Webrtc在lan＆amp; amp;本地但不在互联网上工作

Question

我制作了一个webrtc程序，它在内部网上工作正常，但没有在互联网上工作。我已经让路由器的防火墙关闭了，现在不知道出了什么问题。

这是我的代码：

<?php
global $db;

if(isset($_REQUEST['room_id'])){
    $room_id=$_REQUEST['room_id'];
    require_once 'modules/Conferencingrooms/Conferencingroom.php';
    $conferencingroom=new  Conferencingroom();
    $conferencingroom->retrieve($room_id);
    $branch_id=$conferencingroom->branch_id;    
}
else{
    sugar_die('Error! Access is Denind');
}    

?>
<script>
if (window.location.protocol != "https:")
    window.location.href = "https:" + window.location.href.substring(window.location.protocol.length);
</script>    

<video src="" id="localVideo" style="border:1px red dotted">Patient Video</video>
<video src=""  id="remoteVideo" style="border:1px red dotted">Doctor Video</video>
<script>

var pc;    
var callNo = '<?php echo $room_id;?>'; // 0 indicates callNo not initialised yet

var caller = <?php if($branch_id==$current_user->branch_id) echo 'true';else echo 'false'?>;
if (caller) {

    $("#ajaxStatusDiv").show();   

  }
var video_constraints={
        mandatory:{
            maxWidth:320,
            maxHeight:320,
            maxAspectRatio:16/8,
            maxFrameRate:500,
            }
}
navigator.getUserMedia = navigator.mozGetUserMedia || navigator.webkitGetUserMedia.bind(navigator);;
navigator.getUserMedia({audio: true, video: true}, onUserMediaSuccess, onUserMediaError);

function onUserMediaSuccess(stream) {

  document.getElementById('localVideo').src = webkitURL.createObjectURL(stream);
  document.getElementById('localVideo').play();
  var PeerConnection = window.RTCPeerConnection || window.mozRTCPeerConnection || window.webkitRTCPeerConnection;
  pc = new PeerConnection({iceServers: [ {url:'stun:stun01.sipphone.com'},
                                         {url:'stun:stun.ekiga.net'},
                                         {url:'stun:stun.fwdnet.net'},
                                         {url:'stun:stun.ideasip.com'},
                                         {url:'stun:stun.iptel.org'},
                                         {url:'stun:stun.rixtelecom.se'},
                                         {url:'stun:stun.schlund.de'},
                                         {url:'stun:stun.l.google.com:19302'},
                                         {url:'stun:stun1.l.google.com:19302'},
                                         {url:'stun:stun2.l.google.com:19302'},
                                         {url:'stun:stun3.l.google.com:19302'},
                                         {url:'stun:stun4.l.google.com:19302'},
                                         {url:'stun:stunserver.org'},
                                         {url:'stun:stun.softjoys.com'},
                                         {url:'stun:stun.voiparound.com'},
                                         {url:'stun:stun.voipbuster.com'},
                                         {url:'stun:stun.voipstunt.com'},
                                         {url:'stun:stun.voxgratia.org'},
                                         {url:'stun:stun.xten.com'},
                                         {
                                            url: 'turn:numb.viagenie.ca',
                                            credential: 'muazkh',
                                            username: 'webrtc@live.com'
                                         },
                                         {
                                            url: 'turn:192.158.29.39:3478?transport=udp',
                                            credential: 'JZEOEt2V3Qb0y27GRntt2u2PAYA=',
                                            username: '28224511:1379330808'
                                         },
                                         {
                                            url: 'turn:192.158.29.39:3478?transport=tcp',
                                            credential: 'JZEOEt2V3Qb0y27GRntt2u2PAYA=',
                                            username: '28224511:1379330808'
                                         }]});
  pc.preventSSLAutoAllowed = false;
  pc.autoReDialOnFailure = true;
  pc.setDefaultEventsForMediaElement = false;

  pc.addStream(stream);

  pc.onicecandidate = function (evt) {
      console.log('On Ice Candidate');
      console.log(evt.candidate);

    if (!evt.candidate) {
          console.log('inner');
      pc.onicecandidate = null;

      if (pc.localDescription) postLocalData();

