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使用websocket来 WebRTC 建立连接时的 数据的传递和交换。
WebRTC 建立连接时#xff0c;通常需要按照以下顺序执行一些步骤#xff1a; 1.创建本地 PeerConnection 对象#xff1a;使用 RTCPeerConnection 构造函数创建本地的 PeerConnection 对象#xff0c;该…基本介绍
使用websocket来 WebRTC 建立连接时的 数据的传递和交换。
WebRTC 建立连接时通常需要按照以下顺序执行一些步骤 1.创建本地 PeerConnection 对象使用 RTCPeerConnection 构造函数创建本地的 PeerConnection 对象该对象用于管理 WebRTC 连接。
2.添加本地媒体流通过调用 getUserMedia 方法获取本地的音视频流并将其添加到 PeerConnection 对象中。这样可以将本地的音视频数据发送给远程对等方。
3.创建和设置本地 SDP使用 createOffer 方法创建本地的 Session Description Protocol (SDP)描述本地对等方的音视频设置和网络信息。然后通过调用 setLocalDescription 方法将本地 SDP 设置为本地 PeerConnection 对象的本地描述。
4.发送本地 SDP将本地 SDP 发送给远程对等方可以使用信令服务器或其他通信方式发送。
5.接收远程 SDP从远程对等方接收远程 SDP可以通过信令服务器或其他通信方式接收。
6.设置远程 SDP使用接收到的远程 SDP调用 PeerConnection 对象的 setRemoteDescription 方法将其设置为远程描述。
7.创建和设置本地 ICE 候选项使用 onicecandidate 事件监听 PeerConnection 对象的 ICE 候选项生成在生成候选项后通过信令服务器或其他通信方式将其发送给远程对等方。
8.接收和添加远程 ICE 候选项从远程对等方接收到 ICE 候选项后调用 addIceCandidate 方法将其添加到本地 PeerConnection 对象中。
9.连接建立一旦本地和远程的 SDP 和 ICE 候选项都设置好并添加完毕连接就会建立起来。此时音视频流可以在本地和远程对等方之间进行传输。
windwos局域网直连的环境测试过程的问题
1.http协议下安全性原因导致无法调用摄像头和麦克风 chrome://flags/ 配置安全策略 或者 配置本地的https环境 2. 开启了防火墙webRTC连接失败 windows防火墙-高级设置-入站规则-新建规则-端口 ,udp UDP: 32355-65535 放行 为了方便测试 直接关闭防火墙也行。
效果如下 局域网0延迟 如下demo级别的代码复制运行就能直接测试简单修改后可实现基于webrtc的 共享桌面、视频录制等功能。
html代码
!DOCTYPE
htmlheadmeta charsetUTF-8titleWebRTC WebSocket/titlemeta nameviewport contentwidthdevice-width,initial-scale1.0,user-scalablenostylehtml,body {margin: 0;padding: 0;}#main {position: absolute;width: 370px;height: 550px;}#localVideo {position: absolute;background: #757474;top: 10px;right: 10px;width: 100px;height: 150px;z-index: 2;}#remoteVideo {position: absolute;top: 0px;left: 0px;width: 100%;height: 100%;background: #222;}#buttons {z-index: 3;bottom: 20px;left: 90px;position: absolute;}#toUser {border: 1px solid #ccc;padding: 7px 0px;border-radius: 5px;padding-left: 5px;margin-bottom: 5px;}#toUser:focus {border-color: #66afe9;outline: 0;-webkit-box-shadow: inset 0 1px 1px rgba(0, 0, 0, .075), 0 0 8px rgba(102, 175, 233, .6);box-shadow: inset 0 1px 1px rgba(0, 0, 0, .075), 0 0 8px rgba(102, 175, 233, .6)}#call {width: 70px;height: 35px;background-color: #00BB00;border: none;margin-right: 25px;color: white;border-radius: 5px;}#hangup {width: 70px;height: 35px;background-color: #FF5151;border: none;color: white;border-radius: 5px;}/style
/headbodydiv idmainvideo idremoteVideo playsinline autoplay/videovideo idlocalVideo playsinline autoplay muted/videodiv idbuttonsinput idmyid /input idtoUser placeholder输入在线好友账号 /br /button idcall视频通话/buttonbutton idhangup挂断/button/div/div
/body
!-- 可引可不引 --
!--script th:src{/js/adapter-2021.js}/script--
script typetext/javascript th:inlinejavascriptfunction generateRandomLetters(length) {let result ;const characters abcdefghijklmnopqrstuvwxyz; // 字母表for (let i 0; i length; i) {const randomIndex Math.floor(Math.random() * characters.length);const randomLetter characters[randomIndex];result randomLetter;}return result;}let username generateRandomLetters(2);document.getElementById(myid).value username;let localVideo document.getElementById(localVideo);let remoteVideo document.getElementById(remoteVideo);let websocket null;let peer null;let candidate null;/* WebSocket */function WebSocketInit() {//判断当前浏览器是否支持WebSocketif (WebSocket in window) {websocket new WebSocket(ws://192.168.31.14:8181/webrtc/ username);} else {alert(当前浏览器不支持WebSocket);}//连接发生错误的回调方法websocket.onerror function (e) {alert(WebSocket连接发生错误);};//连接关闭的回调方法websocket.onclose function () {console.error(WebSocket连接关闭);};//连接成功建立的回调方法websocket.onopen