OpenAI WebRTC Python: a Python library for voice interaction with OpenAI real-time APIs

General Introduction

OpenAI Realtime WebRTC Python is a specialized Python library that provides developers with a complete solution for voice interaction with the OpenAI realtime API. The project is based on WebRTC technology, which realizes low-latency real-time audio transmission function. It not only supports automatic audio device management and sample rate conversion , but also provides a sound audio buffer management mechanism. The project is open source under the MIT license and supports multiple operating system platforms such as Windows, macOS and Linux. Through the library , developers can easily implement real-time speech recognition , audio stream processing and other advanced features , especially suitable for building applications that require real-time voice interaction .

Function List

• WebRTC-based low-latency real-time audio communication
• Support for OpenAI's latest Realtime API interface
• Automated management and configuration of intelligent audio devices
• Adaptive audio sample rate conversion
• Professional audio buffer management system
• Supports pause and resume control of audio streams
• Asynchronous audio processing and event callback mechanism
• Built-in audio to text function

Using Help

environmental preparation

1. system requirements
   • Python 3.7 or higher
   • Supports Windows, macOS, Linux operating systems
   • Ensure that the system has audio equipment available
2. installation process

   # 克隆项目代码
   git clone https://github.com/realtime-ai/openai-realtime-webrtc-python.git
   cd openai-realtime-webrtc-python
   # 创建并激活虚拟环境
   python -m venv venv
   source venv/bin/activate  # Linux/macOS系统
   # 或在Windows系统使用：
   # .\venv\Scripts\activate
   # 安装依赖包
   pip install -r requirements.txt
   # 开发模式安装
   pip install -e .
   

Configuration settings

1. Environment variable configuration
   • In the project root directory, create the.envfile
   • Add the OpenAI API key:

   OPENAI_API_KEY=your-api-key-here
   

Basic use process

1. Creating a Client Instance

   import asyncio
   from openai_realtime_webrtc import OpenAIWebRTCClient
   async def main():
   client = OpenAIWebRTCClient(
   api_key="your-api-key",
   model="gpt-4o-realtime-preview-2024-12-17"
   )
   

2. Setting the callback function

   def on_transcription(text: str):
   print(f"转录文本: {text}")
   client.on_transcription = on_transcription
   

3. Start audio streaming

   try:
   # 开始音频流传输
   await client.start_streaming()
   # 保持连接运行
   while True:
   await asyncio.sleep(1)
   except KeyboardInterrupt:
   # 终止音频流
   await client.stop_streaming()
   

Advanced Function Use

1. Audio Device Management
   • The system automatically detects and manages available audio input devices
   • Supports dynamic switching of audio devices
   • Automatic handling of sample rate conversion
2. Audio Flow Control
   • Supports pausing/resuming audio streaming at any time
   • Provides audio buffer management
   • Automatic handling of network latency and jitter
3. Error handling and monitoring
   • Built-in error detection and exception handling mechanisms
   • Supports audio quality monitoring
   • Provide detailed debugging information

caveat

• Ensure stable network connectivity
• Periodically check the validity of the API key
• Monitor the status of your audio devices.
• Reasonable control of the timing of starting and stopping the audio stream