vue+springboot+webtrc+websocket实现双人音视频通话会议(最新推荐)
作者:相与还
这篇文章主要介绍了vue+springboot+webtrc+websocket实现双人音视频通话会议,本文给大家介绍的非常详细,对大家的学习或工作具有一定的参考借鉴价值,需要的朋友参考下吧
前言
最近一些时间我有研究,如何实现一个视频会议功能,但是找了好多资料都不太理想,最终参考了一个文章
WebRTC实现双端音视频聊天(Vue3 + SpringBoot)
只不过,它的实现效果里面只会播放本地的mp4视频文件,但是按照它的原理是可以正常的实现音视频通话的
它的最终效果是这样的
然后我的实现逻辑在它的基础上进行了优化
实现了如下效果,如下是我部署项目到服务器
之后,和朋友验证之后的截图
针对它的逻辑,我优化了如下几点
第一个人可以输入房间号创建房间,需要注意的是,当前第二个人还没加入进来的时候,视频两边都不展示
第二个人根据第一个人的房间号输入进行加入房间,等待视频流的加载就可以互相看到两边的视频和听到音频
添加了关闭/开启麦克风和摄像头功能
ps: 需要注意的是,我接下来分享的代码逻辑,如果某个人突然加入别的房间,原房间它视频分享还是在的,我没有额外进行处理关闭原房间的音视频流,大家可根据这个进行调整
题外话,根据如上的原理,你可以进一步优化,将其开发一个视频会议功能,当前我有开发一个类似的,但是本次只分享双人音视频通话会议项目
VUE逻辑
如下为前端部分逻辑,需要注意的是,本次项目还是沿用参考文章的
VUE3
项目
前端项目结构如下:
package.json
{ "name": "webrtc_test", "private": true, "version": "0.0.0", "type": "module", "scripts": { "dev": "vite", "build": "vite build", "preview": "vite preview" }, "dependencies": { "axios": "^1.7.7", "vue": "^3.5.12" }, "devDependencies": { "@vitejs/plugin-vue": "^5.1.4", "vite": "^5.4.10" } }
换言之,你需要使用npm安装如上依赖
npm i axios@1.7.7
vite.config.js
import { defineConfig } from 'vite' import vue from '@vitejs/plugin-vue' import fs from 'fs'; // https://vite.dev/config/ export default defineConfig({ plugins: [vue()], server: { // 如果需要部署服务器,需要申请SSL证书,然后下载证书到指定文件夹 https: { key: fs.readFileSync('src/certs/www.springsso.top.key'), cert: fs.readFileSync('src/certs/www.springsso.top.pem'), } }, })
main.js
import { createApp } from 'vue' import App from './App.vue' createApp(App).mount('#app')
App.vue
<template> <div class="video-chat"> <div v-if="isRoomEmpty"> <p>{{ roomStatusText }}</p> </div> <!-- 视频双端显示 --> <div class="video_box"> <div class="self_video"> <div class="text_tip">我:<span class="userId">{{ userId }}</span></div> <video ref="localVideo" autoplay playsinline></video> </div> <div class="remote_video"> <div class="text_tip">对方:<span class="userId">{{ oppositeUserId }}</span></div> <video ref="remoteVideo" autoplay playsinline></video> </div> </div> <!-- 加入房间按钮 --> <div class="room-controls"> <div class="room-input"> <input v-model="roomId" placeholder="请输入房间号" /> <button @click="createRoom">创建房间</button> <button @click="joinRoomWithId">加入房间</button> </div> <div class="media-controls"> <button @click="toggleAudio"> {{ isAudioEnabled ? '关闭麦克风' : '打开麦克风' }} </button> <button @click="toggleVideo"> {{ isVideoEnabled ? '关闭摄像头' : '打开摄像头' }} </button> </div> </div> <!-- 日志打印 --> <div class="log_box"> <pre> <div v-for="(item, index) of logData" :key="index">{{ item }}</div> </pre> </div> </div> </template>
<script setup> import { ref, onMounted, nextTick } from "vue"; import axios from "axios"; // WebRTC 相关变量 const localVideo = ref(null); const remoteVideo = ref(null); const isRoomEmpty = ref(true); // 判断房间是否为空 let localStream; // 本地流数据 let peerConnection; // RTC连接对象 let signalingSocket; // 信令服务器socket对象 let userId; // 当前用户ID let oppositeUserId; // 对方用户ID let logData = ref(["日志初始化..."]); // 请求根路径,如果需要部署服务器,把对应ip改成自己服务器ip let BaseUrl = "https://localhost:8095/meetingV1s" let wsUrl = "wss://localhost:8095/meetingV1s"; // candidate信息 let candidateInfo = ""; // 发起端标识 let offerFlag = false; // 房间状态文本 let roomStatusText = ref("点击'加入房间'开始音视频聊天"); // STUN 服务器, // const iceServers = [ // { // urls: "stun:stun.l.google.com:19302" // Google 的 STUN 服务器 // }, // { // urls: "stun:自己的公网IP:3478" // 自己的Stun服务器 // }, // { // urls: "turn:自己的公网IP:3478", // 自己的 TURN 服务器 // username: "maohe", // credential: "maohe" // } // ]; // ============< 看这 >================ // 没有搭建STUN和TURN服务器的使用如下ice配置即可 const iceServers = [ { urls: "stun:stun.l.google.com:19302" // Google 的 STUN 服务器 } ]; // 在 script setup 中添加新的变量声明 const roomId = ref(''); // 房间号 const isAudioEnabled = ref(true); // 音频状态 const isVideoEnabled = ref(true); // 视频状态 onMounted(() => { generateRandomId(); }) // 加入房间,开启本地摄像头获取音视频流数据。 