一小时教你用SpringBoot+WebSocket+WebRTC实现视频通话
一小时教你用SpringBoot+WebSocket+WebRTC实现视频通话
- 1. 运行结果
- 2. 实现
- 2.1 后端实现
- 2.2 前端页面实现
- 3. 总结
1. 运行结果
SpringBoot+WebSocket+WebRTC实现视频通话
(图片来源网络,侵删)上述运行结果中是有声音(比较小而已)及动的画面的(画面不是静止的)。
网上关于webrtc的文档(文章)和视频也挺多的,但是用SpringBoot结合WebRTC的却屈指可数,前一段时间小编我学习了一下WebRTC的相关知识,于是用SpringBoot+WebRTC实现了一个多人的线上自习室(有画面,但是没有声音的那种,开启声音也挺简单,在js代码里设置一下即可[运行结果在最后的总结里])。最近CSDN有活动,正好把前一段时间学习的知识运用起来(下述代码只是实现了,但是其中的逻辑是存在一定问题的,所以如果读者用下述代码,切记需要改动改动哈!)。既然是WebRTC,为什么又和WebSocket扯上关系了呢?因为利用WebSocket技术来发送消息具有实时性,你看我在这端发送一个消息出去,只要另一端处于连接状态,那么就可以接收到这个消息。而如果使用的是http、https等的话,这一端你发送一个消息,另外一段需要刷新一下页面才能看到消息(当然可以搞个定时器)。结合WebSocket技术,能很快速地实现一个视频通话功能。
2. 实现
导入相关jar包的依赖,如下:
4.0.0 org.springframework.boot spring-boot-starter-parent 2.7.10 com.example demo 0.0.1-SNAPSHOT demo Demo project for Spring Boot 1.8 org.springframework.boot spring-boot-starter-web org.springframework.boot spring-boot-starter-test test org.springframework.boot spring-boot-devtools runtime true org.springframework.boot spring-boot-starter-thymeleaf org.springframework.boot spring-boot-starter-websocket org.projectlombok lombok org.springframework.boot spring-boot-maven-plugin上述jar包可能有一些不需要的喔!
2.1 后端实现
websocket 配置类
GetHttpSessionConfig.class
package com.example.demo.websocket2; import javax.servlet.http.HttpSession; import javax.websocket.HandshakeResponse; import javax.websocket.server.HandshakeRequest; import javax.websocket.server.ServerEndpointConfig; public class GetHttpSessionConfig extends ServerEndpointConfig.Configurator { @Override public void modifyHandshake(ServerEndpointConfig sec, HandshakeRequest request, HandshakeResponse response) { HttpSession httpSession = (HttpSession) request.getHttpSession(); // 获取httpsession对象 sec.getUserProperties().put(HttpSession.class.getName(), httpSession); } }ServerEndpointExporter Bean的定义 Config.class
package com.example.demo.websocket2; import org.springframework.context.annotation.Bean; import org.springframework.context.annotation.Configuration; import org.springframework.web.socket.server.standard.ServerEndpointExporter; @Configuration public class Config { @Bean public ServerEndpointExporter serverEndpointExporter() { return new ServerEndpointExporter(); } }*websocket服务器类 WebSocketServer *
package com.example.demo.websocket2; import org.springframework.stereotype.Component; import javax.servlet.http.HttpSession; import javax.websocket.*; import javax.websocket.server.ServerEndpoint; import java.io.IOException; import java.util.Map; import java.util.Set; import java.util.concurrent.ConcurrentHashMap; @Component @ServerEndpoint(value = "/video",configurator = GetHttpSessionConfig.class) public class WebSocketServer { //存储客户端的连接对象,每个客户端连接都会产生一个连接对象 private static ConcurrentHashMap map = new ConcurrentHashMap(); //每个连接都会有自己的会话 private Session session; private String account; @OnOpen public void open(Session session,EndpointConfig config){ HttpSession httpSession = (HttpSession) config.getUserProperties().get(HttpSession.class.getName()); String account = String.valueOf(httpSession.getAttribute("account")); map.put(account,this); this.session = session; this.account = account; } @OnClose public void close(){ map.remove(account); } @OnError public void error(Throwable error){ error.printStackTrace(); } @OnMessage public void getMessage(String message) throws IOException { Set entries = map.entrySet(); for (Map.Entry entry : entries) { if(!entry.getKey().equals(account)){//将消息转发到其他非自身客户端 entry.getValue().send(message); } } } public void send(String message) throws IOException { if(session.isOpen()){ session.getBasicRemote().sendText(message); } } public int getConnetNum(){ return map.size(); } }2.2 前端页面实现
登录界面的代码就不在这儿粘贴了,下面主要展示视频通话界面的代码(包括css样式和js代码都在的)
