webRTC demo

阅读原文时间：2023年07月08日阅读：1

准备：

信令服务
前端页面用于视频通话

demo github 地址。

为了使 demo 尽量简单，功能页面如下，即包含登录、通过对方手机号拨打电话的功能。在实际生成过程中，未必使用的手机号，可能是任何能代表用户身份的字符串。

代码如下：

<!DOCTYPE html>
<html lang="en">
<head>
    <meta charset="UTF-8">
    <title>Title</title>
</head>
<body>
<div style="margin: 20px">
    <label for="loginAccount">登录账号</label><input id="loginAccount" name="loginAccount" placeholder="请输入手机号"
                                                     type="text">
    <button id="login" onclick="login()" type="button">登录</button>
</div>
<div style="margin: 20px">
    <video autoplay controls height="360px" id="localVideo" width="640px"></video>
    <video autoplay controls height="360px" id="remoteVideo" width="640px"></video>
</div>  

<div style="margin: 20px">
    <label for="toAccount">对方账号</label>
    <input id="toAccount" name="toAccount" placeholder="请输入对方手机号" type="text">
    <button id="requestVideo" onclick="requestVideo()" type="button">请求视频通话</button>
</div>  

<div style="margin: 20px">
    <fieldset>
        <button id="accept" type="button">接通</button>
        <button id="hangup" type="button">挂断</button>
    </fieldset>
</div>  

<div style="margin: 20px">
    <fieldset>
        <div>
            录制格式： <select disabled id="codecPreferences"></select>
        </div>
        <button id="startRecord" onclick="startRecording()" type="button">开始录制视频</button>
        <button id="stopRecord" onclick="stopRecording()" type="button">停止录制视频</button>
        <button id="downloadRecord" onclick="download()" type="button">下载</button>
    </fieldset>
</div>  

</body>  

<script>
    let config = {
        iceServers: [
            {
                'urls': 'turn:turn.wildfirechat.cn:3478',
                'credential': 'wfchat',
                'username': 'wfchat'
            }
        ]
    }  

    const localVideo = document.getElementById('localVideo');
    const remoteVideo = document.getElementById('remoteVideo');  

    const requestVideoButton = document.getElementById('requestVideo');
    const acceptButton = document.getElementById('accept');
    const hangupButton = document.getElementById('hangup');  

    const codecPreferences = document.querySelector('#codecPreferences');  

    const recordButton = document.getElementById('startRecord')
    const stopRecordButton = document.getElementById('stopRecord')
    const downloadButton = document.getElementById('downloadRecord')  

    const wsAddress = 'ws://localhost:9113/ws';
    let loginAttemptCount = 0;
    let myId, toId;
    let pc, localStream, ws;  

    let mediaRecorder;
    let recordedBlobs;  

    function login() {
        loginAttemptCount = 0;  

        myId = document.getElementById('loginAccount').value;  

        ws = new WebSocket(wsAddress);
        ws.onopen = function () {
            console.log("WebSocket is open now.");
            connect();
            alert("登录成功");
        };  

        ws.onmessage = function (message) {
            let msg = JSON.parse(message.data);
            console.log("ws 收到消息：" + msg.type);
            switch (msg.type) {
                case "offline": {
                    if (loginAttemptCount < 10) {
                        setTimeout(() => {
                            loginAttemptCount++;
                            watch();
                        }, 1000);
                    }
                    break;
                }
                case "watch": {
                    handleWatch(msg);
                    break;
                }
                case "offer": {
                    handleOffer(msg);
                    break;
                }
                case "answer": {
                    handleAnswer(msg);
                    break;
                }
                case "candidate": {
                    handleCandidate(msg);
                    break;
                }
                case "hangup": {
                    handleHangup(msg);
                    break;
                }
            }
        };
    }  

    requestVideoButton.onclick = async () => {
        toId = document.getElementById('toAccount').value;  

        if (!myId) {
            alert('请先登录');
            return;
        }  

        if (!toId) {
            alert('请输入对方手机号');
            return;
        }  

        watch();  

        localStream = await navigator.mediaDevices.getUserMedia({audio: true, video: true});
        localVideo.srcObject = localStream;  

