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Spring Boot集成websocket实现webrtc功能

1.什么是webrtc？

WebRTC 是 Web 实时通信（Real-Time Communication）的缩写，它既是 API 也是协议。WebRTC 协议是两个 WebRTC Agent 协商双向安全实时通信的一组规则。开发人员可以通过 WebRTC API 使用 WebRTC 协议。目前 WebRTC API 仅有 JavaScript 版本。 可以用 HTTP 和 Fetch API 之间的关系作为类比。WebRTC 协议就是 HTTP，而 WebRTC API 就是 Fetch API。 除了 JavaScript 语言，WebRTC 协议也可以在其他 API 和语言中使用。你还可以找到 WebRTC 的服务器和特定领域的工具。所有这些实现都使用 WebRTC 协议，以便它们可以彼此交互。 WebRTC 协议由 IETF 工作组在rtcweb中维护。WebRTC API 的 W3C 文档在webrtc。

WebSocket

WebSocket是一种在单个TCP连接上进行全双工通信的协议。WebSocket通信协议于2011年被IETF定为标准RFC 6455，并由RFC7936补充规范。WebSocket API也被W3C定为标准。WebSocket使得客户端和服务器之间的数据交换变得更加简单，允许服务端主动向客户端推送数据。在WebSocket API中，浏览器和服务器只需要完成一次握手，两者之间就直接可以创建持久性的连接，并进行双向数据传输

webrtc架构

2.代码工程

实验目标

实现视频通话功能

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 http://maven.apache.org/xsd/maven-4.0.0.xsd">
    <parent>
        <artifactId>springboot-demo</artifactId>
        <groupId>com.et</groupId>
        <version>1.0-SNAPSHOT</version>
    </parent>
    <modelVersion>4.0.0</modelVersion>

    <artifactId>WebRTC</artifactId>

    <properties>
        <maven.compiler.source>8</maven.compiler.source>
        <maven.compiler.target>8</maven.compiler.target>
    </properties>
    <dependencies>

        <dependency>
            <groupId>org.springframework.boot</groupId>
            <artifactId>spring-boot-starter-web</artifactId>
        </dependency>

        <dependency>
            <groupId>org.springframework.boot</groupId>
            <artifactId>spring-boot-autoconfigure</artifactId>
        </dependency>
        <dependency>
            <groupId>org.springframework.boot</groupId>
            <artifactId>spring-boot-starter-test</artifactId>
            <scope>test</scope>
        </dependency>
        <dependency>
            <groupId>org.projectlombok</groupId>
            <artifactId>lombok</artifactId>
        </dependency>
        <dependency>
            <groupId>org.springframework.boot</groupId>
            <artifactId>spring-boot-starter-websocket</artifactId>
        </dependency>
        <dependency>
            <groupId>org.springframework.boot</groupId>
            <artifactId>spring-boot-starter-thymeleaf</artifactId>
        </dependency>
    </dependencies>
</project>

controller

package com.et.webrtc.controller;

import org.springframework.web.bind.annotation.PathVariable;
import org.springframework.web.bind.annotation.RequestMapping;
import org.springframework.web.bind.annotation.RestController;
import org.springframework.web.servlet.ModelAndView;

import java.util.HashMap;
import java.util.Map;

@RestController
public class HelloWorldController {
    @RequestMapping("/hello")
    public Map<String, Object> showHelloWorld(){
        Map<String, Object> map = new HashMap<>();
        map.put("msg", "HelloWorld");
        return map;
    }
    /**
     * WebRTC + WebSocket
     */
    @RequestMapping("webrtc/{username}.html")
    public ModelAndView socketChartPage(@PathVariable String username) {
        ModelAndView modelAndView = new ModelAndView();
        modelAndView.setViewName("webrtc.html");
        modelAndView.addObject("username",username);
        return modelAndView;
    }
}

config

package com.et.webrtc.config;

import com.fasterxml.jackson.databind.DeserializationFeature;
import com.fasterxml.jackson.databind.ObjectMapper;
import lombok.extern.slf4j.Slf4j;
import org.springframework.stereotype.Component;

import javax.websocket.*;
import javax.websocket.server.PathParam;
import javax.websocket.server.ServerEndpoint;
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 Map<String, Session> sessionMap = new ConcurrentHashMap<>();

    /**
     * 连接建立成功调用的方法
     */
    @OnOpen
    public 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
     */
    @OnClose
    public void onClose(Session session) {
        for (Map.Entry<String, Session> entry : sessionMap.entrySet()) {
            if (entry.getValue() == session) {
                sessionMap.remove(entry.getKey());
                break;
            }
        }
    }

