Home Backend Development PHP Tutorial How to use WebSockets API in PHP for live video and audio chat

How to use WebSockets API in PHP for live video and audio chat

Jun 17, 2023 pm 02:37 PM
php websockets real-time interaction

WebSockets API is an important part of realizing real-time video and audio chat. It provides a communication method based on event-driven mechanism, which can realize two-way communication, making the communication between browser and server simpler, faster and more secure. . This article will introduce how to use WebSockets API in PHP for real-time video and audio chat.

  1. Install WebSocket server

To use the WebSockets API in PHP, you first need to install the WebSocket server. It is recommended to use Rachet, which is the most popular WebSocket server in PHP. It can be installed using Composer:

composer require cboden/ratchet
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  1. Creating a WebSocket server

Creating a WebSocket server using Rachet is very simple and only requires a few lines of code:

require dirname(__DIR__) . '/vendor/autoload.php';

use RatchetMessageComponentInterface;
use RatchetConnectionInterface;
use RatchetServerIoServer;
use RatchetHttpHttpServer;
use RatchetWebSocketWsServer;

class Chat implements MessageComponentInterface {
    protected $clients;

    public function __construct() {
        $this->clients = new SplObjectStorage;
    }

    public function onOpen(ConnectionInterface $conn) {
        $this->clients->attach($conn);
        echo "New connection! ({$conn->resourceId})
";
    }

    public function onMessage(ConnectionInterface $from, $msg) {
        broadcast($msg);
    }

    public function onClose(ConnectionInterface $conn) {
        $this->clients->detach($conn);
        echo "Connection {$conn->resourceId} has disconnected
";
    }

    public function onError(ConnectionInterface $conn, Exception $e) {
        echo "An error has occurred: {$e->getMessage()}
";
        $conn->close();
    }

    public function broadcast($msg) {
        foreach ($this->clients as $client) {
            $client->send($msg);
        }
    }
}

$server = IoServer::factory(
    new HttpServer(
        new WsServer(
            new Chat()
        )
    ),
    8080
);

echo "Server started!
";

$server->run();
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This server accepts all connections and broadcasts messages as they arrive. It uses Grad stream objects to store all connections.

  1. Create front-end applications

Using the WebSockets API, creating front-end applications can be done cross-platform and cross-browser. Here's how to create a front-end application using JavaScript:

<!doctype html>

<html lang="en">
<head>
    <meta charset="utf-8">

    <title>Chat</title>
    <meta name="description" content="Chat">
    <meta name="author" content="Chat">

    <script src="//code.jquery.com/jquery-1.11.0.min.js"></script>
    <script src="//cdnjs.cloudflare.com/ajax/libs/socket.io/1.4.5/socket.io.min.js"></script>

    <style>
        #log {
            height: 200px;
            overflow: auto;
        }
    </style>
</head>

<body>
    <div id="log"></div>
    <input type="text" id="message">
    <button id="send">Send</button>
    <script>
        var socket = io('http://localhost:8080');

        socket.on('message', function (data) {
            $('#log').append('<p>' + data + '</p>');
        });

        $('#send').click(function() {
            var message = $('#message').val();
            socket.emit('message', message);
        });
    </script>
</body>
</html>
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Using Socket.io to connect, the application receives and sends messages.

  1. Implement video and audio chat

Through Ratchet and WebSockets API, two-way real-time video and audio chat can be achieved, which requires the use of some additional libraries and tools. It is recommended to use WebRTC, a modern standard for real-time communication that allows peer-to-peer communication between browsers.

Using a WebSocket server and a WebRTC DSP in PHP, you can create a two-way real-time video and audio chat application. This requires some additional JavaScript code, which can be implemented using SimpleWebRTC:

<!doctype html>

<html lang="en">
<head>
    <meta charset="utf-8">

    <title>Chat</title>
    <meta name="description" content="Chat">
    <meta name="author" content="Chat">

    <script src="//code.jquery.com/jquery-1.11.0.min.js"></script>
    <script src="//cdnjs.cloudflare.com/ajax/libs/socket.io/1.4.5/socket.io.min.js"></script>
    <script src="//simplewebrtc.com/latest-v3.js"></script>

    <style>
        #log {
            height: 200px;
            overflow: auto;
        }

        #localVideo {
            width: 200px;
            height: 150px;
            margin-bottom: 10px;
        }

        #remoteVideos video {
            width: 400px;
            height: 300px;
            margin-bottom: 10px;
        }
    </style>
</head>

<body>
    <div id="log"></div>
    <div id="localVideo"></div>
    <div id="remoteVideos"></div>
    <script>
        var server = {
            url: 'http://localhost:8080',
            options: {},
            // Use media servers for production
            signalingServerUrl: 'https://localhost:8888'
        };

        var webrtc = new SimpleWebRTC({
            localVideoEl: 'localVideo',
            remoteVideosEl: 'remoteVideos',
            autoRequestMedia: true,
            url: server.url,
            socketio: {'force new connection': true},
            debug: false,
            detectSpeakingEvents: true,
            autoAdjustMic: false,
            peerConnectionConfig: {
                iceServers: [
                    {url: 'stun:stun.l.google.com:19302'},
                    {url:'stun:stun1.l.google.com:19302'},
                    {url:'stun:stun2.l.google.com:19302'}
                ]
            },
            receiveMedia: {
                mandatory: {
                    OfferToReceiveAudio: true,
                    OfferToReceiveVideo: true
                }
            }
        });

        webrtc.on('readyToCall', function () {
            console.log('readyToCall');
            webrtc.joinRoom('chat');
        });

        webrtc.on('localStream', function (stream) {
            console.log('localStream');
            $('#localVideo').show();
        });

        webrtc.on('videoAdded', function (video, peer) {
            console.log('videoAdded');
            $('#remoteVideos').append('<video id="' + peer.id + '" autoplay></video>');
            webrtc.attachMediaStream($('#' + peer.id), video);
        });

        webrtc.on('videoRemoved', function (video, peer) {
            console.log('videoRemoved');
            $('#' + peer.id).remove();
        });

        webrtc.on('channelMessage', function (peer, label, message) {
            console.log('channelMessage');
            console.log('peer: ' + peer);
            console.log('label: ' + label);
            console.log('message: ' + message);
        });
    </script>
</body>
</html>
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SimpleWebRTC is used here to implement video and audio chat. The code contains client and server code, when a user visits the page, the client tries to connect to the WebSocket server and join the room. The server delivers WebSocket events to SimpleWebRTC.

Summary

Using Rachet and the WebSockets API, two-way real-time video and audio chat can be achieved. Applications can be easily extended to support real-time audio and video chat using SimpleWebRTC. WebRTC is a powerful technology that can be used in a variety of applications, including online education systems, collaboration applications, and online games. Using WebSockets API and WebRTC in PHP, you can create powerful real-time applications.

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