I am trying to implement a Voice-only WebRTC app. I am running it on Chrome Version 29.0.1547.0 dev. My app uses Socket.IO for the signaling mechanism.
peerConnection.addIceCandidate() is giving me this error: Uncaught SyntaxError: An invalid or illegal string was specified.
and separately, peerConnection.setRemoteDescription(); is giving me this error: Uncaught TypeMismatchError: The type of an object was incompatible with the expected type of the parameter associated to the object.
Here's my code:
SERVER (in CoffeeScript)
app = require("express")()
server = require("http").createServer(app).listen(3000)
io = require("socket.io").listen(server)
app.get "/", (req, res) -> res.sendfile("index.html")
app.get "/client.js", (req, res) -> res.sendfile("client.js")
io.sockets.on "connection", (socket) ->
socket.on "message", (data) ->
socket.broadcast.emit "message", data
CLIENT (in JavaScript)
var socket = io.connect("http://localhost:3000");
var pc = new webkitRTCPeerConnection({
"iceServers": [{"url": "stun:stun.l.google.com:19302"}]
});
navigator.getUserMedia = navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia;
navigator.getUserMedia({audio: true}, function (stream) {
pc.addStream(stream);
}, function (error) { console.log(error); });
pc.onicecandidate = function (event) {
if (!event || !event.candidate) return;
socket.emit("message", {
type: "iceCandidate",
"candidate": event.candidate
});
};
pc.onaddstream = function(event) {
var audioElem = document.createElement("audio");
audioElem.src = webkitURL.createObjectURL(event.stream);
audioElem.autoplay = true;
document.appendChild(audioElem);
console.log("Got Remote Stream");
};
socket.on("message", function(data) {
if (data.type === "iceCandidate") {
console.log(data.candidate);
candidate = new RTCIceCandidate(data.candidate);
console.log(candidate);
pc.addIceCandidate(candidate);
} else if (data.type === "offer") {
pc.setRemoteDescription(data.description);
pc.createAnswer(function(description) {
pc.setLocalDescription(description);
socket.emit("message", {type: "answer", description: description});
});
} else if (data.type === "answer") {
pc.setRemoteDescription(data.description);
}
});
function offer() {
pc.createOffer( function (description) {
pc.setLocalDescription(description);
socket.emit("message", {type: "offer", "description": description});
});
};
The HTML just contains a button that calls offer().
I can confirm the ICECandidates and SessionDescriptions are transferring successfully from one client to the other.
What am I doing wrong? And how should I fix these and any other errors so that I can transfer audio from one client to the other?
PS: If you know about a good source documenting the WebRTC API (except the W3C documentation), please tell me about it!
Thanks!
For that error the point is, ICE Candidates must be added only after successfully setting remote description.
Note that just after creating Offer (by Offerer), ice candidates are produced immediately. So, if somehow the answerer receives those candidates, before setting remote description (which in theory would arrive before candidates), you get error.
The same is true for offerer. It must set remote description before adding any ice candidate.
I see that in your javascript code you are not guaranteeing that remote description is set before adding ice candidates.
First of all you can check just before pc.addIceCandidate(candidate); if pc's remoteDescription is set. If you see that it is null (or undefined), you can locally store received ice candidates to add them after setting remoteDescription (or wait in offerer to send them in proper time.)
Related
I'm building a WebApp that creates a Google Document from a template and contains some user supplied data as well as data fetched from a 3rd party service. Since all of the Class (DriveApp, DocumentApp, etc) methods make a request to the Google server, a server glitch could cause a simple statement like var doc = DocumentApp.openById(DOC_ID); to fail, throw an error and stop the entire process dead in its tracks... without the user having any idea why everything appears to be "frozen" (unless he is savvy enough to check the console).
For that reason, would it be appropriate to wrap any/all functions using those methods in a try/catch? Something like:
function createDoc(DOC_ID) {
try {
var doc = DocumentApp.openById(DOC_ID);
DriveApp.someMethod(...);
doc.someOtherDocumentMethod();
...
