How to use Java Websocket to implement online audio and video calls?
How to use Java Websocket to implement online audio and video calls?
In today’s digital age, real-time communication is becoming more and more common. Whether it is remote collaboration at work or remote communication with relatives and friends at home, real-time audio and video calls have become an indispensable part of people. This article will introduce how to use Java Websocket to implement online audio and video calls, and provide specific code examples.
1. Understand Websocket
Websocket is a new protocol in HTML5, which provides full-duplex communication capabilities between the browser and the server. Compared with traditional HTTP requests, Websocket has lower latency and higher efficiency, and is suitable for real-time communication scenarios.
2. Build a Websocket server
First, we need to build a Websocket server to handle audio and video call requests. You can choose to use the WebSocket API in Java EE, or use third-party libraries such as Jetty. The following is a sample code for setting up a server using the WebSocket API in Java EE:
import javax.websocket.*;
import javax.websocket.server.ServerEndpoint;
import java.io.IOException;
import java.util.HashSet;
import java.util.Set;
@ServerEndpoint("/video-call")
public class VideoCallServer {
private static Set<Session> sessions = new HashSet<>();
@OnOpen
public void onOpen(Session session) {
sessions.add(session);
}
@OnMessage
public void onMessage(String message, Session session) throws IOException {
for (Session s : sessions) {
if (!s.equals(session)) {
s.getBasicRemote().sendText(message);
}
}
}
@OnClose
public void onClose(Session session) {
sessions.remove(session);
}
}The above code creates a WebSocket server endpoint named /video-call and implements ## The #onOpen, onMessage, and onClose methods handle the logic of connections, message sending and receiving, and connection closing.
<!DOCTYPE html>
<html>
<head>
<title>视频通话</title>
</head>
<body>
<video id="local-video" autoplay></video>
<video id="remote-video" autoplay></video>
<script>
var localVideo = document.getElementById("local-video");
var remoteVideo = document.getElementById("remote-video");
// 获取本地媒体流，即摄像头和麦克风的输入
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
navigator.getUserMedia({ video: true, audio: true }, function(localStream) {
// 在本地视频元素上显示本地媒体流
localVideo.srcObject = localStream;
var serverUrl = "ws://localhost:8080/video-call";
var websocket = new WebSocket(serverUrl);
websocket.onopen = function(event) {
// 将本地媒体流发送到服务器
websocket.send(JSON.stringify({ type: "stream", stream: localStream }));
};
websocket.onmessage = function(event) {
var data = JSON.parse(event.data);
if (data.type === "stream") {
// 在远程视频元素上显示远程媒体流
remoteVideo.srcObject = data.stream;
}
};
}, function(error) {
console.log("获取本地媒体流失败：" + error);
});
</script>
</body>
</html>The above code uses the getUserMedia method to obtain input from the local camera and microphone and send the local media stream to the server through Websocket. At the same time, receive the remote media stream sent by the server and display it on the corresponding
http://localhost:8080/video-call to make online audio and video calls.
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