A Guide to WebRTC Video Call

In today’s digital landscape, real-time communication is more important than ever, and WebRTC has become a go-to technology for building seamless video call applications. With ZEGOCLOUD’s WebRTC SDK, developers can easily integrate high-quality, low-latency video calling features into their apps without the complexity of managing peer-to-peer connections from scratch. This guide will walk you through the key steps to get started, from setting up your environment to implementing a fully functional video call feature.

What is WebRTC?

WebRTC (Web Real-Time Communication) is an open-source technology that enables real-time communication of audio, video, and data directly between browsers and devices, without requiring plugins or additional software. It allows for peer-to-peer connections, making it ideal for applications like video calls, voice chats, file sharing, and live streaming.

WebRTC works by capturing media from a user’s device, establishing a peer-to-peer connection for streaming audio and video, and enabling data transmission between users. Its ability to provide low-latency, secure communication makes WebRTC a powerful tool for building interactive, real-time applications across web and mobile platforms.

How Does WebRTC Work?

WebRTC works by enabling real-time communication between two or more devices through peer-to-peer connections, allowing users to share audio, video, and data directly. The process starts with media capture, where WebRTC accesses the user’s camera and microphone using the browser’s getUserMedia API.

Once the media is captured, WebRTC establishes a peer-to-peer connection through the RTCPeerConnection API. To create this connection, the two peers exchange network information and connection details (such as IP addresses) via a signaling server. WebRTC then handles NAT traversal using ICE (Interactive Connectivity Establishment) to find the best path for data to travel between devices, even when they are behind firewalls or NATs.

After the connection is established, media (audio and video) and data are streamed directly between the peers using encrypted communication channels, ensuring privacy and security. For sending non-media data, WebRTC also uses the RTCDataChannel, which allows for real-time data sharing, such as file transfers or gaming data. This architecture minimizes latency and reduces the need for intermediaries, making WebRTC ideal for live, real-time interactions like video calls, conferencing, and streaming.

Flow of a WebRTC Video Call

The flow of a WebRTC video call involves several steps, from establishing a connection to maintaining real-time communication between peers. Here’s an overview of the key stages in the process:

1. Media Capture:

The process starts by using the getUserMedia() API to access the user’s camera and microphone, capturing audio and video streams.

2. Signaling:

Peers exchange connection information (like SDP and ICE candidates) through a signaling server, such as WebSockets, to establish communication.

3. Peer Connection:

WebRTC uses RTCPeerConnection and ICE to find the best path for a peer-to-peer connection, even behind firewalls or NATs.

4. Media Streaming:

Once connected, audio and video are streamed directly between peers using RTP, ensuring real-time, low-latency communication. RTCDataChannel can also be used for data sharing.

5. Connection Maintenance and Termination:

WebRTC adapts to network conditions during the call, and once the call ends, the connection is closed and resources are released.

How to Make a WebRTC Video Calling App with ZEGOCLOUD

Creating a video calling app from scratch using WebRTC can be complex, requiring careful attention to peer-to-peer connections, media streaming, and real-time communication protocols. However, with ZEGOCLOUD’s WebRTC video call SDK, building a video calling app becomes much simpler, as it handles the heavy lifting involved in WebRTC implementation, allowing developers to focus on the core functionalities of their app. In this guide, we’ll walk you through how to make a video calling app with WebRTC using ZEGOCLOUD.

Why Use ZEGOCLOUD for WebRTC Video Calling?

ZEGOCLOUD offers a powerful WebRTC SDK that simplifies the process of building a video calling app. Instead of managing the intricate details of WebRTC—such as connection setup, media streaming, and security—you can leverage ZEGOCLOUD’s easy-to-use SDK to accelerate development. Here are some key advantages:

• Scalability: ZEGOCLOUD supports one-on-one video calls and multi-party conferencing, making it ideal for both personal and business applications.
• Low Latency: ZEGOCLOUD provides a high-quality, low-latency video experience, ensuring smooth communication.
• Cross-platform Compatibility: Build your app for web, iOS, and Android with a unified SDK that works across platforms.
• Pre-built UIKits: ZEGOCLOUD offers customizable UIKits for video calls, so you don’t need to design interfaces from scratch.

Preparation for WebRTC SDK

• A ZEGOCLOUD developer account–Sign up
• The latest version of Chrome.
• Basic understanding of Web development

ZEGOCLOUD SDK Integration for WebRTC Video Calling

Importer Call Kit SDK

• Create a new HTML file and add the following code to the file.
• Then, replace the appID and serverSecret parameters in the following code with the AppID and Server Secret of your project.

