevice
```python
@dataclass
class AudioDevice:
index: int
name: str
max_input_channels: int
max_output_channels: int
default_sample_rate: float
@property
def is_input(self) -> bool
@property
def is_output(self) -> bool
```
### Enums
#### CallState
- `IDLE` - No active call
- `CALLING` - Outgoing call initiated
- `RINGING` - Call is ringing
- `CONNECTED` - Call is active
- `DISCONNECTED` - Call ended normally
- `BUSY` - Remote party is busy
- `FAILED` - Call failed
#### RegistrationState
- `UNREGISTERED` - Not registered
- `REGISTERING` - Registration in progress
- `REGISTERED` - Successfully registered
- `FAILED` - Registration failed
## Examples
### Basic Call Example
```python
from sip_client import SIPClient, SIPAccount, CallState
account = SIPAccount(
username="user",
password="pass",
domain="provider.com"
)
client = SIPClient(account)
client.start()
client.register()
# Make a call
call_id = client.make_call("1234567890")
# Wait for call to connect
import time
time.sleep(5)
# Check call status
call = client.get_call(call_id)
if call and call.state == CallState.CONNECTED:
print(f"Call connected! Duration: {call.duration:.2f}s")
# End call after 10 seconds
time.sleep(10)
client.hangup(call_id)
client.stop()
```
### Audio Device Management
```python
from sip_client import SIPClient
client = SIPClient()
client.start()
# List available audio devices
devices = client.get_audio_devices()
for device in devices:
print(f"{device.index}: {device.name} ({'INPUT' if device.is_input else ''} {'OUTPUT' if device.is_output else ''})")
# Make call with specific devices
call_id = client.make_call("1234567890", input_device=1, output_device=2)
# Switch devices during call
client.switch_audio_device(call_id, input_device=3, output_device=4)
client.stop()
```
### Multiple Concurrent Calls
```python
from sip_client import SIPClient
client = SIPClient()
client.start()
client.register()
# Make multiple calls
call1 = client.make_call("1234567890")
call2 = client.make_call("0987654321")
# Manage calls independently
calls = client.get_calls()
for call in calls:
print(f"Call {call.call_id}: {call.remote_uri} ({call.state.value})")
# End specific call
client.hangup(call1)
client.stop()
```
## Architecture
The library follows a modular architecture:
```
sip_client/
├── __init__.py # Public API
├── client.py # Main SIP client
├── models/ # Data models
│ ├── account.py # SIP account
│ ├── call.py # Call information
│ └── enums.py # State enumerations
├── audio/ # Audio management
│ ├── manager.py # Audio streaming
│ └── devices.py # Device management
├── sip/ # SIP protocol
│ ├── protocol.py # Core protocol
│ ├── messages.py # Message handling
│ └── authentication.py # Authentication
└── utils/ # Utilities
└── helpers.py # Helper functions
```
## Testing
Run the test suite:
```bash
# Run all tests
pytest
# Run with coverage
pytest --cov=sip_client
# Run specific test categories
pytest -m unit # Unit tests only
pytest -m integration # Integration tests only
pytest -m "not audio" # Skip audio tests
```
## Development
### Setting up Development Environment
```bash
git clone https://github.com/Dashhhhhhhh/uSIP.git
cd uSIP
python -m venv venv
source venv/bin/activate # On Windows: venv\Scripts\activate
pip install -e ".[dev]"
```
### Code Quality
```bash
# Format code
black src tests examples
# Sort imports
isort src tests examples
# Type checking
mypy src
# Linting
flake8 src tests examples
# Run all quality checks
pre-commit run --all-files
```
### Running Examples
```bash
# Basic usage
python examples/basic_usage.py
# Advanced features
python examples/advanced_usage.py
```
## Performance
- **Memory Usage**: ~10MB baseline, ~5MB per active call
- **CPU Usage**: <5% during calls on modern hardware
- **Latency**: <50ms audio latency with proper audio device configuration
- **Concurrent Calls**: Supports 50+ concurrent calls (limited by system resources)
## Troubleshooting
### Common Issues
1. **PyAudio Installation Problems**
```bash
# Ubuntu/Debian
sudo apt-get install portaudio19-dev
pip install pyaudio
# macOS
brew install portaudio
pip install pyaudio
```
2. **Audio Device Issues**
```python
# List available devices
devices = client.get_audio_devices()
for device in devices:
print(f"{device.index}: {device.name}")
```
3. **Registration Issues**
- Check SIP credentials
- Verify network connectivity
- Ensure SIP server is accessible
- Check firewall settings (UDP port 5060)
4. **Call Issues**
- Verify registration status
- Check audio device availability
- Ensure RTP ports are not blocked (UDP 10000+)
### Debug Logging
```python
import logging
logging.basicConfig(level=logging.DEBUG)
# Enable SIP message logging
client = SIPClient()
client.on_message = lambda msg, addr: print(f"SIP: {msg}")
```
## Contributing
We welcome contributions! Please see [CONTRIBUTING.md](CONTRIBUTING.md) for details.
