ReferenceError:room is not defined, RTCpeerconnection not working. Clients do not connect

I am making a WebRTC video chat application and it was working before i started to add or subtract more code and in the process i deleted or changed the order in a way that now i am getting this error. Sadly i don't have a backup code and it has consumed so much of my time.

You will need socket.io and node-static packages installed on your node server.

Okay i managed to solve the issue with the error but now clients are not connecting with each other, it seems two clients are unable to exchange messages via the server.

My server.js code is below

var static = require('node-static');
var http = require('http');
var file = new(static.Server)();
var app = http.createServer(function (req, res) {
  file.serve(req, res);

}).listen(8080, '127.0.0.1');

var io = require('socket.io').listen(app);
        io.sockets.on('connection', function (socket){
        socket.emit('emit(): client ' + socket.id + ' joined room ' + room);
        socket.broadcast.emit('broadcast(): client ' + socket.id + ' joined room ' + room);


    function log(){
        var array = [">>> Message from server: "];
      for (var i = 0; i < arguments.length; i++) {
        array.push(arguments[i]);
      }
        socket.emit('log', array);
    }

    socket.on('message', function (message) {
        log('Got message: ', message);
    // For a real app, should be room only (not broadcast)
        socket.broadcast.emit('message', message);
    });

    socket.on('create or join', function (room) {
        var numClients = io.sockets.clients(room).length;

        log('Room ' + room + ' has ' + numClients + ' client(s)');
        log('Request to create or join room', room);

        if (numClients == 0){
            socket.join(room);
            socket.emit('created', room);
        } else if (numClients == 1) {
            io.sockets.in(room).emit('join', room);
            socket.join(room);
            socket.emit('joined', room);
        } else { // max two clients
            socket.emit('full', room);
        }

    });

});

my application.js file is as follows

'use strict';

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


var pc_config = webrtcDetectedBrowser === 'firefox' ?
  {'iceServers':[{'url':'stun:23.21.150.121'}]} : // number IP
  {'iceServers': [{'url': 'stun:stun.l.google.com:19302'}]};

var pc_constraints = {'optional': [{'DtlsSrtpKeyAgreement': true}]};

// Set up audio and video regardless of what devices are present.
var sdpConstraints = {'mandatory': {
  'OfferToReceiveAudio':true,
  'OfferToReceiveVideo':true }};


var room = location.pathname.substring(1);
if (room === '') {
room = window.prompt('Enter room name:');
room = '';
} 

var socket = io.connect();



if (room !== '') {
  console.log('Create or join room', room);
  socket.emit('create or join', 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('This peer has joined room ' + room);
  isChannelReady = true;
});

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

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

function sendMessage(message){
    console.log('Client sending message: ', message);
  // if (typeof message === 'object') {
  //   message = JSON.stringify(message);
  // }
  socket.emit('message', 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();
  }
});

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



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

function handleUserMediaError(error){
  console.log('getUserMedia error: ', error);
}

var constraints = {video: true, audio:true};
getUserMedia(constraints, handleUserMedia, handleUserMediaError);

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



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

window.onbeforeunload = function(e){
    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('handleIceCandidate 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: ', e);
}

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

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

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 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 = null;
  // 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;
}

Index.html

<!DOCTYPE html>
<html>
<head>

<meta name='keywords' content='WebRTC, HTML5, JavaScript' />
<meta name='description' content='WebRTC Reference App' />
<meta name='viewport' content='width=device-width,initial-scale=1,minimum-scale=1,maximum-scale=1'>

<base target='_blank'>

<title>WebRTC client</title>

<link rel='stylesheet' href='css/style.css' /> 

</head>

<body id='body'>

<p style=" font-size:24px" align="center">WebRTC Video Share</p>

<div id='container'>

<div>
    <video id='localVideo' autoplay muted></video>
    <video id='remoteVideo' autoplay></video>
</div>

</div>

<script src='/socket.io/socket.io.js'></script>
<script src='js/lib/adapter.js'></script>
<script src='js/main.js'></script>

</body>
</html>

In your server.js code:

var io = require('socket.io').listen(app);
    io.sockets.on('connection', function (socket){
    socket.emit('emit(): client ' + socket.id + ' joined room ' + room);

at the end you refer to a variable room, but you haven't created it yet.