azure-ai-voicelive-py – DevOps & Infrastructure skill for AI agents

Bundled upstream support

Supporting files are included at skill-relative paths. See [UPSTREAM-NOTICE.md](UPSTREAM-NOTICE.md) for their pinned sources, licenses, hashes, and import changes.

Azure AI Voice Live SDK

Build real-time voice AI applications with bidirectional WebSocket communication.

Installation

pip install azure-ai-voicelive aiohttp azure-identity

Environment Variables

AZURE_COGNITIVE_SERVICES_ENDPOINT=https://<region>.api.cognitive.microsoft.com
# For API key auth (not recommended for production)
AZURE_COGNITIVE_SERVICES_KEY=<api-key>

Runtime and privacy boundaries

These snippets are version-sensitive sketches, not complete audio clients. Verify installed SDK/API version, resource endpoint, model availability and token audience together; reference defaults conflict with some explicit scopes below. Close async credentials explicitly, bound session duration/audio memory and cancel+await capture tasks in finally. Audio/transcripts leave the machine for Azure: obtain recording/upload authorization, minimize retention/logs and keep secrets in configured credentials. Validate tool names/JSON schemas and user authorization server-side; generated function calls are requests, not permission.

Authentication

DefaultAzureCredential (preferred):

from azure.ai.voicelive.aio import connect
from azure.identity.aio import DefaultAzureCredential

async with connect(
    endpoint=os.environ["AZURE_COGNITIVE_SERVICES_ENDPOINT"],
    credential=DefaultAzureCredential(),
    model="gpt-4o-realtime-preview",
    credential_scopes=["https://cognitiveservices.azure.com/.default"]
) as conn:
    ...

API Key:

from azure.ai.voicelive.aio import connect
from azure.core.credentials import AzureKeyCredential

async with connect(
    endpoint=os.environ["AZURE_COGNITIVE_SERVICES_ENDPOINT"],
    credential=AzureKeyCredential(os.environ["AZURE_COGNITIVE_SERVICES_KEY"]),
    model="gpt-4o-realtime-preview"
) as conn:
    ...

Quick Start

import asyncio
import os
from azure.ai.voicelive.aio import connect
from azure.identity.aio import DefaultAzureCredential

async def main():
    async with connect(
        endpoint=os.environ["AZURE_COGNITIVE_SERVICES_ENDPOINT"],
        credential=DefaultAzureCredential(),
        model="gpt-4o-realtime-preview",
        credential_scopes=["https://cognitiveservices.azure.com/.default"]
    ) as conn:
        # Update session with instructions
        await conn.session.update(session={
            "instructions": "You are a helpful assistant.",
            "modalities": ["text", "audio"],
            "voice": "alloy"
        })
        
        # Supply a user turn, then explicitly request a response.
        await conn.conversation.item.create(item={"type": "message", "role": "user",
            "content": [{"type": "input_text", "text": "Hello"}]})
        await conn.response.create()
        # Listen for events
        async for event in conn:
            print(f"Event: {event.type}")
            if event.type == "response.audio_transcript.done":
                print(f"Transcript: {event.transcript}")
            elif event.type == "response.done":
                break

asyncio.run(main())

Core Architecture

Connection Resources

The VoiceLiveConnection exposes these resources:

| Resource | Purpose | Key Methods |

|----------|---------|-------------|

| conn.session | Session configuration | update(session=...) |

| conn.response | Model responses | create(), cancel() |

| conn.input_audio_buffer | Audio input | append(), commit(), clear() |

| conn.output_audio_buffer | Audio output | clear() |

| conn.conversation | Conversation state | item.create(), item.delete(), item.truncate() |

| conn.transcription_session | Transcription config | update(session=...) |

Session Configuration

from azure.ai.voicelive.models import RequestSession, FunctionTool

await conn.session.update(session=RequestSession(
    instructions="You are a helpful voice assistant.",
    modalities=["text", "audio"],
    voice="alloy",  # or "echo", "shimmer", "sage", etc.
    input_audio_format="pcm16",
    output_audio_format="pcm16",
    turn_detection={
        "type": "server_vad",
        "threshold": 0.5,
        "prefix_padding_ms": 300,
        "silence_duration_ms": 500
    },
    tools=[
        FunctionTool(
            type="function",
            name="get_weather",
            description="Get current weather",
            parameters={
                "type": "object",
                "properties": {
                    "location": {"type": "string"}
                },
                "required": ["location"]
            }
        )
    ]
))

Audio Streaming

Send Audio (Base64 PCM16)

import base64

# Read audio chunk (16-bit PCM, 24kHz mono)
audio_chunk = await read_audio_from_microphone()
b64_audio = base64.b64encode(audio_chunk).decode()

await conn.input_audio_buffer.append(audio=b64_audio)

Receive Audio

async for event in conn:
    if event.type == "response.audio.delta":
        audio_bytes = base64.b64decode(event.delta)
        await play_audio(audio_bytes)
    elif event.type == "response.audio.done":
        print("Audio complete")

Event Handling

async for event in conn:
    match event.type:
        # Session events
        case "session.created":
            print(f"Session: {event.session}")
        case "session.updated":
            print("Session updated")
        
        # Audio input events
        case "input_audio_buffer.speech_started":
            print(f"Speech started at {event.audio_start_ms}ms")
        case "input_audio_buffer.speech_stopped":
            print(f"Speech stopped at {event.audio_end_ms}ms")
        
        # Transcription events
        case "conversation.item.input_audio_transcription.completed":
            print(f"User said: {event.transcript}")
        case "conversation.item.input_audio_transcription.delta":
            print(f"Partial: {event.delta}")
        
        # Response events
        case "response.created":
            print(f"Response started: {event.response.id}")
        case "response.audio_transcript.delta":
            print(event.delta, end="", flush=True)
        case "response.audio.delta":
            audio = base64.b64decode(event.delta)
        case "response.done":
            print(f"Response complete: {event.response.status}")
        
        # Function calls
        case "response.function_call_arguments.done":
            result = handle_function(event.name, event.arguments)
            await conn.conversation.item.create(item={
                "type": "function_call_output",
                "call_id": event.call_id,
                "output": json.dumps(result)
            })
            await conn.response.create()
        
        # Errors
        case "error":
            print(f"Error: {event.error.message}")

Common Patterns

Manual Turn Mode (No VAD)

await conn.session.update(session={"turn_detection": None})

# Manually control turns
await conn.input_audio_buffer.append(audio=b64_audio)
await conn.input_audio_buffer.commit()  # End of user turn
await conn.response.create()  # Trigger response

Interrupt Handling

async for event in conn:
    if event.type == "input_audio_buffer.speech_started":
        # User interrupted - cancel current response
        await conn.response.cancel()
        await conn.output_audio_buffer.clear()

Conversation History

# Add system message
await conn.conversation.item.create(item={
    "type": "message",
    "role": "system",
    "content": [{"type": "input_text", "text": "Be concise."}]
})

# Add user message
await conn.conversation.item.create(item={
    "type": "message",
    "role": "user", 
    "content": [{"type": "input_text", "text": "Hello!"}]
})

await conn.response.create()

Voice Options

| Voice | Description |

|-------|-------------|

| alloy | Neutral, balanced |

| echo | Warm, conversational |

| shimmer | Clear, professional |

| sage | Calm, authoritative |

| coral | Friendly, upbeat |

| ash | Deep, measured |

| ballad | Expressive |

| verse | Storytelling |

Azure voices: Use AzureStandardVoice, AzureCustomVoice, or AzurePersonalVoice models.

Audio Formats

| Format | Sample Rate | Use Case |

|--------|-------------|----------|

| pcm16 | 24kHz | Default, high quality |

| pcm16-8000hz | 8kHz | Telephony |

| pcm16-16000hz | 16kHz | Voice assistants |

| g711_ulaw | 8kHz | Telephony (US) |

| g711_alaw | 8kHz | Telephony (EU) |

Turn Detection Options

# Server VAD (default)
{"type": "server_vad", "threshold": 0.5, "silence_duration_ms": 500}

# Azure Semantic VAD (smarter detection)
{"type": "azure_semantic_vad"}
{"type": "azure_semantic_vad_en"}  # English optimized

… (full content is in the download package)