How to Build SIP.js WebRTC App with JavaScript?

Learn how to integrate SIP.js with WebRTC in this step-by-step guide. From setting up your development environment to deploying a fully functional SIP.js WebRTC application, this article covers everything you need to know.

Introduction to SIP.js WebRTC

What is SIP.js?

SIP.js is a powerful JavaScript library that enables developers to incorporate real-time communications into web applications using the SIP (Session Initiation Protocol) over WebRTC (Web Real-Time Communication). It simplifies the process of setting up and managing WebRTC connections, making it easier for developers to build robust, scalable, and high-quality communication solutions.

The Importance of WebRTC in Modern Web Applications

WebRTC is a groundbreaking technology that allows for real-time audio, video, and data sharing directly between browsers without the need for plugins or external applications. This has revolutionized web communication, enabling features such as video conferencing, voice calls, and file sharing to be integrated seamlessly into web applications.

Benefits of Using SIP.js for WebRTC Signaling

Using SIP.js for WebRTC signaling offers several key advantages:
  1. Ease of Use: SIP.js abstracts much of the complexity involved in WebRTC signaling, providing a straightforward API for developers.
  2. Interoperability: It supports a wide range of SIP servers and clients, ensuring that applications built with SIP.js can communicate with other SIP-based systems.
  3. Flexibility: SIP.js is highly customizable, allowing developers to tailor the signaling process to meet specific application requirements.
  4. Scalability: The library is designed to handle high volumes of traffic, making it suitable for both small-scale applications and large, enterprise-level deployments.
By leveraging SIP.js, developers can quickly and efficiently integrate WebRTC capabilities into their applications, providing users with a seamless and high-quality communication experience. This introduction serves as the foundation for understanding how to harness the power of SIP.js and WebRTC, setting the stage for the detailed, step-by-step implementation guide that follows.

Getting Started with SIP.js WebRTC

Create a New SIP.js WebRTC App

To kickstart your journey with SIP.js and WebRTC, we'll guide you through creating a new SIP.js WebRTC application from scratch. This section covers setting up the development environment, installing SIP.js, structuring the project, and understanding the app architecture.

Setting Up the Development Environment

Before diving into the code, ensure your development environment is ready. You'll need Node.js and npm (Node Package Manager) installed on your machine. You can download and install them from the official Node.js website.
Once Node.js and npm are installed, create a new directory for your project and navigate into it:

bash

1mkdir sipjs-webrtc-app
2cd sipjs-webrtc-app

Installing SIP.js

To use SIP.js in your project, you need to install it via npm. Run the following command to add SIP.js to your project dependencies:

bash

1npm install sip.js
This command installs SIP.js and adds it to your project's package.json file.

Structure of the Project

A well-structured project is crucial for maintainability and scalability. Here's a basic structure for your SIP.js WebRTC app:
1sipjs-webrtc-app/
2├── node_modules/
3├── src/
4│   ├── components/
5│   ├── styles/
6│   ├── App.js
7│   ├── index.js
8├── .gitignore
9├── package.json
10├── README.md
  • src/components/: This directory will contain your React components.
  • src/styles/: This directory will store your CSS files.
  • src/App.js: The main React component that will serve as the entry point for your app.
  • src/index.js: The JavaScript file that renders your React app to the DOM.

App Architecture

sipjs-webrtc
The architecture of your SIP.js WebRTC app consists of several key components:

User Interface (UI)

This includes all the visual elements of your app, such as buttons, forms, and displays for audio/video streams. You'll build the UI using React components.

SIP.js Integration

SIP.js handles the signaling part of the WebRTC connection. This involves establishing and managing communication sessions between clients.

WebRTC Media Streams

WebRTC takes care of capturing and transmitting audio and video streams. You'll use the WebRTC API to manage these media streams.

