Implementing Real-Time Streaming with VAPI for Live Support Chat Systems

Discover how to build a live support chat system using VAPI and Twilio. Learn about real-time voice transcription and low-latency TTS integration.

Misal Azeem
Misal Azeem

Voice AI Engineer & Creator

Implementing Real-Time Streaming with VAPI for Live Support Chat Systems

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Implementing Real-Time Streaming with VAPI for Live Support Chat Systems

TL;DR

Most live support systems fail when voice and text streams desynchronize. Here's how to build one that doesn't: VAPI handles real-time voice transcription via WebSocket streaming while Twilio manages SIP trunking. Use Server-Sent Events (SSE) for low-latency TTS integration and bidirectional audio routing. Result: sub-200ms transcription latency, zero dropped packets, agents see live captions while customers hear responses instantly.

Prerequisites

API Keys & Credentials

  • VAPI API key (generate at dashboard.vapi.ai)
  • Twilio Account SID and Auth Token (from console.twilio.com)
  • OpenAI API key for LLM inference (gpt-4 or gpt-3.5-turbo)
  • ElevenLabs API key for TTS (optional, if using custom voice provider)

System Requirements

  • Node.js 18+ with npm or yarn
  • WebSocket support (native in modern browsers and Node.js)
  • HTTPS endpoint for webhook callbacks (ngrok or production domain)
  • Minimum 2GB RAM for concurrent session handling

SDK Versions

  • vapi-js SDK v0.8.0+
  • twilio v4.0.0+
  • axios v1.6.0+ (for HTTP requests)

Network & Infrastructure

  • Stable internet connection (WebSocket streaming requires persistent TCP)
  • Firewall rules allowing outbound HTTPS to api.vapi.ai and api.twilio.com
  • Server capable of handling 100+ concurrent WebSocket connections (production deployments)

Knowledge Requirements

  • Familiarity with async/await and event-driven architecture
  • Basic understanding of WebSocket protocols and real-time bidirectional communication
  • Experience with REST APIs and webhook handling

VAPI: Get Started with VAPI → Get VAPI

Step-by-Step Tutorial

Most live support systems break when voice transcription lags behind user speech. Here's how to build a production-grade streaming chat system that handles real-time voice with sub-200ms latency.

Configuration & Setup

Start with your server infrastructure. You need two separate responsibilities: VAPI handles voice-to-text streaming, Twilio manages the phone call transport layer.

javascript
// Server setup - Express with WebSocket support for real-time updates
const express = require('express');
const WebSocket = require('ws');
const crypto = require('crypto');
const app = express();

// Production configuration with environment variables
const config = {
  vapi: {
    apiKey: process.env.VAPI_API_KEY,
    webhookSecret: process.env.VAPI_WEBHOOK_SECRET,
    baseUrl: 'https://api.vapi.ai'
  },
  twilio: {
    accountSid: process.env.TWILIO_ACCOUNT_SID,
    authToken: process.env.TWILIO_AUTH_TOKEN,
    phoneNumber: process.env.TWILIO_PHONE_NUMBER
  },
  server: {
    port: process.env.PORT || 3000,
    webhookUrl: process.env.WEBHOOK_URL // Your ngrok/production URL
  }
};

// Session state with TTL cleanup - prevents memory leaks
const activeSessions = new Map();
const SESSION_TTL = 30 * 60 * 1000; // 30 minutes

function initializeSession(sessionId) {
  const session = {
    id: sessionId,
    transcripts: [],
    isProcessing: false,
    startTime: Date.now(),
    metadata: {}
  };
  activeSessions.set(sessionId, session);
  
  // Auto-cleanup to prevent memory bloat
  setTimeout(() => {
    if (activeSessions.has(sessionId)) {
      activeSessions.delete(sessionId);
      console.log(`Session ${sessionId} expired and cleaned up`);
    }
  }, SESSION_TTL);
  
  return session;
}

// Webhook signature validation - security is not optional
function validateSignature(signature, body, secret) {
  const hmac = crypto.createHmac('sha256', secret);
  const digest = hmac.update(JSON.stringify(body)).digest('hex');
  return crypto.timingSafeEqual(
    Buffer.from(signature),
    Buffer.from(digest)
  );
}

app.use(express.json());
app.use(express.urlencoded({ extended: true }));

Critical: Do NOT mix VAPI's assistant configuration with Twilio's call handling. VAPI processes the voice stream, Twilio routes the call. Trying to configure voice synthesis in both creates double audio.