function joinRoomHandle() { roomStatusText.value = "等待对方加入房间..." getVideoStream(); } // 获取本地视频 模拟从本地摄像头获取音视频流数据 async function getVideoStream() { try { localStream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true }); localVideo.value.srcObject = localStream; wlog(`获取本地流成功~`) createPeerConnection(); // 创建RTC对象,监听candidate } catch (err) { console.error('获取本地媒体流失败:', err); } } // 初始化 WebSocket 连接 function initWebSocket() { wlog("开始连接websocket") // 连接ws时携带用户ID和房间号 signalingSocket = new WebSocket(`${wsUrl}/rtc?userId=${userId}&roomId=${roomId.value}`); signalingSocket.onopen = () => { wlog('WebSocket 已连接'); }; // 消息处理 signalingSocket.onmessage = (event) => { handleSignalingMessage(event.data); }; }; // 消息处理器 - 解析器 function handleSignalingMessage(message) { wlog("收到ws消息,开始解析...") wlog(message) let parseMsg = JSON.parse(message); wlog(`解析结果:${parseMsg}`); if (parseMsg.type == "join") { joinHandle(parseMsg.data); } else if (parseMsg.type == "offer") { wlog("收到发起端offer,开始解析..."); offerHandle(parseMsg.data); } else if (parseMsg.type == "answer") { wlog("收到接收端的answer,开始解析..."); answerHandle(parseMsg.data); }else if(parseMsg.type == "candidate"){ wlog("收到远端candidate,开始解析..."); candidateHandle(parseMsg.data); } } // 远端Candidate处理器 async function candidateHandle(candidate){ peerConnection.addIceCandidate(new RTCIceCandidate(JSON.parse(candidate))); wlog("+++++++ 本端candidate设置完毕 ++++++++"); } // 接收端的answer处理 async function answerHandle(answer) { wlog("将answer设置为远端信息"); peerConnection.setRemoteDescription(new RTCSessionDescription(JSON.parse(answer))); // 设置远端SDP } // 发起端offer处理器 async function offerHandle(offer) { wlog("将发起端的offer设置为远端媒体信息"); await peerConnection.setRemoteDescription(new RTCSessionDescription(JSON.parse(offer))); wlog("创建Answer 并设置到本地"); let answer = await peerConnection.createAnswer() await peerConnection.setLocalDescription(answer); wlog("发送answer给发起端"); // 构造answer消息发送给对端 let paramObj = { userId: oppositeUserId, type: "answer", data: JSON.stringify(answer) } // 执行发送 const res = await axios.post(`${BaseUrl}/rtcs/sendMessage`, paramObj); } // 加入处理器 function joinHandle(userIds) { // 判断连接的用户个数 if (userIds.length == 1 && userIds[0] == userId) { wlog("标识为发起端,等待对方加入房间...") isRoomEmpty.value = true; // 存在一个连接并且是自身,标识我们是发起端 offerFlag = true; } else if (userIds.length > 1) { // 对方加入了 wlog("对方已连接...") isRoomEmpty.value = false; // 取出对方ID for (let id of userIds) { if (id != userId) { oppositeUserId = id; } } wlog(`对端ID: ${oppositeUserId}`) // 开始交换SDP和Candidate swapVideoInfo() } } // 交换SDP和candidate async function swapVideoInfo() { wlog("开始交换Sdp和Candidate..."); // 检查是否为发起端,如果是创建offer设置到本地,并发送给远端 if (offerFlag) { wlog(`发起端创建offer`) let offer = await peerConnection.createOffer() await peerConnection.setLocalDescription(offer); // 将媒体信息设置到本地 wlog("发启端设置SDP-offer到本地"); // 