function () {console.log(WebSocket连接成功);};//接收到消息的回调方法websocket.onmessage async function (event) {let { type, fromUser, msg, sdp, iceCandidate } JSON.parse(event.data.replace(/\n/g, \\n).replace(/\r/g, \\r));console.log(type, fromUser, msg, sdp, iceCandidate);if (type hangup) {console.log(msg);document.getElementById(hangup).click();return;}if (type call_start) {let msg 0if (confirm(fromUser 发起视频通话确定接听吗) true) {document.getElementById(toUser).value fromUser;WebRTCInit();msg 1}websocket.send(JSON.stringify({type: call_back,toUser: fromUser,fromUser: username,msg: msg}));return;}if (type call_back) {if (msg 1) {console.log(document.getElementById(toUser).value 同意视频通话);//创建本地视频并发送offerlet stream await navigator.mediaDevices.getUserMedia({ video: true, audio: true })localVideo.srcObject stream;console.log(peer);stream.getTracks().forEach(track {peer.addTrack(track, stream);});let offer await peer.createOffer();await peer.setLocalDescription(offer);let newOffer offer.toJSON();newOffer[fromUser] username;newOffer[toUser] document.getElementById(toUser).value;websocket.send(JSON.stringify(newOffer));} else if (msg 0) {alert(document.getElementById(toUser).value 拒绝视频通话);document.getElementById(hangup).click();} else {alert(msg);document.getElementById(hangup).click();}return;}if (type offer) {let stream await navigator.mediaDevices.getUserMedia({ video: true, audio: true });localVideo.srcObject stream;stream.getTracks().forEach(track {peer.addTrack(track, stream);});await peer.setRemoteDescription(new RTCSessionDescription({ type, sdp }));let answer await peer.createAnswer();let newAnswer answer.toJSON();newAnswer[fromUser] username;newAnswer[toUser] document.getElementById(toUser).value;websocket.send(JSON.stringify(newAnswer));await peer.setLocalDescription(answer);return;}if (type answer) {peer.setRemoteDescription(new RTCSessionDescription({ type, sdp }));return;}if (type _ice) {peer.addIceCandidate(iceCandidate);return;}}}/* WebRTC */function WebRTCInit() {peer new RTCPeerConnection();//icepeer.onicecandidate function (e) {if (e.candidate) {websocket.send(JSON.stringify({type: _ice,toUser: document.getElementById(toUser).value,fromUser: username,iceCandidate: e.candidate}));}};//trackpeer.ontrack function (e) {if (e e.streams) {remoteVideo.srcObject e.streams[0];}};}/* 按钮事件 */function ButtonFunInit() {//视频通话document.getElementById(call).onclick function (e) {document.getElementById(toUser).style.visibility hidden;let toUser document.getElementById(toUser).value;if (!toUser) {alert(请先指定好友账号再发起视频通话);return;}if (peer null) {WebRTCInit();}websocket.send(JSON.stringify({type: call_start,fromUser: username,toUser: toUser,}));}//挂断document.getElementById(hangup).onclick function (e) {document.getElementById(toUser).style.visibility unset;if (localVideo.srcObject) {const videoTracks localVideo.srcObject.getVideoTracks();videoTracks.forEach(videoTrack {videoTrack.stop();localVideo.srcObject.removeTrack(videoTrack);});}if (remoteVideo.srcObject) {const videoTracks remoteVideo.srcObject.getVideoTracks();videoTracks.forEach(videoTrack {videoTrack.stop();remoteVideo.srcObject.removeTrack(videoTrack);});//挂断同时通知对方websocket.send(JSON.stringify({type: hangup,fromUser: username,toUser: document.getElementById(toUser).value,}));}if (peer) {peer.ontrack null;peer.onremovetrack null;peer.onremovestream null;peer.onicecandidate null;peer.oniceconnectionstatechange null;peer.onsignalingstatechange null;peer.onicegatheringstatechange null;peer.onnegotiationneeded null;peer.close();peer null;}localVideo.srcObject null;remoteVideo.srcObject null;}}WebSocketInit();ButtonFunInit();/script/htmlwebsocket java代码