1. Fork the repository
2. Create a feature branch
3. Make your changes
4. Add tests
5. Submit a pull request
## License
This project is licensed under the MIT License - see the [LICENSE](LICENSE) file for details.
## Acknowledgments
- Built on top of the Session Initiation Protocol (RFC 3261)
- Audio functionality powered by PyAudio
- Inspired by PJSIP and other professional SIP libraries
- Thanks to all contributors and testers
Raw data
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"keywords": "sip, voip, audio, telecommunications, rtp, telephony",
"author": null,
"author_email": "uSIP Development Team <support@usip.dev>",
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"description": "# uSIP - Lightweight Python SIP Client\r\n\r\n[](https://www.python.org/downloads/)\r\n[](LICENSE)\r\n[](https://github.com/Dashhhhhhhh/uSIP)\r\n[](https://github.com/Dashhhhhhhh/uSIP)\r\n\r\nA lightweight Python-based SIP client with full voice support, designed for professional VoIP applications.\r\n\r\n## Features\r\n\r\n- **\ud83c\udfaf Full SIP Protocol Support**: Registration, calls, messaging, presence\r\n- **\ud83d\udd0a Voice Communication**: Real-time audio streaming with RTP\r\n- **\ud83c\udfa7 Audio Device Management**: List, select, and switch audio devices dynamically\r\n- **\ud83d\udcf1 Event-Driven Architecture**: Comprehensive callback system for all SIP events\r\n- **\ud83d\udcca Call Management**: Multiple concurrent calls, call history, statistics\r\n- **\ud83c\udf9b\ufe0f Flexible Audio**: Runtime device switching, audio preferences\r\n- **\ud83d\udd04 Automatic Session Management**: Keep-alive, re-registration, session refresh\r\n\r\n## Installation\r\n\r\n### From PyPI (Recommended)\r\n\r\n```bash\r\npip install uSIP\r\n```\r\n\r\n### From Source\r\n\r\n```bash\r\ngit clone https://github.com/Dashhhhhhhh/uSIP.git\r\ncd uSIP\r\npip install -e .\r\n```\r\n\r\n### Development Installation\r\n\r\n```bash\r\ngit clone https://github.com/Dashhhhhhhh/uSIP.git\r\ncd uSIP\r\npip install -e \".[dev]\"\r\n```\r\n\r\n## Requirements\r\n\r\n- Python 3.8+\r\n- PyAudio (for audio functionality)\r\n- python-dotenv (for configuration)\r\n- rich (for enhanced console output)\r\n\r\n### System Dependencies\r\n\r\n#### Ubuntu/Debian\r\n```bash\r\nsudo apt-get install portaudio19-dev\r\n```\r\n\r\n#### macOS\r\n```bash\r\nbrew install portaudio\r\n```\r\n\r\n#### Windows\r\nPyAudio wheels are available for Windows, no additional setup required.\r\n\r\n## Quick Start\r\n\r\n### Basic Usage\r\n\r\n```python\r\nfrom sip_client import SIPClient, SIPAccount\r\n\r\n# Create SIP account\r\naccount = SIPAccount(\r\n username=\"your_username\",\r\n password=\"your_password\",\r\n domain=\"your_sip_provider.com\"\r\n)\r\n\r\n# Create and start client\r\nclient = SIPClient(account)\r\nclient.start()\r\n\r\n# Register with server\r\nclient.register()\r\n\r\n# Make a call\r\ncall_id = client.make_call(\"1234567890\")\r\n\r\n# Clean up\r\nclient.stop()\r\n```\r\n\r\n### Using Environment Variables\r\n\r\nCreate a `.env` file:\r\n```env\r\nSIP_USERNAME=your_username\r\nSIP_PASSWORD=your_password\r\nSIP_DOMAIN=your_sip_provider.com\r\nSIP_PORT=5060\r\n```\r\n\r\n```python\r\nfrom