        createPeerConnection();
    }  

    function connect() {
        send({
            type: "connect",
            from: myId
        });
    }  

    function handleWatch(msg) {
        toId = msg.from;
    }  

    acceptButton.onclick = async () => {
        localStream = await navigator.mediaDevices.getUserMedia({audio: true, video: true});
        localVideo.srcObject = localStream;
        createPeerConnection();  

        pc.createOffer().then(offer => {
            pc.setLocalDescription(offer);
            send({
                type: 'offer',
                from: myId,
                to: toId,
                data: offer
            });
        });
    }  

    function handleOffer(msg) {
        pc.setRemoteDescription(msg.data);  

        pc.createAnswer().then(answer => {
            pc.setLocalDescription(answer);
            send({
                type: "answer",
                from: myId,
                to: toId,
                data: answer
            });
        });
    }  

    function watch() {
        send({
            type: 'watch',
            from: myId,
            to: toId
        });
    }  

    function handleAnswer(msg) {
        if (!pc) {
            console.error('no peer connection');
            return;
        }
        pc.setRemoteDescription(msg.data);
    }  

    function handleCandidate(msg) {
        if (!pc) {
            console.error('no peer connection');
            return;
        }
        pc.addIceCandidate(new RTCIceCandidate(msg.data)).then(() => {
            console.log('candidate添加成功')
        }).catch(handleError)
    }  

    function handleError(error) {
        console.log(error);
    }  

    function createPeerConnection() {
        pc = new RTCPeerConnection(config);
        pc.onicecandidate = e => {
            if (e.candidate) {
                send({
                    type: "candidate",
                    from: myId,
                    to: toId,
                    data: e.candidate
                });
            }
        };  

        pc.ontrack = e => remoteVideo.srcObject = e.streams[0];
        localStream.getTracks().forEach(track => pc.addTrack(track, localStream));
    }  

    hangupButton.onclick = async () => {
        if (pc) {
            pc.close();
            pc = null;
        }
        if (localStream) {
            localStream.getTracks().forEach(track => track.stop());
            localStream = null;
        }
        send({
            type: "hangup",
            from: myId,
            to: toId
        });
    }  

    function handleHangup() {
        if (!pc) {
            console.error('no peer connection');
            return;
        }
        pc.close();
        pc = null;
        if (localStream) {
            localStream.getTracks().forEach(track => track.stop());
            localStream = null;
        }
        console.log('hangup');
    }  

    function send(msg) {
        ws.send(JSON.stringify(msg));
    }  

    function getSupportedMimeTypes() {
        const possibleTypes = [
            'video/webm;codecs=vp9,opus',
            'video/webm;codecs=vp8,opus',
            'video/webm;codecs=h264,opus',
            'video/mp4;codecs=h264,aac',
        ];
        return possibleTypes.filter(mimeType => {
            return MediaRecorder.isTypeSupported(mimeType);
        });
    }  

    function startRecording() {
        recordedBlobs = [];
        getSupportedMimeTypes().forEach(mimeType => {
            const option = document.createElement('option');
            option.value = mimeType;
            option.innerText = option.value;
            codecPreferences.appendChild(option);
        });
        const mimeType = codecPreferences.options[codecPreferences.selectedIndex].value;
        const options = {mimeType};  

        try {
            mediaRecorder = new MediaRecorder(remoteVideo.srcObject, options);
        } catch (e) {
            console.error('Exception while creating MediaRecorder:', e);
            alert('Exception while creating MediaRecorder: ' + e);
            return;
        }  

        console.log('Created MediaRecorder', mediaRecorder, 'with options', options);
        recordButton.textContent = 'Stop Recording';
        mediaRecorder.onstop = (event) => {
            console.log('Recorder stopped: ', event);
            console.log('Recorded Blobs: ', recordedBlobs);
        };
        mediaRecorder.ondataavailable = handleDataAvailable;
        mediaRecorder.start();
        console.log('MediaRecorder started', mediaRecorder);
    }  

    function handleDataAvailable(event) {
        console.log('handleDataAvailable', event);
        if (event.data && event.data.size > 0) {
            recordedBlobs.push(event.data);
        }
    }  

    function stopRecording() {
        mediaRecorder.stop();
    }  

    function download() {
        const blob = new Blob(recordedBlobs, {type: 'video/webm'});
        const url = window.URL.createObjectURL(blob);
        const a = document.createElement('a');
        a.style.display = 'none';
        a.href = url;
        a.download = 'test.webm';
        document.body.appendChild(a);
        a.click();
        setTimeout(() => {
            document.body.removeChild(a);
            window.URL.revokeObjectURL(url);
        }, 100);
    }  

