problem about webrtc-Collection of common programming errors
Imran Sheikh
webrtc
I am using the same code of AppRTC hosted by WebRTC.ORG but still my appRTC does not works.kindly let me know, what am I doing wrong?this is my appRTC app: http://qoon.azurewebsites.net
Geoffrey Hing
node.js audio-streaming red5 rtmp webrtc
This is more a conceptual question rather than a direct “how to do this”. Is it generally possible to implement a flash-like solution to stream the audio (independent of where we get the stream data from, e.g. webRTC or other) in HTML5 and nodeJS/binaryJS. If so, how would you go about this?There has been only one inquire on stackoverflow found here and its from 2010. NodeJS and HTML5 have grown and matured since then.What people usually do: When using multimedia streaming (i.e. video or audio)
Leo Lou
node.js websocket localhost port webrtc
I get the source code of a great webrtc demo from https://github.com/webRTC/webrtc.io-demoand the demo site is: webrtc.dennis.isIt works fine when I start a local node server, but when I upload the demo to a node server online( www.appfog.com ), I come into an unexpected bug which says :WebSocket connection to ‘ws://talktosomeone4.ap01.aws.af.cm/’ failed: Unexpected response code: 200 talktosomeone4.ap01.aws.af.cm:1At first I think maybe it’s caused by missing port,so I try to complete the webso
Konga Raju
html5 webrtc
How many peer connections can I create on a single client? Is there any limit?
user3026495
video-streaming raspberry-pi webrtc beagleboneblack
I want to build a standalone video chat application using Raspberry pi/ Beaglebone board (linux based). I have compiled the WebRTC API (libjingle_peerconnection library) but during runtime, it sayslibjingle_peerconnection_so.so : cannot open shared object file: No such file or directoryKindly provide the solution for this or any other alternative to achieve the objective.
Shih-En Chou
html5 html5-video webrtc
I run my webrtc code with chrome 21.If I open two tabs in the same chrome, and then open page with webrtc code inside. One tab is for sending video stream; one tab is for receiving video stream It works fine.However, If I open page with two Incognito mode or two different chrome browser, I can get the sdp and candidate information correctly. It seems that video can decode the information.In remote video, I only can see Besides, it seems crash. I tried to click “close chrome” but useless. Anyone
Jeet
node.js deployment express production-environment webrtc
I am using Webrtc, nodejs, Expressjs as the framework to create a audio, video and chat application. I have used Forever so that the script runs continuously. As my application deals with audio, video and chat. User presence plays an important role. We need to have the system up and running always and avoid system crashes and restart. If it happens we are going to loose all information regarding the users who were online. Need Suggestions what are the best approaches to avoid such situations.Als
nickd717
ios ipad webrtc avcapture opentok
I have a beta tester getting this error when trying to start a video session on our iPad app using the OpenTok iOS WebRTC SDK. The app crashes every time. User is on an iPad 2 with iOS 6.1.3. Does anyone have a clue what could be causing such an error? My best guess is something involving camera access, but I’m at a loss for what it could be.
user2773279
ios webrtc
I just downloaded the lukeweber version of webrtcjingle, which I believe is just a branch of the google one. I change the server and account setting on the demo and run it. It was able to log in and send/receive messages. However, When I tried to logout. The app crash.I had a little debug on it, and found out the line of code that crash was in file txmppump.cc, function:void TXmppPump::WakeTasks() {talk_base::Thread::Current()->Post(this);} Does anyone have an idea what the problem is? I not
Tako
node.js socket.io webrtc
I’m actually working on an application based on NodeJS, ws, WebRTC.io and socket.io. The aim of this application is to broadcast the video of an Host to multiple attendees.The host and all the attendees load the web application, then the host launch his video and stream it to all the attendees. For this case I don’t have any problem, but if the host launch his video, then an attendee join the room, this attendee is not able to retrieve the stream, and I don’t how to get this stream.I’ve tried ma
user1735617
node.js express socket.io webrtc video-conferencing
first sorry for my bad english. I’m trying to make a video call with WebRTC but it doesn’t work. I’m using node.js+socket.io+express for the server. I appreciate any help or suggestions you can give me. Thank you very much. This is my code.SERVERvar express = require(‘express’), io = require(‘socket.io’), app = express(), server = require(‘http’).createServer(app), io = io.listen(server);server.listen(8080);app.get(‘/’, function (req, res) {res.sendfile(__dirname + ‘/index.html’); });var usuari
ecr
php webrtc livechat opentok
Recently I tried to use OpenTok’s WebRTC demo for a web application requiring peer-to-peer video/audio communication. I went through the tutorials on http://www.tokbox.com/opentok/webrtc/docs/js/tutorials/ and http://www.tokbox.com/opentok/docs/server/server_side_libraries.html and I got something working using JavaScript (client-side) and PHP’s library (server-side) for token/session generation. I managed to code a basic peer-to-peer chat room locally. However, as soon as I upload it to a web s
vaibhav
video html5-video livechat webrtc
