SIP Voice Call Control

Voice interface for OpenClaw using Telnyx Call Control API. Call via SIP and have natural conversations with your AI assistant.

How It Works

Your Phone (SIP) → Telnyx Call Control → Webhooks → This Server → Commands back
                         ↓                              ↓
                    STT/TTS/Media              LLM + Tool Execution

This is not WebRTC — it's server-side SIP call handling. Telnyx handles all audio; we just receive webhooks and send commands.

Setup

1. Install Dependencies

cd skills/sip-voice-call-control
npm install

2. Configure

cp .env.example .env

Edit .env:

TELNYX_API_KEY=KEY_your_api_key_here

3. Start Server

npm run start

Console shows:

🎙️  Assistant Voice Server ready on port 3000
📞 Dial: sip:openclaw@yourconnection.sip.telnyx.com

Agent Setup Flow

When setting up for a user, agents should:

1. Collect Configuration

Ask for:

1. Telnyx API Key (required)

   • Get from: https://portal.telnyx.com/#/app/auth/v2
   • Needs Voice API + Inference permissions

2. TTS Voice (optional)

   • Telnyx.NaturalHD.astra — Female, warm and clear (default, recommended)
   • Telnyx.NaturalHD.atlas — Male, deep and authoritative
   • Polly.Amy-Neural — British female
   • Polly.Brian-Neural — British male
   • Polly.Joanna-Neural — American female
   • Polly.Matthew-Neural — American male
   • Azure.en-US-JennyNeural — American female
   • Azure.en-GB-RyanNeural — British male
   • See .env.example for full list

3. Voice Model (optional)

   • Qwen/Qwen3-235B-A22B — Best for function calling (default)
   • meta-llama/Meta-Llama-3.1-8B-Instruct — Fastest
   • meta-llama/Llama-3.3-70B-Instruct — Balanced

Personalization (assistant name, user name, timezone) is pulled automatically from workspace files (IDENTITY.md, USER.md).

2. Write .env File

cat > .env << 'EOF'
TELNYX_API_KEY=<user_api_key>
VOICE_MODEL=Qwen/Qwen3-235B-A22B
TTS_VOICE=Telnyx.NaturalHD.astra
EOF

3. Start in Background (Persistent)

The server must run persistently to receive calls. Use nohup to keep it alive:

cd /path/to/sip-voice-call-control
nohup npm run start > sip-voice-call-control.log 2>&1 &

Or from an agent:

// Use nohup to keep process alive after session ends
exec({ 
  command: "cd /path/to/sip-voice-call-control && nohup npm run start > sip-voice-call-control.log 2>&1 &",
  background: true 
})

Important: Without nohup, the process will die when the parent session ends. Always use nohup or a process manager for production.

To check if running:

ps aux | grep "tsx.*dev" | grep -v grep

To stop:

pkill -f "tsx.*dev.ts"

To view logs:

tail -f /path/to/sip-voice-call-control/sip-voice-call-control.log

4. Extract SIP Address

Poll the process logs and give the user the SIP dial-in:

📞 Dial: sip:openclaw@<connection>.sip.telnyx.com

Environment Variables

Available Tools

Features

• Low-latency — 500ms-1.5s response time with enable_thinking: false
• Barge-in — Interrupt the assistant anytime by speaking
• Function calling — Native tool support with Qwen
• Auto-setup — Cloudflare tunnel and Call Control app created automatically
• Personalization — Reads IDENTITY.md and USER.md for context

Troubleshooting

No response after speaking:

• Check Telnyx API key has Voice API + Inference permissions
• Verify webhook URL is reachable (tunnel must be active)

Slow responses (>3s):

• Ensure using function-calling branch (not main)
• Check model availability on your Telnyx account

Tool not executing:

• Ensure openclaw CLI is in PATH
• Check WORKSPACE_DIR is set correctly

Port already in use:

• Kill existing server: pkill -f "tsx.*dev.ts"
• Or change PORT in .env

Resources

• Telnyx Call Control: https://developers.telnyx.com/docs/voice/call-control
• Telnyx Inference: https://developers.telnyx.com/docs/inference
• See ARCHITECTURE.md for technical details