Architecture & Flow

mermaid
flowchart LR
    A[Customer Calls] --> B[Twilio Receives Call]
    B --> C[Forward to VAPI Assistant]
    C --> D[Real-time STT Stream]
    D --> E[Your Webhook Handler]
    E --> F[Process & Route]
    F --> G[TTS Response via VAPI]
    G --> H[Stream to Customer]

The flow is unidirectional for audio: Twilio → VAPI → Your Server → VAPI → Twilio. Never try to inject audio mid-stream from your server.

Step-by-Step Implementation

Step 1: Create the VAPI Assistant

Configure streaming transcription with aggressive barge-in detection. Most systems fail here by using default thresholds.

javascript
// Assistant config for live support - optimized for interruptions
const assistantConfig = {
  model: {
    provider: "openai",
    model: "gpt-4",
    temperature: 0.7,
    messages: [{
      role: "system",
      content: "You are a live support agent. Keep responses under 20 words. If customer interrupts, stop immediately."
    }]
  },
  voice: {
    provider: "11labs",
    voiceId: "21m00Tcm4TlvDq8ikWAM", // Rachel voice
    stability: 0.5,
    similarityBoost: 0.75,
    optimizeStreamingLatency: 3 // Max optimization for real-time
  },
  transcriber: {
    provider: "deepgram",
    model: "nova-2",
    language: "en",
    smartFormat: true,
    endpointing: 150 // Aggressive - detect silence after 150ms
  },
  recordingEnabled: true,
  firstMessage: "Hi, I'm here to help. What can I assist you with today?",
  serverUrl: config.server.webhookUrl + "/webhook/vapi",
  serverUrlSecret: config.vapi.webhookSecret
};

// Create assistant via VAPI API - this is YOUR server calling VAPI
async function createAssistant() {
  try {
    const response = await fetch(`${config.vapi.baseUrl}/assistant`, {
      method: 'POST',
      headers: {
        'Authorization': `Bearer ${config.vapi.apiKey}`,
        'Content-Type': 'application/json'
      },
      body: JSON.stringify(assistantConfig)
    });
    
    if (!response.ok) {
      const error = await response.json();
      throw new Error(`VAPI API error ${response.status}: ${error.message}`);
    }
    
    const assistant = await response.json();
    console.log(`Assistant created: ${assistant.id}`);
    return assistant.id;
  } catch (error) {
    console.error('Assistant creation failed:', error);
    throw error;
  }
}

Step 2: Handle Streaming Transcripts

Process partial transcripts as they arrive. This is where race conditions kill most implementations.

javascript
// Webhook handler - processes real-time events from VAPI
app.post('/webhook/vapi', async (req, res) => {
  const event = req.body;
  
  // Validate webhook signature - prevents replay attacks
  const signature = req.headers['x-vapi-signature'];
  if (!signature || !validateSignature(signature, req.body, config.vapi.webhookSecret)) {
    console.error('Invalid webhook signature');
    return res.status(401).json({ error: 'Invalid signature' });
  }

  const sessionId = event.call?.id;
  if (!sessionId) {
    return res.status(400).json({ error: 'Missing call ID' });
  }
  
  // Initialize session if new
  if (!activeSessions.has(sessionId)) {
    initializeSession(sessionId);
  }
  
  const session = activeSessions.get(sessionId);
  
  // Guard against race conditions - only one handler per session at a time
  if (session.isProcessing) {
    console.log(`Session ${sessionId} already processing, queuing event`);
    return res.status(200).json({ queued: true });
  }

