FAQ Database Discussion Community


Automated testing of webrtc application?

javascript,testing,automated-tests,mocha,webrtc
I am developing a conferencing app, details: target: chrome browser server: node.js ( currently windows env) simplest test scenario would be: open two browser tabs( open browser if need be) emulate button click on both. emulate accept getUserMedia request( hardest part) more emulation stuff and reading JavaScript variable values and...

Where can I find official and up-to-date documentation for the WebRTC API?

google-chrome,firefox,documentation,webrtc
I know about adapter.js, which tries to: insulate apps from spec changes and prefix differences. But adpater.js only covers the very basic WebRTC API's. I'll just use setRemoteDescription as an example. In 2013 it was called like this: pc.setRemoteDescription(offer); According to https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection, the current API is? pc.setRemoteDescription(new RTCSessionDescription(offer), function() {...

How does OpenGL render YUV video on Android?

java,android,opengl-es-2.0,webrtc
I am working on some webrtc staff on Android and trying to figure out how does VideoRendererGui.java works. Unfortunately, I have some trouble with understanding how the following OpenGL code work: private final String VERTEX_SHADER_STRING = "varying vec2 interp_tc;\n" + "attribute vec4 in_pos;\n" + "attribute vec2 in_tc;\n" + "\n" +...

Multi peer connection in webrtc

javascript,webrtc,p2p
I use array pc to save RTCPeerConnections from 1 client to other clients then call createOffer. pc is global variable but have an error: inside createOffer function, I get pc[list[i]] is undefined. How do I fix it? for (var i = 0; i<list.length - 1; i++) { pc[list[i]] = createPC(list[i]);...

Why is null almost always passed to RTCPeerConnection?

javascript,html5,webrtc
Nearly every example of WebRTC I've seen on the Internet involves creating a RTCPeerConnection with a single parameter of null. The spec calls for two parameters to be passed: an ICE server configuration and your media constraints. I've noticed that technically everything still works if the two requests are coming...

Why does nothing happen when calling createAnswer?

java,android,webrtc
I have this class I use as SdpObserver: public class CallFragmentObserverCategory implements SdpObserver { public void onCreateSuccess(org.webrtc.SessionDescription sessionDescription) { parent.peerConnection.setLocalDescription(this, sessionDescription); //I send the sdp to the remote peer here as well (on a worker thread) } public void onSetSuccess() { } public void onCreateFailure(String s) { localDatabase.log(s, ERROR); //ERROR...

Android WebRTC DataChannel binary transfer mode

android,file,webrtc,file-transfer,rtcdatachannel
I achieved to transfer data between two android phones using WebRTC's DataChannel : On one side, I send the data: boolean isBinaryFile = false; File file = new File(path); // let's assume path is a .whatever file's path (txt, jpg, pdf..) ByteBuffer byteBuffer = ByteBuffer.wrap(convertFileToByteArray(file)); DataChannel.Buffer buf = new DataChannel.Buffer(byteBuffer,...

Why can't webrtc find the capturer it suggested?

c++,windows,winapi,webrtc
I'm creating a webrtc-based voip app for windows in C++. I'm trying to initialize a peerconnection. I'm stuck at the part to fetch a camera. I'm using the following code to find a camera to start streaming media from (copied from the peerconnection client example): rtc::scoped_ptr<cricket::DeviceManagerInterface> dev_manager(cricket::DeviceManagerFactory::Create()); if (!dev_manager->Init()) {...

WebRTC firefox constraints

javascript,firefox,webrtc
I currently use WebRTC in my personal development, everything works fine. I get the stream from my webcam, but now I want to use constraints for getUserMedia(). var constraints = { audio: false, video: { mandatory : { minWidth: 1280, minHeight: 720 } } }; When I test this in...

webrtc - Disable the use of libcmt

c++,webrtc
I'm using webrtc and I want to avoid using libcmt because an error in linking is reported. This is the error: LIBCMT.lib(invarg.obj) : error LNK2005: __invalid_parameter already defined in MSVCRTD.lib(MSVCR120D.dll) I have read this in common.gypi but I don't know how to perform that because I cannot find where include.gypi...

Is it possible to check if the user has a camera and microphone and if the permissions have been granted with Javascript?

javascript,html5,html5-video,webrtc,html5-audio
I would like to find out if the user's device has an attached camera and microphone, and if so, has permissions been granted to get the audio and video stream using Javascript. I want to make this check to be made across Chrome and Firefox at the very least. What's...

WebRTC: Unable to successfully complete signalling process using DataChannel

javascript,webrtc
I've been having trouble establishing a WebRTC session and am trying to simplify the issue as much as possible. So I've written up a simple copy & paste example, where you just paste the offer/answer into webforms and click submit. The HTML+JS, all in one file, can be found here:...

Rails Signalling Server

ruby-on-rails,signals,server,message,webrtc
I am creating an app with Ruby on Rails (4.2). I want to add a WebRTC functionality to it so that two users can choose to videochat with each other. I'd appreciate if you can help with: What are some of the options to create a signalling server with Rails...

Why does video resolution change when streaming from Android via WebRTC

android,google-chrome,video,webrtc,video-processing
I'm trying to stream at 640x480 from Chrome on Android using WebRTC, and the video starts off at that, but then the resolution drops to 320x240. Here are the getUserMedia parameters that are sent: "getUserMedia": [ { "origin": "http://webrtc.example.com:3001", "pid": 30062, "rid": 15, "video": "mandatory: {minWidth:640, maxWidth:640, minHeight:480, maxHeight:480}" }...