构造消息ws发送给远端 let paramObj = { userId: oppositeUserId, type: "offer", data: JSON.stringify(offer) }; wlog(`构造offer信息发送给远端:${paramObj}`) // 执行发送 const res = await axios.post(`${BaseUrl}/rtcs/sendMessage`, paramObj); } } // 将candidate信息发送给远端 async function sendCandidate(candidate) { // 构造消息ws发送给远端 let paramObj = { userId: oppositeUserId, type: "candidate", data: JSON.stringify(candidate) }; wlog(`构造candidate信息发送给远端:${paramObj}`); // 执行发送 const res = await axios.post(`${BaseUrl}/rtcs/sendMessage`, paramObj); } // 创建RTC连接对象并监听和获取condidate信息 function createPeerConnection() { wlog("开始创建PC对象...") peerConnection = new RTCPeerConnection(iceServers); wlog("创建PC对象成功") // 创建RTC连接对象后连接websocket initWebSocket(); // 监听网络信息(ICE Candidate) peerConnection.onicecandidate = (event) => { if (event.candidate) { candidateInfo = event.candidate; wlog("candidate信息变化..."); // 将candidate信息发送给远端 setTimeout(()=>{ sendCandidate(event.candidate); }, 150) } }; // 监听远端音视频流 peerConnection.ontrack = (event) => { nextTick(() => { wlog("====> 收到远端数据流 <=====") if (!remoteVideo.value.srcObject) { remoteVideo.value.srcObject = event.streams[0]; remoteVideo.value.play(); // 强制播放 } }); }; // 监听ice连接状态 peerConnection.oniceconnectionstatechange = () => { wlog(`RTC连接状态改变:${peerConnection.iceConnectionState}`); }; // 添加本地音视频流到 PeerConnection localStream.getTracks().forEach(track => { peerConnection.addTrack(track, localStream); }); } // 日志编写 function wlog(text) { logData.value.unshift(text); } // 给用户生成随机ID. function generateRandomId() { userId = Math.random().toString(36).substring(2, 12); // 生成10位的随机ID wlog(`分配到ID:${userId}`) } // 创建房间 async function createRoom() { if (!roomId.value) { alert('请输入房间号'); return; } try { const res = await axios.post(`${BaseUrl}/rtcs/createRoom`, { roomId: roomId.value, userId: userId }); if (res.data.success) { wlog(`创建房间成功:${roomId.value}`); joinRoomHandle(); } } catch (error) { wlog(`创建房间失败:${error}`); } } // 加入指定房间 async function joinRoomWithId() { if (!roomId.value) { alert('请输入房间号'); return; } try { const res = await axios.post(`${BaseUrl}/rtcs/joinRoom`, { roomId: roomId.value, userId: userId }); if (res.data.success) { wlog(`加入房间成功:${roomId.value}`); joinRoomHandle(); } } catch (error) { wlog(`加入房间失败:${error}`); } } // 切换音频 function toggleAudio() { if (localStream) { const audioTrack = localStream.getAudioTracks()[0]; if (audioTrack) { audioTrack.enabled = !audioTrack.enabled; isAudioEnabled.value = audioTrack.enabled; wlog(`麦克风已${audioTrack.enabled ? '打开' : '关闭'}`); } } } // 切换视频 function toggleVideo() { if (localStream) { const videoTrack = localStream.getVideoTracks()[0]; if (videoTrack) { videoTrack.enabled = !videoTrack.enabled; isVideoEnabled.value = videoTrack.enabled; wlog(`摄像头已${videoTrack.enabled ? '打开' : '关闭'}`); } } } </script>
<style scoped> .video-chat { display: flex; flex-direction: column; align-items: center; } video { width: 300px; height: 200px; margin: 10px; } .remote_video { border: solid rgb(30, 40, 226) 1px; margin-left: 20px; } .self_video { border: solid red 1px; } .video_box { display: flex; } .video_box div { border-radius: 10px; } .join_room_btn button { border: none; background-color: rgb(119 178 63); height: 30px; width: 80px; border-radius: 10px; color: white; margin-top: 10px; cursor: pointer; font-size: 13px; } .text_tip { font-size: 13px; color: #484848; padding: 6px; } pre { width: 600px; height: 300px; background-color: #d4d4d4; border-radius: 10px; padding: 10px; overflow-y: auto; } pre div { padding: 4px 0px; font-size: 15px; } .userId{ color: #3669ad; } .video-chat p{ font-weight: 600; color: #b24242; } .room-controls { margin: 20px 0; display: flex; flex-direction: column; gap: 10px; } .room-input { display: flex; gap: 10px; align-items: center; } .room-input input { padding: 5px 10px; border: 1px solid #ccc; border-radius: 5px; } .media-controls { display: flex; gap: 10px; } .room-controls button { border: none; background-color: rgb(119 178 63); height: 30px; padding: 0 15px; border-radius: 5px; color: white; cursor: pointer; font-size: 13px; } .media-controls button { background-color: #3669ad; } </style>
SpringBoot逻辑
如下为后端逻辑,项目结构如下:
pom.xml
<?xml version="1.0" encoding="UTF-8"?> <project xmlns="http://maven.apache.org/POM/4.0.0" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="http://maven.apache.org/POM/4.0.0 https://maven.apache.org/xsd/maven-4.0.0.xsd"> <modelVersion>4.0.0</modelVersion> <parent> <groupId>org.springframework.boot</groupId> <artifactId>spring-boot-starter-parent</artifactId> <version>2.7.9</version> <relativePath/> <!-- lookup parent from repository --> </parent> <groupId>com.mh</groupId> <artifactId>webrtc-backend</artifactId> <version>0.0.1-SNAPSHOT</version> <name>webrtc-backend</name> <description>webrtc-backend</description> <properties> <java.version>1.8</java.version> </properties> <dependencies> <dependency> <groupId>org.springframework.boot</groupId> <artifactId>spring-boot-starter-web</artifactId> </dependency> <dependency> <groupId>org.springframework.boot</groupId> <artifactId>spring-boot-starter-test</artifactId> <scope>test</scope> </dependency> <dependency> <groupId>org.springframework.boot</groupId> <artifactId>spring-boot-starter-websocket</artifactId> </dependency> <dependency> <groupId>org.projectlombok</groupId> <artifactId>lombok</artifactId> <version>1.18.34</version> </dependency> </dependencies> <build> <plugins> <plugin> <groupId>org.springframework.boot</groupId> <artifactId>spring-boot-maven-plugin</artifactId> <version>2.6.2</version> <configuration> <mainClass>com.mh.WebrtcBackendApplication</mainClass> <layout>ZIP</layout> </configuration> <executions> <execution> <goals> <goal>repackage</goal> </goals> </execution> </executions> </plugin> </plugins> </build> </project>
application.yml
server: port: 8095 servlet: context-path: /meetingV1s ssl: #ssl配置 enabled: true # 默认为true #key-alias: alias-key # 别名(可以不进行配置) # 保存SSL证书的秘钥库的路径,如果部署到服务器,必须要开启ssl才能获取到摄像头和麦克风 key-store: classpath:www.springsso.top.jks # ssl证书密码 key-password: gf71v8lf key-store-password: gf71v8lf key-store-type: JKS tomcat: uri-encoding: UTF-8
入口文件
// 这个是自己实际项目位置 package com.mh; import org.springframework.boot.SpringApplication; import org.springframework.boot.autoconfigure.SpringBootApplication; @SpringBootApplication public class WebrtcBackendApplication { public static void main(String[] args) { SpringApplication.run(WebrtcBackendApplication.class, args); } }
WebSocket处理器