Server-Side Components

While this guide focuses on the client-side, a SIP server (like Asterisk or FreeSWITCH) is required to facilitate the signaling between clients. Ensure you have a SIP server set up and configured properly.
By the end of this section, you should have a new project directory set up with SIP.js installed and a basic project structure in place. This foundation will allow you to build out the rest of your SIP.js WebRTC application with a clear and organized approach. In the next sections, we'll dive into the detailed implementation steps, starting with setting up the basic configuration and getting connected to a SIP server.

Step 1: Get Started with Basic Setup

In this section, we'll walk you through the initial setup for your SIP.js WebRTC application. We'll cover creating the initial project files, configuring basic settings, and connecting to a SIP server. By the end of this step, you'll have a solid foundation to build upon.

Setting Up the Project

First, ensure you are in your project directory:

bash

1cd sipjs-webrtc-app
Now, let's create the initial project files. We'll start by setting up the index.html, index.js, and App.js files.

[a] Create index.html

In the src directory, create an index.html file:

HTML

1<!DOCTYPE html>
2<html lang="en">
3<head>
4    <meta charset="UTF-8">
5    <meta name="viewport" content="width=device-width, initial-scale=1.0">
6    <title>SIP.js WebRTC App</title>
7</head>
8<body>
9    <div id="root"></div>
10    <script src="index.js"></script>
11</body>
12</html>
This basic HTML file sets up a div with an id of root where our React app will be rendered.

[b] Create index.js

Next, create an index.js file in the src directory. This file will render our main React component into the root div:

JavaScript

1import React from 'react';
2import ReactDOM from 'react-dom';
3import App from './App';
4
5ReactDOM.render(<App />, document.getElementById('root'));

[c] Create App.js

Now, create an App.js file in the src directory. This file will be the main component of our React app:

JavaScript

1import React, { useState } from 'react';
2import { UA } from 'sip.js';
3
4function App() {
5    const [userAgent, setUserAgent] = useState(null);
6
7    const handleConnect = () => {
8        const ua = new UA({
9            uri: 'sip:your_username@sip_server.com',
10            transportOptions: {
11                wsServers: ['wss://sip_server.com/ws'],
12            },
13            authorizationUser: 'your_username',
14            password: 'your_password',
15        });
16
17        ua.start();
18        setUserAgent(ua);
19    };
20
21    return (
22        <div>
23            <h1>SIP.js WebRTC App</h1>
24            <button onClick={handleConnect}>Connect</button>
25        </div>
26    );
27}
28
29export default App;

Configuring Basic Settings

In the App.js file, we have a basic configuration for the SIP.js user agent (UA). Here’s a breakdown of the configuration options:
  • uri: This is the SIP address of the user. Replace your_username with your actual SIP username and sip_server.com with your SIP server’s domain.
  • transportOptions: This includes the WebSocket server address used for signaling. Replace sip_server.com with your actual SIP server’s domain.
  • authorizationUser and password: These are your SIP credentials. Replace your_username and your_password with your actual SIP username and password.

Connecting to a SIP Server

The handleConnect function creates a new instance of the SIP.js UA and starts it. When the user clicks the "Connect" button, this function is called, initiating the connection to the SIP server.
Before running the app, make sure you have a SIP server (like Asterisk or FreeSWITCH) set up and configured properly. This server will handle the signaling for WebRTC connections.

Running the Application

To run the application, you need a development server. You can use a tool like

create-react-app

 to quickly set up a development server if you haven't already.
If you used create-react-app, start the development server with:

bash

1npm start
Open your browser and navigate to http://localhost:3000. You should see a simple interface with a "Connect" button. Clicking this button will initiate the connection to your SIP server.
By following these steps, you have set up the initial configuration for your SIP.js WebRTC application. You created the necessary project files, configured basic settings, and connected to a SIP server. This groundwork prepares you for the subsequent steps, where you'll add more functionality and refine your app. Next, we'll dive into designing the user interface and wiring up all the components.

Step 2: Wireframe All the Components

In this step, we'll focus on designing the user interface (UI) of your SIP.js WebRTC application. A well-structured UI is essential for providing a smooth user experience. We'll create a basic wireframe, identify key components, and structure the HTML layout.

