voice-sdk-dialer

Voice SDK Dialer

A powerful WebRTC SIP client library for making and receiving voice calls in web browsers. Built with modern JavaScript and TypeScript support.

🚀 Features

• WebRTC SIP Integration - Full SIP protocol support over WebSocket
• Call Management - Make, receive, hold, mute, and transfer calls
• Real-time Events - Comprehensive callback system for call states
• TypeScript Support - Complete type definitions included
• Browser Compatible - Works in all modern browsers
• Audio Controls - Mute, hold, speaker controls
• Call Duration Tracking - Built-in call timer

📦 Installation

npm install voice-sdk-dialer

Or with yarn:

yarn add voice-sdk-dialer

🏗️ Quick Start

Basic Usage

import VoiceSDKDialer from 'voice-sdk-dialer';

// Initialize the dialer
const dialer = new VoiceSDKDialer({
  token: 'your-token',
  domain: 'your-domain.com'
});

// Start the dialer with SIP configuration
dialer.start({
  sipConfig: {
    server: 'sip.yourdomain.com',
    username: 'your-sip-username',
    password: 'your-sip-password',
    displayName: 'Your Display Name'
  },
  callbacks: {
    onConnected: () => console.log('Connected to SIP server'),
    onCallAnswered: (callInfo) => console.log('Call answered:', callInfo),
    onError: (error) => console.error('Error:', error)
  }
});

// Make a call
dialer.makeCall('[phone]');

TypeScript Usage

import VoiceSDKDialer, { 
  VoiceSDKOptions, 
  CallInfo, 
  CALL_STATES 
} from 'voice-sdk-dialer';

const options: VoiceSDKOptions = {
  sipConfig: {
    server: 'sip.yourdomain.com',
    username: 'user',
    password: 'pass',
    displayName: 'User Name'
  },
  callbacks: {
    onCallAnswered: (callInfo: CallInfo) => {
      console.log(`Call with ${callInfo.caller} is active`);
    }
  }
};

const dialer = new VoiceSDKDialer({
  token: 'your-token',
  domain: 'your-domain'
});

dialer.start(options);

🔧 API Reference

Constructor

new VoiceSDKDialer(options)

Parameters:

• options.token (string) - Authentication token
• options.domain (string) - Your domain
• options.defaultCountryCode (string, optional) - Default country code (default: '+84')

Methods

start(options)

Initialize and start the dialer with SIP configuration.

dialer.start({
  sipConfig: {
    server: 'sip.example.com',
    username: 'user',
    password: 'pass',
    displayName: 'Display Name',
    port: '8089' // optional
  },
  callbacks: {
    // Event callbacks
  }
});

makeCall(number)

Make an outgoing call to the specified number.

options = {
  'X-[name]' = value
}
await dialer.makeCall('[phone]', options);

answerCall()

Answer an incoming call.

dialer.answerCall();

hangupCall()

End or decline the current call.

await dialer.hangupCall();

hold() / unhold()

Put the current call on hold or resume it.

await dialer.hold();
await dialer.unhold();

mute() / unmute()

Mute or unmute the microphone.

await dialer.mute();
await dialer.unmute();

destroy()

Clean up resources and destroy the dialer instance.

dialer.destroy();

📡 Events & Callbacks

The dialer supports comprehensive event callbacks:

const callbacks = {
  // Connection events
  onReady: (dialer) => console.log('Dialer ready'),
  onConnected: () => console.log('Connected to SIP server'),
  onDisconnected: () => console.log('Disconnected from server'),
  onRegistered: () => console.log('SIP registration successful'),
  
  // Call events
  onIncomingCall: (callInfo) => console.log('Incoming call from:', callInfo.caller),
  onOutgoingCall: (callInfo) => console.log('Calling:', callInfo.number),
  onCallAnswered: (callInfo) => console.log('Call answered'),
  onCallEnded: () => console.log('Call ended'),
  onCallInitiated: (callInfo) => console.log('Call initiated'),
  
  // Call controls
  onMuteToggle: (isMuted) => console.log('Mute:', isMuted),
  onHoldToggle: (isHeld) => console.log('Hold:', isHeld),
  onDurationUpdate: (seconds) => console.log('Duration:', seconds),
  
  // Error handling
  onError: (error) => console.error('Error:', error)
};

🌐 Browser Support

• Chrome/Chromium 70+
• Firefox 65+
• Safari 12+
• Edge 79+

📋 Requirements

• Modern browser with WebRTC support
• SIP server with WebSocket support
• Valid SSL certificate (required for WebRTC)
• Microphone permissions

📄 License

MIT License - see the LICENSE file for details.

📞 Support

• 📧 Email: [e-mail]

🏷️ Changelog

v1.0.0

• Initial release
• Basic SIP calling functionality
• WebRTC integration
• TypeScript support
• Call management features

v1.0.1

• How to fix index.js

v1.0.2

• How to fix hold/unhold

v1.0.3

• How to fix not register

v1.0.4

v1.0.5

v1.0.6

v1.0.7

v1.0.8

• How to fix cannot imported VoiceSDKDialer
• Add more callback event

v1.0.9

• Add more exported interface

v1.1.0

• Add more exported interface
• Re-export type for interface
• Add check reconnect

v1.1.1

v1.1.2

v1.1.3

v1.1.4

v1.1.5

v1.1.6

v1.1.7

• How to fix cannot imported VoiceSDKDialer

v1.1.8

• How to fix cannot imported VoiceSDKDialer
• Add check reconnect

v1.1.9

• How to fix cannot imported VoiceSDKDialer
• Add check reconnect
• Remove check audio, add function InitAudio

v1.2.0

v1.2.1

v1.2.2

v1.2.3

v1.2.4

v1.2.5

v1.2.6

v1.2.7

• How to fix cannot imported VoiceSDKDialer
• Reactjs yarn add alias with link folder ../node_modules/voice-sdk-dialer

v1.2.8

• How to fix hangup event

v1.2.9

• How to fix hangup event
• Add declineCall

v1.3.0

• How to fix hangup event

v1.3.1

• Update required SessionState

v1.3.2

• Update required Inviter

v1.3.3

• Replace hangupCall and DeclineCall

v1.3.4

• Update callback object call details

v1.3.5

• Update callback object call details
• Add options make call

v1.3.6

• Add options make call
• Add on makecall in ts

v1.3.7

• Fix error add options in make call

v1.3.8

• Add options in hangupCall and declineCall

v1.3.9

• Fixed Microphone permission

v1.4.0

• Remove and not function isRegistered()

v1.4.1

• Fixed cannot call outbound/inbound when fix permission microphone

v1.4.2

• Fixed call in speaker