package com.mh.common; import com.mh.dto.bo.UserManager; import com.mh.dto.vo.MessageOut; import lombok.RequiredArgsConstructor; import lombok.extern.slf4j.Slf4j; import org.springframework.stereotype.Component; import org.springframework.web.socket.CloseStatus; import org.springframework.web.socket.TextMessage; import org.springframework.web.socket.WebSocketSession; import org.springframework.web.socket.handler.TextWebSocketHandler; import com.fasterxml.jackson.databind.ObjectMapper; import java.net.URI; import java.util.ArrayList; import java.util.Set; /** * Date:2024/11/14 * author:zmh * description: WebSocket处理器 **/ @Component @RequiredArgsConstructor @Slf4j public class RtcWebSocketHandler extends TextWebSocketHandler { // 管理用户的加入和退出... private final UserManager userManager; private final ObjectMapper objectMapper = new ObjectMapper(); // 用户连接成功 @Override public void afterConnectionEstablished(WebSocketSession session) throws Exception { // 获取用户ID和房间ID String userId = getParameterByName(session.getUri(), "userId"); String roomId = getParameterByName(session.getUri(), "roomId"); if (userId != null && roomId != null) { // 保存用户会话 userManager.addUser(userId, session); log.info("用户 {} 连接成功,房间:{}", userId, roomId); // 获取房间中的所有用户 Set<String> roomUsers = userManager.getRoomUsers(roomId); // 通知房间内所有用户(包括新加入的用户) for (String uid : roomUsers) { WebSocketSession userSession = userManager.getUser(uid); if (userSession != null && userSession.isOpen()) { MessageOut messageOut = new MessageOut(); messageOut.setType("join"); messageOut.setData(new ArrayList<>(roomUsers)); String message = objectMapper.writeValueAsString(messageOut); userSession.sendMessage(new TextMessage(message)); log.info("向用户 {} 发送房间更新消息", uid); } } } } // 接收到客户端消息,解析消息内容进行分发 @Override protected void handleTextMessage(WebSocketSession session, TextMessage message) throws Exception { // 转换并分发消息 log.info("收到消息"); } // 处理断开的连接 @Override public void afterConnectionClosed(WebSocketSession session, CloseStatus status) throws Exception { String userId = getParameterByName(session.getUri(), "userId"); String roomId = getParameterByName(session.getUri(), "roomId"); if (userId != null && roomId != null) { // 从房间和会话管理中移除用户 userManager.removeUser(userId); userManager.leaveRoom(roomId, userId); // 获取更新后的房间用户列表 Set<String> remainingUsers = userManager.getRoomUsers(roomId); // 通知房间内的其他用户 for (String uid : remainingUsers) { WebSocketSession userSession = userManager.getUser(uid); if (userSession != null && userSession.isOpen()) { MessageOut messageOut = new MessageOut(); messageOut.setType("join"); messageOut.setData(new ArrayList<>(remainingUsers)); String message = objectMapper.writeValueAsString(messageOut); userSession.sendMessage(new TextMessage(message)); log.info("向用户 {} 发送用户离开更新消息", uid); } } log.info("用户 {} 断开连接,房间:{}", userId, roomId); } } // 辅助方法:从URI中获取参数值 private String getParameterByName(URI uri, String paramName) { String query = uri.getQuery(); if (query != null) { String[] pairs = query.split("&"); for (String pair : pairs) { String[] keyValue = pair.split("="); if (keyValue.length == 2 && keyValue[0].equals(paramName)) { return keyValue[1]; } } } return null; } }
WebSocket配置类