Designing the User Interface

The UI of a SIP.js WebRTC application typically includes elements such as login screens, call controls, and participant views. We'll start by outlining these components and their functions.

Key Components

  1. Join Screen: Allows users to enter their credentials and connect to the SIP server.
  2. Call Controls: Provides buttons for initiating, muting, holding, and ending calls.
  3. Participant View: Displays a list of active participants in the call.

Structuring the HTML Layout

Now, let's translate this wireframe into a structured HTML layout using React components. We'll update the App.js file to include placeholders for each component.

Update App.js

JavaScript

1import React, { useState } from 'react';
2import { UA } from 'sip.js';
3
4function App() {
5    const [userAgent, setUserAgent] = useState(null);
6
7    const handleConnect = () => {
8        const ua = new UA({
9            uri: `sip:${username}@${server}`,
10            transportOptions: {
11                wsServers: [`wss://${server}/ws`],
12            },
13            authorizationUser: username,
14            password: password,
15        });
16
17        ua.start();
18        setUserAgent(ua);
19    };
20
21    return (
22        <div>
23            <h1>SIP.js WebRTC App</h1>
24            <JoinScreen onConnect={handleConnect} />
25            {userAgent && <CallControls userAgent={userAgent} />}
26            {userAgent && <ParticipantView userAgent={userAgent} />}
27        </div>
28    );
29}
30
31function JoinScreen({ onConnect }) {
32    const [username, setUsername] = useState('');
33    const [password, setPassword] = useState('');
34    const [server, setServer] = useState('');
35
36    return (
37        <div>
38            <h2>Join Screen</h2>
39            <input 
40                type="text" 
41                placeholder="Username" 
42                value={username}
43                onChange={(e) => setUsername(e.target.value)} 
44            />
45            <input 
46                type="password" 
47                placeholder="Password" 
48                value={password}
49                onChange={(e) => setPassword(e.target.value)} 
50            />
51            <input 
52                type="text" 
53                placeholder="Server" 
54                value={server}
55                onChange={(e) => setServer(e.target.value)} 
56            />
57            <button onClick={() => onConnect(username, password, server)}>Connect</button>
58        </div>
59    );
60}
61
62function CallControls({ userAgent }) {
63    return (
64        <div>
65            <h2>Call Controls</h2>
66            <button onClick={() => userAgent.invite('sip:callee@sip_server.com')}>Start Call</button>
67            <button onClick={() => userAgent.mute()}>Mute</button>
68            <button onClick={() => userAgent.hold()}>Hold</button>
69            <button onClick={() => userAgent.terminateSessions()}>End Call</button>
70        </div>
71    );
72}
73
74function ParticipantView({ userAgent }) {
75    return (
76        <div>
77            <h2>Participant View</h2>
78            <ul>
79                {userAgent.sessions.map((session, index) => (
80                    <li key={index}>{session.remoteIdentity.displayName || session.remoteIdentity.uri.toString()}</li>
81                ))}
82            </ul>
83        </div>
84    );
85}
86
87export default App;

Explanation of the Code

  • JoinScreen Component: This component includes input fields for username, password, and server, and a "Connect" button to initiate the connection to the SIP server.
  • CallControls Component: This component provides buttons to start a call, mute, hold, and end the call. The userAgent object is used to control the call actions.
  • ParticipantView Component: This component displays the list of active participants in the call. It maps over the userAgent.sessions array to display each participant.
By structuring your application in this way, you separate concerns and make your code more modular and maintainable.
In this section, you designed the UI of your SIP.js WebRTC application by creating a basic wireframe and structuring the HTML layout using React components. This sets the stage for implementing the join screen, call controls, and participant view in subsequent steps. Next, we'll dive deeper into implementing the join screen functionality.

Step 3: Implement Join Screen

In this step, we will focus on implementing the join screen, which allows users to enter their credentials and connect to the SIP server. This involves designing the interface, handling user inputs, and validating user credentials.

Creating the Join Screen

The join screen is the first interaction point for users. It collects necessary information such as username, password, and server address, and initiates the connection to the SIP server.