    /**
     * 发生错误时调用
     */
    @OnError
    public 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
     */
    @OnMessage
    public void onMessage(String message, Session session) {
        try{
            //jackson
            ObjectMapper 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 HashMap
            HashMap hashMap = mapper.readValue(message, HashMap.class);

            //消息类型
            String type = (String) hashMap.get("type");

            //to user
            String toUser = (String) hashMap.get("toUser");
            Session toUserSession = sessionMap.get(toUser);
            String fromUser = (String) hashMap.get("fromUser");

            //msg
            String msg = (String) hashMap.get("msg");

            //sdp
            String sdp = (String) hashMap.get("sdp");

            //ice
            Map iceCandidate  = (Map) hashMap.get("iceCandidate");

            HashMap<String, 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);
            }

            //offer
            if ("offer".equals(type)) {
                map.put("fromUser",toUser);
                map.put("sdp",sdp);
            }

            //answer
            if ("answer".equals(type)) {
                map.put("fromUser",toUser);
                map.put("sdp",sdp);
            }

            //ice
            if ("_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();
        }
    }
}

package com.et.webrtc.config;

import org.springframework.context.annotation.Bean;
import org.springframework.context.annotation.Configuration;
import org.springframework.web.socket.config.annotation.EnableWebSocket;
import org.springframework.web.socket.server.standard.ServerEndpointExporter;

@Configuration
@EnableWebSocket
public class WebSocketConfiguration {

    @Bean
    public ServerEndpointExporter serverEndpointExporter() {
        return new ServerEndpointExporter();
    }
}

前端页面

<!DOCTYPE>
<!--解决idea thymeleaf 表达式模板报红波浪线-->
<!--suppress ALL -->
<html xmlns:th="http://www.thymeleaf.org">
<head>
    <meta charset="UTF-8">
    <title>WebRTC + WebSocket</title>
    <meta name="viewport" content="width=device-width,initial-scale=1.0,user-scalable=no">
    <style>
        html,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>
</head>
<body>
<div id="main">
    <video id="remoteVideo" playsinline autoplay></video>
    <video id="localVideo" playsinline autoplay muted></video>

    <div id="buttons">
        <input id="toUser" placeholder="输入在线好友账号"/><br/>
        <button id="call">视频通话</button>
        <button id="hangup">挂断</button>
    </div>
</div>
</body>
<!-- 可引可不引 -->
<!--<script th:src="@{/js/adapter-2021.js}"></script>-->
<script type="text/javascript" th:inline="javascript">
    let username = /*[[${username}]]*/'';
    let localVideo = document.getElementById('localVideo');
    let remoteVideo = document.getElementById('remoteVideo');
    let websocket = null;
    let peer = null;

    WebSocketInit();
    ButtonFunInit();

    /* WebSocket */
    function WebSocketInit(){
        //判断当前浏览器是否支持WebSocket
        if ('WebSocket' in window) {
            websocket = new WebSocket("wss://192.168.0.104/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);

            if (type === 'hangup') {
                console.log(msg);
                document.getElementById('hangup').click();
                return;
            }

            if (type === 'call_start') {
                let msg = "0"
                if(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offer
                    let stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true })
                    localVideo.srcObject = stream;
                    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();

        //ice
        peer.onicecandidate = function (e) {
            if (e.candidate) {
                websocket.send(JSON.stringify({
                    type: '_ice',
                    toUser:document.getElementById('toUser').value,
                    fromUser:username,
                    iceCandidate: e.candidate
                }));
            }
        };

        //track
        peer.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;
        }
    }
</script>
</html>

DemoAppliciation.java

package com.et.webrtc;

import org.springframework.boot.SpringApplication;
import org.springframework.boot.autoconfigure.SpringBootApplication;

@SpringBootApplication
public class DemoApplication {

   public static void main(String[] args) {
      SpringApplication.run(DemoApplication.class, args);
   }
}

以上只是一些关键代码，所有代码请参见下面代码仓库

代码仓库

  • GitHub - Harries/springboot-demo: a simple springboot demo with some components for example: redis,solr,rockmq and so on.

3.测试

启动Spring Boot应用

测试视频通话

前置条件：必须是https协议，不然无法打开视频和语音权限

输入对方id，进行视屏通话

4.引用

标签： spring boot websocket webrtc
本文转载自: https://blog.csdn.net/dot_life/article/details/139759752
版权归原作者 HBLOGA 所有， 如有侵权，请联系我们删除。
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