  // Return 200 immediately - process async to avoid webhook timeout
  res.status(200).json({ received: true });
  
  // Process event asynchronously
  setImmediate(async () => {
    session.isProcessing = true;
    
    try {
      switch(event.message.type) {
        case 'transcript':
          // Handle both partial and final transcripts
          if (!event.message.transcriptType || event.message.transcriptType === 'partial') {
            await handlePartialTranscript(sessionId, event.message.transcript);
          } else if (event.message.transcriptType === 'final') {
            await handleFinalTranscript(sessionId, event.message.transcript);
          }
          break;
          
        case 'function-call':
          // Customer needs escalation or specific action
          await handleFunctionCall(sessionId, event.message);
          break;
          
        case 'speech-update':
          // Real-time speech status for UI indicators
          await handleSpeechUpdate(sessionId, event.message);
          break

### System Diagram

System architecture for vapi integration with your application.

```mermaid
graph TB
    User[User Device]
    VAPI[VAPI Service]
    ASR[Automatic Speech Recognition]
    NLP[NLP Processor]
    TTS[Text-to-Speech Engine]
    Webhook[Webhook Server]
    DB[(Database)]
    ErrorHandler[Error Handler]

    User-->|Voice Input|VAPI
    VAPI-->ASR
    ASR-->NLP
    NLP-->TTS
    TTS-->|Voice Output|User
    VAPI-->|Event Data|Webhook
    Webhook-->DB
    VAPI-->|Error Events|ErrorHandler
    ErrorHandler-->|Log|DB
    NLP-->|Error|ErrorHandler
    ASR-->|Error|ErrorHandler
    TTS-->|Error|ErrorHandler

Testing & Validation

Most streaming implementations fail in production because devs skip local validation. Here's how to catch race conditions before they hit users.

Local Testing

Test the WebSocket connection and session lifecycle with a simple client script. This catches buffer issues and timing problems that break real calls.

javascript
// Test WebSocket connection and session handling
const WebSocket = require('ws');

const ws = new WebSocket('ws://localhost:3000');
const testSessionId = crypto.randomBytes(16).toString('hex');

ws.on('open', () => {
  console.log('WebSocket connected');
  
  // Simulate VAPI session start
  ws.send(JSON.stringify({
    type: 'session.start',
    sessionId: testSessionId,
    timestamp: Date.now()
  }));
  
  // Test partial transcript handling
  setTimeout(() => {
    ws.send(JSON.stringify({
      type: 'transcript.partial',
      sessionId: testSessionId,
      text: 'I need help with my account'
    }));
  }, 1000);
});

ws.on('message', (data) => {
  const message = JSON.parse(data);
  console.log('Server response:', message);
  
  // Verify session exists in activeSessions
  if (message.type === 'session.created') {
    console.log('✓ Session initialized:', message.sessionId);
  }
});

ws.on('error', (error) => {
  console.error('WebSocket error:', error.message);
});

Run this before deploying. If activeSessions doesn't populate within 500ms, your session initialization is too slow for real-time streaming.

Webhook Validation

Validate webhook signatures to prevent replay attacks. VAPI sends x-vapi-signature headers that MUST match your computed HMAC.

javascript
// Test webhook signature validation locally
const testPayload = {
  type: 'call.started',
  sessionId: testSessionId,
  timestamp: Date.now()
};

const testSignature = crypto
  .createHmac('sha256', config.server.webhookSecret)
  .update(JSON.stringify(testPayload))
  .digest('hex');

console.log('Expected signature:', testSignature);

// Send test webhook with curl
// curl -X POST http://localhost:3000/webhook \
//   -H "x-vapi-signature: <testSignature>" \
//   -H "Content-Type: application/json" \
//   -d '<testPayload>'

If validateSignature() returns false, check that your webhookSecret matches the value in your VAPI dashboard. Signature mismatches cause silent webhook failures—no errors, just dropped events.

