Integrating VaxVoIP WebPhone SDK into React and Angular Apps

Getting Started: VaxVoIP WebPhone SDK Tutorial for DevelopersVaxVoIP WebPhone SDK is a JavaScript library designed to bring SIP-based voice, video, and messaging capabilities directly into web applications using WebRTC. This tutorial walks through the essentials: installation, core concepts, initialization, handling calls and messages, common integration patterns (React/Angular), security considerations, debugging tips, and production deployment recommendations. By the end you’ll have a working knowledge to integrate a browser softphone or in-app calling feature.

What the SDK provides (at a glance)

Browser SIP over WebRTC: Register and handle SIP signaling in the browser while using secure WebRTC media.
Call control: Make, receive, hold, transfer, and end calls.
Video and audio: Negotiation and media stream handling for 1:1 and conference scenarios.
Messaging: In-band SIP MESSAGE support for text chat.
Events and hooks: Fine-grained event model to wire UI and application logic.
Pluggable transport/backends: Works with standard SIP servers and popular VoIP backends.

Prerequisites

Intermediate familiarity with JavaScript/TypeScript and modern front-end frameworks (React/Angular/Vue).
Basic understanding of SIP and WebRTC concepts (SIP registration, SDP, ICE, STUN/TURN).
A SIP account (username, password, SIP server/edge) or access to a SIP backend for testing.
HTTPS hosting (browsers require secure context for getUserMedia and WebRTC).
Optional: TURN server credentials if testing across NATs/firewalls.

Installation

Install via npm (recommended):

npm install vaxvoip-webphone-sdk

Or include via a CDN/script tag:

<script src="https://cdn.example.com/vaxvoip-webphone-sdk/latest/vaxvoip-webphone-sdk.min.js"></script>

(Adjust URL to actual CDN path provided by VaxVoIP.)

Core Concepts

User Agent (UA): The SDK typically exposes a UA/client object which manages SIP registration, transports, and active sessions.
Session: Represents an active call (incoming or outgoing). Sessions expose media streams, state, and control methods (answer, hangup, hold, transfer).
Events: The SDK emits events for registration state, incoming call, call progress, track added/removed, message received, and errors.
Media constraints: Control over audio/video capture, codecs, and device selection.
ICE/DTLS: WebRTC uses ICE for connectivity checks and DTLS/SRTP for encrypting media. The SDK surfaces configuration for STUN/TURN and ICE options.

Quick start: Minimal example

Below is a concise example showing initialization, registration, making and receiving a call, and attaching local/remote streams to the DOM.

import VaxWebPhone from 'vaxvoip-webphone-sdk'; const config = {   uri: 'sip:[email protected]',   authorizationUser: 'alice',   password: 'alicePassword',   wsServers: ['wss://sip-ws.example.com'],   displayName: 'Alice',   stunServers: ['stun:stun.l.google.com:19302'],   turnServers: [{ urls: 'turn:turn.example.com:3478', username: 'turnuser', credential: 'turnpass' }],   iceTransportPolicy: 'all',   media: { audio: true, video: true } }; const ua = new VaxWebPhone.UA(config); // DOM elements const localVideo = document.getElementById('localVideo'); const remoteVideo = document.getElementById('remoteVideo'); ua.on('registered', () => console.log('Registered')); ua.on('registrationFailed', (err) => console.error('Registration failed', err)); ua.on('incomingCall', async (session) => {   console.log('Incoming call from', session.remoteIdentity.uri.toString());   // auto-answer example:   await session.answer({ media: { audio: true, video: true } });   attachSessionStreams(session); }); function attachSessionStreams(session) {   session.on('track', (event) => {     // event.streams[0] is remote stream     remoteVideo.srcObject = event.streams[0];   });   // get local stream and show   navigator.mediaDevices.getUserMedia({ audio: true, video: true })     .then(stream => {       localVideo.srcObject = stream;       session.sessionDescriptionHandler?.peerConnection.getSenders().forEach((s) => {         // senders already attached by SDK typically; code depends on SDK specifics       });     }); } async function call(target) {   const session = ua.invite(`sip:${target}`, {     media: { audio: true, video: true }   });   session.on('progress', () => console.log('Ringing...'));   session.on('accepted', () => attachSessionStreams(session));   session.on('terminated', () => console.log('Call ended')); }

Note: exact API names (UA, invite, events) depend on the SDK’s implementation; consult the SDK reference for exact method signatures.

