voice-ai-development

---
name: voice-ai-development
description: Expert in building voice AI applications - from real-time voice
  agents to voice-enabled apps. Covers OpenAI Realtime API, Vapi for voice
  agents, Deepgram for transcription, ElevenLabs for synthesis, LiveKit for
  real-time infrastructure, and WebRTC fundamentals.
risk: unknown
source: vibeship-spawner-skills (Apache 2.0)
date_added: 2026-02-27
---

# Voice AI Development

Expert in building voice AI applications - from real-time voice agents to voice-enabled apps.
Covers OpenAI Realtime API, Vapi for voice agents, Deepgram for transcription, ElevenLabs
for synthesis, LiveKit for real-time infrastructure, and WebRTC fundamentals. Knows how to
build low-latency, production-ready voice experiences.

**Role**: Voice AI Architect

You are an expert in building real-time voice applications. You think in terms of
latency budgets, audio quality, and user experience. You know that voice apps feel
magical when fast and broken when slow. You choose the right combination of providers
for each use case and optimize relentlessly for perceived responsiveness.

### Expertise

- Real-time audio streaming
- Voice agent architecture
- Provider selection
- Latency optimization
- Audio quality tuning

## Capabilities

- OpenAI Realtime API
- Vapi voice agents
- Deepgram STT/TTS
- ElevenLabs voice synthesis
- LiveKit real-time infrastructure
- WebRTC audio handling
- Voice agent design
- Latency optimization

## Prerequisites

- 0: Async programming
- 1: WebSocket basics
- 2: Audio concepts (sample rate, codec)
- Required skills: Python or Node.js, API keys for providers, Audio handling knowledge

## Scope

- 0: Latency varies by provider
- 1: Cost per minute adds up
- 2: Quality depends on network
- 3: Complex debugging

## Ecosystem

### Primary

- OpenAI Realtime API
- Vapi
- Deepgram
- ElevenLabs

### Infrastructure

- LiveKit
- Daily.co
- Twilio

### Common_integrations

- WebRTC
- WebSockets
- Telephony (SIP/PSTN)

### Platforms

- Web applications
- Mobile apps
- Call centers
- Voice assistants

## Patterns

### OpenAI Realtime API

Native voice-to-voice with GPT-4o

**When to use**: When you want integrated voice AI without separate STT/TTS

import asyncio
import websockets
import json
import base64

OPENAI_API_KEY = "sk-..."

async def voice_session():
    url = "wss://api.openai.com/v1/realtime?model=gpt-4o-realtime-preview"
    headers = {
        "Authorization": f"Bearer {OPENAI_API_KEY}",
        "OpenAI-Beta": "realtime=v1"
    }

    async with websockets.connect(url, extra_headers=headers) as ws:
        # Configure session
        await ws.send(json.dumps({
            "type": "session.update",
            "session": {
                "modalities": ["text", "audio"],
                "voice": "alloy",  # alloy, echo, fable, onyx, nova, shimmer
                "input_audio_format": "pcm16",
                "output_audio_format": "pcm16",
                "input_audio_transcription": {
                    "model": "whisper-1"
                },
                "turn_detection": {
                    "type": "server_vad",  # Voice activity detection
                    "threshold": 0.5,
                    "prefix_padding_ms": 300,
                    "silence_duration_ms": 500
                },
                "tools": [
                    {
                        "type": "function",
                        "name": "get_weather",
                        "description": "Get weather for a location",
                        "parameters": {
                            "type": "object",
                            "properties": {
                                "location": {"type": "string"}
                            }
                        }
                    }
                ]
            }
        }))

        # Send audio (PCM16, 24kHz, mono)
        async def send_audio(audio_bytes):
            await ws.send(json.dumps({
                "type": "input_audio_buffer.append",
                "audio": base64.b64encode(audio_bytes).decode()
            }))

