wavoice 0.1.0__tar.gz
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- wavoice-0.1.0/PKG-INFO +254 -0
- wavoice-0.1.0/README.md +235 -0
- wavoice-0.1.0/pyproject.toml +36 -0
- wavoice-0.1.0/pyproject.toml.orig +35 -0
- wavoice-0.1.0/src/wavoice/__init__.py +7 -0
- wavoice-0.1.0/src/wavoice/agents/gemini_live.py +138 -0
- wavoice-0.1.0/src/wavoice/cli.py +335 -0
- wavoice-0.1.0/src/wavoice/core/config.py +101 -0
- wavoice-0.1.0/src/wavoice/core/ice.py +72 -0
- wavoice-0.1.0/src/wavoice/core/lifecycle.py +72 -0
- wavoice-0.1.0/src/wavoice/core/meta.py +346 -0
- wavoice-0.1.0/src/wavoice/core/telemetry.py +154 -0
- wavoice-0.1.0/src/wavoice/core/tunnel.py +179 -0
- wavoice-0.1.0/src/wavoice/py.typed +0 -0
- wavoice-0.1.0/src/wavoice/sdk.py +196 -0
- wavoice-0.1.0/src/wavoice/server/app.py +298 -0
- wavoice-0.1.0/src/wavoice/server/security.py +43 -0
wavoice-0.1.0/PKG-INFO
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Metadata-Version: 2.3
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Name: wavoice
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Version: 0.1.0
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Summary: One-command WhatsApp Voice AI with Pipecat and automated Cloudflare Quick Tunnels
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Author: Amit Bhakt
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Author-email: Amit Bhakt <yashubhakt@gmail.com>
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Requires-Dist: aiohttp>=3.14.5
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Requires-Dist: fastapi>=0.143.0
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Requires-Dist: loguru>=0.7.3
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Requires-Dist: pipecat-ai[google,webrtc]>=1.12.0
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Requires-Dist: pydantic>=2.14.1
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Requires-Dist: pydantic-settings>=2.15.0
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Requires-Dist: python-dotenv>=1.2.4
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Requires-Dist: rich>=15.0.0
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Requires-Dist: typer>=0.27.3
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Requires-Dist: uvicorn[standard]>=0.54.0
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Requires-Python: >=3.12
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Description-Content-Type: text/markdown
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# WAVoice: Set Up WhatsApp Voice AI in 90 Seconds
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[](https://www.python.org/)
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[](https://opensource.org/licenses/MIT)
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[](https://github.com/pipecat-ai/pipecat)
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[](https://github.com/amitbhakt)
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> **The zero-friction, one-command pipeline for WhatsApp Voice AI.**
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> Go from zero to an active AI voice agent answering live WhatsApp calls in under **90 seconds**. No manual Meta dashboard clicking, no ngrok account setup, and no WebRTC configuration required.
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Created by [Amit Bhakt](https://github.com/amitbhakt), `WAVoice` solves the hardest part of building WhatsApp voice agents: **the entire telephony and networking plumbing**.
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---
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## Why Developers Use WAVoice
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Setting up WhatsApp voice calling traditionally requires hours of manual work: navigating Meta Business Manager, registering phone number IDs, manually creating ngrok tunnels, pasting webhook verification tokens, handling WebRTC SDP offer/answers, and diagnosing why audio drops over NAT.
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**`WAVoice` collapses all of that into a single 90-second command (`wavoice init`):**
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1. **Automated Cloudflare Quick Tunnel**: Ephemeral public HTTPS TLS gateway with zero tokens or account sign-ups.
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2. **Auto-Discovery**: Extracts your WABA ID and Phone Number ID directly from your token via Meta Graph API.
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3. **Programmatic Calling Activation**: Enables WhatsApp Voice Calling and makes the Call button visible automatically.
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4. **Full 3-Tier Webhook Sync**: Programmatically configures Phone Number overrides, WABA subscriptions, and App subscriptions.
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5. **Production WebRTC Infrastructure**: Pre-configured STUN/TURN, SRTP audio handling, and atomic call claim locks (preventing duplicate webhook deliveries).
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---
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## How It Works: Instant Testing vs. Custom Pipelines
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Whether you want a complete out-of-the-box agent or a custom multi-model architecture, `WAVoice` separates the **telephony infrastructure** from your **AI logic**:
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```
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┌────────────────────────────────────────┐
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│ Telephony Infrastructure Layer │
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│ (Handled automatically by WAVoice) │
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│ │
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│ - Cloudflare Quick Tunnel │
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│ - Meta Graph API Webhooks │
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│ - WebRTC SDP Offer / Answer │
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│ - SRTP Audio Stream Decoupling │
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└───────────────────┬────────────────────┘
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│
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┌───────────────────┴────────────────────┐
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▼ ▼
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┌─────────────────────────────────┐ ┌─────────────────────────────────┐
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│ Mode A: Out-of-the-Box (90s) │ │ Mode B: Custom Layers │
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│ │ │ (Pluggable Pipecat Pipeline) │
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│ Run: `wavoice start` │ │ │
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│ Native Speech-to-Speech via │ │ - STT: Deepgram, Whisper, Groq │
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│ Gemini 2.0 Flash Live │ │ - LLM: OpenAI, Claude, Ollama │
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│ (~300ms latency, zero code) │ │ - TTS: Cartesia, ElevenLabs │
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└─────────────────────────────────┘ └─────────────────────────────────┘
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```
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### How does the one-command setup help developers who want separate layers?
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When you want to build a cascaded stack (**STT $\rightarrow$ LLM $\rightarrow$ TTS**), you still face the exact same frustrating hurdle: connecting WhatsApp's WebRTC audio stream to your server.
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By running `wavoice init`, **your WhatsApp number and webhook plumbing are 100% configured in 90 seconds**. Then, instead of maintaining complex SIP/WebRTC servers, you simply write standard, modular Python:
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```python
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from wavoice import WhatsAppVoiceApp
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app = WhatsAppVoiceApp()
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@app.on_call
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async def handle_call(transport, call_info):
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# 'transport' already receives raw audio from WhatsApp WebRTC
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# Plug in any separate STT, LLM, or TTS processor:
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pipeline = Pipeline([
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transport.input(),
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DeepgramSTTService(...),
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OpenAILLMService(...),
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CartesiaTTSService(...),
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transport.output(),
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])
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...
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```
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---
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## 90-Second Quickstart
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### Step 1: Install
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```bash
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git clone https://github.com/amitbhakt/wavoice.git
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cd wavoice
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uv sync
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```
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*(Or install via `pip install wavoice`)*
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### Step 2: One-Command Setup
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```bash
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uv run wavoice init
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```
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The interactive wizard asks for only 3 credentials:
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1. `WHATSAPP_TOKEN`: Meta System User access token.
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2. `WHATSAPP_APP_SECRET`: Meta App Secret.
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3. `GEMINI_API_KEY`: Google Gemini API key.
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`WAVoice` automatically discovers your business accounts, activates WhatsApp voice calling, saves `.env`, and starts the server immediately.
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### Step 3: Dial & Talk
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Open WhatsApp on your phone and dial your business number. Your AI agent answers immediately.
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---
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## Pre-Flight Diagnostics
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To inspect your setup, token scopes, voice calling flags, and webhook routing tiers at any time:
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```bash
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uv run wavoice doctor
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```
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Output:
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```text
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╦ ╦╔═╗╦ ╦╔═╗╦╔═╗╔═╗ WAVoice v0.1.0
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║║║╠═╣╚╗╔╝║ ║║║ ║╣ One-command WhatsApp Voice AI runtime
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╚╩╝╩ ╩ ╚╝ ╚═╝╩╚═╝╚═╝
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╭────────────────────────────────╮
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│ WAVoice Pre-Flight Diagnostics │
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╰────────────────────────────────╯
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┏━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━┳━━━━━━━━━━━━━━┳━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━┓
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┃ Check ┃ Status ┃ Details ┃
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┡━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━╇━━━━━━━━━━━━━━╇━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━┩
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│ Meta Credentials │ PASS │ Token & Phone ID configured │
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│ Phone Number Reachability │ PASS │ Verified number: +1 555-188 │
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│ WhatsApp Voice Calling API │ PASS │ Voice calling is ENABLED │
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│ Tier 1: Phone Override │ ACTIVE │ Cloudflare Tunnel Verified │
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│ Tier 2: WABA Subscription │ ACTIVE │ Webhook Subscribed │
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│ Tier 3: App Subscriptions │ ACTIVE │ calls, messages Active │
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│ Default AI Agent │ PASS │ Gemini 2.0 Flash Live ready │
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└────────────────────────────────┴──────────────┴──────────────────────────────┘
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```
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---
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## Building Custom Multi-Model Pipelines
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For developers building with independent STT, LLM, and TTS providers, `WAVoice` exposes the complete [Pipecat](https://github.com/pipecat-ai/pipecat) pipeline:
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```python
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import os
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from wavoice import WhatsAppVoiceApp
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from pipecat.audio.vad.silero import SileroVADAnalyzer
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from pipecat.pipeline.pipeline import Pipeline
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from pipecat.pipeline.runner import PipelineRunner
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from pipecat.pipeline.task import PipelineParams, PipelineTask
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from pipecat.processors.aggregators.llm_context import LLMContext
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from pipecat.processors.aggregators.llm_response_universal import (
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LLMContextAggregatorPair,
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LLMUserAggregatorParams,
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)
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from pipecat.services.deepgram import DeepgramSTTService
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from pipecat.services.openai import OpenAILLMService
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from pipecat.services.cartesia import CartesiaTTSService
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app = WhatsAppVoiceApp()
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@app.on_call
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async def handle_incoming_call(transport, call_info):
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caller = getattr(call_info, "from_", "caller")
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print(f"Incoming call from {caller}")
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# 1. Custom Layer Services
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stt = DeepgramSTTService(api_key=os.environ["DEEPGRAM_API_KEY"])
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llm = OpenAILLMService(api_key=os.environ["OPENAI_API_KEY"], model="gpt-4o-mini")
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tts = CartesiaTTSService(api_key=os.environ["CARTESIA_API_KEY"], voice_id="...")
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# 2. Context Aggregation & Silero Turn Detection
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context = LLMContext([{"role": "system", "content": "You are a concise voice assistant."}])
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user_agg, assistant_agg = LLMContextAggregatorPair(
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context,
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user_params=LLMUserAggregatorParams(vad_analyzer=SileroVADAnalyzer()),
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)
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# 3. Assemble Pipeline
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pipeline = Pipeline([
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transport.input(), # WebRTC Audio In from WhatsApp
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stt, # Speech-to-Text
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user_agg, # Turn Aggregator
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llm, # Large Language Model
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tts, # Text-to-Speech
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transport.output(), # WebRTC Audio Out to WhatsApp
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assistant_agg,
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])
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task = PipelineTask(pipeline=pipeline, params=PipelineParams(allow_interruptions=True))
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runner = PipelineRunner()
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await runner.run(task)
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if __name__ == "__main__":
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app.run() # Launches tunnel, registers Meta webhooks, and serves pipeline
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```
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*(See [examples/custom_stt_llm_tts.py](examples/custom_stt_llm_tts.py) for the complete runnable file)*
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---
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## CLI Command Reference
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| Command | Description |
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| :--- | :--- |
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| `wavoice init` | Complete 90-second setup: prompts for 3 credentials, auto-discovers IDs, turns on voice calling, and launches. Pass `--no-telemetry` to opt out of usage metrics. |
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| `wavoice doctor` | Runs diagnostics across Meta credentials, calling status, and 3-tier webhooks. |
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| `wavoice start` | Starts the server, establishes the Cloudflare tunnel, and begins accepting calls. Pass `--no-telemetry` to run with telemetry disabled. |
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---
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## Technical Details
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- **WebRTC Protocol**: WhatsApp Calling exchanges SDP offers/answers over HTTPS and streams media via SRTP over UDP. `WAVoice` uses `aiortc` and Pipecat's `SmallWebRTCTransport` to negotiate codecs and stream PCM audio.
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- **Webhook Delivery Deduplication**: Meta delivers webhooks at least once, causing race conditions where multiple workers answer the same call. `WAVoice` uses atomic in-memory claiming per `call_id` to guarantee calls are answered exactly once.
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- **Bi-directional Termination**: If the caller hangs up, `WAVoice` tears down the pipeline. If the bot hangs up or times out, `WAVoice` calls `POST /v21.0/{phone_number_id}/calls` with `action: "terminate"` to end the call on the user's phone.
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- **Telemetry & Privacy**: WAVoice collects anonymous usage metrics (command runs, answered calls, and coarse country/timezone) to improve SDK stability. No personal info, phone numbers, tokens, or audio data are ever collected. Telemetry is enabled by default, and you can opt out at any time by:
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- Running `wavoice init --no-telemetry` or `wavoice start --no-telemetry`
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- Setting `WAVOICE_TELEMETRY=false` in `.env`
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- Setting `DO_NOT_TRACK=1` in your environment
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---
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## Maintainer & Credits
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Crafted by **Amit Bhakt** ([@amitbhakt](https://github.com/amitbhakt)).
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Contributions, issues, and feature suggestions are welcome on GitHub at [https://github.com/amitbhakt/wavoice](https://github.com/amitbhakt/wavoice).
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---
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## License
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MIT License. Copyright (c) 2026 Amit Bhakt.
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wavoice-0.1.0/README.md
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# WAVoice: Set Up WhatsApp Voice AI in 90 Seconds
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[](https://www.python.org/)
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[](https://opensource.org/licenses/MIT)
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[](https://github.com/pipecat-ai/pipecat)
|
|
6
|
+
[](https://github.com/amitbhakt)
|
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7
|
+
|
|
8
|
+
> **The zero-friction, one-command pipeline for WhatsApp Voice AI.**
|
|
9
|
+
> Go from zero to an active AI voice agent answering live WhatsApp calls in under **90 seconds**. No manual Meta dashboard clicking, no ngrok account setup, and no WebRTC configuration required.
|
|
10
|
+
|
|
11
|
+
Created by [Amit Bhakt](https://github.com/amitbhakt), `WAVoice` solves the hardest part of building WhatsApp voice agents: **the entire telephony and networking plumbing**.
|
|
12
|
+
|
|
13
|
+
---
|
|
14
|
+
|
|
15
|
+
## Why Developers Use WAVoice
|
|
16
|
+
|
|
17
|
+
Setting up WhatsApp voice calling traditionally requires hours of manual work: navigating Meta Business Manager, registering phone number IDs, manually creating ngrok tunnels, pasting webhook verification tokens, handling WebRTC SDP offer/answers, and diagnosing why audio drops over NAT.
|
|
18
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+
|
|
19
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+
**`WAVoice` collapses all of that into a single 90-second command (`wavoice init`):**
|
|
20
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+
|
|
21
|
+
1. **Automated Cloudflare Quick Tunnel**: Ephemeral public HTTPS TLS gateway with zero tokens or account sign-ups.
|
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22
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+
2. **Auto-Discovery**: Extracts your WABA ID and Phone Number ID directly from your token via Meta Graph API.
|
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23
|
+
3. **Programmatic Calling Activation**: Enables WhatsApp Voice Calling and makes the Call button visible automatically.
|
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24
|
+
4. **Full 3-Tier Webhook Sync**: Programmatically configures Phone Number overrides, WABA subscriptions, and App subscriptions.
|
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25
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+
5. **Production WebRTC Infrastructure**: Pre-configured STUN/TURN, SRTP audio handling, and atomic call claim locks (preventing duplicate webhook deliveries).
|
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26
|
+
|
|
27
|
+
---
|
|
28
|
+
|
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29
|
+
## How It Works: Instant Testing vs. Custom Pipelines
|
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30
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+
|
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31
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+
Whether you want a complete out-of-the-box agent or a custom multi-model architecture, `WAVoice` separates the **telephony infrastructure** from your **AI logic**:
|
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|
+
|
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+
```
|
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34
|
+
┌────────────────────────────────────────┐
|
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35
|
+
│ Telephony Infrastructure Layer │
|
|
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|
+
│ (Handled automatically by WAVoice) │
|
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|
+
│ │
|
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38
|
+
│ - Cloudflare Quick Tunnel │
|
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39
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+
│ - Meta Graph API Webhooks │
|
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+
│ - WebRTC SDP Offer / Answer │
|
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|
+
│ - SRTP Audio Stream Decoupling │
|
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|
+
└───────────────────┬────────────────────┘
|
|
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|
+
│
|
|
44
|
+
┌───────────────────┴────────────────────┐
|
|
45
|
+
▼ ▼
|
|
46
|
+
┌─────────────────────────────────┐ ┌─────────────────────────────────┐
|
|
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|
+
│ Mode A: Out-of-the-Box (90s) │ │ Mode B: Custom Layers │
|
|
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|
+
│ │ │ (Pluggable Pipecat Pipeline) │
|
|
49
|
+
│ Run: `wavoice start` │ │ │
|
|
50
|
+
│ Native Speech-to-Speech via │ │ - STT: Deepgram, Whisper, Groq │
|
|
51
|
+
│ Gemini 2.0 Flash Live │ │ - LLM: OpenAI, Claude, Ollama │
|
|
52
|
+
│ (~300ms latency, zero code) │ │ - TTS: Cartesia, ElevenLabs │
|
|
53
|
+
└─────────────────────────────────┘ └─────────────────────────────────┘
|
|
54
|
+
```
|
|
55
|
+
|
|
56
|
+
### How does the one-command setup help developers who want separate layers?
|
|
57
|
+
When you want to build a cascaded stack (**STT $\rightarrow$ LLM $\rightarrow$ TTS**), you still face the exact same frustrating hurdle: connecting WhatsApp's WebRTC audio stream to your server.
|
|
58
|
+
|
|
59
|
+
By running `wavoice init`, **your WhatsApp number and webhook plumbing are 100% configured in 90 seconds**. Then, instead of maintaining complex SIP/WebRTC servers, you simply write standard, modular Python:
|
|
60
|
+
|
|
61
|
+
```python
|
|
62
|
+
from wavoice import WhatsAppVoiceApp
|
|
63
|
+
|
|
64
|
+
app = WhatsAppVoiceApp()
|
|
65
|
+
|
|
66
|
+
@app.on_call
|
|
67
|
+
async def handle_call(transport, call_info):
|
|
68
|
+
# 'transport' already receives raw audio from WhatsApp WebRTC
|
|
69
|
+
# Plug in any separate STT, LLM, or TTS processor:
|
|
70
|
+
pipeline = Pipeline([
|
|
71
|
+
transport.input(),
|
|
72
|
+
DeepgramSTTService(...),
|
|
73
|
+
OpenAILLMService(...),
|
|
74
|
+
CartesiaTTSService(...),
|
|
75
|
+
transport.output(),
|
|
76
|
+
])
|
|
77
|
+
...
|
|
78
|
+
```
|
|
79
|
+
|
|
80
|
+
---
|
|
81
|
+
|
|
82
|
+
## 90-Second Quickstart
|
|
83
|
+
|
|
84
|
+
### Step 1: Install
|
|
85
|
+
```bash
|
|
86
|
+
git clone https://github.com/amitbhakt/wavoice.git
|
|
87
|
+
cd wavoice
|
|
88
|
+
uv sync
|
|
89
|
+
```
|
|
90
|
+
*(Or install via `pip install wavoice`)*
|
|
91
|
+
|
|
92
|
+
### Step 2: One-Command Setup
|
|
93
|
+
```bash
|
|
94
|
+
uv run wavoice init
|
|
95
|
+
```
|
|
96
|
+
|
|
97
|
+
The interactive wizard asks for only 3 credentials:
|
|
98
|
+
1. `WHATSAPP_TOKEN`: Meta System User access token.
|
|
99
|
+
2. `WHATSAPP_APP_SECRET`: Meta App Secret.
|
|
100
|
+
3. `GEMINI_API_KEY`: Google Gemini API key.
|
|
101
|
+
|
|
102
|
+
`WAVoice` automatically discovers your business accounts, activates WhatsApp voice calling, saves `.env`, and starts the server immediately.
|
|
103
|
+
|
|
104
|
+
### Step 3: Dial & Talk
|
|
105
|
+
Open WhatsApp on your phone and dial your business number. Your AI agent answers immediately.
|
|
106
|
+
|
|
107
|
+
---
|
|
108
|
+
|
|
109
|
+
## Pre-Flight Diagnostics
|
|
110
|
+
|
|
111
|
+
To inspect your setup, token scopes, voice calling flags, and webhook routing tiers at any time:
|
|
112
|
+
|
|
113
|
+
```bash
|
|
114
|
+
uv run wavoice doctor
|
|
115
|
+
```
|
|
116
|
+
|
|
117
|
+
Output:
|
|
118
|
+
```text
|
|
119
|
+
╦ ╦╔═╗╦ ╦╔═╗╦╔═╗╔═╗ WAVoice v0.1.0
|
|
120
|
+
║║║╠═╣╚╗╔╝║ ║║║ ║╣ One-command WhatsApp Voice AI runtime
|
|
121
|
+
╚╩╝╩ ╩ ╚╝ ╚═╝╩╚═╝╚═╝
|
|
122
|
+
|
|
123
|
+
╭────────────────────────────────╮
|
|
124
|
+
│ WAVoice Pre-Flight Diagnostics │
|
|
125
|
+
╰────────────────────────────────╯
|
|
126
|
+
┏━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━┳━━━━━━━━━━━━━━┳━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━┓
|
|
127
|
+
┃ Check ┃ Status ┃ Details ┃
|
|
128
|
+
┡━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━╇━━━━━━━━━━━━━━╇━━━━━━━━━━━━━━━━━━━━━━━━━━━━━━┩
|
|
129
|
+
│ Meta Credentials │ PASS │ Token & Phone ID configured │
|
|
130
|
+
│ Phone Number Reachability │ PASS │ Verified number: +1 555-188 │
|
|
131
|
+
│ WhatsApp Voice Calling API │ PASS │ Voice calling is ENABLED │
|
|
132
|
+
│ Tier 1: Phone Override │ ACTIVE │ Cloudflare Tunnel Verified │
|
|
133
|
+
│ Tier 2: WABA Subscription │ ACTIVE │ Webhook Subscribed │
|
|
134
|
+
│ Tier 3: App Subscriptions │ ACTIVE │ calls, messages Active │
|
|
135
|
+
│ Default AI Agent │ PASS │ Gemini 2.0 Flash Live ready │
|
|
136
|
+
└────────────────────────────────┴──────────────┴──────────────────────────────┘
|
|
137
|
+
```
|
|
138
|
+
|
|
139
|
+
---
|
|
140
|
+
|
|
141
|
+
## Building Custom Multi-Model Pipelines
|
|
142
|
+
|
|
143
|
+
For developers building with independent STT, LLM, and TTS providers, `WAVoice` exposes the complete [Pipecat](https://github.com/pipecat-ai/pipecat) pipeline:
|
|
144
|
+
|
|
145
|
+
```python
|
|
146
|
+
import os
|
|
147
|
+
from wavoice import WhatsAppVoiceApp
|
|
148
|
+
from pipecat.audio.vad.silero import SileroVADAnalyzer
|
|
149
|
+
from pipecat.pipeline.pipeline import Pipeline
|
|
150
|
+
from pipecat.pipeline.runner import PipelineRunner
|
|
151
|
+
from pipecat.pipeline.task import PipelineParams, PipelineTask
|
|
152
|
+
from pipecat.processors.aggregators.llm_context import LLMContext
|
|
153
|
+
from pipecat.processors.aggregators.llm_response_universal import (
|
|
154
|
+
LLMContextAggregatorPair,
|
|
155
|
+
LLMUserAggregatorParams,
|
|
156
|
+
)
|
|
157
|
+
from pipecat.services.deepgram import DeepgramSTTService
|
|
158
|
+
from pipecat.services.openai import OpenAILLMService
|
|
159
|
+
from pipecat.services.cartesia import CartesiaTTSService
|
|
160
|
+
|
|
161
|
+
app = WhatsAppVoiceApp()
|
|
162
|
+
|
|
163
|
+
@app.on_call
|
|
164
|
+
async def handle_incoming_call(transport, call_info):
|
|
165
|
+
caller = getattr(call_info, "from_", "caller")
|
|
166
|
+
print(f"Incoming call from {caller}")
|
|
167
|
+
|
|
168
|
+
# 1. Custom Layer Services
|
|
169
|
+
stt = DeepgramSTTService(api_key=os.environ["DEEPGRAM_API_KEY"])
|
|
170
|
+
llm = OpenAILLMService(api_key=os.environ["OPENAI_API_KEY"], model="gpt-4o-mini")
|
|
171
|
+
tts = CartesiaTTSService(api_key=os.environ["CARTESIA_API_KEY"], voice_id="...")
|
|
172
|
+
|
|
173
|
+
# 2. Context Aggregation & Silero Turn Detection
|
|
174
|
+
context = LLMContext([{"role": "system", "content": "You are a concise voice assistant."}])
|
|
175
|
+
user_agg, assistant_agg = LLMContextAggregatorPair(
|
|
176
|
+
context,
|
|
177
|
+
user_params=LLMUserAggregatorParams(vad_analyzer=SileroVADAnalyzer()),
|
|
178
|
+
)
|
|
179
|
+
|
|
180
|
+
# 3. Assemble Pipeline
|
|
181
|
+
pipeline = Pipeline([
|
|
182
|
+
transport.input(), # WebRTC Audio In from WhatsApp
|
|
183
|
+
stt, # Speech-to-Text
|
|
184
|
+
user_agg, # Turn Aggregator
|
|
185
|
+
llm, # Large Language Model
|
|
186
|
+
tts, # Text-to-Speech
|
|
187
|
+
transport.output(), # WebRTC Audio Out to WhatsApp
|
|
188
|
+
assistant_agg,
|
|
189
|
+
])
|
|
190
|
+
|
|
191
|
+
task = PipelineTask(pipeline=pipeline, params=PipelineParams(allow_interruptions=True))
|
|
192
|
+
runner = PipelineRunner()
|
|
193
|
+
await runner.run(task)
|
|
194
|
+
|
|
195
|
+
if __name__ == "__main__":
|
|
196
|
+
app.run() # Launches tunnel, registers Meta webhooks, and serves pipeline
|
|
197
|
+
```
|
|
198
|
+
|
|
199
|
+
*(See [examples/custom_stt_llm_tts.py](examples/custom_stt_llm_tts.py) for the complete runnable file)*
|
|
200
|
+
|
|
201
|
+
---
|
|
202
|
+
|
|
203
|
+
## CLI Command Reference
|
|
204
|
+
|
|
205
|
+
| Command | Description |
|
|
206
|
+
| :--- | :--- |
|
|
207
|
+
| `wavoice init` | Complete 90-second setup: prompts for 3 credentials, auto-discovers IDs, turns on voice calling, and launches. Pass `--no-telemetry` to opt out of usage metrics. |
|
|
208
|
+
| `wavoice doctor` | Runs diagnostics across Meta credentials, calling status, and 3-tier webhooks. |
|
|
209
|
+
| `wavoice start` | Starts the server, establishes the Cloudflare tunnel, and begins accepting calls. Pass `--no-telemetry` to run with telemetry disabled. |
|
|
210
|
+
|
|
211
|
+
---
|
|
212
|
+
|
|
213
|
+
## Technical Details
|
|
214
|
+
|
|
215
|
+
- **WebRTC Protocol**: WhatsApp Calling exchanges SDP offers/answers over HTTPS and streams media via SRTP over UDP. `WAVoice` uses `aiortc` and Pipecat's `SmallWebRTCTransport` to negotiate codecs and stream PCM audio.
|
|
216
|
+
- **Webhook Delivery Deduplication**: Meta delivers webhooks at least once, causing race conditions where multiple workers answer the same call. `WAVoice` uses atomic in-memory claiming per `call_id` to guarantee calls are answered exactly once.
|
|
217
|
+
- **Bi-directional Termination**: If the caller hangs up, `WAVoice` tears down the pipeline. If the bot hangs up or times out, `WAVoice` calls `POST /v21.0/{phone_number_id}/calls` with `action: "terminate"` to end the call on the user's phone.
|
|
218
|
+
- **Telemetry & Privacy**: WAVoice collects anonymous usage metrics (command runs, answered calls, and coarse country/timezone) to improve SDK stability. No personal info, phone numbers, tokens, or audio data are ever collected. Telemetry is enabled by default, and you can opt out at any time by:
|
|
219
|
+
- Running `wavoice init --no-telemetry` or `wavoice start --no-telemetry`
|
|
220
|
+
- Setting `WAVOICE_TELEMETRY=false` in `.env`
|
|
221
|
+
- Setting `DO_NOT_TRACK=1` in your environment
|
|
222
|
+
|
|
223
|
+
---
|
|
224
|
+
|
|
225
|
+
## Maintainer & Credits
|
|
226
|
+
|
|
227
|
+
Crafted by **Amit Bhakt** ([@amitbhakt](https://github.com/amitbhakt)).
|
|
228
|
+
|
|
229
|
+
Contributions, issues, and feature suggestions are welcome on GitHub at [https://github.com/amitbhakt/wavoice](https://github.com/amitbhakt/wavoice).