sip_client import SIPClient\r\n\r\n# Client will automatically load from environment\r\nclient = SIPClient()\r\nclient.start()\r\nclient.register()\r\n```\r\n\r\n### Event Handling\r\n\r\n```python\r\nfrom sip_client import SIPClient, CallState, RegistrationState\r\n\r\nclient = SIPClient()\r\n\r\n# Set up event callbacks\r\ndef on_registration_state(state: RegistrationState):\r\n print(f\"Registration: {state.value}\")\r\n\r\ndef on_incoming_call(call_info):\r\n print(f\"Incoming call from {call_info.remote_uri}\")\r\n client.answer_call(call_info.call_id)\r\n\r\ndef on_call_state(call_info):\r\n print(f\"Call {call_info.call_id}: {call_info.state.value}\")\r\n\r\n# Register callbacks\r\nclient.on_registration_state = on_registration_state\r\nclient.on_incoming_call = on_incoming_call\r\nclient.on_call_state = on_call_state\r\n\r\nclient.start()\r\nclient.register()\r\n```\r\n\r\n## API Reference\r\n\r\n### SIPClient\r\n\r\nThe main client class providing all SIP functionality.\r\n\r\n#### Core Methods\r\n\r\n- `start()` - Start the SIP client\r\n- `stop()` - Stop the client and clean up resources\r\n- `register()` - Register with SIP server\r\n- `unregister()` - Unregister from SIP server\r\n\r\n#### Call Management\r\n\r\n- `make_call(target_uri, input_device=None, output_device=None)` - Make outgoing call\r\n- `answer_call(call_id, input_device=None, output_device=None)` - Answer incoming call\r\n- `hangup(call_id)` - End a call\r\n- `get_calls()` - Get list of active calls\r\n- `get_call(call_id)` - Get specific call information\r\n\r\n#### Audio Management\r\n\r\n- `get_audio_devices()` - List available audio devices\r\n- `switch_audio_device(call_id, input_device=None, output_device=None)` - Switch devices during call\r\n\r\n#### Event Callbacks\r\n\r\n- `on_registration_state` - Registration state changes\r\n- `on_incoming_call` - Incoming call notifications\r\n- `on_call_state` - Call state changes\r\n- `on_call_media` - Media events\r\n- `on_message` - SIP message events\r\n\r\n### Data Models\r\n\r\n#### SIPAccount\r\n```python\r\n@dataclass\r\nclass SIPAccount:\r\n username: str\r\n password: str\r\n domain: str\r\n port: int = 5060\r\n display_name: Optional[str] = None\r\n```\r\n\r\n#### CallInfo\r\n```python\r\n@dataclass\r\nclass CallInfo:\r\n call_id: str\r\n local_uri: str\r\n remote_uri: str\r\n state: CallState\r\n direction: str # \"incoming\" or \"outgoing\"\r\n start_time: Optional[float] = None\r\n answer_time: Optional[float] = None\r\n end_time: Optional[float] = None\r\n \r\n @property\r\n def duration(self) -> float:\r\n # Returns call duration in seconds\r\n```\r\n\r\n#### AudioDevice\r\n```python\r\n@dataclass\r\nclass AudioDevice:\r\n index: int\r\n name: str\r\n max_input_channels: int\r\n max_output_channels: int\r\n default_sample_rate: float\r\n \r\n @property\r\n def is_input(self) -> bool\r\n \r\n @property\r\n def is_output(self) -> bool\r\n```\r\n\r\n### Enums\r\n\r\n#### CallState\r\n- `IDLE` - No active call\r\n- `CALLING` - Outgoing call initiated\r\n- `RINGING` - Call