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With the continuous development of the Internet, real-time video and audio communication have increasingly become an important way for people to communicate, and it has penetrated into every aspect of our daily life and work. PHP and WebRTC are very useful tools when implementing real-time video and audio communication. So, how to use PHP and WebRTC for real-time video and audio communication? Next, we will give detailed operation instructions step by step.
First of all, we need to understand some basic knowledge of WebRTC. WebRTC is a real-time communication protocol that enables video and audio communication on the browser. It uses JavaScript API, which can be used without installing any plug-ins. Communicate with the browser. However, WebRTC is not an independent technology and needs to work together with other technologies to achieve real-time video and audio communication. Then, before using PHP and WebRTC to achieve real-time video and audio communication, we need to master the following technical foundation:
Next, we will introduce the specific steps to use PHP and WebRTC to achieve real-time video and audio communication:
Using WebRTC API to create audio and video streams is the first step to achieve real-time video and audio communication. Through the WebRTC API, a local audio and video stream can be created and sent to a remote device. We can use the following code to create local audio and video streams:
<script> const getUserMedia = (navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia); const constraints = { audio: true, video: { width: 1280, height: 720 } }; getUserMedia(constraints, (stream) => { const video = document.querySelector('video'); video.srcObject = stream; video.onloadedmetadata = (e) => { video.play(); }; }, (err) => { console.log(err); }); </script>
In the above code, we use the getUserMedia ()
method to obtain the local audio and video streams and assign them to video
tag.
WebSocket is a two-way data transmission protocol based on the TCP protocol. Using WebSocket we can establish a long-lasting connection between client and server. We can use the following code to establish a WebSocket connection:
<script> var ws = new WebSocket('ws://yourserveraddress:port/'); ws.onopen = function() { console.log('WebSocket已打开'); }; ws.onmessage = function(e) { console.log('收到消息', e.data); }; ws.onclose = function() { console.log('WebSocket已关闭'); }; ws.onerror = function(e) { console.log('WebSocket发生错误:', e); }; </script>
In the above code, we use the WebSocket constructor to create a WebSocket object, specify the connected server address and port, and use the onopen, onmessage, onclose, and onerror methods respectively. Handles open, receive message, close, and error events. When WebSocket is opened, we can send and receive messages to the server through WebSocket.
WebSocket can use binary data transmission, and we can pass local audio and video streams to remote devices through WebSocket. We can use the following code to send audio and video streams:
<script> var ws = new WebSocket('ws://yourserveraddress:port/'); ... function sendData() { var video = document.getElementById('video'); var canvas = document.createElement('canvas'); canvas.width = video.videoWidth; canvas.height = video.videoHeight; canvas.getContext('2d').drawImage(video, 0, 0, canvas.width, canvas.height); canvas.toBlob(function(blob) { ws.send(blob); }); } </script>
In the above code, we convert the video stream into a canvas canvas, use the toBlob()
function to convert it into a Blob object, and then pass WebSocket sends it to the remote device.
We can use the onmessage() function to receive audio and video streams from remote devices. We can use the following code to receive audio and video streams:
<script> var video2 = document.getElementById('video2'); function startVideo(stream) { video2.srcObject = stream; video2.onloadedmetadata = function(e) { video2.play(); }; } ... ws.onmessage = function(e) { var blob = e.data; var videoUrl = window.URL.createObjectURL(blob); video2.src = videoUrl; }; </script>
In the above code, we use the URL.createObjectURL()
function to convert the browser's Blob object into a URL address, and then Assign video2
tag.
To sum up, it is not difficult to use PHP and WebRTC to implement real-time video and audio communication. We only need to master some basic knowledge and then follow the above steps step by step. Real-time video and audio communication is a very practical technology that will greatly help our lives and work.
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