i’m trying out a simple video chat in a single web page using webrtc and am getting the following error Uncaught Error: SYNTAX_ERR: DOM Exception 12my code is<button id=’btn2′ onclick=’call();’>call</button>…… function call(){pc1 = new webkitPeerConnection00(null,icecallback1);console.log(‘created local peer connection object pc1’);pc2 = new webkitPeerConnection00(null,icecallback2);console.log(‘created local peer connection object pc2’);pc2.onaddstream = gotremotestream;pc1.adds
lrussell
webrtc channel peer
I’m trying to connect 2 peers with webrtc and datachannel without camera and microphone.try {socket = new WebSocket(“ws://localhost:1337/”);var servers = {iceServers:[{url:”stun:stun.l.google.com:19302″}]};peerConn = new webkitRTCPeerConnection(servers, {optional:[{RtpDataChannels: true}]});channel = peerConn.createDataChannel(“abcd1234”, {reliable: false});peerConn.onicecandidate = function(evt) {if(evt.candidate) {socket.send(JSON.stringify({“candidate”: evt.candidate}));}};channel.onopen = fu
b3by
javascript dom binary-data webrtc arraybuffer
I’m trying to send some data over a datachannel between two browsers. In my code I have:var asString = JSON.stringify(this); var buffer = new ArrayBuffer(asString.length*2); // 2 bytes for each char var bufView = new Uint16Array(buffer); for (var i=0; i<asString.length; i++) {bufView[i] = asString.charCodeAt(i); }channel.send(buffer);In this example, datachannel is open and correctly working (its type is WebKitDatachannel). When I send it, i receive this exception:Uncaught Error: SyntaxError:
vaibhav
javascript google-app-engine channel-api webrtc videochat
I want to create an app like https://apprtc.appspot.com/?r=04188292. I know a little bit about webrtc but cannot get a hold of google app-engine.How to set it up for javascript and start coding??I tried this but it did not work<script src=”https://talkgadget.google.com/talkgadget/channel.js”></script> //for inclusion of channel api … console.log(create_channel(1)); //gave an error “Uncaught ReferenceError: create_channel is not definedAny tutorial/help would be appreciated Is there
user1424120
google-chrome webcam webrtc
When I try and record webcam video using an instance of LocalMediaStream as per the WebRTC spec, I get the following error: Uncaught TypeError: Object #<LocalMediaStream> has no method ‘record’ I have tried stable/beta/dev & canary builds of Google Chrome.Is it currently possible to record webcam video using any currently available version of Google Chrome?
konga raju
webrtc
I work on media server by using part of webrtc technology. I create a local peerconnection as a role of VideoReceiver(Answerer), and it needn’t offer any media stream, and no data channel, just setRemoteDescription and ICE candidates. So I supposed it could be work just by createObjectUrl when onaddstream called, but it failed as the states of peerconnection always ‘have-remote-offer’, so what’s wrong with following code: `var d = eval(“(” + data + “)”);if (d.sdp && d.type == “offer”) {t
AbigailW
meteor webrtc opentok
Got a tough one here. So, we’re trying to upgrade an OpenTok video chat application from Flash to WebRTC, and are running into socket errors as we try to implement the ‘helloworld’ WebRTC sample. The errors occur when we try to do a session.connect() call, not when we request a sessionId or a token. And the error basically looks like this (session_id and partner_id anonymized):SessionInfo Response: #document<sessions>?<Session>?<session_id>?asfgdagbasdfovnwoinvcwoinvoiandfvoi
AbigailW
webrtc opentok tokbox
So, we have a WebRTC enabled app that is actually working in the sense that it’s establishing sessions, we’re able to connect to them, people are publishing streams, other people are subscribing and unsubscribing to those streams, etc. etc. But we keep getting errors from WebRTC which say ‘PeerConnection.getStats. Already getting the stats!’ The error occurs in the video stream itself. White text on a black background, instead of the video. Also, in the console log, we’re getting the following
Madis Nõmme
webrtc getusermedia
Background:I’m developing a system, part of which is a WebRTC video (or audio) calling. The solution includes quite a bit business logic built on top of what the getUserMedia and RTCPeerConnection API-s support. Especially since Firefox and Chrome implement some of the API-s differently(e.g. MediaStream#getAudioTracks always returns [] on Firefox). Example of a requirement to test:Two parties have a running audio-audio call (both sides have been asked and allowed audio stream(access to microphon
Rampart
google-chrome p2p webrtc
I want to make a video platform. I am experimenting with WebRTC, running nodejs as a server. Now the problem is that I have Chrome 21 in Ubuntu goes as it should, no errors at all, but in Chrome 23 (in Windows), I hace an error at client side.Here is my codeif(typeof webkitPeerConnection === ‘function’) pc = new webkitPeerConnection(“NONE”, onSignalingMessage); else pc = new webkitDeprecatedPeerConnection(“NONE”, onSignalingMessage); The error happens to try to use the function webkitDeprecat
Arjun Bajaj
html5 node.js websocket socket.io webrtc
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
mark
javascript html5 html5-video webrtc
I’m on Chrome 25 successfully using getUserMedia and RTCPeerConnection to connect audio from a web page to another party, but I’m unable to get the API to stop the red blinking indication icon in the Chrome tab that media is being used on that page. My question is essentially a duplicate of Stop/Close webcam which is opened by navigator.getUserMedia except that the resolution there isn’t working. If I have a page that just uses getUserMedia with no remote media (no peer), then stopping the cam
Yotam Omer
javascript webrtc
I got a RTCPeerConnection, when the connection is established I want to disconnect from the web-server.How do I check for an established connection?readyState was always undefined and onopen never fired.
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