Real-World Example

Barge-In Scenario

Most live support systems break when users interrupt the agent mid-sentence. The agent keeps talking, the user repeats themselves, and you end up with overlapping audio chaos. Here's what actually happens in production:

User calls in, agent starts explaining refund policy. User interrupts at 2.3 seconds with "I just need my order number." Without proper barge-in handling, the TTS buffer continues playing the refund explanation while STT processes the interruption. Result: agent talks over user, user gets frustrated, session quality tanks.

javascript
// Barge-in handler with buffer flush
const handleInterruption = async (sessionId, partialTranscript) => {
  const session = activeSessions.get(sessionId);
  if (!session) return;

  // Race condition guard - prevent multiple interrupts
  if (session.isProcessing) {
    console.log(`[${sessionId}] Already processing, queuing interrupt`);
    session.pendingInterrupt = partialTranscript;
    return;
  }

  session.isProcessing = true;

  try {
    // Flush TTS buffer immediately
    if (session.audioBuffer && session.audioBuffer.length > 0) {
      console.log(`[${sessionId}] Flushing ${session.audioBuffer.length} audio chunks`);
      session.audioBuffer = [];
    }

    // Cancel ongoing TTS request
    if (session.ttsController) {
      session.ttsController.abort();
      session.ttsController = null;
    }

    // Send interrupt signal to VAPI
    const response = await fetch('https://api.vapi.ai/chat', {
      method: 'POST',
      headers: {
        'Authorization': `Bearer ${process.env.VAPI_API_KEY}`,
        'Content-Type': 'application/json'
      },
      body: JSON.stringify({
        assistantId: session.assistantId,
        message: {
          type: 'interrupt',
          transcript: partialTranscript,
          timestamp: Date.now()
        }
      })
    });

    if (!response.ok) {
      throw new Error(`Interrupt failed: ${response.status}`);
    }

    const result = await response.json();
    session.lastInterruptAt = Date.now();
    
  } catch (error) {
    console.error(`[${sessionId}] Interrupt handling failed:`, error);
  } finally {
    session.isProcessing = false;
    
    // Process queued interrupt if exists
    if (session.pendingInterrupt) {
      const queued = session.pendingInterrupt;
      session.pendingInterrupt = null;
      await handleInterruption(sessionId, queued);
    }
  }
};

Event Logs

Real production logs from a support session with multiple interruptions. Timestamps show the actual latency impact:

[2024-01-15T14:32:18.234Z] Session 7a3f initialized [2024-01-15T14:32:18.891Z] Agent TTS started: "Thank you for calling TechFlow support..." [2024-01-15T14:32:20.456Z] STT partial: "I just" (confidence: 0.72) [2024-01-15T14:32:20.623Z] STT partial: "I just need my" (confidence: 0.84) [2024-01-15T14:32:20.789Z] Barge-in detected, flushing 12 audio chunks [2024-01-15T14:32:20.801Z] TTS cancelled mid-sentence [2024-01-15T14:32:21.034Z] STT final: "I just need my order number" (confidence: 0.91) [2024-01-15T14:32:21.156Z] Agent response latency: 122ms [2024-01-15T14:32:21.289Z] Agent TTS started: "I can help you find that..." [2024-01-15T14:32:22.567Z] STT partial: "it's" (confidence: 0.68) - IGNORED (< 0.7 threshold) [2024-01-15T14:32:23.891Z] STT partial: "it's order" (confidence: 0.79) [2024-01-15T14:32:24.023Z] False positive check: gap since last interrupt = 3.2s (> 2.5s threshold) [2024-01-15T14:32:24.034Z] Barge-in detected, flushing 8 audio chunks

The critical metric: 122ms from final transcript to agent response. Anything over 300ms feels laggy. The false positive at 22.567s shows why confidence thresholds matter—breathing sounds and background noise trigger STT partials constantly.

Edge Cases

Multiple rapid interrupts: User says "wait wait wait" in quick succession. Without the isProcessing guard, you get three concurrent barge-in handlers racing to flush the same buffer. Solution: queue subsequent interrupts until the first completes.

