如何发送和接收通过 getUsermedia() 生成的桌面捕获流

我正在使用 WebRTC + Socket.io 制作一个屏幕共享应用程序，但卡在一个地方。 使用 WebRTC + Socket.io 连接两个浏览器并可以发送文字

我正在寻求支持codelab https://codelabs.developers.google.com/codelabs/webrtc-web/#8但它不适用于流。（如果解决方案基于此链接，则非常有帮助）

如何发送 getUserMedia() 流：

dataChannel.send(stream);

并在channel.onmessage()上接收相同的流： 我将 event.data 作为“[object MediaStream]”而不是流获取。

channel.onmessage = function(event){
  // unable to get correct stream
  // event.data is "[object MediaStream]"  in string
}

function createPeerConnection(isInitiator, config) {
    console.log('Creating Peer connection as initiator?', isInitiator, 'config:', config);
    peerConn = new RTCPeerConnection(config);

    // send any ice candidates to the other peer
    peerConn.onicecandidate = function (event) {
        console.log('onIceCandidate event:', event);
        if (event.candidate) {
            sendMessage({
                type: 'candidate',
                label: event.candidate.sdpMLineIndex,
                id: event.candidate.sdpMid,
                candidate: event.candidate.candidate
            });
        } else {
            console.log('End of candidates.');
        }
    };

    if (isInitiator) {
        console.log('Creating Data Channel');
        dataChannel = peerConn.createDataChannel("screen");
        onDataChannelCreated(dataChannel);

        console.log('Creating an offer');
        peerConn.createOffer(onLocalSessionCreated, logError);
    } else {
        peerConn.ondatachannel = function (event) {
            console.log('ondatachannel:', event.channel);
            dataChannel = event.channel;
            onDataChannelCreated(dataChannel);
        };
    }
}

它适用于字符串或 json，即 dataChannel.send('Hello');

我为此创建了一个维基页面：wiki https://github.com/muaz-khan/WebRTC-Experiment/wiki/Unable-to-receive-correctly-desktop-capture-stream-that-got-by-getUserMedia-but-working-fine-for-string

请帮忙。

请尝试这样的事情：（代码末尾有解释)

var btnShareYourCamera = document.querySelector('#share-your-camera');
var localVideo = document.querySelector('#local-video');
var remoteVideo = document.querySelector('#remote-video');

var websocket = new WebSocket('wss://path-to-server:port/');
websocket.onmessage = function(event) {
    var data = JSON.parse(event.data);
    if (data.sdp) {
        if (data.sdp.type === 'offer') {
            getUserMedia(function(video_stream) {
                localVideo.srcObject = video_stream;
                answererPeer(new RTCSessionDescription(data.sdp), video_stream);
            });
        }

        if (data.sdp.type === 'answer') {
            offerer.setRemoteDescription(new RTCSessionDescription(data.sdp));
        }
    }

    if (data.candidate) {
        addIceCandidate((offerer || answerer), new RTCIceCandidate(data.candidate));
    }
};

var iceTransportPolicy = 'all';
var iceTransportLimitation = 'udp';

function addIceCandidate(peer, candidate) {
    if (iceTransportLimitation === 'tcp') {
        if (candidate.candidate.toLowerCase().indexOf('tcp') === -1) {
            return; // ignore UDP
        }
    }

    peer.addIceCandidate(candidate);
}

var offerer, answerer;

var iceServers = {
    iceServers: [{
        'urls': [
            'stun:stun.l.google.com:19302',
            'stun:stun1.l.google.com:19302',
            'stun:stun2.l.google.com:19302',
            'stun:stun.l.google.com:19302?transport=udp',
        ]
    }],
    iceTransportPolicy: iceTransportPolicy,
    rtcpMuxPolicy: 'require',
    bundlePolicy: 'max-bundle'
};

// https://https;//cdn.webrtc-experiment.com/IceServersHandler.js
if (typeof IceServersHandler !== 'undefined') {
    iceServers.iceServers = IceServersHandler.getIceServers();
}

var mediaConstraints = {
    OfferToReceiveAudio: true,
    OfferToReceiveVideo: true
};

/* offerer */

function offererPeer(video_stream) {
    offerer = new RTCPeerConnection(iceServers);
    offerer.idx = 1;

    video_stream.getTracks().forEach(function(track) {
        offerer.addTrack(track, video_stream);
    });

    offerer.ontrack = function(event) {
        remoteVideo.srcObject = event.streams[0];
    };

    offerer.onicecandidate = function(event) {
        if (!event || !event.candidate) return;
        websocket.send(JSON.stringify({
            candidate: event.candidate
        }));
    };

    offerer.createOffer(mediaConstraints).then(function(offer) {
        offerer.setLocalDescription(offer).then(function() {
            websocket.send(JSON.stringify({
                sdp: offer
            }));
        });
    });
}

/* answerer */

function answererPeer(offer, video_stream) {
    answerer = new RTCPeerConnection(iceServers);
    answerer.idx = 2;

    video_stream.getTracks().forEach(function(track) {
        answerer.addTrack(track, video_stream);
    });

    answerer.ontrack = function(event) {
        remoteVideo.srcObject = event.streams[0];
    };

    answerer.onicecandidate = function(event) {
        if (!event || !event.candidate) return;
        websocket.send(JSON.stringify({
            candidate: event.candidate
        }));
    };

    answerer.setRemoteDescription(offer).then(function() {
        answerer.createAnswer(mediaConstraints).then(function(answer) {
            answerer.setLocalDescription(answer).then(function() {
                websocket.send(JSON.stringify({
                    sdp: answer
                }));
            });
        });
    });
}

var video_constraints = {
    mandatory: {},
    optional: []
};

function getUserMedia(successCallback) {
    function errorCallback(e) {
        alert(JSON.stringify(e, null, '\t'));
    }

    var mediaConstraints = {
        video: true,
        audio: true
    };

    navigator.mediaDevices.getUserMedia(mediaConstraints).then(successCallback).catch(errorCallback);
}

btnShareYourCamera.onclick = function() {
    getUserMedia(function(video_stream) {
        localVideo.srcObject = video_stream;
        offererPeer(video_stream);
    });
};

1. 您必须使用附加流peer.addTrack正如你在上面的例子中看到的
2. 您必须使用接收远程流peer.ontrack正如你在上面的例子中看到的

即使用addTrack连接相机并使用ontrack接收远程摄像头。

您绝不能使用以下方式发送流dataChannel.send。两者是完全不同的协议。 AMediaStream必须使用 RTP 共享；不是 SCTP。仅当您拨打电话时才使用 RTPpeer.addTrack附加相机流的方法。

此过程发生在您打开或加入房间之前。

在这里查看单页演示：https://www.webrtc-experiment.com/getStats/ https://www.webrtc-experiment.com/getStats/

上述代码片段的 HTML：

<button id="share-your-camera"></button>
<video id="local-video" controls autoplay playsinline></video>
<video id="remote-video" controls autoplay playsinline></video>