Question

我使用WebRTC设置我的视频会议服务用于学术目的，除音频捕获外，一切正常。当我访问我的网站时，它只会要求获得摄像机的许可，但不会要求获得音频许可，当然它也不会成为任何音频......

我的main.js

'use strict';

var isChannelReady = false;
var isInitiator = false;
var isStarted = false;
var localStream;
var pc;
var remoteStream;
var turnReady;

var pcConfig = {
    'iceServers': [{
        'urls': 'stun:stun.l.google.com:19302'
    }]
};

// Set up audio and video regardless of what devices are present.
var sdpConstraints = {
    offerToReceiveAudio: true,
    offerToReceiveVideo: true
};

/////////////////////////////////////////////

var room = 'foo';
// Could prompt for room name:
// room = prompt('Enter room name:');

var socket = io.connect();

if (room !== '') {
    socket.emit('create or join', room);
    console.log('Attempted to create or  join room', room);
}

socket.on('created', function(room) {
    console.log('Created room ' + room);
    isInitiator = true;
});

socket.on('full', function(room) {
    console.log('Room ' + room + ' is full');
});

socket.on('join', function (room){
    console.log('Another peer made a request to join room ' + room);
    console.log('This peer is the initiator of room ' + room + '!');
    isChannelReady = true;
});

socket.on('joined', function(room) {
    console.log('joined: ' + room);
    isChannelReady = true;
});

socket.on('log', function(array) {
    console.log.apply(console, array);
});

////////////////////////////////////////////////

function sendMessage(message) {
    console.log('Client sending message: ', message);
    socket.emit('message', message);
}

// This client receives a message
socket.on('message', function(message) {
    console.log('Client received message:', message);
    if (message === 'got user media') {
        maybeStart();
    } else if (message.type === 'offer') {
        if (!isInitiator && !isStarted) {
            maybeStart();
        }
        pc.setRemoteDescription(new RTCSessionDescription(message));
        doAnswer();
    } else if (message.type === 'answer' && isStarted) {
        pc.setRemoteDescription(new RTCSessionDescription(message));
    } else if (message.type === 'candidate' && isStarted) {
        var candidate = new RTCIceCandidate({
            sdpMLineIndex: message.label,
            candidate: message.candidate
        });
        pc.addIceCandidate(candidate);
    } else if (message === 'bye' && isStarted) {
        handleRemoteHangup();
    }
});

////////////////////////////////////////////////////

var localVideo = document.querySelector('#localVideo');
var remoteVideo = document.querySelector('#remoteVideo');

navigator.mediaDevices.getUserMedia({
    audio: true,
    video: true
})
    .then(gotStream)
    .catch(function(e) {
        alert('getUserMedia() error: ' + e.name);
    });

function gotStream(stream) {
    console.log('Adding local stream.');
    localVideo.src = window.URL.createObjectURL(stream);
    localStream = stream;
    sendMessage('got user media');
    if (isInitiator) {
        maybeStart();
    }
}

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

console.log('Getting user media with constraints', constraints);

if (location.hostname !== 'localhost') {
    requestTurn(
        'https://computeengineondemand.appspot.com/turn?username=41784574&key=4080218913'
    );
}

function maybeStart() {
    console.log('>>>>>>> maybeStart() ', isStarted, localStream, isChannelReady);
    if (!isStarted && typeof localStream !== 'undefined' && isChannelReady) {
        console.log('>>>>>> creating peer connection');
        createPeerConnection();
        pc.addStream(localStream);
        isStarted = true;
        console.log('isInitiator', isInitiator);
        if (isInitiator) {
            doCall();
        }
    }
}

window.onbeforeunload = function() {
    sendMessage('bye');
};

/////////////////////////////////////////////////////////

function createPeerConnection() {
    try {
        pc = new RTCPeerConnection(null);
        pc.onicecandidate = handleIceCandidate;
        pc.onaddstream = handleRemoteStreamAdded;
        pc.onremovestream = handleRemoteStreamRemoved;
        console.log('Created RTCPeerConnnection');
    } catch (e) {
        console.log('Failed to create PeerConnection, exception: ' + e.message);
        alert('Cannot create RTCPeerConnection object.');
        return;
    }
}

function handleIceCandidate(event) {
    console.log('icecandidate 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.');
    }
}

function handleRemoteStreamAdded(event) {
    console.log('Remote stream added.');
    remoteVideo.src = window.URL.createObjectURL(event.stream);
    remoteStream = event.stream;
}

