远程视频流完全不显示

时间:2018-12-02 17:50:30

标签: javascript socket.io webrtc

我一直在尝试使用.ontrack来显示远程视频流,该代码在我的代码的对等连接功能下。到目前为止,.ontrack仅在调用方触发,而在被调用方则不会触发,甚至在调用该函数时也不会触发。

检查.ontrack是否触发的日志将显示“ Got Remote Stream”,但仅在调用方显示,这可能是问题所在,但是我不确定为什么另一方不进入包含.ontrack,当它没有该语句要检查的event.stream [0]时。

我在下面添加了来自Caller和Callee的控制台日志。图片中未显示,过一会儿候选人将显示为空,但两个用户仍保持联系。

main.js

'use strict';

var isInitiator;
var configuration = {
  iceServers: [
    {
      urls: 'stun:stun.l.google.com:19302'
    }
  ]
};
var pc = new RTCPeerConnection(configuration);

// Define action buttons.
const callButton = document.getElementById('callButton');
const hangupButton = document.getElementById('hangupButton');

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

window.room = prompt('Enter room name:');

var socket = io.connect();

if (room !== '') {
  console.log('Message from client: Asking to join room ' + room);
  socket.emit('create or join', room);
}

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

socket.on('full', function(room) {
  console.log('Message from client: Room ' + room + ' is full :^(');
});

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

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

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

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

// This client receives a message
socket.on('message', async function(message) {
  try {
    if (message.type === 'offer') {
      await pc.setRemoteDescription(new RTCSessionDescription(message));
      await pc
        .setLocalDescription(await pc.createAnswer())
        .then(function() {
          sendMessage(pc.localDescription);
        })
        .catch(function(err) {
          console.log(err.name + ': ' + err.message);
        });
      createPeerConnection();
    } else if (message.type === 'answer') {
      await pc.setRemoteDescription(new RTCSessionDescription(message));
    } else if (message.type === 'candidate') {
      await pc.addIceCandidate(candidate);
    }
  } catch (err) {
    console.error(err);
  }
});

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

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

// Set up initial action buttons status: disable call and hangup.
callButton.disabled = true;
hangupButton.disabled = true;

// Add click event handlers for buttons.
callButton.addEventListener('click', callStart);
hangupButton.addEventListener('click', hangupCall);

function startVideo() {
  navigator.mediaDevices
    .getUserMedia({
      audio: true,
      video: true
    })
    .then(function(stream) {
      localVideo.srcObject = stream;
      stream.getTracks().forEach(track => pc.addTrack(track, stream));
    })
    .catch(function(err) {
      console.log('getUserMedia() error: ' + err.name);
    });
  callButton.disabled = false;
}

async function callStart() {
  createPeerConnection();
  callButton.disabled = true;
  hangupButton.disabled = false;
  if (isInitiator) {
    console.log('Sending offer to peer');
    await pc
      .setLocalDescription(await pc.createOffer())
      .then(function() {
        sendMessage(pc.localDescription);
      })
      .catch(function(err) {
        console.log(err.name + ': ' + err.message);
      });
  }
}

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

function createPeerConnection() {
  try {
    pc.ontrack = event => {
      if (remoteVideo.srcObject !== event.streams[0]) {
        remoteVideo.srcObject = event.streams[0];
        console.log('Got remote stream');
      }
    };
    pc.onicecandidate = ({ candidate }) => sendMessage({ candidate });
    console.log('Created RTCPeerConnnection');
  } catch (e) {
    console.log('Failed to create PeerConnection, exception: ' + e.message);
    alert('Cannot create RTCPeerConnection object.');
    return;
  }
}

function hangupCall() {
  pc.close();
  pc = null;
  callButton.disabled = false;
  hangupButton.disabled = true;
  console.log('Call Ended');
}

index.js

'use strict';

var express = require('express');
var app = (module.exports.app = express());
var path = require('path');

var server = require('http').createServer(app);
var io = require('socket.io')(server);
const PORT_NO = process.env.APP_PORT || 3000;
server.listen(PORT_NO);

app.get('/', function(request, response) {
  response.sendFile(path.resolve('./index.html'));
});

app.use(express.static('.'));
io.on('connection', socket => {
  function log() {
    const array = ['Message from server:'];
    for (let i = 0; i < arguments.length; i++) {
      array.push(arguments[i]);
    }
    socket.emit('log', array);
  }

  socket.on('message', message => {
    log('Got message:', message);
    socket.broadcast.emit('message', message);
  });

  socket.on('create or join', room => {
    var clientsInRoom = io.sockets.adapter.rooms[room];
    var numClients = clientsInRoom
      ? Object.keys(clientsInRoom.sockets).length
      : 0;

    // max two clients
    if (numClients === 2) {
      socket.emit('full', room);
      return;
    }

    log('Room ' + room + ' now has ' + (numClients + 1) + ' client(s)');

    if (numClients === 0) {
      socket.join(room);
      log('Client ID ' + socket.id + ' created room ' + room);
      socket.emit('created', room, socket.id);
    } else {
      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');
    }
  });
});

Callee side (in Safari)

Caller side (in Firefox)

2 个答案:

答案 0 :(得分:0)

让加入者成为发起者。

我猜'created'发生在'joined'之前?即一方在第二方加入之前创建房间?

由于您的startVideo()所做的不只是开始本地视频,它实际上开始了连接协商,因此我怀疑您在第二方准备就绪之前就开始进行比赛。而是尝试:

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

socket.on('joined', function(room) {
  console.log('joined: ' + room);
  isInitiator = true; // <-- begin negotiating once 2nd party arrives.
  startVideo();
});

答案 1 :(得分:0)

您在答录器侧缺少对createPeerConnection()的呼叫,这意味着答录器未正确设置为向ICE候选人发送信号或触发跟踪事件。

您只能从startCall()进行呼叫,因此只有在几乎完全同时按下两端的通话按钮时,此方法才起作用。

createPeerConnection()是用词不当。相反,只需在页面加载时使用pcontrack回调来初始化onicecandidate

仍然无法正常工作吗?

您向我们展示的与WebRTC相关的其余代码看起来都不错-除非您在答录器端两次致电getUserMedia,这是多余的,但这不成问题。

我怀疑您的服务器逻辑中存在错误。例如。您并没有向我们展示发出'create or join'会如何变成'created''joined'套接字消息。您还试图预先确定要约/答案交换中的哪一方,这很好,除非这意味着您在应答方上的Call按钮不起作用。大多数演示程序只会让任何人只要按一下按钮就可以成为要约人,尽管这可能会引起眩光。仅供参考。

这是双向呼叫。 remoteVideo在哪个方向上不起作用?

此外,您在这里有一个双向呼叫,可以在两个方向上发送视频,但您没有提到您没有看到的 远程视频。

有关工作示例,请查看我的two-way tab demo。在同一浏览器中的两个相邻窗口中将其打开,然后在其中的一个中单击Call按钮以进行连接。您应该看到(相同)双向发送的视频。它依赖于使用localStorage的localSocket黑客。