</script>
</html>

基于 JDK 1.8 Spring Boot、Netty 搭建，主要用于解决两个问题：

确认参与人，即拨打视频电话的人和接通视频电话的人
提供功能按钮 API，比如：发起视频通话、挂电话、以及 webRTC 建立通信通道

主要功能如下：

switch (event.getType()) {
    case "connect": {
        USER_MAP.put(event.getFrom(), ctx);
        break;
    }
    case "watch": {
        WebRtcEvent watchRequest = new WebRtcEvent();
        if (USER_MAP.containsKey(event.getTo())) {
            watchRequest.setType("watch");
            watchRequest.setFrom(event.getFrom());
            watchRequest.setTo(event.getTo());
            USER_MAP.get(event.getTo()).writeAndFlush(new TextWebSocketFrame(JSONObject.toJSONString(watchRequest)));
        } else {
            watchRequest.setType("offline");
            USER_MAP.get(event.getFrom()).writeAndFlush(new TextWebSocketFrame(JSONObject.toJSONString(watchRequest)));
        }
        break;
    }
    case "offer": {
        WebRtcEvent offerRequest = new WebRtcEvent();
        offerRequest.setType("offer");
        offerRequest.setFrom(event.getFrom());
        offerRequest.setTo(event.getTo());
        offerRequest.setData(event.getData());
        USER_MAP.get(event.getTo()).writeAndFlush(new TextWebSocketFrame(JSONObject.toJSONString(offerRequest)));
        break;
    }
    case "answer": {
        WebRtcEvent answerRequest = new WebRtcEvent();
        answerRequest.setType("answer");
        answerRequest.setFrom(event.getFrom());
        answerRequest.setData(event.getData());
        USER_MAP.get(event.getTo()).writeAndFlush(new TextWebSocketFrame(JSONObject.toJSONString(answerRequest)));
        break;
    }
    case "candidate": {
        WebRtcEvent candidateRequest = new WebRtcEvent();
        candidateRequest.setType("candidate");
        candidateRequest.setFrom(event.getFrom());
        candidateRequest.setData(event.getData());
        USER_MAP.get(event.getTo()).writeAndFlush(new TextWebSocketFrame(JSONObject.toJSONString(candidateRequest)));
        break;
    }
    case "hangup": {
        WebRtcEvent hangupRequest = new WebRtcEvent();
        hangupRequest.setType("hangup");
        hangupRequest.setFrom(event.getFrom());
        hangupRequest.setTo(event.getTo());
        USER_MAP.get(event.getTo()).writeAndFlush(new TextWebSocketFrame(JSONObject.toJSONString(hangupRequest)));
        break;
    }
}

connect -> 登录

与 html 页面中的“登录”按钮对应，当输入手机号后，点击登录，手机号将会在信令服务中存到 map 中，以待后续操作使用。

如下图所示，至少两个客户端登录以后，才能正常视频通话。

watch -> 请求视频通话

点击 watch 按钮后，前端将发送一个事件到信令服务中，结构如下：

{
    type: 'watch',      //事件类型
    from: 13789122381,  // 我的账号，比如 13789122381
    to: 1323493929      // 对方的账号，比如 1323493929
}

此时输入的对方账号对应 “to” 字段。

信令服务器收到 watch 事件后，从 map 中找出对应的在线客户端，将该事件转发至相应的客户端中。

offer -> 接通

对于接收者来说，点击“接通”按钮以后，webRTC 将开始建立通信隧道。

接通的 json 结构如下：

{
    type: 'offer',
    from: myId,
    to: toId,
    data: offer
}

整个拨打电话、接通的流程如下：

在 html 中还需要配置 coturn TURN 服务地址，我在 demo 中使用的地址是测试地址，所以请不要在生产中使用。

手机扫一扫

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