Update the JoinScreen Component

We will enhance the JoinScreen component to handle user input and connect to the SIP server upon clicking the "Connect" button.
Updated JoinScreen Component:

JavaScript

1import React, { useState } from 'react';
2
3function JoinScreen({ onConnect }) {
4    const [username, setUsername] = useState('');
5    const [password, setPassword] = useState('');
6    const [server, setServer] = useState('');
7    const [error, setError] = useState('');
8
9    const handleConnect = () => {
10        if (!username || !password || !server) {
11            setError('All fields are required');
12            return;
13        }
14        setError('');
15        onConnect(username, password, server);
16    };
17
18    return (
19        <div>
20            <h2>Join Screen</h2>
21            {error && <p style={{ color: 'red' }}>{error}</p>}
22            <input 
23                type="text" 
24                placeholder="Username" 
25                value={username}
26                onChange={(e) => setUsername(e.target.value)} 
27            />
28            <input 
29                type="password" 
30                placeholder="Password" 
31                value={password}
32                onChange={(e) => setPassword(e.target.value)} 
33            />
34            <input 
35                type="text" 
36                placeholder="Server" 
37                value={server}
38                onChange={(e) => setServer(e.target.value)} 
39            />
40            <button onClick={handleConnect}>Connect</button>
41        </div>
42    );
43}
44
45export default JoinScreen;
Explanation
  • State Management: We use React's useState hook to manage the input values (username, password, server) and any error messages.
  • Input Fields: The component includes three input fields for the username, password, and server. Each input updates its corresponding state.
  • Error Handling: If any input field is empty when the user tries to connect, an error message is displayed.
  • Handle Connect: The handleConnect function validates the inputs and invokes the onConnect callback with the input values if validation passes.

Update the App.js to Handle Connection Logic

We need to modify the App.js file to handle the connection logic when the user submits their credentials.
Updated App.js:

JavaScript

1import React, { useState } from 'react';
2import { UA } from 'sip.js';
3import JoinScreen from './components/JoinScreen';
4import CallControls from './components/CallControls';
5import ParticipantView from './components/ParticipantView';
6
7function App() {
8    const [userAgent, setUserAgent] = useState(null);
9
10    const handleConnect = (username, password, server) => {
11        const ua = new UA({
12            uri: `sip:${username}@${server}`,
13            transportOptions: {
14                wsServers: [`wss://${server}/ws`],
15            },
16            authorizationUser: username,
17            password: password,
18        });
19
20        ua.start();
21        setUserAgent(ua);
22    };
23
24    return (
25        <div>
26            <h1>SIP.js WebRTC App</h1>
27            {!userAgent && <JoinScreen onConnect={handleConnect} />}
28            {userAgent && <CallControls userAgent={userAgent} />}
29            {userAgent && <ParticipantView userAgent={userAgent} />}
30        </div>
31    );
32}
33
34export default App;
Explanation
  • State Management: The userAgent state holds the instance of the SIP.js user agent.
  • Handle Connect: The handleConnect function creates and starts a new UA instance with the provided credentials and server address. It then updates the userAgent state with the new instance.
  • Conditional Rendering: The JoinScreen component is displayed only if userAgent is null. Once connected, the CallControls and ParticipantView components are rendered.

Styling the Join Screen

To improve the user experience, we'll add some basic styles to the join screen. Create a styles.css file in the src/styles directory and add the following CSS:

CSS

1/* src/styles/styles.css */
2body {
3    font-family: Arial, sans-serif;
4    background-color: #f0f0f0;
5    display: flex;
6    justify-content: center;
7    align-items: center;
8    height: 100vh;
9    margin: 0;
10}
11
12div {
13    background-color: #fff;
14    padding: 20px;
15    border-radius: 8px;
16    box-shadow: 0 2px 4px rgba(0, 0, 0, 0.1);
17    width: 300px;
18}
19
20input {
21    display: block;
22    width: 100%;
23    margin-bottom: 10px;
24    padding: 8px;
25    border: 1px solid #ccc;
26    border-radius: 4px;
27}
28
29button {
30    width: 100%;
31    padding: 10px;
32    background-color: #007bff;
33    color: #fff;
34    border: none;
35    border-radius: 4px;
36    cursor: pointer;
37}
38
39button:hover {
40    background-color: #0056b3;
41}
Import the CSS file in your index.js:

JavaScript

1import './styles/styles.css';
In this step, we implemented the join screen for our SIP.js WebRTC application. We designed the user interface, handled user inputs, validated credentials, and connected to the SIP server. This join screen serves as the entry point for users to access the application's real-time communication features. In the next steps, we will implement the call controls and participant view to enhance the application's functionality.

Step 4: Implement Controls

In this step, we'll implement the call controls for your SIP.js WebRTC application. These controls will allow users to start calls, mute, hold, and end calls. This section involves designing the interface for the controls, writing the logic for call actions, and integrating these functionalities into the application.

Adding Call Controls

The call controls are essential for managing the state of the call and providing users with necessary actions to handle their communication effectively.

[a] Create CallControls Component

First, let's create a new CallControls component. This component will include buttons for starting a call, muting, holding, and ending a call.
Create CallControls.js:

JavaScript

1import React from 'react';
2
3function CallControls({ userAgent }) {
4    const handleStartCall = () => {
5        const target = prompt('Enter the SIP address of the callee:');
6        if (target) {
7            userAgent.invite(target);
8        }
9    };
10
11    const handleMute = () => {
12        userAgent.getSession().mute();
13    };
14
15    const handleHold = () => {
16        userAgent.getSession().hold();
17    };
18
19    const handleEndCall = () => {
20        userAgent.getSession().terminate();
21    };
22
23    return (
24        <div>
25            <h2>Call Controls</h2>
26            <button onClick={handleStartCall}>Start Call</button>
27            <button onClick={handleMute}>Mute</button>
28            <button onClick={handleHold}>Hold</button>
29            <button onClick={handleEndCall}>End Call</button>
30        </div>
31    );
32}
33
34export default CallControls;
Explanation
  • handleStartCall: Prompts the user to enter the SIP address of the callee and initiates a call using userAgent.invite().
  • handleMute: Mutes the current session using userAgent.getSession().mute().
  • handleHold: Puts the current session on hold using userAgent.getSession().hold().
  • handleEndCall: Terminates the current session using userAgent.getSession().terminate().

[b] Integrate CallControls Component

Now, we need to integrate the CallControls component into the main application. Update the App.js file to include this component.
Updated App.js:

JavaScript

1import React, { useState } from 'react';
2import { UA } from 'sip.js';
3import JoinScreen from './components/JoinScreen';
4import CallControls from './components/CallControls';
5import ParticipantView from './components/ParticipantView';
6
7function App() {
8    const [userAgent, setUserAgent] = useState(null);
9
10    const handleConnect = (username, password, server) => {
11        const ua = new UA({
12            uri: `sip:${username}@${server}`,
13            transportOptions: {
14                wsServers: [`wss://${server}/ws`],
15            },
16            authorizationUser: username,
17            password: password,
18        });
19
20        ua.start();
21        setUserAgent(ua);
22    };
23
24    return (
25        <div>
26            <h1>SIP.js WebRTC App</h1>
27            {!userAgent && <JoinScreen onConnect={handleConnect} />}
28            {userAgent && <CallControls userAgent={userAgent} />}
29            {userAgent && <ParticipantView userAgent={userAgent} />}
30        </div>
31    );
32}
33
34export default App;
Explanation
Conditional Rendering: The CallControls component is rendered only if the userAgent is not null.