package com.mh.config; import com.mh.common.RtcWebSocketHandler; import lombok.RequiredArgsConstructor; import org.springframework.context.annotation.Configuration; import org.springframework.web.socket.config.annotation.EnableWebSocket; import org.springframework.web.socket.config.annotation.WebSocketConfigurer; import org.springframework.web.socket.config.annotation.WebSocketHandlerRegistry; /** * Date:2024/11/14 * author:zmh * description: WebSocket配置类 **/ @Configuration @EnableWebSocket @RequiredArgsConstructor public class WebSocketConfig implements WebSocketConfigurer { private final RtcWebSocketHandler rtcWebSocketHandler; @Override public void registerWebSocketHandlers(WebSocketHandlerRegistry registry) { registry.addHandler(rtcWebSocketHandler, "/rtc") .setAllowedOrigins("*"); } }
webRtc相关接口
package com.mh.controller; import com.mh.dto.bo.UserManager; import com.mh.dto.vo.MessageReceive; import lombok.RequiredArgsConstructor; import lombok.extern.slf4j.Slf4j; import org.springframework.http.ResponseEntity; import org.springframework.web.bind.annotation.*; import java.util.HashMap; import java.util.Map; /** * Date:2024/11/15 * author:zmh * description: rtc 相关接口 **/ @RestController @Slf4j @CrossOrigin @RequiredArgsConstructor @RequestMapping("/rtcs") public class RtcController { private final UserManager userManager; /** * 给指定用户发送执行类型消息 * @param messageReceive 消息参数接收Vo * @return · */ @PostMapping("/sendMessage") public Boolean sendMessage(@RequestBody MessageReceive messageReceive){ userManager.sendMessage(messageReceive); return true; } @PostMapping("/createRoom") public ResponseEntity<?> createRoom(@RequestBody Map<String, String> params) { String roomId = params.get("roomId"); String userId = params.get("userId"); // 在 UserManager 中实现房间创建逻辑 boolean success = userManager.createRoom(roomId, userId); Map<String, Object> response = new HashMap<>(); response.put("success", success); return ResponseEntity.ok(response); } @PostMapping("/joinRoom") public ResponseEntity<?> joinRoom(@RequestBody Map<String, String> params) { String roomId = params.get("roomId"); String userId = params.get("userId"); // 在 UserManager 中实现加入房间逻辑 boolean success = userManager.joinRoom(roomId, userId); Map<String, Object> response = new HashMap<>(); response.put("success", success); return ResponseEntity.ok(response); } }
用户管理器单例对象
package com.mh.dto.bo; import com.fasterxml.jackson.core.JsonProcessingException; import com.fasterxml.jackson.databind.ObjectMapper; import com.mh.dto.vo.MessageOut; import com.mh.dto.vo.MessageReceive; import java.util.stream.Collectors; import lombok.Data; import lombok.extern.slf4j.Slf4j; import org.springframework.stereotype.Component; import org.springframework.web.socket.TextMessage; import org.springframework.web.socket.WebSocketSession; import java.io.IOException; import java.util.HashMap; import java.util.List; import java.util.Map; import java.util.Set; import java.util.HashSet; import java.util.concurrent.ConcurrentHashMap; /** * Date:2024/11/14 * author:zmh * description: 用户管理器单例对象 **/ @Data @Component @Slf4j public class UserManager { // 管理连接用户信息 private final HashMap<String, WebSocketSession> userMap = new HashMap<>(); // 添加房间管理的Map private final Map<String, Set<String>> roomUsers = new ConcurrentHashMap<>(); // 加入用户 public void addUser(String userId, WebSocketSession session) { userMap.put(userId, session); log.info("用户 {} 加入", userId); } // 移除用户 public void removeUser(String userId) { userMap.remove(userId); log.info("用户 {} 退出", userId); } // 获取用户 public WebSocketSession getUser(String userId) { return