<html>
  <head>
    <style>
      #root {
        width: 100vw;
        height: 100vh;
      }
    </style>
  </head>

  <body>
    <div id="root"></div>
  </body>
  <script src="https://unpkg.com/@zegocloud/zego-uikit-prebuilt/zego-uikit-prebuilt.js"></script>
  <script>
    function getUrlParams(url = window.location.href) {
      let urlStr = url.split("?")[1];
      return new URLSearchParams(urlStr);
    }
    // Generate a Kit Token by calling a method.
    // @param 1: appID
    // @param 2: serverSecret
    // @param 3: Room ID
    // @param 4: User ID
    // @param 5: Username
    const roomID =
      getUrlParams().get("roomID") ||
      "room_" + Math.floor(Math.random() * 1000);
    const userID = Math.floor(Math.random() * 10000) + "";
    const userName = "userName" + userID;
    const appID = 221417685;
    const serverSecret = "be15e8ea06d70c26c704f87ed661b6aa";
    const kitToken = ZegoUIKitPrebuilt.generateKitTokenForTest(
      appID,
      serverSecret,
      roomID,
      userID,
      userName
    );

    const zp = ZegoUIKitPrebuilt.create(kitToken);
    zp.joinRoom({
      container: document.querySelector("#root"),
      sharedLinks: [
        {
          url:
            window.location.origin +
            window.location.pathname +
            "?roomID=" +
            roomID,
        },
      ],
      scenario: {
        mode: ZegoUIKitPrebuilt.OneONoneCallGroupCall,
      },
    });
  </script>
</html>

Group-call configuration

Group-call scenarios can be implemented by configuring the parameters of the joinRoom function.

• Set scenario.mode to ZegoUIKitPrebuilt.GroupCall

// ...
zp.joinRoom({
        //...
        scenario: {
           mode: ZegoUIKitPrebuilt.GroupCall 
        },
        //...
});
//...

Audio-only call configuration

Audio-only call scenarios can also be implemented by configuring the parameters of the join room function.

• Set the turnOnCameraWhenJoining to false to turn off the camera.
• Set the showMyCameraToggleButton to false to hide the toggle button for enabling/disabling the camera.
• Set the showAudioVideoSettingsButton to false to hide the advanced setting button for audio and video.
• Set the showScreenSharingButton to false to hide the screen-sharing button.

// ....
zp.joinRoom({
     // ...,
    turnOnCameraWhenJoining: false,
    showMyCameraToggleButton: false,
    showAudioVideoSettingsButton: false,
    showScreenSharingButton: false; 
     // ...
});

Run a Demo

Using the Web Server for Chrome plugin or any web server, start the html created in the previous step.

Conclusion

We hope our article helped you understand the major points associated with a WebRTC video calling app. Wait no more! Use ZEGOCLOUD to start your video-calling app with the WebRTC development process today. We wish you all success!

Read More:

• How to Build iOS Apps using WebRTC
• How to Build Android Apps using WebRTC

FAQ

Q1: What are the advantages of using WebRTC for video calls?

WebRTC offers several advantages for video calls, including ultra-low latency, high-quality audio and video transmission, and built-in encryption for secure communication. It’s also supported natively in most modern browsers, eliminating the need for plugins. WebRTC’s peer-to-peer architecture reduces the need for central servers, making it efficient and scalable for applications like video conferencing, live streaming, and interactive webinars.

Q2: Can WebRTC video calls handle group conferences?

Yes, WebRTC can support group video conferences, but the peer-to-peer nature of WebRTC means that each participant must maintain connections with all other participants, which can become bandwidth-intensive for larger groups. To handle group video calls more efficiently, many platforms use a server-based architecture, such as an SFU (Selective Forwarding Unit) or MCU (Multipoint Control Unit), to manage media streams and reduce the bandwidth load on individual participants.

Q3: What factors can affect the quality of a WebRTC video call?

Several factors can affect the quality of a WebRTC video call, including the user’s network conditions, device performance, and bandwidth availability. Poor network conditions can cause issues like video freezing, low resolution, and lag. WebRTC handles these issues using adaptive bitrate streaming, which adjusts the quality of the video in real time to match the available network bandwidth, ensuring the best possible experience for users.