main body { background: #eee; padding: 5% 0; } video { background: black; border: 1px solid gray; } .call-page { position: relative; display: block; margin: 0 auto; width: 500px; height: 500px; } #localVideo { width: 150px; height: 150px; position: absolute; top: 15px; right: 15px; } #remoteVideo { width: 500px; height: 500px; }//our username var connectedUser; //connecting to our signaling server var conn = new WebSocket("ws://localhost:9999/video"); conn.onopen = function () { console.log("Connected to the signaling server"); }; //when we got a message from a signaling server conn.onmessage = function (msg) { console.log("Got message", msg.data); var data = JSON.parse(msg.data); switch(data.type) { case "login": handleLogin(data.success); break; //when somebody wants to call us case "offer": handleOffer(data.offer, data.name); break; case "answer": handleAnswer(data.answer); break; //when a remote peer sends an ice candidate to us case "candidate": handleCandidate(data.candidate); break; case "leave": handleLeave(); break; default: break; } }; conn.onerror = function (err) { console.log("Got error", err); }; //alias for sending JSON encoded messages function send(message) { //attach the other peer username to our messages if (connectedUser) { message.name = connectedUser; } conn.send(JSON.stringify(message)); } //****** //UI selectors block //****** var callPage = document.querySelector("#callPage"); var callToUsernameInput = document.querySelector("#callToUsernameInput"); var callBtn = document.querySelector("#callBtn"); var hangUpBtn = document.querySelector("#hangUpBtn"); var localVideo = document.querySelector("#localVideo"); var remoteVideo = document.querySelector("#remoteVideo"); var yourConn; var stream; // callPage.style.display = "none"; var PeerConnection = (window.webkitRTCPeerConnection || window.mozRTCPeerConnection || window.RTCPeerConnection || undefined); var RTCSessionDescription = (window.webkitRTCSessionDescription || window.mozRTCSessionDescription || window.RTCSessionDescription || undefined); navigator.getUserMedia = (navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia); //********************** //Starting a peer connection //********************** //getting local video stream navigator.getUserMedia({ video: true, audio: true }, function (myStream) { stream = myStream; //displaying local video stream on the page localVideo.srcObject = stream; //using Google public stun server var configuration = { "iceServers": [] }; yourConn = new PeerConnection(configuration); // setup stream listening yourConn.addStream(stream); //when a remote user adds stream to the peer connection, we display it yourConn.onaddstream = function (e) { remoteVideo.srcObject = e.stream; }; // Setup ice handling yourConn.onicecandidate = function (event) { if (event.candidate) { send({ type: "candidate", candidate: event.candidate }); } }; }, function (error) { console.log(error); }); //initiating a call callBtn.addEventListener("click", function () { var callToUsername = callToUsernameInput.value; if (callToUsername.length > 0) { connectedUser = callToUsername; // create an offer yourConn.createOffer(function (offer) { send({ type: "offer", offer: offer }); yourConn.setLocalDescription(offer); }, function (error) { alert("Error when creating an offer"); }); } }); //when somebody sends us an offer function handleOffer(offer, name) { connectedUser = name; yourConn.setRemoteDescription(new RTCSessionDescription(offer)); //create an answer to an offer yourConn.createAnswer(function (answer) { yourConn.setLocalDescription(answer); send({ type: "answer", answer: answer }); }, function (error) { alert("Error when creating an answer"); }); } //when we got an answer from a remote user function handleAnswer(answer) { yourConn.setRemoteDescription(new RTCSessionDescription(answer)); } //when we got an ice candidate from a remote user function handleCandidate(candidate) { yourConn.addIceCandidate(new RTCIceCandidate(candidate)); } //hang up hangUpBtn.addEventListener("click", function () { send({ type: "leave" }); handleLeave(); }); function handleLeave() { connectedUser = null; remoteVideo.src = null; yourConn.close(); yourConn.onicecandidate = null; yourConn.onaddstream = null; }Call Hang Up3. 总结
上述前端代码参考来自这里:webrtc视频演示,上述代码中如果有不懂的读者可以去仔细看看这个链接里的知识,里面关于webrtc有详细的介绍及实现,不过,没有讲多人的,它只讲了一对一的,不过,前一段时间小编在参考一些大佬的实现思路及自己思考下,也实现了一个多人的,运行结果如下:
基于SpringBoot,WebSocket,WebRTC实现多人自习室功能