} catch(e) {
handleServerError(e);
return false;
}
return doc;
}
or is there a better way to handle any errors that may be out of the developer's control?
EDIT
Here's the request I send from my HTML page...
$('form').submit(function(e) {
e.preventDefault();
var obj = $('form').serializeObject();
var gurl = 'https://script.google.com/macros/s/AKfycbzmhaub3ojPARA-B-Y2uVC2BJZPaRvbgMwMTH9pd7R9aHuAD5M/exec';
$.ajax({
url: gurl,
type : "GET",
data: obj,
dataType: 'jsonp',
success : function (data, status, xhr) {
console.log("success");
console.log(data);
},
complete : function (xhr, status) {
console.log("complete");
}
});
});
I'm currently unsuccessfully trying to make my PWA installable. I have registered a SertviceWorker and linked a manifest as well as I am listening on the beforeInstallPromt event.
My ServiceWorker is listening to any fetch event.
My problem is, that the created beforeInstall banner is just being shown on Chrome desktop but on mobile I get a warning in Chrome inspection tab "Application" in the "Manifest" section:
Installability
Service worker does not have the 'fetch' handler
You can check the message on https://dev.testapp.ga/
window.addEventListener('beforeinstallprompt', (e) => {
// Stash the event so it can be triggered later.
deferredPrompt = e;
mtShowInstallButton();
});
manifest.json
{"name":"TestApp","short_name":"TestApp","start_url":"https://testapp.ga/loginCheck","icons":[{"src":"https://testapp.ga/assets/icons/launcher-ldpi.png","sizes":"36x36","density":0.75},{"src":"https://testapp.ga/assets/icons/launcher-mdpi.png","sizes":"48x48","density":1},{"src":"https://testapp.ga/assets/icons/launcher-hdpi.png","sizes":"72x72","density":1.5},{"src":"https://testapp.ga/assets/icons/launcher-xhdpi.png","sizes":"96x96","density":2},{"src":"https://testapp.ga/assets/icons/launcher-xxhdpi.png","sizes":"144x144","density":3},{"src":"https://testapp.ga/assets/icons/launcher-xxxhdpi.png","sizes":"192x192","density":4},{"src":"https://testapp.ga/assets/icons/launcher-web.png","sizes":"512x512","density":10}],"display":"standalone","background_color":"#ffffff","theme_color":"#0288d1","orientation":"any"}
ServiceWorker:
//This array should NEVER contain any file which doesn't exist. Otherwise no single file can be cached.
var preCache=[
'/favicon.png',
'/favicon.ico',
'/assets/Bears/bear-standard.png',
'/assets/jsInclude/mathjax.js',
'/material.js',
'/main.js',
'functions.js',
'/material.css',
'/materialcolors.css',
'/user.css',
'/translations.json',
'/roboto.css',
'/sw.js',
'/'
];
//Please specify the version off your App. For every new version, any files are being refreched.
var appVersion="v0.2.1";
//Please specify all files which sould never be cached
var noCache=[
'/api/'
];
//On installation of app, all files from preCache are being stored automatically.
self.addEventListener('install', function(event) {
event.waitUntil(
caches.open(appVersion+'-offline').then(function(cache) {
return cache.addAll(preCache).then(function(){
console.log('mtSW: Given files were successfully pre-cached')
});
})
);
});
function shouldCache(url) {
//Checking if url is market as noCache
var isNoCache=noCache.includes(url.substr(8).substr(url.substr(8).indexOf("/")))||noCache.includes((url.substr(8).substr(url.substr(8).indexOf("/"))).substr(0,(url.substr(8).substr(url.substr(8).indexOf("/"))).indexOf("?")));
//Checking of hostname of request != current hostname
var isOtherHost=url.substr(8).substr(0,url.substr(8).indexOf("/"))!=location.hostname&&url.substr(7).substr(0,url.substr(7).indexOf("/"))!=location.hostname;
return((url.substr(0,4)=="http"||url.substr(0,3)=="ftp") && isNoCache==false && isOtherHost==false);
}
//If any fetch fails, it will look for the request in the cache and serve it from there first
self.addEventListener('fetch', function(event) {
//Trying to answer with "online" version if fails, using cache.