package com.coco.boot.config;import com.fasterxml.jackson.databind.DeserializationFeature;
import com.fasterxml.jackson.databind.ObjectMapper;
import jakarta.websocket.*;
import jakarta.websocket.server.PathParam;
import jakarta.websocket.server.ServerEndpoint;
import lombok.extern.slf4j.Slf4j;
import org.springframework.stereotype.Component;import java.text.SimpleDateFormat;
import java.util.HashMap;
import java.util.Map;
import java.util.concurrent.ConcurrentHashMap;/*** WebRTC WebSocket*/
Slf4j
Component
ServerEndpoint(value /webrtc/{username})
public class WebRtcWSServer {/*** 连接集合*/private static final MapString, Session sessionMap new ConcurrentHashMap();/*** 连接建立成功调用的方法*/OnOpenpublic void onOpen(Session session, PathParam(username) String username, PathParam(publicKey) String publicKey) {sessionMap.put(username, session);}/*** 连接关闭调用的方法*/OnClosepublic void onClose(Session session) {for (Map.EntryString, Session entry : sessionMap.entrySet()) {if (entry.getValue() session) {sessionMap.remove(entry.getKey());break;}}}/*** 发生错误时调用*/OnErrorpublic void onError(Session session, Throwable error) {error.printStackTrace();}/*** 服务器接收到客户端消息时调用的方法*/OnMessagepublic void onMessage(String message, Session session) {try{//jacksonObjectMapper mapper new ObjectMapper();mapper.setDateFormat(new SimpleDateFormat(yyyy-MM-dd HH:mm:ss));mapper.configure(DeserializationFeature.FAIL_ON_UNKNOWN_PROPERTIES, false);//JSON字符串转 HashMapHashMap hashMap mapper.readValue(message, HashMap.class);//消息类型String type (String) hashMap.get(type);//to userString toUser (String) hashMap.get(toUser);Session toUserSession sessionMap.get(toUser);String fromUser (String) hashMap.get(fromUser);//msgString msg (String) hashMap.get(msg);//sdpString sdp (String) hashMap.get(sdp);//iceMap iceCandidate (Map) hashMap.get(iceCandidate);HashMapString, Object map new HashMap();map.put(type,type);//呼叫的用户不在线if(toUserSession null){toUserSession session;map.put(type,call_back);map.put(fromUser,系统消息);map.put(msg,Sorry呼叫的用户不在线);send(toUserSession,mapper.writeValueAsString(map));return;}//对方挂断if (hangup.equals(type)) {map.put(fromUser,fromUser);map.put(msg,对方挂断);}//视频通话请求if (call_start.equals(type)) {map.put(fromUser,fromUser);map.put(msg,1);}//视频通话请求回应if (call_back.equals(type)) {map.put(fromUser,toUser);map.put(msg,msg);}//offerif (offer.equals(type)) {map.put(fromUser,toUser);map.put(sdp,sdp);}//answerif (answer.equals(type)) {map.put(fromUser,toUser);map.put(sdp,sdp);}//iceif (_ice.equals(type)) {map.put(fromUser,toUser);map.put(iceCandidate,iceCandidate);}send(toUserSession,mapper.writeValueAsString(map));}catch(Exception e){e.printStackTrace();}}/*** 封装一个send方法发送消息到前端*/private void send(Session session, String message) {try {System.out.println(message);session.getBasicRemote().sendText(message);} catch (Exception e) {e.printStackTrace();}}
}springboot 相关依赖和配置
// 1.pom
dependencygroupIdorg.springframework.boot/groupIdartifactIdspring-boot-starter-websocket/artifactId
/dependency// 2.启用功能
EnableWebSocket
public class CocoBootApplication3.config
package com.coco.boot.config;
import org.springframework.context.annotation.Bean;
import org.springframework.context.annotation.Configuration;
import org.springframework.web.socket.server.standard.ServerEndpointExporter;Configuration
public class WebSocketConfig {Beanpublic ServerEndpointExporter serverEndpointExporter() {return new ServerEndpointExporter();}
}
主要参考: WebRTC WebSocket 实现视频通话 WebRTC穿透服务器防火墙配置问题 WebRT音视频录制