Handling media devices

List devices:

const devices = await navigator.mediaDevices.enumerateDevices(); const audioInputs = devices.filter(d => d.kind === 'audioinput'); const videoInputs = devices.filter(d => d.kind === 'videoinput');

Select a specific camera/microphone when creating/getting media:

const stream = await navigator.mediaDevices.getUserMedia({   audio: { deviceId: selectedAudioId ? { exact: selectedAudioId } : undefined },   video: { deviceId: selectedVideoId ? { exact: selectedVideoId } : undefined } });

Switch devices mid-call by obtaining a new track and replacing the sender’s track:

const newStream = await navigator.mediaDevices.getUserMedia({ video: { deviceId: { exact: newVideoId } } }); const videoTrack = newStream.getVideoTracks()[0]; const sender = session.sessionDescriptionHandler.peerConnection.getSenders()   .find(s => s.track && s.track.kind === 'video'); sender.replaceTrack(videoTrack);

Call control features

Answer: session.answer({ media: { audio: true, video: false } })
Hangup: session.terminate() or session.bye()
Hold/resume: session.hold() / session.unhold() (may use re-INVITE/SDP a=sendonly/a=recvonly)
Transfer: session.transfer(target) or attended transfer flow via REFER
Mute/unmute: stop local track or use RTCRtpSender.setParameters() to adjust encoding

Messaging (SIP MESSAGE)

Send a text message via the UA/session:

ua.sendMessage('sip:[email protected]', 'Hello from VaxWebPhone SDK');

Listen for incoming messages:

ua.on('message', (msg) => {   console.log('Message from', msg.remoteIdentity.uri.toString(), msg.body); });

Integration patterns

React

Keep UA as a singleton or in React context so components can access registration state and sessions.
Use useEffect to initialize/cleanup UA on mount/unmount.
Store active sessions in state; render call UI from session events.

Example pattern:

// PhoneContext.js import React, { createContext, useEffect, useState } from 'react'; import VaxWebPhone from 'vaxvoip-webphone-sdk'; export const PhoneContext = createContext(); export function PhoneProvider({ children }) {   const [ua, setUa] = useState(null);   useEffect(() => {     const client = new VaxWebPhone.UA(config);     setUa(client);     return () => client.stop();   }, []);   return <PhoneContext.Provider value={{ ua }}>{children}</PhoneContext.Provider>; }

Angular

Create an injectable service that manages the UA lifecycle and exposes Observables for registration and sessions.
Use ngOnDestroy to unregister and cleanup.

Security best practices

Always use WSS/WSS+TURN over HTTPS. Avoid plain WS/WSS without TLS.
Use short-lived credentials and OAuth/IMS tokens rather than long-lived static passwords where possible.
Use TURN servers to handle NAT traversal securely; restrict TURN credentials per user/session if supported.
Sanitize and validate incoming SIP headers/messages if exposing any UI that reflects them.
Ensure proper content security policy (CSP) headers and HTTPS configuration on your hosting.

Debugging tips

Enable SDK logging (if available) and route logs to console.
Use browser WebRTC internals: chrome://webrtc-internals (Chrome) to inspect ICE candidates, DTLS, RTP, and stats.
Check STUN/TURN connectivity with samples like testRTC or simple WebRTC test pages.
Monitor SIP signaling with your SIP server logs; use ngrep/tcpdump on the SIP edge if permitted.
Verify device permissions and HTTPS context if getUserMedia fails.

Testing and staging

Test on multiple browsers — Chrome and Edge use Chromium engine, Firefox has different codec/DTLS behavior, and Safari has specific constraints for autoplay and media capture.
Validate mobile browsers and in-app webviews; some webviews restrict WebRTC or require special configuration.
Use network throttling and simulated packet loss to ensure call quality under adverse conditions.

Production deployment checklist

Use HTTPS and WSS endpoints with valid certificates.
Configure TURN with geographic redundancy and authentication.
Implement rate-limiting and access controls on SIP edge servers.
Monitor call quality (MOS, packet loss, jitter) via periodic stats collection and alerts.
Implement fallback UX for when media or signaling fails (e.g., offer callback or PSTN fallback).

Advanced topics & next steps

Conferencing: server-mixed (MCU) vs. client-mixed (SFU) approaches; WebPhone SDK can be used as a client to both models—choose based on scale and bandwidth.
Recording: capture media streams server-side or use media-server integrations (Asterisk, FreeSWITCH, Janus, Jitsi, mediasoup) to record calls.
Analytics: collect RTCP stats and use them to display per-call diagnostics and historical reporting.
Custom codecs and transcoding: understand browser codec support (Opus, VP8, H.264) and transcode server-side if necessary.

Example resources

SDK API reference (consult vendor docs for exact method names and signatures).
SIP and WebRTC specs (RFC 3261, RTCWEB, etc.).
WebRTC debugging guides and chrome://webrtc-internals.

If you want, I can: provide a full React example app using VaxVoIP WebPhone SDK, draft a minimal server-side token auth flow for registration, or generate code snippets for call transfer and hold/resume flows.