      }

      };

  pc.onaddstream = function(evt) {
      document.getElementById('remoteVideo').src = webkitURL.createObjectURL(evt.stream);
      document.getElementById('remoteVideo').play();
    };

  if (caller){
            console.log('Create Offer called');
        pc.createOffer(onDescCreated, onCreateOfferError);
  }
  else {    
    getRemoteData();
    }

  window.onbeforeunload = function() {
/*
    stream.stop();

    if (caller && callNo > 0) navigator.sendBeacon('deleteCall.php', callNo);
*/
    };    
  }

function onUserMediaError(err) {
    if(err.name=='DevicesNotFoundError'){
        alert('Media device is not connected');}
  console.log('User media error: ' + err.name);
  }

function onCreateOfferError(err) {
  logError('Error creating offer: ' + err.name);
  }

function onDescCreated(desc) {

  pc.setLocalDescription(desc, onLocalDescSuccess, onLocalDescError);
}
function onLocalDescSuccess() {
    if(caller)
        console.log('offer create success');
  if (pc.iceGatheringState == 'complete') {
            if(caller)
            {
                console.log('Send for store sdp for caller');
            }
            else{
                console.log('Send for store sdp for callee');
                }
      postLocalData();
      }
  }

function onLocalDescError(err) {
  logError("Local description could not be created: " + err.name);
  }

function postLocalData() {

  console.log('Storing local sdp');
  var xhr = new XMLHttpRequest();
  var sdp=pc.localDescription.sdp;    

  xhr.open('POST', 'index.php?module=Conferencingrooms&action=AjaxCalls&storeSdp=true&caller='+caller);    
  xhr.onload = (caller ? getRemoteData : waitForConnection);    
  xhr.send(callNo + '~' + sdp);      
  }

function getRemoteData() {
  console.log("Checking db for  remote sdp");
  var xhr = new XMLHttpRequest();

  xhr.open('GET', 'index.php?module=Conferencingrooms&action=AjaxCalls&fetchSdp=true&caller='+caller+'&callNo=' + callNo);
  xhr.onload = processRemoteData;
  xhr.send();
  }

function processRemoteData() {

  if (this.response) {

    var SessionDescription = window.RTCSessionDescription || window.mozRTCSessionDescription || window.webkitRTCSessionDescription;

    var sdp = new SessionDescription();     
    sdp.type = (caller ? 'answer' : 'offer');    
    sdp.sdp = this.response;

    console.log('Processing remote sdp: ' + sdp);
    pc.setRemoteDescription(sdp, onRemoteDescSuccess, onRemoteDescError);

    }
  else
    setTimeout(getRemoteData, 1000);
  }

function onRemoteDescSuccess() {
  console.log("Remote sdp successfully set");

  if (caller) {
    waitForConnection();
    }

  else
    pc.createAnswer(onDescCreated, onCreateAnswerError);
  }

function onRemoteDescError(err) {
  logError("Remote description could not be set: " + err.name);
  }

function onCreateAnswerError(err) {
  logError("Error creating answer: " + err.name);
  }

function waitForConnection() {
console.log('Waiting');
  if (pc.iceConnectionState == 'connected' || pc.iceConnectionState == 'completed') {
      console.log('Waitingdsadsd');

    logError('Connection complete');

    caller = null;

    document.getElementById('remoteVideo').ondblclick = function() {

      this.requestFullscreen = this.mozRequestFullScreen || this.webkitRequestFullscreen;
      this.requestFullscreen();
      }
    }

  else
    setTimeout(waitForConnection, 1000);
  }

function logError(msg) {

  console.log(msg);
  if (caller && callNo > 0) navigator.sendBeacon('deleteCall.php', callNo);
  }    
</script>

这是一个php信令版本，其中会在url

中的传递房间ID上创建房间