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…

INSTALLATION
npx skills add https://github.com/davila7/claude-code-templates --skill voice-ai-development
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SKILL.md

Voice AI Development

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.

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

Requirements

  • Python or Node.js
  • API keys for providers
  • Audio handling knowledge

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"] == "resp

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

Anti-Patterns

❌ Non-streaming Pipeline

Why bad: Adds seconds of latency.

User perceives as slow.

Loses conversation flow.

Instead: Stream everything:

  • STT: interim results
  • LLM: token streaming
  • TTS: chunk streaming

Start TTS before LLM finishes.

❌ Ignoring Interruptions

Why bad: Frustrating user experience.

Feels like talking to a machine.

Wastes time.

Instead: Implement barge-in detection.

Use VAD to detect user speech.

Stop TTS immediately.

Clear audio queue.

❌ Single Provider Lock-in

Why bad: May not be best quality.

Single point of failure.

Harder to optimize.

Instead: Mix best providers:

  • Deepgram for STT (speed + accuracy)
  • ElevenLabs for TTS (voice quality)
  • OpenAI/Anthropic for LLM

Limitations

  • Latency varies by provider
  • Cost per minute adds up
  • Quality depends on network
  • Complex debugging

Related Skills

Works well with: langgraph, structured-output, langfuse

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