function handleCreateOfferError(event) {
    console.log('createOffer() error: ', event);
}

function doCall() {
    console.log('Sending offer to peer');
    pc.createOffer(setLocalAndSendMessage, handleCreateOfferError);
}

function doAnswer() {
    console.log('Sending answer to peer.');
    pc.createAnswer().then(
        setLocalAndSendMessage,
        onCreateSessionDescriptionError
    );
}

function setLocalAndSendMessage(sessionDescription) {
    // Set Opus as the preferred codec in SDP if Opus is present.
    //  sessionDescription.sdp = preferOpus(sessionDescription.sdp);
    pc.setLocalDescription(sessionDescription);
    console.log('setLocalAndSendMessage sending message', sessionDescription);
    sendMessage(sessionDescription);
}

function onCreateSessionDescriptionError(error) {
    trace('Failed to create session description: ' + error.toString());
}

function requestTurn(turnURL) {
    var turnExists = false;
    for (var i in pcConfig.iceServers) {
        if (pcConfig.iceServers[i].url.substr(0, 5) === 'turn:') {
            turnExists = true;
            turnReady = true;
            break;
        }
    }
    if (!turnExists) {
        console.log('Getting TURN server from ', turnURL);
        // No TURN server. Get one from computeengineondemand.appspot.com:
        var xhr = new XMLHttpRequest();
        xhr.onreadystatechange = function() {
            if (xhr.readyState === 4 && xhr.status === 200) {
                var turnServer = JSON.parse(xhr.responseText);
                console.log('Got TURN server: ', turnServer);
                pcConfig.iceServers.push({
                    'url': 'turn:' + turnServer.username + '@' + turnServer.turn,
                    'credential': turnServer.password
                });
                turnReady = true;
            }
        };
        xhr.open('GET', turnURL, true);
        xhr.send();
    }
}

function handleRemoteStreamAdded(event) {
    console.log('Remote stream added.');
    remoteVideo.src = window.URL.createObjectURL(event.stream);
    remoteStream = event.stream;
}

function handleRemoteStreamRemoved(event) {
    console.log('Remote stream removed. Event: ', event);
}

function hangup() {
    console.log('Hanging up.');
    stop();
    sendMessage('bye');
}

function handleRemoteHangup() {
    console.log('Session terminated.');
    stop();
    isInitiator = false;
}

function stop() {
    isStarted = false;
    // isAudioMuted = false;
    // isVideoMuted = false;
    pc.close();
    pc = null;
}

///////////////////////////////////////////

// Set Opus as the default audio codec if it's present.
function preferOpus(sdp) {
    var sdpLines = sdp.split('\r\n');
    var mLineIndex;
    // Search for m line.
    for (var i = 0; i < sdpLines.length; i++) {
        if (sdpLines[i].search('m=audio') !== -1) {
            mLineIndex = i;
            break;
        }
    }
    if (mLineIndex === null) {
        return sdp;
    }

    // If Opus is available, set it as the default in m line.
    for (i = 0; i < sdpLines.length; i++) {
        if (sdpLines[i].search('opus/48000') !== -1) {
            var opusPayload = extractSdp(sdpLines[i], /:(\d+) opus\/48000/i);
            if (opusPayload) {
                sdpLines[mLineIndex] = setDefaultCodec(sdpLines[mLineIndex],
                    opusPayload);
            }
            break;
        }
    }

    // Remove CN in m line and sdp.
    sdpLines = removeCN(sdpLines, mLineIndex);

    sdp = sdpLines.join('\r\n');
    return sdp;
}

function extractSdp(sdpLine, pattern) {
    var result = sdpLine.match(pattern);
    return result && result.length === 2 ? result[1] : null;
}

// Set the selected codec to the first in m line.
function setDefaultCodec(mLine, payload) {
    var elements = mLine.split(' ');
    var newLine = [];
    var index = 0;
    for (var i = 0; i < elements.length; i++) {
        if (index === 3) { // Format of media starts from the fourth.
            newLine[index++] = payload; // Put target payload to the first.
        }
        if (elements[i] !== payload) {
            newLine[index++] = elements[i];
        }
    }
    return newLine.join(' ');
}