        # Receive events
        async for message in ws:
            event = json.loads(message)

            if event["type"] == "response.audio.delta":
                # Play audio chunk
                audio = base64.b64decode(event["delta"])
                play_audio(audio)

            elif event["type"] == "response.audio_transcript.done":
                print(f"Assistant said: {event['transcript']}")

            elif event["type"] == "input_audio_buffer.speech_started":
                print("User started speaking")

            elif event["type"] == "response.function_call_arguments.done":
                # Handle tool call
                name = event["name"]
                args = json.loads(event["arguments"])
                result = call_function(name, args)
                await ws.send(json.dumps({
                    "type": "conversation.item.create",
                    "item": {
                        "type": "function_call_output",
                        "call_id": event["call_id"],
                        "output": json.dumps(result)
                    }
                }))

### Vapi Voice Agent

Build voice agents with Vapi platform

**When to use**: Phone-based agents, quick deployment

# Vapi provides hosted voice agents with webhooks

from flask import Flask, request, jsonify
import vapi

app = Flask(__name__)
client = vapi.Vapi(api_key="...")

# Create an assistant
assistant = client.assistants.create(
    name="Support Agent",
    model={
        "provider": "openai",
        "model": "gpt-4o",
        "messages": [
            {
                "role": "system",
                "content": "You are a helpful support agent..."
            }
        ]
    },
    voice={
        "provider": "11labs",
        "voiceId": "21m00Tcm4TlvDq8ikWAM"  # Rachel
    },
    firstMessage="Hi! How can I help you today?",
    transcriber={
        "provider": "deepgram",
        "model": "nova-2"
    }
)

# Webhook for conversation events
@app.route("/vapi/webhook", methods=["POST"])
def vapi_webhook():
    event = request.json

    if event["type"] == "function-call":
        # Handle tool call
        name = event["functionCall"]["name"]
        args = event["functionCall"]["parameters"]

        if name == "check_order":
            result = check_order(args["order_id"])
            return jsonify({"result": result})

    elif event["type"] == "end-of-call-report":
        # Call ended - save transcript
        transcript = event["transcript"]
        save_transcript(event["call"]["id"], transcript)

    return jsonify({"ok": True})

# Start outbound call
call = client.calls.create(
    assistant_id=assistant.id,
    customer={
        "number": "+1234567890"
    },
    phoneNumber={
        "twilioPhoneNumber": "+0987654321"
    }
)

# Or create web call
web_call = client.calls.create(
    assistant_id=assistant.id,
    type="web"
)
# Returns URL for WebRTC connection

### Deepgram STT + ElevenLabs TTS

Best-in-class transcription and synthesis

**When to use**: High quality voice, custom pipeline

import asyncio
from deepgram import DeepgramClient, LiveTranscriptionEvents
from elevenlabs import ElevenLabs

# Deepgram real-time transcription
deepgram = DeepgramClient(api_key="...")

async def transcribe_stream(audio_stream):
    connection = deepgram.listen.live.v("1")

    async def on_transcript(result):
        transcript = result.channel.alternatives[0].transcript
        if transcript:
            print(f"Heard: {transcript}")
            if result.is_final:
                # Process final transcript
                await handle_user_input(transcript)

    connection.on(LiveTranscriptionEvents.Transcript, on_transcript)

    await connection.start({
        "model": "nova-2",  # Best quality
        "language": "en",
        "smart_format": True,
        "interim_results": True,  # Get partial results
        "utterance_end_ms": 1000,
        "vad_events": True,  # Voice activity detection
        "encoding": "linear16",
        "sample_rate": 16000
    })

    # Stream audio
    async for chunk in audio_stream:
        await connection.send(chunk)

    await connection.finish()

# ElevenLabs streaming synthesis
eleven = ElevenLabs(api_key="...")

def text_to_speech_stream(text: str):
    """Stream TTS audio chunks."""
    audio_stream = eleven.text_to_speech.convert_as_stream(
        voice_id="21m00Tcm4TlvDq8ikWAM",  # Rachel
        model_id="eleven_turbo_v2_5",  # Fastest
        text=text,
        output_format="pcm_24000"  # Raw PCM for low latency
    )

    for chunk in audio_stream:
        yield chunk

# Or with WebSocket for lowest latency
async def tts_websocket(text_stream):
    async with eleven.text_to_speech.stream_async(
        voice_id="21m00Tcm4TlvDq8ikWAM",
        model_id="eleven_turbo_v2_5"
    ) as tts:
        async for text_chunk in text_stream:
            audio = await tts.send(text_chunk)
            yield audio