False positive from background noise: Call center environment, someone sneezes nearby. STT fires with confidence 0.68. Agent stops mid-sentence for nothing. Solution: require confidence > 0.7 AND minimum 2.5s gap since last interrupt.

Network jitter on mobile: User on 4G, packet loss causes STT delay. Partial transcript arrives 400ms late, AFTER agent already started next sentence. Solution: track lastInterruptAt timestamp, ignore partials older than 500ms.

javascript
// Edge case: Stale interrupt detection
if (Date.now() - session.lastInterruptAt < 500) {
  console.log(`[${sessionId}] Ignoring stale interrupt (${Date.now() - session.lastInterruptAt}ms old)`);
  return;
}

Buffer not fully flushed: TTS chunks still in WebSocket send queue when interrupt fires. Agent voice "bleeds through" for 200-300ms after interrupt. This will bite you. Solution: implement proper WebSocket drain before sending new audio.

Common Issues & Fixes

Most streaming implementations break under production load. Here's what actually fails and how to fix it.

Race Condition: Overlapping TTS Streams

Problem: User interrupts mid-sentence, but TTS buffer isn't flushed. Old audio plays after the new response starts → bot talks over itself.

javascript
// WRONG: No cancellation logic
ws.on('message', (data) => {
  const event = JSON.parse(data);
  if (event.type === 'transcript-partial') {
    generateTTSResponse(event.text); // Queues audio without checking state
  }
});

// CORRECT: Cancel in-flight TTS on barge-in
let currentTTSStream = null;

ws.on('message', async (data) => {
  const event = JSON.parse(data);
  
  if (event.type === 'speech-start') {
    // User started speaking - kill active TTS immediately
    if (currentTTSStream) {
      currentTTSStream.abort();
      currentTTSStream = null;
    }
  }
  
  if (event.type === 'transcript-final') {
    const controller = new AbortController();
    currentTTSStream = controller;
    
    try {
      await fetch('https://api.elevenlabs.io/v1/text-to-speech/' + config.voice.voiceId, {
        method: 'POST',
        headers: {
          'xi-api-key': process.env.ELEVENLABS_API_KEY,
          'Content-Type': 'application/json'
        },
        body: JSON.stringify({ text: event.text }),
        signal: controller.signal
      });
    } catch (error) {
      if (error.name === 'AbortError') return; // Expected on interruption
      console.error('TTS Error:', error);
    }
  }
});

Why this breaks: ElevenLabs streams take 800-1200ms to complete. Without abort handling, you get audio overlap when users interrupt quickly.

WebSocket Timeout on Mobile Networks

Problem: Mobile carriers drop idle WebSocket connections after 30-60 seconds. Session dies silently, no reconnection.

Fix: Implement ping/pong with 20-second intervals:

javascript
const PING_INTERVAL = 20000; // 20s - below carrier timeout thresholds

ws.on('open', () => {
  const session = activeSessions.get(testSessionId);
  
  session.pingTimer = setInterval(() => {
    if (ws.readyState === WebSocket.OPEN) {
      ws.ping();
    } else {
      clearInterval(session.pingTimer);
    }
  }, PING_INTERVAL);
});

ws.on('pong', () => {
  const session = activeSessions.get(testSessionId);
  session.lastPong = Date.now();
});

ws.on('close', () => {
  const session = activeSessions.get(testSessionId);
  if (session?.pingTimer) clearInterval(session.pingTimer);
});

Twilio Media Stream Desync

Problem: Twilio sends audio in 20ms chunks (mulaw 8kHz). If your transcriber expects 16kHz PCM, you get garbled transcripts or silence.

Fix: Match Twilio's exact format in transcriber config:

javascript
const assistantConfig = {
  transcriber: {
    provider: 'deepgram',
    model: 'nova-2-phonecall', // Optimized for telephony
    language: 'en',
    encoding: 'mulaw', // CRITICAL: Must match Twilio's codec
    sampleRate: 8000,  // Twilio's native rate
    endpointing: 800   // Longer for phone latency
  }
};

