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

1. 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
2. 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
3. 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IDlet 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IDfor (let id of userIds) {if (id != userId) {oppositeUserId = id;}}wlog(`对端ID: ${oppositeUserId}`)// 开始交换SDP和CandidateswapVideoInfo()}
}// 交换SDP和candidate
async function swapVideoInfo() {wlog("开始交换Sdp和Candidate...");// 检查是否为发起端，如果是创建offer设置到本地，并发送给远端if (offerFlag) {wlog(`发起端创建offer`)let offer = await peerConnection.createOffer()await peerConnection.setLocalDescription(offer); // 将媒体信息设置到本地wlog("发启端设置SDP-offer到本地");// 构造消息ws发送给远端let paramObj = {userId: oppositeUserId,type: "offer",data: JSON.stringify(offer)};wlog(`构造offer信息发送给远端：${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(`${BaseUrl}/rtcs/sendMessage`, paramObj);}// 创建RTC连接对象并监听和获取condidate信息
function createPeerConnection() {wlog("开始创建PC对象...")peerConnection = new RTCPeerConnection(iceServers);wlog("创建PC对象成功")// 创建RTC连接对象后连接websocketinitWebSocket();// 监听网络信息（ICE Candidate）peerConnection.onicecandidate = (event) => {if (event.candidate) {candidateInfo = event.candidate;wlog("candidate信息变化...");// 将candidate信息发送给远端setTimeout(()=>{sendCandidate(event.candidate);}, 150)}};// 监听远端音视频流peerConnection.ontrack = (event) => {nextTick(() => {wlog("====> 收到远端数据流 <=====")if (!remoteVideo.value.srcObject) {remoteVideo.value.srcObject = event.streams[0];remoteVideo.value.play();  // 强制播放}});};// 监听ice连接状态peerConnection.oniceconnectionstatechange = () => {wlog(`RTC连接状态改变：${peerConnection.iceConnectionState}`);};// 添加本地音视频流到 PeerConnectionlocalStream.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IDwlog(`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: 8095servlet:context-path: /meetingV1sssl: #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: gf71v8lfkey-store-password: gf71v8lfkey-store-type: JKStomcat: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@Overridepublic void afterConnectionEstablished(WebSocketSession session) throws Exception {// 获取用户ID和房间IDString 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@Overrideprotected 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@Overridepublic 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;@Overridepublic 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Mapprivate 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