[c] Enhance Call Control Functionalities

To handle the call actions correctly, we need to ensure the userAgent has an active session. We will add session management in the App.js.
Update App.js for Session Management:

JavaScript

1import React, { useState } from 'react';
2import { UA } from 'sip.js';
3import JoinScreen from './components/JoinScreen';
4import CallControls from './components/CallControls';
5import ParticipantView from './components/ParticipantView';
6
7function App() {
8    const [userAgent, setUserAgent] = useState(null);
9    const [session, setSession] = useState(null);
10
11    const handleConnect = (username, password, server) => {
12        const ua = new UA({
13            uri: `sip:${username}@${server}`,
14            transportOptions: {
15                wsServers: [`wss://${server}/ws`],
16            },
17            authorizationUser: username,
18            password: password,
19        });
20
21        ua.start();
22        ua.on('invite', (incomingSession) => {
23            incomingSession.accept();
24            setSession(incomingSession);
25        });
26        setUserAgent(ua);
27    };
28
29    const handleStartCall = (target) => {
30        const outgoingSession = userAgent.invite(target);
31        setSession(outgoingSession);
32    };
33
34    const handleMute = () => {
35        if (session) {
36            session.mute();
37        }
38    };
39
40    const handleHold = () => {
41        if (session) {
42            session.hold();
43        }
44    };
45
46    const handleEndCall = () => {
47        if (session) {
48            session.terminate();
49            setSession(null);
50        }
51    };
52
53    return (
54        <div>
55            <h1>SIP.js WebRTC App</h1>
56            {!userAgent && <JoinScreen onConnect={handleConnect} />}
57            {userAgent && (
58                <CallControls
59                    onStartCall={handleStartCall}
60                    onMute={handleMute}
61                    onHold={handleHold}
62                    onEndCall={handleEndCall}
63                />
64            )}
65            {userAgent && <ParticipantView userAgent={userAgent} />}
66        </div>
67    );
68}
69
70export default App;
Explanation:
  • Session Management: The session state is used to manage the current SIP session.
  • Event Handling: The userAgent listens for incoming calls and accepts them automatically, setting the session state.
  • Updated Call Control Functions: The call control functions now check for the existence of a session before performing actions.

[d] Update CallControls Component for New Props

Update the CallControls component to use the new props for handling call actions.
Updated CallControls.js:

JavaScript

1import React from 'react';
2
3function CallControls({ onStartCall, onMute, onHold, onEndCall }) {
4    const handleStartCall = () => {
5        const target = prompt('Enter the SIP address of the callee:');
6        if (target) {
7            onStartCall(target);
8        }
9    };
10
11    return (
12        <div>
13            <h2>Call Controls</h2>
14            <button onClick={handleStartCall}>Start Call</button>
15            <button onClick={onMute}>Mute</button>
16            <button onClick={onHold}>Hold</button>
17            <button onClick={onEndCall}>End Call</button>
18        </div>
19    );
20}
21
22export default CallControls;
Explanation: Prop Handling: The component now uses props (onStartCall, onMute, onHold, onEndCall) to handle call actions, ensuring the main logic resides in App.js.
In this step, we implemented the call controls for the SIP.js WebRTC application. We created a CallControls component, integrated it into the main application, and ensured proper session management for handling call actions. These controls allow users to start, mute, hold, and end calls, enhancing the application's functionality and user experience. In the next step, we will focus on implementing the participant view to display active call participants.

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Step 5: Implement Participant View

In this step, we'll implement the participant view for your SIP.js WebRTC application. This feature will display a list of active participants in the call, providing a clear overview of who is currently connected. We will update the ParticipantView component to reflect this functionality.

Displaying Call Participants

The participant view will dynamically update to show the current call participants. We need to ensure that our application listens for changes in the session state and updates the participant list accordingly.

