A guide to implementing real-time video chat with PHP and WebRTC

In today's information age, people are increasingly dependent on the Internet, and network transmission content is gradually changing from single content such as text, pictures, and audio to more colorful forms such as video and live broadcast. Under such demand, real-time video chat has become a standard feature of many applications, such as social media, online conferencing software, etc. How to implement a stable and efficient real-time video chat system? This article will introduce a guide to implementing real-time video chat using PHP and WebRTC.

1. What is WebRTC

WebRTC (Web Real-Time Communications) is a real-time communication technology. It can realize audio, video, file sharing, screen sharing and other applications directly in the browser. WebRTC is an open source project led by Google.

The advantage of WebRTC is that it is based on and executed within the browser, making real-time audio and video communication more convenient than ever. Moreover, WebRTC supports PCs, mobile devices and IoT devices, enabling real-time communication between various devices.

2. The basic process of using WebRTC to implement real-time video chat

1. Establishing a connection

Using WebRTC to establish a connection requires the use of three technologies:

• Signaling/WebSockets: The basic framework for establishing connections;
• SDP (Session Description Protocol): used to describe session information;
• ICE (Interactive Connectivity Establishment): used to overcome Network obstacles, including NAT, firewalls, etc.

2. Establish a stream

When using WebRTC for real-time video chat, you need to establish a stream for audio and video transmission. When setting up a stream, it is important to ensure that the audio and video are synchronized for optimal results.

3. Media reconnection

If the media stream is now interrupted, SDP needs to be resent to re-establish the media stream.

4. Close the connection

After the WebRTC communication ends, you need to use a JavaScript function to close the connection.

3. The specific process of using PHP and WebRTC to implement real-time video chat

1. Preparation work

In order to implement an efficient and stable WebRTC application, the following preparations are required:

• Server space: can be deployed using cloud servers and other methods;
• SSL certificate: WebRTC must use an encrypted communication stack, therefore, an SSL certificate is required;
• STUN/TURN server: In order to solve the NAT problem, STUN (Simple Traversal of UDP through NATs) and TURN (Traversal Using Relays around NATs) are technologies that WebRTC must use.

2. Real-time video chat using PHP and WebRTC

First, you need to install and configure the PHP environment on the web server. Then use the WebSocket server (Ratchet) provided by PHP to implement WebRTC.

The following is the specific process for WebRTC to implement real-time video chat:

• Establish a WebSocket connection;
• Establish a standard WebRTC connection;
• Send a message Command to allow WebRTC to start communication;
• Send and receive media data in RTCDataChannel;
• Close the WebRTC connection;
• Close the WebSocket connection.

The specific implementation process is as follows:

Establish a WebSocket connection:

Establish a WebSocket connection in the PHP code, the code is as follows:

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

$server->run();

This code will listen for WebSocket connection requests from the browser and create a WebSocket object. The core of the WebSocket class is the onMessage() function, in which the basic configuration of WebRTC communication is performed and the transmission of audio and video data is completed.

Establish a standard WebRTC connection:

Use JavaScript code to establish a WebRTC connection. The code is as follows:

var pcConfig = {
    "iceServers": [
        {
            "urls": "stun:stun.l.google.com:19302"
        },
        {
            "urls": "turn:myusername:mypassword@turn.bigtalk.com:3478",
            "credentialType": "password"
        }
    ]
};

var pc = new RTCPeerConnection(pcConfig);

In the configuration of the WebRTC connection, a STUN/TURN server is required. These servers support rejecting packets from NAT (Network Address Translation) firewalls. If NAT does not allow these IP packets, audio and video data will not be transmitted.

Send signaling to allow WebRTC to start communication:

During the WebRTC communication process, a signaling server (signaling server) must be used to establish a peer-to-peer communication connection. In PHP and WebSocket we can use Ratchet/Hanlebars/PHP as signaling server. The code is as follows:

        case 'signal':
            $to = $jsonData->to;
            unset($jsonData->to);

            $conn = null;
            foreach ($this->clients as $client) {
                if ($client->resourceId === (string)$to) {
                    $conn = $client;
                    break;
                }
            }

            if (!$conn) {
                return;
            }

            $msg = json_encode(array(
                'type' => 'signal',
                'data' => $jsonData,
            ));

            $conn->send($msg);
            break;

In this code, the sending of WebRTC signaling is implemented through broadcast information. This will allow point-to-point communication connections to be established.

Sending and receiving media data in RTCDataChannel:

After establishing a point-to-point communication connection, audio and video data need to be sent and received in RTCDataChannel. The following is the core code to implement this process:

            case 'stream':
                $to = $jsonData->to;
                unset($jsonData->to);

                $conn = null;
                foreach ($this->clients as $client) {
                    if ($client->resourceId === (string)$to) {
                        $conn = $client;
                        break;
                    }
                }

                if (!$conn) {
                    return;
                }

                $msg = json_encode(array(
                    'type' => 'stream',
                    'data' => $jsonData->data,
                ));

                $conn->send($msg);
                break;

In this code, WebRTC's RTCDataChannel object is used to send and receive media data.

Close the WebRTC connection:

After completing the live video chat, you need to close the WebRTC connection. The code is as follows:

        case 'close':
            $to = $jsonData->to;
            unset($jsonData->to);
            $conn = null;
            foreach ($this->clients as $client) {
                if ($client->resourceId === (string)$to) {
                    $conn = $client;
                    break;
                }
            }

            if (!$conn) {
                return;
            }

            $msg = json_encode(array(
                'type' => 'close',
            ));

            $conn->send($msg);
            break;

Close the WebSocket connection:

Once the WebRTC connection is closed, you need to close the PHP Ratchet server on the WebSocket connection. The code is as follows:

$conn->close();

4. Summary

The idea of ​​using PHP and WebRTC to implement real-time video chat is not complicated, but the specific implementation process needs to be understood step by step. The key lies in setting up basic configurations such as WebSocket connections and WebRTC connections, and using a signaling server and RTCDataChannel for the transmission of audio and video data. This article introduces the basic PHP and WebRTC process to implement real-time video chat, hoping to provide readers with a more comprehensive guide based on known technologies.

The above is the detailed content of A guide to implementing real-time video chat with PHP and WebRTC. For more information, please follow other related articles on the PHP Chinese website!