Click To Call SDK

A standalone, framework-agnostic SDK for SIP-based voice calling with WebRTC. Built on JsSIP.

Features

• 🚀 Framework Agnostic — works with React, Angular, Vue, or vanilla JS.
• 🔒 Type Safe — full TypeScript support.
• 📞 SIP/WebRTC — built on JsSIP for reliable communication.
• 🎛️ Full Control — start, end, mute, unmute, send info messages.
• 🔄 Event Driven — 17 events for real-time state updates.
• 🎵 Auto Audio — remote audio plays automatically; raw stream available via captureAudio event.
• 💬 Transcription — real-time transcription events, auto-separated from info messages.

Installation

npm install @cognigy/click-to-call-sdk

Quick Start

import { createWebRTCClient, checkWebRTCSupport } from '@cognigy/click-to-call-sdk';

// 1. Check browser support
const support = checkWebRTCSupport();
if (!support.supported) throw new Error('Missing: ' + support.missing);

// 2. Create client
const client = await createWebRTCClient({
  endpointUrl: 'https://your-cognigy-environment.com/token',
  userId: 'user-123',
});

// 3. Listen to events
client.on('answered', (session) => console.log('Call answered:', session.id));
client.on('ended', (session, endInfo) => console.log('Call ended:', endInfo.cause));
client.on('error', (error) => console.error('Error:', error.message));

// 4. Connect and call
await client.connectAndCall();

// 5. Cleanup on page unload
window.addEventListener('beforeunload', () => {
  void client.destroy().catch(() => {
    // Ignore errors during page unload
  });
});

Note: The call must be initiated from a user gesture (e.g. button click) for browser autoplay policies to allow audio.

Configuration

interface WebRTCClientConfig {
  endpointUrl: string;        // URL to fetch SIP configuration
  userId?: string;            // Optional user identifier
  pcConfig?: RTCConfiguration; // WebRTC peer connection config
  captureAudio?: boolean;     // Enable captureAudio event to receive raw MediaStream
}

API

| Method | Description | |-------------------------|----------------------------------------------| | connect() | Connect to SIP server and register | | disconnect() | Disconnect from SIP server | | connectAndCall() | Connect + start call in one step | | startCall() | Start a call (must be connected first) | | endCall() | End the current call | | mute() / unmute() | Toggle microphone | | sendInfo(text, data?) | Send info message during a call | | isConnected() | Check connection state | | getCurrentSession() | Get active CallSession or null | | on(event, callback) | Add event listener (returns this) | | off(event, callback) | Remove event listener (returns this) | | destroy() | Disconnect, end calls, release all resources |

Events

| Event | Callback | |-----------------|-------------------------------------------------------------------------| | connecting | () | | connected | () | | disconnected | () | | registered | () | | unregistered | () | | ringing | (session: CallSession) | | answered | (session: CallSession) | | ended | (session: CallSession, endInfo: CallEndInfo) | | failed | (session: CallSession, endInfo: CallEndInfo) | | muted | (session: CallSession) | | unmuted | (session: CallSession) | | captureAudio | Remote audio stream available (requires opt-in) (stream: MediaStream) | | audioEnded | () | | infoSent | (text: string, data: Record<string, any>) | | infoReceived | (data: { originator: string; info: { body: string } }) | | transcription | (transcription: { originator: string; messages: { text: string }[] }) | | error | (error: Error) |

Browser Compatibility

Chrome 93+ · Firefox 92+ · Safari 15.4+ · Edge 93+

Documentation

For the full guide, API reference, event reference, and troubleshooting:

• Getting Started
• API Reference
• Event Reference
• Custom Audio
• Security
• Troubleshooting

License

MIT