[a] Update ParticipantView Component

First, we'll create a ParticipantView component that lists the active participants in the call.
Create ParticipantView.js:

JavaScript

1import React, { useEffect, useState } from 'react';
2
3function ParticipantView({ session }) {
4    const [participants, setParticipants] = useState([]);
5
6    useEffect(() => {
7        if (session) {
8            const updateParticipants = () => {
9                const newParticipants = [];
10                if (session.remoteIdentity.displayName) {
11                    newParticipants.push(session.remoteIdentity.displayName);
12                } else {
13                    newParticipants.push(session.remoteIdentity.uri.toString());
14                }
15                setParticipants(newParticipants);
16            };
17
18            updateParticipants();
19
20            session.on('accepted', updateParticipants);
21            session.on('bye', () => setParticipants([]));
22            session.on('terminated', () => setParticipants([]));
23
24            return () => {
25                session.off('accepted', updateParticipants);
26                session.off('bye', () => setParticipants([]));
27                session.off('terminated', () => setParticipants([]));
28            };
29        }
30    }, [session]);
31
32    return (
33        <div>
34            <h2>Participant View</h2>
35            <ul>
36                {participants.map((participant, index) => (
37                    <li key={index}>{participant}</li>
38                ))}
39            </ul>
40        </div>
41    );
42}
43
44export default ParticipantView;
Explanation:
  • State Management: We use the useState hook to manage the list of participants.
  • Effect Hook: The useEffect hook listens for changes in the session and updates the participant list accordingly.
  • Event Handling: The component listens for accepted, bye, and terminated events to update the participant list.

[b] Integrate ParticipantView Component

Next, we'll integrate the ParticipantView component into the main application. We need to pass the current session to the ParticipantView component.
Updated App.js:

JavaScript

1import React, { useState } from 'react';
2import { UA } from 'sip.js';
3import JoinScreen from './components/JoinScreen';
4import CallControls from './components/CallControls';
5import ParticipantView from './components/ParticipantView';
6
7function App() {
8    const [userAgent, setUserAgent] = useState(null);
9    const [session, setSession] = useState(null);
10
11    const handleConnect = (username, password, server) => {
12        const ua = new UA({
13            uri: `sip:${username}@${server}`,
14            transportOptions: {
15                wsServers: [`wss://${server}/ws`],
16            },
17            authorizationUser: username,
18            password: password,
19        });
20
21        ua.start();
22        ua.on('invite', (incomingSession) => {
23            incomingSession.accept();
24            setSession(incomingSession);
25        });
26        setUserAgent(ua);
27    };
28
29    const handleStartCall = (target) => {
30        const outgoingSession = userAgent.invite(target);
31        setSession(outgoingSession);
32    };
33
34    const handleMute = () => {
35        if (session) {
36            session.mute();
37        }
38    };
39
40    const handleHold = () => {
41        if (session) {
42            session.hold();
43        }
44    };
45
46    const handleEndCall = () => {
47        if (session) {
48            session.terminate();
49            setSession(null);
50        }
51    };
52
53    return (
54        <div>
55            <h1>SIP.js WebRTC App</h1>
56            {!userAgent && <JoinScreen onConnect={handleConnect} />}
57            {userAgent && (
58                <CallControls
59                    onStartCall={handleStartCall}
60                    onMute={handleMute}
61                    onHold={handleHold}
62                    onEndCall={handleEndCall}
63                />
64            )}
65            {session && <ParticipantView session={session} />}
66        </div>
67    );
68}
69
70export default App;
Explanation:
  • Session Prop: The session state is passed to the ParticipantView component, allowing it to display the active participants.
  • Conditional Rendering: The ParticipantView component is rendered only if there is an active session.

[c] Enhancing Participant Management

For a more comprehensive participant view, you might want to display more details or manage multiple participants if your application supports group calls. The basic implementation can be extended as needed to suit your application's requirements.
Extended ParticipantView for Group Calls:
If your application supports group calls, you might need to handle multiple participants. Here's an example extension for the ParticipantView component:

JavaScript

1import React, { useEffect, useState } from 'react';
2
3function ParticipantView({ session }) {
4    const [participants, setParticipants] = useState([]);
5
6    useEffect(() => {
7        if (session) {
8            const updateParticipants = () => {
9                const newParticipants = session.dialogs.map(dialog => dialog.remoteIdentity.displayName || dialog.remoteIdentity.uri.toString());
10                setParticipants(newParticipants);
11            };
12
13            updateParticipants();
14
15            session.on('progress', updateParticipants);
16            session.on('accepted', updateParticipants);
17            session.on('bye', updateParticipants);
18            session.on('terminated', updateParticipants);
19
20            return () => {
21                session.off('progress', updateParticipants);
22                session.off('accepted', updateParticipants);
23                session.off('bye', updateParticipants);
24                session.off('terminated', updateParticipants);
25            };
26        }
27    }, [session]);
28
29    return (
30        <div>
31            <h2>Participant View</h2>
32            <ul>
33                {participants.map((participant, index) => (
34                    <li key={index}>{participant}</li>
35                ))}
36            </ul>
37        </div>
38    );
39}
40
41export default ParticipantView;
Explanation:
  • Group Calls: The updated ParticipantView handles multiple participants by mapping over the session dialogs.
  • Event Handling: The component listens for additional events (progress, accepted, bye, terminated) to keep the participant list updated.
In this step, we implemented the participant view for the SIP.js WebRTC application. This component dynamically displays the list of active participants in a call, providing users with clear visibility of who is connected. We ensured that the participant view updates in real-time based on session events. This enhancement improves the overall user experience by keeping them informed about the current call participants. In the next step, we will focus on running and testing the application to ensure everything works as expected.

Step 6: Run Your Code Now

In this final step, we will focus on running and testing your SIP.js WebRTC application to ensure everything works as expected. We'll cover how to start the application, test its functionality, and troubleshoot common issues. This section will also include some tips for preparing your app for production deployment.

Running the Application

To run your application, you'll need to start your development server. If you used a tool like create-react-app to set up your project, you can use the following command:

bash

1npm start
This command will start the development server and open your application in the default web browser. The application should be accessible at http://localhost:3000.

Testing the Application

Once your application is running, you can test its functionality by following these steps:

Open the Application

Navigate to http://localhost:3000 in your web browser.

Connect to SIP Server

  • Enter your SIP username, password, and server address in the Join Screen.
  • Click the "Connect" button.
  • If the connection is successful, the call controls and participant view should appear.

Test Call Controls

  • Start a Call: Click the "Start Call" button and enter the SIP address of the callee. Verify that the call is initiated.
  • Mute: Click the "Mute" button to mute the call. Verify that the call is muted.
  • Hold: Click the "Hold" button to put the call on hold. Verify that the call is on hold.
  • End Call: Click the "End Call" button to terminate the call. Verify that the call is ended.

Check Participant View

  • Ensure that the participant view displays the active call participants.
  • Verify that the participant list updates dynamically based on the call status.

Troubleshooting Common Issues

If you encounter any issues while running your application, here are some common troubleshooting tips:

Connection Issues

  • Ensure that your SIP server is running and accessible.
  • Verify the SIP username, password, and server address are correct.
  • Check for any network issues or firewall settings that might be blocking the connection.

Call Controls Not Working

  • Ensure that the session object is properly initialized and available.
  • Check the console for any errors related to SIP.js or WebRTC.

Participant View Not Updating

  • Verify that the event listeners are correctly set up in the ParticipantView component.
  • Ensure that the session events (accepted, bye, terminated) are being triggered correctly.

Preparing for Production Deployment

Once you have tested your application and ensured it works as expected, you can prepare it for production deployment. Here are some steps to consider:

Build the Application

Use the following command to create a production build of your application:

bash

1     npm run build
This will generate optimized static files for deployment.

Configure SIP Server

  • Ensure that your SIP server is properly configured and secured for production use.
  • Consider using a reliable SIP provider to handle production traffic.

Deploy the Application

  • Deploy the built files to a web server or a cloud service provider like AWS, Google Cloud, or Azure.
  • Ensure that your deployment environment supports WebSocket connections required for SIP.js.

Monitor and Maintain

  • Set up monitoring and logging to track the performance and usage of your application.
  • Regularly update dependencies and security patches to keep your application secure.

Conclusion

By following this guide, you have successfully built and tested a SIP.js WebRTC application. You have learned how to create a join screen, implement call controls, and display call participants. With these features in place, your application is ready for production deployment.

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