## Introduction to SIP.js WebRTC ### What is SIP.js? SIP.js is a powerful JavaScript library that enables developers to incorporate real-time communication (RTC) 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. But how does SIP.js actually simplify WebRTC? For example, instead of writing complex code to manage session negotiation, SIP.js offers a straightforward API, significantly reducing development time.

Launch Your AI Voice Agent in 5 Minutes

Build, customize, and scale AI voice agents with VideoSDK’s developer-friendly APIs and SDKs.

🚀 Get Started Now

### 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 browser 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. How does this work in practice? SIP.js handles the intricate details of session management, allowing developers to focus on building the user interface and application logic. 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. This makes it ideal for scalable WebRTC solutions. Beyond voice calls and video conferencing, SIP.js and WebRTC can power various applications, including screen sharing for remote support, collaborative document editing with real-time updates, and interactive broadcasting platforms. However, when implementing real-time communication, security is paramount. Consider implementing TLS for secure signaling and DTLS for secure media streams. Regularly update SIP.js to benefit from the latest security patches and be aware of potential vulnerabilities like man-in-the-middle attacks. While SIP.js offers many benefits, it's worth considering alternatives like Socket.IO or other custom signaling solutions. However, SIP.js provides a standardized approach, ensuring interoperability and a robust feature set specifically designed for real-time communication. 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 title="bash" mkdir sipjs-webrtc-app cd 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 title="bash" npm 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: ``` sipjs-webrtc-app/ ├── node_modules/ ├── src/ │ ├── components/ │ ├── styles/ │ ├── App.js │ ├── index.js ├── .gitignore ├── package.json ├── 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](https://strapi.videosdk.live/uploads/sequence_diagram_of_how_SIP_js_Web_RTC_work_7e362354ce.png) 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 title="bash" cd 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 title="HTML" SIP.js WebRTC App

``` 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 title="JavaScript" import React from 'react'; import ReactDOM from 'react-dom'; import App from './App'; ReactDOM.render(, 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 title="JavaScript" import React, { useState } from 'react'; import { UA } from 'sip.js'; function App() { const [userAgent, setUserAgent] = useState(null); const handleConnect = () => { const ua = new UA({ uri: 'sip:your_username@sip_server.com', transportOptions: { wsServers: ['wss://sip_server.com/ws'], }, authorizationUser: 'your_username', password: 'your_password', }); ua.start(); setUserAgent(ua); }; return (

SIP.js WebRTC App

); } export 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. When troubleshooting common issues with SIP.js and WebRTC, start by verifying your SIP server configuration. Ensure the WebSocket address is correct and that the server supports the necessary protocols. Also, check your browser's console for error messages that can provide clues about the problem. Network connectivity issues can also prevent the connection from being established, so test your internet connection. ### Running the Application To run the application, you need a development server. You can use a tool like [create-react-app](https://reactjs.org/docs/create-a-new-react-app.html#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 title="bash" npm 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 title="JavaScript" import React, { useState } from 'react'; import { UA } from 'sip.js'; function App() { const [userAgent, setUserAgent] = useState(null); const handleConnect = () => { const ua = new UA({ uri: `sip:${username}@${server}`, transportOptions: { wsServers: [`wss://${server}/ws`], }, authorizationUser: username, password: password, }); ua.start(); setUserAgent(ua); }; return (

SIP.js WebRTC App

{userAgent && } {userAgent && }

); } function JoinScreen({ onConnect }) { const [username, setUsername] = useState(''); const [password, setPassword] = useState(''); const [server, setServer] = useState(''); return (

Join Screen

setUsername(e.target.value)} /> setPassword(e.target.value)} /> setServer(e.target.value)} />

); } function CallControls({ userAgent }) { return (

Call Controls

); } function ParticipantView({ userAgent }) { return (

Participant View

{session.remoteIdentity.displayName || session.remoteIdentity.uri.toString()}