Validation: Check Twilio's <Stream> payload - if mediaFormat.encoding is audio/x-mulaw, your transcriber MUST use mulaw + 8000 sample rate. Mismatch = 100% failure rate.

Complete Working Example

Most live support chat implementations fail in production because they treat streaming as an afterthought. Here's the full server that handles VAPI WebSocket streaming, Twilio voice bridging, and real-time voice transcription without race conditions.

Full Server Code

This is production-grade code that handles session lifecycle, webhook validation, and bidirectional audio streaming. Copy-paste this into server.js:

javascript
const express = require('express');
const WebSocket = require('ws');
const crypto = require('crypto');
const fetch = require('node-fetch');

const app = express();
app.use(express.json());

// Configuration from previous sections
const config = {
  vapi: {
    apiKey: process.env.VAPI_API_KEY,
    baseUrl: 'https://api.vapi.ai'
  },
  twilio: {
    accountSid: process.env.TWILIO_ACCOUNT_SID,
    authToken: process.env.TWILIO_AUTH_TOKEN
  },
  server: {
    port: process.env.PORT || 3000,
    webhookSecret: process.env.WEBHOOK_SECRET
  }
};

// Session management with TTL
const activeSessions = new Map();
const SESSION_TTL = 30 * 60 * 1000; // 30 minutes

function initializeSession(sessionId) {
  const session = {
    id: sessionId,
    transcripts: [],
    metadata: { startTime: Date.now() },
    isProcessing: false,
    currentTTSStream: null
  };
  activeSessions.set(sessionId, session);
  
  // Auto-cleanup to prevent memory leaks
  setTimeout(() => {
    if (activeSessions.has(sessionId)) {
      const session = activeSessions.get(sessionId);
      if (session.currentTTSStream) {
        session.currentTTSStream.abort();
      }
      activeSessions.delete(sessionId);
      console.log(`Session ${sessionId} expired and cleaned up`);
    }
  }, SESSION_TTL);
  
  return session;
}

// Webhook signature validation (security is not optional)
function validateSignature(payload, signature) {
  const hmac = crypto.createHmac('sha256', config.server.webhookSecret);
  hmac.update(JSON.stringify(payload));
  const digest = hmac.digest('hex');
  return crypto.timingSafeEqual(Buffer.from(signature), Buffer.from(digest));
}

// Create assistant with streaming configuration
async function createAssistant() {
  const assistantConfig = {
    model: {
      provider: "openai",
      model: "gpt-4",
      temperature: 0.7,
      messages: [{
        role: "system",
        content: "You are a live support agent. Keep responses under 30 words. Handle interruptions gracefully."
      }]
    },
    voice: {
      provider: "elevenlabs",
      voiceId: "21m00Tcm4TlvDq8ikWAM",
      stability: 0.5,
      similarityBoost: 0.75,
      optimizeStreamingLatency: 2 // Critical for low-latency TTS integration
    },
    transcriber: {
      provider: "deepgram",
      model: "nova-2",
      language: "en",
      endpointing: 200 // Barge-in detection threshold
    },
    firstMessage: "Hi, I'm your support agent. How can I help?"
  };

  try {
    const response = await fetch(`${config.vapi.baseUrl}/assistant`, {
      method: 'POST',
      headers: {
        'Authorization': `Bearer ${config.vapi.apiKey}`,
        'Content-Type': 'application/json'
      },
      body: JSON.stringify(assistantConfig)
    });

    if (!response.ok) {
      const error = await response.json();
      throw new Error(`Assistant creation failed: ${error.message}`);
    }

    const assistant = await response.json();
    console.log(`Assistant created: ${assistant.id}`);
    return assistant;
  } catch (error) {
    console.error('Failed to create assistant:', error);
    throw error;
  }
}

// WebSocket handler for VAPI WebSocket streaming
const wss = new WebSocket.Server({ noServer: true });

wss.on('connection', (ws, request) => {
  const sessionId = request.url.split('/').pop();
  const session = initializeSession(sessionId);
  
  console.log(`WebSocket connected: ${sessionId}`);

  // Handle streaming transcripts