// Strip CN from sdp before CN constraints is ready.
function removeCN(sdpLines, mLineIndex) {
    var mLineElements = sdpLines[mLineIndex].split(' ');
    // Scan from end for the convenience of removing an item.
    for (var i = sdpLines.length - 1; i >= 0; i--) {
        var payload = extractSdp(sdpLines[i], /a=rtpmap:(\d+) CN\/\d+/i);
        if (payload) {
            var cnPos = mLineElements.indexOf(payload);
            if (cnPos !== -1) {
                // Remove CN payload from m line.
                mLineElements.splice(cnPos, 1);
            }
            // Remove CN line in sdp
            sdpLines.splice(i, 1);
        }
    }

    sdpLines[mLineIndex] = mLineElements.join(' ');
    return sdpLines;
}

和我的index.js（服务器）

var fs = require('fs');
var http = require('http');
var https = require('https');
var cors = require('cors');
var privateKey  = fs.readFileSync('host.key', 'utf8');
var certificate = fs.readFileSync('host.cert', 'utf8');
var os = require('os');

var credentials = {key: privateKey, cert: certificate};
var express = require('express');
var app = express();

app.use(cors());
var path = require('path');
app.use(express.static(path.join(__dirname, 'public')));
app.get('/', function(req, res){
    res.sendFile(__dirname + '/index.html');
    //console.log('Servidor en marxa.');
});

console.log('Servidor ON');

var httpServer = http.createServer(app);
var httpsServer = https.createServer(credentials,app);

httpServer.listen(80,"0.0.0.0");
httpsServer.listen(443,"0.0.0.0");
var io = require("socket.io");
io = io.listen(httpServer);
io = io.listen(httpsServer);

//EL SERVER ESTA CONFIGURAT I FUNCIONA//

var usersConnected = [];
var nUsersConnected = 0;

io.sockets.on('connection',function(socket){
    function log() {
        var array = ['Message from server:'];
        array.push.apply(array, arguments);
        socket.emit('log', array);
    }

    usersConnected.push(socket.id);
    nUsersConnected++;

    log('Welcome to WebRTC Server: Miguel & Pavel');
    console.log('New user connected: '+socket.id+' #Users connected = '+nUsersConnected);

    socket.on('message', function(message) {
        log('Client said: ', message);
        // for a real app, would be room-only (not broadcast)
        socket.broadcast.emit('message', message);
    });

    socket.on('create or join', function(room) {
        log('Received request to create or join room ' + room);

        var numClients = io.sockets.sockets.length;
        log('Room ' + room + ' now has ' + numClients + ' client(s)');

        if (numClients === 1) {
            socket.join(room);
            log('Client ID ' + socket.id + ' created room ' + room);
            socket.emit('created', room, socket.id);

        } else if (numClients === 2) {
            log('Client ID ' + socket.id + ' joined room ' + room);
            io.sockets.in(room).emit('join', room);
            socket.join(room);
            socket.emit('joined', room, socket.id);
            io.sockets.in(room).emit('ready');
        } else { // max two clients
            socket.emit('full', room);
        }
    });

    socket.on('ipaddr', function() {
        var ifaces = os.networkInterfaces();
        for (var dev in ifaces) {
            ifaces[dev].forEach(function(details) {
                if (details.family === 'IPv4' && details.address !== '127.0.0.1') {
                    socket.emit('ipaddr', details.address);
                }
            });
        }
    });

    socket.on('disconnect', function (message) {
        nUsersConnected--;
        console.log('User: '+socket.id+' disconnected #Users connected = '+nUsersConnected);
        /*if(usuari1 == socket.id){
         io.to(usuari2).emit('message', {type:'hangup'});
         }else{
         io.to(usuari1).emit('message', {type:'hangup'});
         }*/


        //tots els usuaris s'han desconnectat
        if(nUsersConnected == 0){
            console.log('All users have been deleted');
            nUsersConnected = 0;
            usersConnected = [];
        }
    })
    socket.on('bye', function(){
        console.log('received bye');
    });
});

谢谢！

Answer 1

当我访问我的网站时，它只会要求获得摄像机的许可，但不会要求获得音频许可，当然它只是赢得了音频

我怀疑系统麦克风有问题如果你的麦克风是正确的，那么你的本地流应该有audioTracks localStream.getAudioTracks()。如果我们取消静音localVideo元素，它应该回显音频。

试试此演示https://webrtc.github.io/samples/src/content/devices/input-output/来测试您的麦克风。用结果更新你的问题。

Google停止提供低于网址的转弯服务器，因此您需要设置TurnServer。试试CoTURN。

requestTurn(
    'https://computeengineondemand.appspot.com/turn?username=41784574&key=4080218913'
);

音频不起作用，但视频在WebRTC上有效

1 个答案: