Robert Helewka
ecf37658ce
Complete project scaffolding and core implementation of an AI-powered telephony system that calls companies, navigates IVR menus, waits on hold, and transfers to the user when a human answers. Key components: - FastAPI server with REST API, WebSocket, and MCP (SSE) interfaces - SIP/VoIP call management via PJSUA2 with RTP audio streaming - LLM-powered IVR navigation using OpenAI/Anthropic with tool calling - Hold detection service combining audio analysis and silence detection - Real-time STT (Whisper/Deepgram) and TTS (OpenAI/Piper) pipelines - Call recording with per-channel and mixed audio capture - Event bus (asyncio pub/sub) for real-time client updates - Web dashboard with live call monitoring - SQLite persistence via SQLAlchemy with call history and analytics - Notification support (email, SMS, webhook, desktop) - Docker Compose deployment with Opal VoIP and Opal Media containers - Comprehensive test suite with unit, integration, and E2E tests - Simplified .gitignore and full project documentation in README
166 lines
5.2 KiB
Markdown
166 lines
5.2 KiB
Markdown
# Configuration
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All configuration is via environment variables, loaded through Pydantic Settings. Copy `.env.example` to `.env` and edit.
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## Environment Variables
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### SIP Trunk
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| Variable | Description | Default | Required |
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|----------|-------------|---------|----------|
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| `SIP_TRUNK_HOST` | Your SIP provider hostname | — | Yes |
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| `SIP_TRUNK_PORT` | SIP signaling port | `5060` | No |
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| `SIP_TRUNK_USERNAME` | SIP auth username | — | Yes |
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| `SIP_TRUNK_PASSWORD` | SIP auth password | — | Yes |
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| `SIP_TRUNK_DID` | Your phone number (E.164) | — | Yes |
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| `SIP_TRUNK_TRANSPORT` | Transport protocol (`udp`, `tcp`, `tls`) | `udp` | No |
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### Gateway
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| Variable | Description | Default | Required |
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|----------|-------------|---------|----------|
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| `GATEWAY_SIP_PORT` | Port for device SIP registration | `5080` | No |
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| `GATEWAY_RTP_PORT_MIN` | Minimum RTP port | `10000` | No |
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| `GATEWAY_RTP_PORT_MAX` | Maximum RTP port | `20000` | No |
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| `GATEWAY_HOST` | Bind address | `0.0.0.0` | No |
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### LLM
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| Variable | Description | Default | Required |
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|----------|-------------|---------|----------|
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| `LLM_BASE_URL` | OpenAI-compatible API endpoint | `http://localhost:11434/v1` | No |
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| `LLM_MODEL` | Model name for IVR analysis | `llama3` | No |
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| `LLM_API_KEY` | API key (if required) | `not-needed` | No |
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| `LLM_TIMEOUT` | Request timeout in seconds | `30.0` | No |
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| `LLM_MAX_TOKENS` | Max tokens per response | `1024` | No |
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| `LLM_TEMPERATURE` | Sampling temperature | `0.3` | No |
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### Speech-to-Text
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| Variable | Description | Default | Required |
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|----------|-------------|---------|----------|
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| `SPEACHES_URL` | Speaches/Whisper STT endpoint | `http://localhost:22070` | No |
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| `SPEACHES_MODEL` | Whisper model name | `whisper-large-v3` | No |
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### Database
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| Variable | Description | Default | Required |
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|----------|-------------|---------|----------|
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| `DATABASE_URL` | PostgreSQL or SQLite connection string | `sqlite+aiosqlite:///./hold_slayer.db` | No |
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### Notifications
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| Variable | Description | Default | Required |
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|----------|-------------|---------|----------|
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| `NOTIFY_SMS_NUMBER` | Phone number for SMS alerts (E.164) | — | No |
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### Audio Classifier
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| Variable | Description | Default | Required |
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|----------|-------------|---------|----------|
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| `CLASSIFIER_WINDOW_SECONDS` | Audio window size for classification | `3.0` | No |
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| `CLASSIFIER_SILENCE_THRESHOLD` | RMS below this = silence | `0.85` | No |
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| `CLASSIFIER_MUSIC_THRESHOLD` | Spectral flatness below this = music | `0.7` | No |
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| `CLASSIFIER_SPEECH_THRESHOLD` | Spectral flatness above this = speech | `0.6` | No |
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### Hold Slayer
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| Variable | Description | Default | Required |
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|----------|-------------|---------|----------|
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| `MAX_HOLD_TIME` | Maximum seconds to wait on hold | `7200` | No |
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| `HOLD_CHECK_INTERVAL` | Seconds between audio checks | `2.0` | No |
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| `DEFAULT_TRANSFER_DEVICE` | Device to transfer to | `sip_phone` | No |
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### Recording
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| Variable | Description | Default | Required |
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|----------|-------------|---------|----------|
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| `RECORDING_DIR` | Directory for WAV recordings | `recordings` | No |
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| `RECORDING_MAX_SECONDS` | Maximum recording duration | `7200` | No |
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| `RECORDING_SAMPLE_RATE` | Audio sample rate | `16000` | No |
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## Settings Architecture
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Configuration is managed by Pydantic Settings in `config.py`:
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```python
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from config import get_settings
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settings = get_settings()
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settings.sip_trunk_host # "sip.provider.com"
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settings.llm.base_url # "http://localhost:11434/v1"
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settings.llm.model # "llama3"
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settings.speaches_url # "http://localhost:22070"
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settings.database_url # "sqlite+aiosqlite:///./hold_slayer.db"
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```
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LLM settings are nested under `settings.llm` as a `LLMSettings` sub-model.
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## Deployment
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### Development
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```bash
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# 1. Clone and install
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git clone <repo-url>
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cd hold-slayer
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python -m venv .venv
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source .venv/bin/activate
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pip install -e ".[dev]"
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# 2. Configure
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cp .env.example .env
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# Edit .env
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# 3. Start Ollama (for LLM)
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ollama serve
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ollama pull llama3
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# 4. Start Speaches (for STT)
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docker run -p 22070:8000 ghcr.io/speaches-ai/speaches
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# 5. Run
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uvicorn main:app --host 0.0.0.0 --port 8000 --reload
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```
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### Production
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```bash
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# Use PostgreSQL instead of SQLite
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DATABASE_URL=postgresql+asyncpg://user:pass@localhost/hold_slayer
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# Use vLLM for faster inference
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LLM_BASE_URL=http://localhost:8000/v1
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LLM_MODEL=meta-llama/Llama-3-8B-Instruct
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# Run with multiple workers (note: each worker is independent)
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uvicorn main:app --host 0.0.0.0 --port 8000 --workers 1
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```
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Note: Hold Slayer is designed as a single-process application. Multiple workers would each have their own SIP engine and call state. For high availability, run behind a load balancer with sticky sessions.
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### Docker
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```dockerfile
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FROM python:3.13-slim
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# Install system dependencies for PJSUA2 and Sippy
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RUN apt-get update && apt-get install -y \
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build-essential \
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libpjproject-dev \
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&& rm -rf /var/lib/apt/lists/*
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WORKDIR /app
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COPY . .
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RUN pip install -e .
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EXPOSE 8000 5080/udp 10000-20000/udp
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CMD ["uvicorn", "main:app", "--host", "0.0.0.0", "--port", "8000"]
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```
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Port mapping:
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- `8000` — HTTP API + WebSocket + MCP
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- `5080/udp` — SIP device registration
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- `10000-20000/udp` — RTP media ports
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