  ws.on('message', (data) => {
    try {
      const message = JSON.parse(data);
      
      if (message.type === 'transcript-partial') {
        // Real-time voice transcription - update UI immediately
        session.transcripts.push({
          type: 'partial',
          text: message.text,
          timestamp: Date.now()
        });
      }
      
      if (message.type === 'transcript-final') {
        // Replace partial with final transcript
        session.transcripts = session.transcripts.filter(t => t.type !== 'partial');
        session.transcripts.push({
          type: 'final',
          text: message.text,
          timestamp: Date.now()
        });
      }

      if (message.type === 'interruption') {
        // Handle barge-in: cancel current TTS stream
        if (session.currentTTSStream && !session.currentTTSStream.aborted) {
          session.currentTTSStream.abort();
          session.currentTTSStream = null;
          console.log(`TTS stream cancelled for session ${sessionId}`);
        }
      }

      // Broadcast to all connected clients (Server-Sent Events pattern)
      ws.send(JSON.stringify({
        sessionId: session.id,
        transcripts: session.transcripts,
        metadata: session.metadata
      }));

    } catch (error) {
      console.error('WebSocket message error:', error);
    }
  });

  ws.on('close', () => {
    console.log(`WebSocket disconnected: ${sessionId}`);
  });

  // Keep-alive ping to prevent connection drops
  const PING_INTERVAL = setInterval(() => {
    if (ws.readyState === WebSocket.OPEN) {
      ws.ping();
    }
  }, 30000);

  ws.on('close', () => clearInterval(PING_INTERVAL));
});

// Webhook endpoint for VAPI events
app.post('/webhook/vapi', async (req, res) => {
  const signature = req.headers['x-vapi-signature'];
  
  if (!validateSignature(req.body, signature)) {
    console.error('Invalid webhook signature');
    return res.status(401).json({ error: 'Invalid signature' });
  }

  const event = req.body;
  const sessionId = event.call?.id || event.message?.call?.id;

  if (!sessionId) {
    return res.status(400).json({ error: 'Missing session ID' });
  }

  const session = activeSessions.get(sessionId);
  if (!session) {
    console.warn(`Received event for unknown session: ${sessionId}`);
    return res.status(404).json({ error: 'Session not found' });
  }

  // Handle different event types
  switch (event.type) {
    case 'call-started':
      session.metadata.callStarted = Date.now();
      break;
      
    case 'speech-started':
      session.isProcessing = true;
      break;
      
    case 'speech-ended':
      session.isProcessing = false;
      break;
      
    case 'call-ended':
      session.metadata.callEnded = Date.now();
      session.metadata.duration = session.metadata.callEnded - session.metadata.callStarted;
      break;
  }

  res.status

## FAQ

### Technical Questions

**How does VAPI handle WebSocket streaming for real-time voice transcription?**

VAPI maintains a persistent WebSocket connection that receives audio chunks (typically PCM 16kHz, 16-bit) from the client and streams partial transcripts back in real-time. The transcriber processes audio frames asynchronously, emitting `transcript.partial` events before the final `transcript.final` event fires. This dual-event pattern lets you display live captions while the user is still speaking. The key is buffering incoming audio chunks in a queue (not dropping frames) and processing them sequentially to avoid race conditions between partial and final transcripts.

**What's the latency impact of adding Twilio integration to VAPI?**

Twilio adds ~200-400ms of additional latency due to SIP signaling and media gateway routing. VAPI's native latency is ~150-300ms (STT + LLM + TTS). Combined, expect 350-700ms end-to-end for a user utterance to trigger a bot response. Mitigate this by: (1) enabling `optimizeStreamingLatency: true` in the TTS config to stream audio chunks instead of waiting for full synthesis, (2) using partial transcripts to start LLM processing before the user finishes speaking, (3) reducing model inference time by using smaller models (gpt-3.5-turbo vs gpt-4).

