What problem does it solve?
Enable developers to build low-latency, bidirectional voice AI applications without managing raw WebSocket protocols or event plumbing, reducing integration complexity for real-time assistants, voice chatbots, and transcription pipelines.
Core Features & Use Cases
- Bidirectional WebSocket Streaming: Stream base64 PCM16 audio to Azure, receive audio deltas and transcripts in real time for responsive voice assistants.
- Turn Detection & Interrupts: Supports server VAD, Azure semantic VAD, and manual turn control for push-to-talk and interruption handling.
- Tooling & Integration: Function calling, MCP tools, avatar/webrtc integration, transcription-only flows, and robust typed event handling for production workflows.
- Use Case: Build a Python voice assistant that transcribes user speech, triggers a backend function for events like "get_weather", and returns synthesized audio with low latency.
Quick Start
Connect to Azure Voice Live using DefaultAzureCredential or an API key, stream microphone audio as base64 PCM16, commit the input buffer, and create a response to receive the assistant's real-time audio and transcript.