        # Flush remaining audio
        final_audio = await tts.flush()
        yield final_audio

### LiveKit Real-time Infrastructure

WebRTC infrastructure for voice apps

**When to use**: Building custom real-time voice apps

from livekit import api, rtc
import asyncio

# Server-side: Create room and tokens
lk_api = api.LiveKitAPI(
    url="wss://your-livekit.livekit.cloud",
    api_key="...",
    api_secret="..."
)

async def create_room(room_name: str):
    room = await lk_api.room.create_room(
        api.CreateRoomRequest(name=room_name)
    )
    return room

def create_token(room_name: str, participant_name: str):
    token = api.AccessToken(
        api_key="...",
        api_secret="..."
    )
    token.with_identity(participant_name)
    token.with_grants(api.VideoGrants(
        room_join=True,
        room=room_name
    ))
    return token.to_jwt()

# Agent-side: Connect and process audio
async def voice_agent(room_name: str):
    room = rtc.Room()

    @room.on("track_subscribed")
    def on_track(track, publication, participant):
        if track.kind == rtc.TrackKind.KIND_AUDIO:
            # Process incoming audio
            audio_stream = rtc.AudioStream(track)
            asyncio.create_task(process_audio(audio_stream))

    token = create_token(room_name, "agent")
    await room.connect("wss://your-livekit.livekit.cloud", token)

    # Publish agent's audio
    source = rtc.AudioSource(sample_rate=24000, num_channels=1)
    track = rtc.LocalAudioTrack.create_audio_track("agent-voice", source)
    await room.local_participant.publish_track(track)

    # Send audio from TTS
    async def speak(text: str):
        for audio_chunk in text_to_speech(text):
            await source.capture_frame(rtc.AudioFrame(
                data=audio_chunk,
                sample_rate=24000,
                num_channels=1,
                samples_per_channel=len(audio_chunk) // 2
            ))

    return room, speak

# Process audio with STT
async def process_audio(audio_stream):
    async for frame in audio_stream:
        # Send to Deepgram or other STT
        await transcriber.send(frame.data)

### Full Voice Agent Pipeline

Complete voice agent with all components

**When to use**: Custom production voice agent

import asyncio
from dataclasses import dataclass
from typing import AsyncIterator

@dataclass
class VoiceAgentConfig:
    stt_provider: str = "deepgram"
    tts_provider: str = "elevenlabs"
    llm_provider: str = "openai"
    vad_enabled: bool = True
    interrupt_enabled: bool = True

class VoiceAgent:
    def __init__(self, config: VoiceAgentConfig):
        self.config = config
        self.is_speaking = False
        self.conversation_history = []

    async def process_audio_stream(
        self,
        audio_in: AsyncIterator[bytes],
        audio_out: asyncio.Queue
    ):
        """Main audio processing loop."""

        # STT streaming
        async def transcribe():
            transcript_buffer = ""
            async for audio_chunk in audio_in:
                # Check for interruption
                if self.is_speaking and self.config.interrupt_enabled:
                    if await self.detect_speech(audio_chunk):
                        await self.stop_speaking()

                result = await self.stt.transcribe(audio_chunk)
                if result.is_final:
                    yield result.transcript

        # Process transcripts
        async for user_text in transcribe():
            if not user_text.strip():
                continue

            self.conversation_history.append({
                "role": "user",
                "content": user_text
            })

            # Generate response with streaming
            self.is_speaking = True
            async for audio_chunk in self.generate_response(user_text):
                await audio_out.put(audio_chunk)
            self.is_speaking = False

    async def generate_response(self, text: str) -> AsyncIterator[bytes]:
        """Stream LLM response through TTS."""

        # Stream LLM tokens
        llm_stream = self.llm.stream_chat(self.conversation_history)

        # Buffer for TTS (need ~50 chars for good prosody)
        text_buffer = ""
        full_response = ""

        async for token in llm_stream:
            text_buffer += token
            full_response += token

            # Send to TTS when we have enough text
            if len(text_buffer) > 50 or token in ".!?":
                async for audio in self.tts.synthesize_stream(text_buffer):
                    yield audio
                text_buffer = ""

        # Flush remaining
        if text_buffer:
            async for audio in self.tts.synthesize_stream(text_buffer):
                yield audio

        self.conversation_history.append({
            "role": "assistant",
            "content": full_response
        })

    async def detect_speech(self, audio: bytes) -> bool:
        """Voice activity detection."""
        # Use WebRTC VAD or Silero VAD
        return self.vad.is_speech(audio)

    async def stop_speaking(self):
        """Handle interruption."""
        self.is_speaking = False
        # Clear audio queue
        # Stop TTS generation

# Latency optimization tips:
# 1. Use streaming everywhere (STT, LLM, TTS)
# 2. Start TTS before LLM finishes (~50 char buffer)
# 3. Use PCM audio format (no encoding overhead)
# 4. Keep WebSocket connections alive
# 5. Use regional endpoints close to users

## Validation Checks

### Non-Streaming TTS

Severity: HIGH

Message: Non-streaming TTS adds significant latency.

Fix action: Use tts.synthesize_stream() or tts.convert_as_stream()

### Hardcoded Sample Rate

Severity: MEDIUM

Message: Hardcoded sample rate may cause format mismatches.

Fix action: Define sample rates as constants, document expected formats

### WebSocket Without Reconnection

Severity: HIGH

Message: WebSocket connections need reconnection logic.

Fix action: Add retry loop with exponential backoff

### Missing VAD Configuration

Severity: MEDIUM

Message: VAD needs tuning for good user experience.

Fix action: Configure threshold and silence_duration_ms

### Blocking Audio Processing

Severity: HIGH

Message: Audio processing should be async to avoid blocking.

Fix action: Use async def and await for audio operations

### Missing Interruption Handling

Severity: MEDIUM

Message: Voice agents should handle user interruptions.

Fix action: Add barge-in detection and cancel current response

### Audio Queue Without Clear

Severity: LOW

Message: Audio queues should be clearable for interruptions.

Fix action: Add method to clear queue on interruption

### WebSocket Without Error Handling

Severity: HIGH

Message: WebSocket operations need error handling.

Fix action: Wrap in try/except for ConnectionClosed

## Collaboration

### Delegation Triggers

- agent graph|workflow|state -> langgraph (Need complex agent logic behind voice)
- extract|structured|json -> structured-output (Need to extract structured data from voice)
- observability|tracing|monitoring -> langfuse (Need to monitor voice agent quality)
- frontend|web|react -> nextjs-app-router (Need web interface for voice agent)

### Intelligent Voice Agent

Skills: voice-ai-development, langgraph, structured-output

Workflow:

```
1. Design agent graph with tools
2. Add voice interface layer
3. Use structured output for tool responses
4. Optimize for voice latency
```

### Monitored Voice Agent

Skills: voice-ai-development, langfuse

Workflow:

```
1. Build voice agent with provider of choice
2. Add Langfuse callbacks
3. Track latency, quality, conversation flow
4. Iterate based on metrics
```

### Phone-based Agent

Skills: voice-ai-development, twilio

Workflow:

```
1. Set up Vapi or custom agent
2. Connect to Twilio for PSTN
3. Handle inbound/outbound calls
4. Implement call routing logic
```

## Related Skills

Works well with: `langgraph`, `structured-output`, `langfuse`

## When to Use
- User mentions or implies: voice ai
- User mentions or implies: voice agent
- User mentions or implies: speech to text
- User mentions or implies: text to speech
- User mentions or implies: realtime voice
- User mentions or implies: vapi
- User mentions or implies: deepgram
- User mentions or implies: elevenlabs
- User mentions or implies: livekit
- User mentions or implies: openai realtime

## Limitations
- Use this skill only when the task clearly matches the scope described above.
- Do not treat the output as a substitute for environment-specific validation, testing, or expert review.
- Stop and ask for clarification if required inputs, permissions, safety boundaries, or success criteria are missing.

使用 Skill 时，您的对话内容将被发送至 AI 模型进行处理。我们会严格保护您的隐私数据，不会将您的对话内容用于模型训练或分享给第三方。 以下为此 Skill 的数据处理说明。