voice-ai-development
/voice-ai-development
Pomaga budować aplikacje głosowe AI od agentów czasu rzeczywistego po aplikacje z głosem: OpenAI Realtime API, Vapi, Deepgram, ElevenLabs, LiveKit i podstawy WebRTC.
- Autor
- davila7 ↗
- Repozytorium
- claude-code-templates ↗
- Licencja
- MIT
- Ostatnia zmiana repozytorium
- 25.10.2026
Ocena i bezpieczeństwo
- Pliki w paczce
- 1
- Skrypty
- 0
- Komentarze HTML
- 0
- Adresy URL w SKILL.md
- 0
Brak sygnałów. Repozytorium zmienione 25.10.2026, bez odwołań do wycofanych modeli i API.
MIT, otwarta: można używać i zmieniać, z zachowaniem informacji o autorze.
Co robi i kiedy
Pomaga budować aplikacje głosowe AI od agentów czasu rzeczywistego po aplikacje z głosem: OpenAI Realtime API, Vapi, Deepgram, ElevenLabs, LiveKit i podstawy WebRTC.
Oryginalny opis (EN)
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. Use when: voice ai, voice agent, speech to text, text to speech, realtime voice.
Jak działa
- Capabilities
- Requirements
- Patterns
- Anti-Patterns
- Limitations
- Related Skills
Podgląd SKILL.md
---
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. Knows how to build low-latency, production-ready voice experiences. Use when: voice ai, voice agent, speech to text, text to speech, realtime voice."
metadata:
source: vibeship-spawner-skills (Apache 2.0)
---
# 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
```python
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
```python
# 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
```python
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`
Wersje
| Repozytorium | Wynik | Licencja |
|---|---|---|
| Ta strona · davila7/claude-code-templates | 100 | MIT |
| kumaran-is/claude-code-onboarding | 100 | MIT |
Identyczna treść występuje jeszcze w 10 innych folderach na GitHubie. Tu pokazujemy jedną, najlepiej ocenioną kopię.
Migawka bazy z 28.10.2026 · wersja paczki 94eb3ced851d (skrót SHA-256 całej paczki)
Pobrania w ostatnich 30 dniach: 0 · ranking
Zgłoś problem z tym skillem: szkodliwa treść, błędny opis albo prośba autora o usunięcie.
Podobne skille
- 95openai-realtime-voice
Buduje produkcyjne agenty głosowe OpenAI Realtime z sesjami audio, wywoływaniem narzędzi i kontrolą opóźnień oraz kosztów. - 84voicetest
Testuje agentów głosowych w narzędziu voicetest, także dla platform Retell, VAPI, LiveKit i Bland, przez grafy rozmów. - 100voice-agent-builder
Projektuje i buduje agentów głosowych AI: potoki speech-to-text, integracje text-to-speech, asystentów głosowych w czasie rzeczywistym, boty telefoniczne (Twilio, Vapi) i wykrywanie wake word z niskim opóźnieniem. - 100openai-agents
Buduje aplikacje AI z OpenAI Agents SDK: agenty tekstowe i głosowe z WebRTC, przekazania między agentami, narzędzia ze schematami Zod, guardrails, strumieniowanie oraz diagnozowanie błędów, takich jak MaxTurnsExceededError. - 100openai-realtime
Buduje agentów głosowych z OpenAI Realtime API przez WebRTC, WebSocket i SIP: strumieniowanie dźwięku z mikrofonu, obsługa przerwań i wywołań funkcji, odbieranie połączeń SIP oraz tymczasowe klucze dla przeglądarki. - 100azure-ai-voicelive-dotnet
Buduje w .NET aplikacje głosowe AI czasu rzeczywistego z dwukierunkową komunikacją WebSocket za pomocą Azure AI Voice Live SDK: asystenci głosowi, speech-to-speech i chatboty głosowe.