event.respondWith(
fetch(event.request).then(function (response) {
if(shouldCache(response.url)) {
console.log('mtSW: Adding file to cache: '+response.url);
caches.open(appVersion+'-offline').then(function(cache) {
cache.add(new Request(response.url));
});
}
return(response);
}).catch(function(error) {
console.log( 'mtSW: Error fetching. Serving content from cache: ' + error );
//Check to see if you have it in the cache
//Return response
//If not in the cache, then return error page
return caches.open(appVersion+'-offline').then(function (cache) {
return cache.match(event.request).then(function (matching) {
var report = !matching || matching.status == 404?Promise.reject('no-match'): matching;
return report
});
});
})
);
})
I checked the mtShowInstallButton function. It's fully working on desktop.
What does this mean? On the Desktop, I never got this warning, just when using a handheld device/emulator.
Fetch function is used to fetch JSon manifest file. Try reading google docs again.
For adding PWA in Mobile you need manifest file to be fetched which is fetched using service-worker using fetch function.
Here is the code :
fetch('examples/example.json')
.then(function(response) {
// Do stuff with the response
})
.catch(function(error) {
console.log('Looks like there was a problem: \n', error);
});
for more about fetch and manifest try this.
I'm trying to make my OpenShift Node.js app working, but the WS connection is not working. Client error is: connection refused.
Client side factory service:
var dataStream = $websocket(localStorageService.get('wsUrl'))
dataStream.onMessage(function(message) {
var call = JSON.parse(message.data)
if (fnMap[call.fn]) {
fnMap[call.fn](call.event, call.data)
}
})
dataStream.onError(function(err) {
console.log(err)
})
dataStream.onClose(function(event){
console.log('event: ' + JSON.stringify(event))
})
var fnMap = {
"broadcastResult": function(event, data) {
$rootScope.$broadcast(event, data)
}
}
var methods = {
callFn: function(paramJSON) {
dataStream.send(JSON.stringify(paramJSON));
}
}
return methods
I'm trying to connect on the following URL: ws://myapp-myname.rhcloud.com:8000
Could you please help?
Thank you in advance,
Csaba
First, you may want to make sure the websocket library, ws, is installed by declaring it in the dependencies section of your package.json and then doing another git push.
Otherwise, try the example provided in the openshift blog: https://blog.openshift.com/paas-websockets/
Specifically:
var websocket = new WebSocket("ws://myapp-myname.rhcloud.com:8000");
websocket.onopen = function(event) {
// The connection was opened
console.log(event)
};
websocket.onclose = function(event) {
// The connection was closed
console.log(event)
};
websocket.onmessage = function(event) {
// New message arrived
message = event.data
console.log(event)
};
websocket.onerror = function(event) {
// There was an error with your WebSocket
console.log(event)
};
The above at least might help point you toward a specific part of your code that's not working. Or, if the example isn't even working, it could be possibly something wrong with your cartridge, and you could try reaching out to RedHat's support.
i have simple web sockets html5 , when the server is up every thing is working fine
the problem is when i close the server ( for testing )
im getting :
WebSocket connection to 'ws://127.0.0.1:7777/api' failed: Error in connection establishment: net::ERR_CONNECTION_REFUSED
which i unable to catch its never jumps to onerror or onclose in case of this error
init: function () {
this.m_wsiSendBinary = new WebSocket("ws://127.0.0.1:7681/wsapi");
this.m_wsiSendBinary.onopen = function(evt) {
cc.log("Send Binary WS was opened.");
};
this.m_wsiSendBinary.onmessage = (function(evt) {
this.handleServerResponse(yStr);
this.m_wsiSendBinary.onerror = function(evt) {
};
this.m_wsiSendBinary.onclose = function(evt) {
cc.log("m_wsiSendBinary websocket instance closed.");
self.m_wsiSendBinary = null;
};
}).bind(this);
I do not have full answer, however I dealt with similar issue and have a partial and not so elegant solution (but may help someone). Unfortunately without the elimination of the error message.
