sip-to-ai

# SIP-to-AI

Why this project exists

Most voice agent frameworks:
- require WebRTC
- rely on heavy dependencies (C / ffmpeg)
- are not telephony-native

This project:
- pure Python SIP + RTP (no C deps)
- direct bridge to realtime AI
- built for call center / telephony scenarios

**SIP-to-AI** — stream RTP audio from **FreeSWITCH / OpenSIPS / Asterisk** directly to **end-to-end realtime voice models**:
- ✅ **OpenAI Realtime API** (gpt-realtime GA)
- ✅ **Deepgram Voice Agent**
- ✅ **Gemini Live** (Gemini 2.5 Flash)

Simple passthrough bridge: **SIP (G.711 μ-law @ 8kHz)** ↔ **AI voice models**. OpenAI and Deepgram support native G.711, Gemini requires PCM16 resampling (8kHz ↔ 16kHz/24kHz).

## Quick Start (OpenAI Realtime)

**Prerequisites:** Python 3.12+, UV package manager

**Pure Python, No External Dependencies:** This project uses a pure Python asyncio implementation of SIP+RTP. No C libraries or compilation required!

1. **Install dependencies:**
   ```bash
   git clone <repository-url>
   cd sip-to-ai
   uv venv && source .venv/bin/activate
   uv sync
   ```

2. **Configure environment:**
   ```bash
   cp .env.example .env
   ```

   Edit `.env` with your OpenAI API key:
   ```bash
   # AI Service
   AI_VENDOR=openai
   OPENAI_API_KEY=sk-proj-your-key-here
   OPENAI_MODEL=gpt-realtime

   # Agent prompt
   AGENT_PROMPT_FILE=agent_prompt.yaml

   # SIP Settings (userless account - receive only)
   SIP_DOMAIN=192.168.1.100
   SIP_TRANSPORT_TYPE=udp
   SIP_PORT=6060
   ```

   **Optional:** Create `agent_prompt.yaml` for custom agent personality:
   ```yaml
   instructions: |
     You are a helpful AI assistant. Be concise and friendly.

   greeting: "Hello! How can I help you today?"
   ```

3. **Run the server:**
   ```bash
   uv run python -m app.main
   ```

   The server will listen on `SIP_DOMAIN:SIP_PORT` for incoming calls. Each call creates an independent OpenAI Realtime WebSocket connection.

4. **Make a test call:**
   ```bash
   # From FreeSWITCH/Aste