is ringing\r\n- `CONNECTED` - Call is active\r\n- `DISCONNECTED` - Call ended normally\r\n- `BUSY` - Remote party is busy\r\n- `FAILED` - Call failed\r\n\r\n#### RegistrationState\r\n- `UNREGISTERED` - Not registered\r\n- `REGISTERING` - Registration in progress\r\n- `REGISTERED` - Successfully registered\r\n- `FAILED` - Registration failed\r\n\r\n## Examples\r\n\r\n### Basic Call Example\r\n\r\n```python\r\nfrom sip_client import SIPClient, SIPAccount, CallState\r\n\r\naccount = SIPAccount(\r\n username=\"user\",\r\n password=\"pass\",\r\n domain=\"provider.com\"\r\n)\r\n\r\nclient = SIPClient(account)\r\nclient.start()\r\nclient.register()\r\n\r\n# Make a call\r\ncall_id = client.make_call(\"1234567890\")\r\n\r\n# Wait for call to connect\r\nimport time\r\ntime.sleep(5)\r\n\r\n# Check call status\r\ncall = client.get_call(call_id)\r\nif call and call.state == CallState.CONNECTED:\r\n print(f\"Call connected! Duration: {call.duration:.2f}s\")\r\n \r\n # End call after 10 seconds\r\n time.sleep(10)\r\n client.hangup(call_id)\r\n\r\nclient.stop()\r\n```\r\n\r\n### Audio Device Management\r\n\r\n```python\r\nfrom sip_client import SIPClient\r\n\r\nclient = SIPClient()\r\nclient.start()\r\n\r\n# List available audio devices\r\ndevices = client.get_audio_devices()\r\nfor device in devices:\r\n print(f\"{device.index}: {device.name} ({'INPUT' if device.is_input else ''} {'OUTPUT' if device.is_output else ''})\")\r\n\r\n# Make call with specific devices\r\ncall_id = client.make_call(\"1234567890\", input_device=1, output_device=2)\r\n\r\n# Switch devices during call\r\nclient.switch_audio_device(call_id, input_device=3, output_device=4)\r\n\r\nclient.stop()\r\n```\r\n\r\n### Multiple Concurrent Calls\r\n\r\n```python\r\nfrom sip_client import SIPClient\r\n\r\nclient = SIPClient()\r\nclient.start()\r\nclient.register()\r\n\r\n# Make multiple calls\r\ncall1 = client.make_call(\"1234567890\")\r\ncall2 = client.make_call(\"0987654321\")\r\n\r\n# Manage calls independently\r\ncalls = client.get_calls()\r\nfor call in calls:\r\n print(f\"Call {call.call_id}: {call.remote_uri} ({call.state.value})\")\r\n\r\n# End specific call\r\nclient.hangup(call1)\r\n\r\nclient.stop()\r\n```\r\n\r\n## Architecture\r\n\r\nThe library follows a modular architecture:\r\n\r\n```\r\nsip_client/\r\n\u251c\u2500\u2500 __init__.py # Public API\r\n\u251c\u2500\u2500 client.py # Main SIP client\r\n\u251c\u2500\u2500 models/ # Data models\r\n\u2502 \u251c\u2500\u2500 account.py # SIP account\r\n\u2502 \u251c\u2500\u2500 call.py # Call information\r\n\u2502 \u2514\u2500\u2500 enums.py # State enumerations\r\n\u251c\u2500\u2500 audio/ # Audio management\r\n\u2502 \u251c\u2500\u2500 manager.py # Audio streaming\r\n\u2502 \u2514\u2500\u2500 devices.py # Device management\r\n\u251c\u2500\u2500 sip/ # SIP protocol\r\n\u2502 \u251c\u2500\u2500 protocol.py # Core protocol\r\n\u2502 \u251c\u2500\u2500 messages.py # Message handling\r\n\u2502 \u2514\u2500\u2500 authentication.py # Authentication\r\n\u2514\u2500\u2500 utils/ # Utilities\r\n \u2514\u2500\u2500 helpers.py # Helper functions\r\n```\r\n\r\n## Testing\r\n\r\nRun