|
|
230
|
+
|
|
231
|
+
---
|
|
232
|
+
|
|
233
|
+
## License
|
|
234
|
+
|
|
235
|
+
MIT License. Copyright (c) 2026 Amit Bhakt.
|
|
@@ -0,0 +1,36 @@
|
|
|
1
|
+
[project]
|
|
2
|
+
name = "wavoice"
|
|
3
|
+
version = "0.1.0"
|
|
4
|
+
description = "One-command WhatsApp Voice AI with Pipecat and automated Cloudflare Quick Tunnels"
|
|
5
|
+
readme = "README.md"
|
|
6
|
+
requires-python = ">=3.12"
|
|
7
|
+
dependencies = [
|
|
8
|
+
"aiohttp>=3.14.5",
|
|
9
|
+
"fastapi>=0.143.0",
|
|
10
|
+
"loguru>=0.7.3",
|
|
11
|
+
"pipecat-ai[google,webrtc]>=1.12.0",
|
|
12
|
+
"pydantic>=2.14.1",
|
|
13
|
+
"pydantic-settings>=2.15.0",
|
|
14
|
+
"python-dotenv>=1.2.4",
|
|
15
|
+
"rich>=15.0.0",
|
|
16
|
+
"typer>=0.27.3",
|
|
17
|
+
"uvicorn[standard]>=0.54.0",
|
|
18
|
+
]
|
|
19
|
+
|
|
20
|
+
[[project.authors]]
|
|
21
|
+
name = "Amit Bhakt"
|
|
22
|
+
email = "yashubhakt@gmail.com"
|
|
23
|
+
|
|
24
|
+
[project.scripts]
|
|
25
|
+
wavoice = "wavoice.cli:cli"
|
|
26
|
+
|
|
27
|
+
[build-system]
|
|
28
|
+
requires = ["uv_build>=0.12.13,<0.13.0"]
|
|
29
|
+
build-backend = "uv_build"
|
|
30
|
+
|
|
31
|
+
[dependency-groups]
|
|
32
|
+
dev = [
|
|
33
|
+
"httpx>=0.28.1",
|
|
34
|
+
"pytest>=9.1.1",
|
|
35
|
+
"pytest-asyncio>=1.4.0",
|
|
36
|
+
]
|
|
@@ -0,0 +1,35 @@
|
|
|
1
|
+
[project]
|
|
2
|
+
name = "wavoice"
|
|
3
|
+
version = "0.1.0"
|
|
4
|
+
description = "One-command WhatsApp Voice AI with Pipecat and automated Cloudflare Quick Tunnels"
|
|
5
|
+
readme = "README.md"
|
|
6
|
+
authors = [
|
|
7
|
+
{ name = "Amit Bhakt", email = "yashubhakt@gmail.com" }
|
|
8
|
+
]
|
|
9
|
+
requires-python = ">=3.12"
|
|
10
|
+
dependencies = [
|
|
11
|
+
"aiohttp>=3.14.5",
|
|
12
|
+
"fastapi>=0.143.0",
|
|
13
|
+
"loguru>=0.7.3",
|
|
14
|
+
"pipecat-ai[google,webrtc]>=1.12.0",
|
|
15
|
+
"pydantic>=2.14.1",
|
|
16
|
+
"pydantic-settings>=2.15.0",
|
|
17
|
+
"python-dotenv>=1.2.4",
|
|
18
|
+
"rich>=15.0.0",
|
|
19
|
+
"typer>=0.27.3",
|
|
20
|
+
"uvicorn[standard]>=0.54.0",
|
|
21
|
+
]
|
|
22
|
+
|
|
23
|
+
[project.scripts]
|
|
24
|
+
wavoice = "wavoice.cli:cli"
|
|
25
|
+
|
|
26
|
+
[build-system]
|
|
27
|
+
requires = ["uv_build>=0.12.13,<0.13.0"]
|
|
28
|
+
build-backend = "uv_build"
|
|
29
|
+
|
|
30
|
+
[dependency-groups]
|
|
31
|
+
dev = [
|
|
32
|
+
"httpx>=0.28.1",
|
|
33
|
+
"pytest>=9.1.1",
|
|
34
|
+
"pytest-asyncio>=1.4.0",
|
|
35
|
+
]
|