**Why does my bot interrupt the user mid-sentence?**

VAD (Voice Activity Detection) threshold is too aggressive. Default endpointing fires after 500-800ms of silence, but network jitter can trigger false positives. Increase the `endpointing` threshold to 1200-1500ms and set `silenceThreshold` to 0.5+ (default 0.3) to reduce breathing-sound false triggers. Also check if your Twilio SIP trunk is dropping RTP packets—packet loss causes the transcriber to misinterpret silence as speech end.

### Performance

**How do I prevent TTS buffer overflow during rapid exchanges?**

Implement a cancellation controller: when a new user message arrives, abort the current TTS stream immediately using `AbortController`. Store the active stream reference in `currentTTSStream` and call `controller.abort()` before queuing new audio. Without this, old TTS audio queues up and plays after the bot has already moved to the next response, creating overlapping speech.

**What's the maximum concurrent sessions VAPI can handle?**

VAPI's free tier supports ~10 concurrent calls; paid tiers scale to 100+ depending on your plan. Per-session memory usage is ~2-5MB (transcript history + session state). If you're storing `activeSessions` in-memory, set `SESSION_TTL` to 3600 seconds (1 hour) and implement cleanup: `setTimeout(() => delete activeSessions[sessionId], SESSION_TTL * 1000)`. For production, use Redis instead of in-memory storage.

### Platform Comparison

**Should I use VAPI's native voice or Twilio's voice synthesis?**

Use VAPI's native voice (ElevenLabs or Google). Twilio's voice synthesis is older (lower naturalness) and adds extra latency. Configure VAPI's `voice.provider: "elevenlabs"` with `optimizeStreamingLatency: true` to stream audio chunks. Twilio's role is media routing only—let VAPI own the voice experience.

**Can I replace Twilio with a different SIP provider?**

Yes, but Twilio is the easiest integration. Other SIP providers (Vonage, Bandwidth) work, but you'll need to handle SIP registration, media negotiation, and RTP routing yourself. Stick with Twilio unless you have specific cost or compliance requirements (e.g., HIPAA-compliant media gateways).

## Resources

**Twilio**: Get Twilio Voice API → [https://www.twilio.com/try-twilio](https://www.twilio.com/try-twilio)

**VAPI Documentation**
- [VAPI API Reference](https://docs.vapi.ai) – WebSocket streaming, real-time voice transcription, low-latency TTS integration
- [VAPI Assistant Configuration](https://docs.vapi.ai/assistants) – Model, voice, transcriber setup for bidirectional audio

**Twilio Integration**
- [Twilio Voice API](https://www.twilio.com/docs/voice) – SIP trunking, call routing for live support
- [Twilio WebSocket Connections](https://www.twilio.com/docs/voice/twiml/stream) – Server-Sent Events (SSE) patterns

**Implementation References**
- [VAPI GitHub Examples](https://github.com/VapiAI) – Production streaming implementations
- [Express WebSocket Middleware](https://github.com/websockets/ws) – Real-time bidirectional communication

## References

1. https://docs.vapi.ai/chat/quickstart
2. https://docs.vapi.ai/quickstart/web
3. https://docs.vapi.ai/workflows/quickstart
4. https://docs.vapi.ai/quickstart/phone
5. https://docs.vapi.ai/server-url/developing-locally
6. https://docs.vapi.ai/quickstart/introduction
7. https://docs.vapi.ai/observability/evals-quickstart
8. https://docs.vapi.ai/assistants/structured-outputs-quickstart
9. https://docs.vapi.ai/tools/custom-tools

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Written by

Misal Azeem
Misal Azeem

Voice AI Engineer & Creator

Building production voice AI systems and sharing what I learn. Focused on VAPI, LLM integrations, and real-time communication. Documenting the challenges most tutorials skip.

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