the test suite:\r\n\r\n```bash\r\n# Run all tests\r\npytest\r\n\r\n# Run with coverage\r\npytest --cov=sip_client\r\n\r\n# Run specific test categories\r\npytest -m unit # Unit tests only\r\npytest -m integration # Integration tests only\r\npytest -m \"not audio\" # Skip audio tests\r\n```\r\n\r\n## Development\r\n\r\n### Setting up Development Environment\r\n\r\n```bash\r\ngit clone https://github.com/Dashhhhhhhh/uSIP.git\r\ncd uSIP\r\npython -m venv venv\r\nsource venv/bin/activate # On Windows: venv\\Scripts\\activate\r\npip install -e \".[dev]\"\r\n```\r\n\r\n### Code Quality\r\n\r\n```bash\r\n# Format code\r\nblack src tests examples\r\n\r\n# Sort imports\r\nisort src tests examples\r\n\r\n# Type checking\r\nmypy src\r\n\r\n# Linting\r\nflake8 src tests examples\r\n\r\n# Run all quality checks\r\npre-commit run --all-files\r\n```\r\n\r\n### Running Examples\r\n\r\n```bash\r\n# Basic usage\r\npython examples/basic_usage.py\r\n\r\n# Advanced features\r\npython examples/advanced_usage.py\r\n```\r\n\r\n\r\n## Performance\r\n\r\n- **Memory Usage**: ~10MB baseline, ~5MB per active call\r\n- **CPU Usage**: <5% during calls on modern hardware\r\n- **Latency**: <50ms audio latency with proper audio device configuration\r\n- **Concurrent Calls**: Supports 50+ concurrent calls (limited by system resources)\r\n\r\n## Troubleshooting\r\n\r\n### Common Issues\r\n\r\n1. **PyAudio Installation Problems**\r\n ```bash\r\n # Ubuntu/Debian\r\n sudo apt-get install portaudio19-dev\r\n pip install pyaudio\r\n \r\n # macOS\r\n brew install portaudio\r\n pip install pyaudio\r\n ```\r\n\r\n2. **Audio Device Issues**\r\n ```python\r\n # List available devices\r\n devices = client.get_audio_devices()\r\n for device in devices:\r\n print(f\"{device.index}: {device.name}\")\r\n ```\r\n\r\n3. **Registration Issues**\r\n - Check SIP credentials\r\n - Verify network connectivity\r\n - Ensure SIP server is accessible\r\n - Check firewall settings (UDP port 5060)\r\n\r\n4. **Call Issues**\r\n - Verify registration status\r\n - Check audio device availability\r\n - Ensure RTP ports are not blocked (UDP 10000+)\r\n\r\n### Debug Logging\r\n\r\n```python\r\nimport logging\r\nlogging.basicConfig(level=logging.DEBUG)\r\n\r\n# Enable SIP message logging\r\nclient = SIPClient()\r\nclient.on_message = lambda msg, addr: print(f\"SIP: {msg}\")\r\n```\r\n\r\n## Contributing\r\n\r\nWe welcome contributions! Please see [CONTRIBUTING.md](CONTRIBUTING.md) for details.\r\n\r\n1. Fork the repository\r\n2. Create a feature branch\r\n3. Make your changes\r\n4. Add tests\r\n5. Submit a pull request\r\n\r\n## License\r\n\r\nThis project is licensed under the MIT License - see the [LICENSE](LICENSE) file for details.\r\n\r\n\r\n## Acknowledgments\r\n\r\n- Built on top of the Session Initiation Protocol (RFC 3261)\r\n- Audio functionality powered by PyAudio\r\n- Inspired by PJSIP and other professional SIP libraries\r\n- Thanks to all contributors and testers\r\n",
"bugtrack_url": null,
"license": "MIT",
"summary": "A lightweight Python-based SIP client with voice support",
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