); } export 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 title="JavaScript" import React, { useState } from 'react'; function JoinScreen({ onConnect }) { const [username, setUsername] = useState(''); const [password, setPassword] = useState(''); const [server, setServer] = useState(''); const [error, setError] = useState(''); const handleConnect = () => { if (!username || !password || !server) { setError('All fields are required'); return; } setError(''); onConnect(username, password, server); }; return (

Join Screen

{error &&

{error}

} setUsername(e.target.value)} /> setPassword(e.target.value)} /> setServer(e.target.value)} />

); } export 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 title="JavaScript" import React, { useState } from 'react'; import { UA } from 'sip.js'; import JoinScreen from './components/JoinScreen'; import CallControls from './components/CallControls'; import ParticipantView from './components/ParticipantView'; function App() { const [userAgent, setUserAgent] = useState(null); const handleConnect = (username, password, server) => { const ua = new UA({ uri: `sip:${username}@${server}`, transportOptions: { wsServers: [`wss://${server}/ws`], }, authorizationUser: username, password: password, }); ua.start(); setUserAgent(ua); }; return (

SIP.js WebRTC App

{!userAgent && } {userAgent && } {userAgent && }

); } export 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 title="CSS" /* src/styles/styles.css */ body { font-family: Arial, sans-serif; background-color: #f0f0f0; display: flex; justify-content: center; align-items: center; height: 100vh; margin: 0; } div { background-color: #fff; padding: 20px; border-radius: 8px; box-shadow: 0 2px 4px rgba(0, 0, 0, 0.1); width: 300px; } input { display: block; width: 100%; margin-bottom: 10px; padding: 8px; border: 1px solid #ccc; border-radius: 4px; } button { width: 100%; padding: 10px; background-color: #007bff; color: #fff; border: none; border-radius: 4px; cursor: pointer; } button:hover { background-color: #0056b3; } ``` Import the CSS file in your `index.js`: ```javascript title="JavaScript" import './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 title="JavaScript" import React from 'react'; function CallControls({ userAgent }) { const handleStartCall = () => { const target = prompt('Enter the SIP address of the callee:'); if (target) { userAgent.invite(target); } }; const handleMute = () => { userAgent.getSession().mute(); }; const handleHold = () => { userAgent.getSession().hold(); }; const handleEndCall = () => { userAgent.getSession().terminate(); }; return (

Call Controls

); } export 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 title="JavaScript" import React, { useState } from 'react'; import { UA } from 'sip.js'; import JoinScreen from './components/JoinScreen'; import CallControls from './components/CallControls'; import ParticipantView from './components/ParticipantView'; function App() { const [userAgent, setUserAgent] = useState(null); const handleConnect = (username, password, server) => { const ua = new UA({ uri: `sip:${username}@${server}`, transportOptions: { wsServers: [`wss://${server}/ws`], }, authorizationUser: username, password: password, }); ua.start(); setUserAgent(ua); }; return (

SIP.js WebRTC App

{!userAgent && } {userAgent && } {userAgent && }

); } export 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 title="JavaScript" import React, { useState } from 'react'; import { UA } from 'sip.js'; import JoinScreen from './components/JoinScreen'; import CallControls from './components/CallControls'; import ParticipantView from './components/ParticipantView'; function App() { const [userAgent, setUserAgent] = useState(null); const [session, setSession] = useState(null); const handleConnect = (username, password, server) => { const ua = new UA({ uri: `sip:${username}@${server}`, transportOptions: { wsServers: [`wss://${server}/ws`], }, authorizationUser: username, password: password, }); ua.start(); ua.on('invite', (incomingSession) => { incomingSession.accept(); setSession(incomingSession); }); setUserAgent(ua); }; const handleStartCall = (target) => { const outgoingSession = userAgent.invite(target); setSession(outgoingSession); }; const handleMute = () => { if (session) { session.mute(); } }; const handleHold = () => { if (session) { session.hold(); } }; const handleEndCall = () => { if (session) { session.terminate(); setSession(null); } }; return (

SIP.js WebRTC App

{!userAgent && } {userAgent && ( )} {userAgent && }

); } export 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 title="JavaScript" import React from 'react'; function CallControls({ onStartCall, onMute, onHold, onEndCall }) { const handleStartCall = () => { const target = prompt('Enter the SIP address of the callee:'); if (target) { onStartCall(target); } }; return (

Call Controls

); } export 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.