Two business requirements:
BR1 - Handle state in initialization when the server is not available.
BR2 - Handle state when the server stops.
Solution for BR1
var global_connection_openned=null;//Here you have the result
init: function () {
global_connection_openned=false;
this.m_wsiSendBinary = new WebSocket("ws://127.0.0.1:7681/wsapi");
this.m_wsiSendBinary.onopen = function(evt)
{
global_connection_openned=true;
};
Solution for BR2 (assumes the BR1)
//somewhere in your project called by setInterval(..) which will detect the connection is lost (and tries to reestablish/reopen the connetion.
{
if (this.m_wsiSendBinary==null || this.m_wsiSendBinary.readyState==3)
this.init();
if (!global_connection_openned)
this.m_wsiSendBinary=null;
}
Anyway, I would be really curious if there is solid and proper solution of this use case.
I have been trying to get WebRTC to function with a broadcaster and multiple listeners but am stuck when it comes to transferal descriptions and candidates via signalling (with nodejs & socket.io).
I can get the process working between two browsers with a simple nodejs socket app which simply broadcasts the descriptions and candidates to other already connected clients, but when I attempt to store a description and connect with a newly opened browser, nothing happens.
What I basically need to understand is what do I need to provide to one browser, in order for it to begin communicating with another? The project I am working on requires the ability for listeners to join rooms, authenticate, and begin listening to whatever media is being sent.
Below is my client side code:
var audioContext = new webkitAudioContext()
var client = null
var configuration =
{
'iceServers':
[{
'url': 'stun:stun.example.org'
}]
}
$(function ()
{
window.RTCPeerConnection = window.RTCPeerConnection || window.webkitRTCPeerConnection || window.mozRTCPeerConnection
client = new RTCPeerConnection(configuration, { optional:[ { RtpDataChannels: true } ]})
client.onnegotiationneeded = function ()
{
console.log('Negotiation needed')
createOffer()
}
client.onicecandidate = function (event)
{
console.log('onicecandidate')
socket.emit('candidate', JSON.stringify({ 'candidate': event.candidate }))
}
client.onaddstream = function (event)
{
console.log('onaddstream')
$('#player').attr('src', URL.createObjectURL(event.stream))
player.play()
}
socket.on('candidate', function (event)
{
candidate(event)
})
socket.on('description', function (message)
{
if(!client) { return }
client.setRemoteDescription(new RTCSessionDescription(message.sdp), function () {
if (client.remoteDescription.type == 'offer')
client.createAnswer(function (description)
{
client.setLocalDescription(description, function ()
{
socket.emit('description', JSON.stringify({ 'sdp':client.localDescription }))
})
}, function (err)
{
console.log('error: ' + err)
})
}, function(err)
{
console.log('error: ' + err)
})
})
addStream()
})
function createOffer ()
{
if(!client) { return; }
client.createOffer(function (description)
{
console.log(description)
client.setLocalDescription(description, function ()
{
socket.emit('description', JSON.stringify({ 'sdp': client.localDescription }))
console.log('set local description')
})
})
}
function candidate (message)
{
if(message.candidate)
{
console.log('candidate')
client.addIceCandidate(new RTCIceCandidate(message.candidate))
}
}
function addStream ()
{
navigator.webkitGetUserMedia({audio: true, video: false}, function(stream)
{
client.addStream(stream)
})
}
And my signalling part of my server as it currently stands:
io.on 'connection', (socket) ->
socket.on 'description', (data) ->
parsed = JSON.parse data
socket.broadcast.emit 'description', parsed
socket.on 'candidate', (candidate) ->
parsed = JSON.parse candidate
socket.broadcast.emit 'candidate', parsed
I'd appreciate any insight into this. Thanks.
The "PeerConnection" as the name indicates can be used with only one other peer. You cannot cache the offer SDP generated by one PeerConnection instance to use it with more than one other peers.
In your case, you must create a PeerConnection for each browser that you want to send/receive audio and video from and then exchange the corresponding SDP offer and answers with those browsers via your signaling mechanism.
Please feel free to go through some of the links I have mentioned here to understand how WebRTC works.