userMap.get(userId); } // 获取所有用户ID构造成list返回 public List<String> getAllUserId() { return userMap.keySet().stream().collect(Collectors.toList()); } // 通知用户加入-广播消息 public void sendMessageAllUser() throws IOException { // 获取所有连接用户ID列表 List<String> allUserId = getAllUserId(); for (String userId : userMap.keySet()) { WebSocketSession session = userMap.get(userId); MessageOut messageOut = new MessageOut("join", allUserId); String messageText = new ObjectMapper().writeValueAsString(messageOut); // 广播消息 session.sendMessage(new TextMessage(messageText)); } } /** * 创建房间 * @param roomId 房间ID * @param userId 用户ID * @return 创建结果 */ public boolean createRoom(String roomId, String userId) { if (roomUsers.containsKey(roomId)) { log.warn("房间 {} 已存在", roomId); return false; } Set<String> users = new HashSet<>(); users.add(userId); roomUsers.put(roomId, users); log.info("用户 {} 创建了房间 {}", userId, roomId); return true; } /** * 加入房间 * @param roomId 房间ID * @param userId 用户ID * @return 加入结果 */ public boolean joinRoom(String roomId, String userId) { Set<String> users = roomUsers.computeIfAbsent(roomId, k -> new HashSet<>()); if (users.size() >= 2) { log.warn("房间 {} 已满", roomId); return false; } users.add(userId); log.info("用户 {} 加入房间 {}", userId, roomId); return true; } /** * 离开房间 * @param roomId 房间ID * @param userId 用户ID */ public void leaveRoom(String roomId, String userId) { Set<String> users = roomUsers.get(roomId); if (users != null) { users.remove(userId); if (users.isEmpty()) { roomUsers.remove(roomId); log.info("房间 {} 已清空并删除", roomId); } log.info("用户 {} 离开了房间 {}", userId, roomId); } } /** * 获取房间用户 * @param roomId 房间ID * @return 用户集合 */ public Set<String> getRoomUsers(String roomId) { return roomUsers.getOrDefault(roomId, new HashSet<>()); } // 修改现有的 sendMessage 方法,考虑房间信息 public void sendMessage(MessageReceive messageReceive) { String userId = messageReceive.getUserId(); String type = messageReceive.getType(); String data = messageReceive.getData(); WebSocketSession session = userMap.get(userId); if (session != null && session.isOpen()) { try { MessageOut messageOut = new MessageOut(); messageOut.setType(type); messageOut.setData(data); String message = new ObjectMapper().writeValueAsString(messageOut); session.sendMessage(new TextMessage(message)); log.info("消息发送成功: type={}, to={}", type, userId); } catch (Exception e) { log.error("消息发送失败", e); } } } }
消息输出前端Vo对象
package com.mh.dto.vo; import lombok.AllArgsConstructor; import lombok.Data; import lombok.NoArgsConstructor; /** * Date:2024/11/15 * author:zmh * description: 消息输出前端Vo对象 **/ @Data @AllArgsConstructor @NoArgsConstructor public class MessageOut { /** * 消息类型【join, offer, answer, candidate, leave】 */ private String type; /** * 消息内容 前端stringFiy序列化后字符串 */ private Object data; }
消息接收Vo对象
package com.mh.dto.vo; import lombok.AllArgsConstructor; import lombok.Data; import lombok.NoArgsConstructor; /** * Date:2024/11/15 * author:zmh * description: 消息接收Vo对象 **/ @Data @AllArgsConstructor @NoArgsConstructor public class MessageReceive { /** * 用户ID,用于获取用户Session */ private String userId; /** * 消息类型【join, offer, answer, candidate, leave】 */ private String type; /** * 消息内容 前端stringFiy序列化后字符串 */ private String data; }
结语
如上为vue+springboot+webtrc+websocket实现双人音视频通话会议的全部逻辑,如有遗漏后续会进行补充
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