WebRTC SDK

The jambonz WebRTC SDK enables browser and mobile applications to make and receive voice calls through a jambonz SBC using SIP over WebSocket.

It supports Web (React) and React Native (iOS + Android) with a unified API. JsSIP is used internally for SIP signaling but is completely hidden from SDK consumers.

Source code: github.com/jambonz/webrtc-sdk

Packages

Both packages share the same API. The only difference is the import path.

What It Does

• SIP registration with a jambonz SBC over WebSocket
• Outbound calls to phone numbers, SIP users, queues, conferences, and applications
• Incoming calls with answer/decline
• In-call controls — mute, hold, DTMF, transfer (blind + attended)
• Call quality monitoring — jitter, packet loss, round-trip time
• Audio device management — enumerate and switch microphones/speakers
• SIP messaging — send and receive SIP MESSAGE
• React hooks — useJambonzClient() and useCall() for declarative React integration

Call Types

The SDK supports five types of outbound calls, matching jambonz SBC routing:

Quick Example

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import { createJambonzClient } from '@jambonz/client-sdk-web';
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const client = createJambonzClient({
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  server: 'wss://sbc.example.com:8443',
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  username: 'user1',
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  password: 'pass123',
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});
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await client.connect();
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// Make a call
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const call = client.call('+15551234567');
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call.on('accepted', () => console.log('Connected!'));
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call.on('ended', () => console.log('Call ended'));
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// Receive incoming calls
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client.on('incoming', (incomingCall) => {
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  incomingCall.answer();
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});