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SIP Setup Instructions

Date: 2025-10-31 Task: Enable SIP outbound calling via SignalWire

Current Status

✅ Code Complete - Commit 694f872

  • inbox.js updated to use SIP instead of LiveKit
  • Migration file created: supabase/migrations/20251031120000_add_sip_credentials.sql

⏸️ Blocked - Need to:

  1. Apply database migration (add SIP credential columns)
  2. Provision SIP endpoints in SignalWire
  3. Update database with SIP credentials

Step 1: Apply Database Migration

Option A: Via Supabase SQL Editor (Recommended)

  1. Open Supabase SQL Editor:

    https://supabase.com/dashboard/project/mtxbiyilvgwhbdptysex/sql/new

  2. Copy and paste this SQL:

    -- Add SIP credential fields to service_numbers table
ALTER TABLE service_numbers
ADD COLUMN IF NOT EXISTS sip_username VARCHAR(255),
ADD COLUMN IF NOT EXISTS sip_password VARCHAR(255),
ADD COLUMN IF NOT EXISTS sip_domain VARCHAR(255) DEFAULT 'erik.signalwire.com',
ADD COLUMN IF NOT EXISTS sip_ws_server VARCHAR(255) DEFAULT 'wss://erik.signalwire.com:7443';

-- Add comments
COMMENT ON COLUMN service_numbers.sip_username IS 'SIP username for WebRTC calling';
COMMENT ON COLUMN service_numbers.sip_password IS 'SIP password (should be encrypted in production)';
COMMENT ON COLUMN service_numbers.sip_domain IS 'SIP domain (SignalWire space)';
COMMENT ON COLUMN service_numbers.sip_ws_server IS 'WebSocket server for SIP over WebRTC';

  3. Click "Run" to execute

Option B: Via Supabase CLI (If connection works)

export SUPABASE_ACCESS_TOKEN=<your-token-from-supabase-dashboard>
npx supabase db push

Verify Migration Applied

Run this command to verify:

node scripts/check-sip-credentials.js

Expected output should show:

⚠️  🟢 Active +1XXXXXXXXXX
   ❌ Missing SIP credentials

Step 2: Provision SIP Endpoints in SignalWire

For Each Service Number:

  1. Log in to SignalWire Dashboard

    https://erik.signalwire.com

  2. Navigate to SIP Section

    • Click "Phone Numbers" in sidebar
    • Click "SIP" tab

  3. Create New SIP Endpoint

    • Click "+ Add Endpoint" or similar
    • Configure:
      • Username: Use phone number without + (e.g., 16042101966)
      • Password: Generate strong password (save it!)
      • Caller ID: The phone number (e.g., +16042101966)
      • Domain: erik.signalwire.com
      • WebSocket: Enable WebRTC/WebSocket support

  4. Link to Phone Number (if needed)

    • Some SignalWire interfaces require linking SIP endpoint to phone number
    • Associate the endpoint with the corresponding phone number

Alternative: SignalWire API

If SignalWire dashboard doesn't have SIP endpoint UI, we may need to:

  • Use SignalWire REST API to create SIP credentials
  • Check SignalWire documentation for "SIP Endpoints" or "WebRTC Endpoints"
  • Or configure via LaML/SWML scripts

Step 3: Update Database with SIP Credentials

Once you have SIP credentials from SignalWire, update the database:

Method 1: Via Supabase SQL Editor

-- For each service number, run:
UPDATE service_numbers SET
  sip_username = '16042101966',  -- Phone number without +
  sip_password = 'YOUR_PASSWORD_FROM_SIGNALWIRE',
  sip_domain = 'erik.signalwire.com',
  sip_ws_server = 'wss://erik.signalwire.com:7443'
WHERE phone_number = '+16042101966';

-- Repeat for each number

Method 2: Via Script

Create a script scripts/update-sip-credentials.js:

#!/usr/bin/env node
import { createClient } from '@supabase/supabase-js';
import dotenv from 'dotenv';

dotenv.config();

const supabase = createClient(
  process.env.VITE_SUPABASE_URL,
  process.env.SUPABASE_SERVICE_ROLE_KEY
);

const credentials = [
  {
    phone_number: '+16042101966',
    sip_username: '16042101966',
    sip_password: 'YOUR_PASSWORD_HERE',
    sip_domain: 'erik.signalwire.com',
    sip_ws_server: 'wss://erik.signalwire.com:7443'
  },
  // Add more numbers here
];

async function updateCredentials() {
  for (const cred of credentials) {
    const { error } = await supabase
      .from('service_numbers')
      .update({
        sip_username: cred.sip_username,
        sip_password: cred.sip_password,
        sip_domain: cred.sip_domain,
        sip_ws_server: cred.sip_ws_server
      })
      .eq('phone_number', cred.phone_number);

    if (error) {
      console.error(`Failed to update ${cred.phone_number}:`, error);
    } else {
      console.log(`✅ Updated ${cred.phone_number}`);
    }
  }
}

updateCredentials();

Then run:

chmod +x scripts/update-sip-credentials.js
node scripts/update-sip-credentials.js

Step 4: Test SIP Calling

  1. Verify SIP Credentials

    node scripts/check-sip-credentials.js

    Should show:

    ✅ 🟢 Active +16042101966
   SIP Username: 16042101966
   SIP Domain: erik.signalwire.com
   WS Server: wss://erik.signalwire.com:7443

  2. Open Inbox in Browser

    http://localhost:3000/inbox

  3. Make Test Call

    • Click on a contact or enter a phone number
    • Click the Call button
    • Watch browser console for SIP debug logs: 
      🔧 Initializing SIP client...
🔌 WebSocket connected successfully
✅ SIP registered successfully
📞 Call ringing...
✅ Call connected

  4. Check for Issues

    • SIP registration fails: Check username/password in database
    • Call doesn't connect: Check SignalWire trunk configuration
    • No audio: Check browser microphone permissions
    • WebSocket fails: Verify wss://erik.signalwire.com:7443 is correct endpoint

Troubleshooting

Error: "SIP registration failed"

  • Verify username/password in database match SignalWire endpoint
  • Check SignalWire endpoint is enabled
  • Verify domain is correct (erik.signalwire.com)

Error: "Call failed: 403 Forbidden"

  • SIP endpoint may not have permission to make outbound calls
  • Check SignalWire trunk allows outbound calling
  • Verify caller ID is authorized

Error: "WebSocket connection failed"

  • Verify WebSocket server URL is correct
  • Check firewall/network allows WSS connections on port 7443
  • Try alternative WebSocket endpoint if SignalWire provides one

No Audio

  • Check browser microphone permissions
  • Verify STUN/TURN servers in sipClient.js (lines 189-193)
  • Check browser console for WebRTC errors

Next Steps After SIP Works

Once basic SIP calling is working:

  1. Phase 2: LiveKit Bridging (specs/Pat-AI/plan.md)

    • Add CXML/SWML to bridge SIP call to LiveKit room
    • Enable recording by recording LiveKit room
    • Add agent to call via LiveKit agent

  2. Phase 3: Speaker Labeling & Tool Tracking

    • Label speakers: User, Agent, Guest
    • Track tool invocations in call records
    • Store in call_records table

Files Modified

  • ✅ src/pages/inbox.js:1499-1594 - initiateCall() uses SIP
  • ✅ src/pages/inbox.js:1447-1459 - hangup uses sipClient
  • ✅ supabase/migrations/20251031120000_add_sip_credentials.sql - Migration
  • ✅ scripts/check-sip-credentials.js - Helper script
  • ✅ scripts/apply-sip-migration.js - Helper script (deprecated)
  • ✅ SESSION-NOTES.md - Updated with current progress

Reference Links

Questions?

If you encounter issues not covered here, check:

  1. Browser console logs (F12 → Console)
  2. SignalWire call logs in dashboard
  3. Supabase Edge Function logs for any backend errors
  4. SESSION-NOTES.md for historical context