WebRTC Video Renderer

c++,winapi,webrtc
We're busy trying to render the frames we get from WebRTC, but we are having problems showing the video correctly. Does anyone have experience in this or is there an guide online that shows how to render frames from WebRTC in win32? We are building out application in Visual Studio...

How to send contents of a webrtc stream directly to my server?

javascript,websocket,webrtc
I've recently started getting into webRTC and would like to stream my webcam to my web server. However, I can only seem to find concrete examples of doing this peer to peer. I know the very basics: navigator.getUserMedia(constraints, successCallback, errorCallback); function successCallback(stream) { // I want to send the output...

RecordRTC with custom sample rate records silence

javascript,angularjs,webrtc,getusermedia
I'm trying to use RecordRTC.js to record audio from a microphone and upload it to a nancyfx server. For testing purposes, I'm just trying to upload the audio stream and save it to a wav file. My requirement is, however, that the stream is saved in 16 bits at 22050Hz....

Quickblox Group Video Calls with Javascript SDK

javascript,webrtc,quickblox
The documentation for group video calls is missing on Quickblox website : Group video / voice calls - Is coming soon - http://quickblox.com/developers/Sample-webrtc-web I have a workaround but it's not a viable solution when there are more than 4 users. For 3 user conference: User 1 calls User 2 and...

Is there a way to access a websever behind NAT?

webrtc,nat,stun,turn
I am trying to access a web server behind NAT. The challenge is: because there could be several web servers coexist, router registration based on port is not a feasible solution here. I apologize this looks more like general question, because I really don't know where to start after reading...

event driven pattern for writing chunks to a file - JS

javascript,design-patterns,file-io,webrtc,event-driven-design
I'm trying to transfer a file over WebRTC, and I'm struggling to figure out a good pattern for writing data as it's coming in. Since file chunks will be coming in at an unknown rate, I need to be able to write each chunk as it becomes available; this means...

Attach data to simpleWebRTC room or video

javascript,php,video,webrtc
I'm using simpleWebRTC for a multi-party video chat. Each user creates his own room. When a user subscribes to another (in order to see this person 0 and registers in the db). When a user chooses another - he gets this users username by XHR from the db. I tried...

RTCDataChannel with Google Channel API

javascript,google-app-engine,webrtc,channel-api,rtcdatachannel
I'm trying to follow this example by Dan Ristic for RTCDataChannel browser p2p communication with Google's Channel API for signaling. It seems to be failing silently - I can't get the RTCDataChannel.onopen, RTCPeerConnection.onicecandidate, or RTCPeerConnection.ondatachannel events to fire. Client JS/HTML: <html> <head> <script src="https://code.jquery.com/jquery-1.11.2.min.js"></script> <script type="text/javascript" src="/_ah/channel/jsapi"></script> <script> $(document).ready(function(){ var...

Clarification about TURN server authentication through REST api

webrtc,turn
I was going through this draft to undertstand usage of REST api to access TURN servics. I am bit confused after going through that. Currently, I am authenicating my TURN server using Long Term Credential Mechanism with Redis database, but instead of using actual username and password, I am using...

WebRTC: View self-view while muting outgoing video in a call

webrtc
Currently, the video mute functionality in webrtc is achieved by setting the enabled property of a video track to false stream.getVideoTracks().forEach(function (track) { track.enabled = false; }); But the above code would not only mute the outgoing video, but the local self-view which is rendered using that local stream, also...

unable to configure apprtc.appspot with own url

python,google-bigquery,webrtc,apprtcdemo
This is the error I get when I try to configure apprtc with my own url. I tried to set up my own Turn Server and also tried to give a client url but it still did not work . <HttpError 404 when requesting https://www.googleapis.com/bigquery/v2/projects/esuioswebrtc/datasets/prod/tables/analytics/insertAll?alt=json returned "Not Found: Table esuioswebrtc:prod.analytics">...

iOS webRTC library supporting both armv7 & arm64

ios,webrtc,armv7,arm64
How can I get the webRTC library which will support for both armv7 & arm64 in iOS?

Minimal WebRTC for native application without audio and video

webrtc,libjingle
I am interested in designing a WebRTC/libjingle that uses DataChannels but does not use the audio and video capability. The audio and video capability adds a lot of dependencies that are large and difficult to cross compile. Is there a minimal subset of the WebRTC build that will separate out...

RecordRTC video upload AmazonS3 Timeout error

javascript,amazon-s3,html5-video,webrtc,video-capture
I am currently developing a component that allows you to make webcam videos and upload them directly to amazon s3. For that purpose I user RecordRTC library and Amazon S3 storage. I have discovered a strange issue, and I am not sure whether it has to do with RecordRTC blobs...

createMediaElementSource plays but getByteFrequencyData returns all 0's

webrtc,web-audio
I am attempting to visualize audio coming out of an element on a webpage. The source for that element is a WebRTC stream connecting to an Asterisk call via sip.js. The audio works as intended. However, when I attempt to get the frequency data using web audio api, it returns...

can't add Remote session description in webrtc android client

java,android,webrtc,libjingle
Response from server: { "rtcid": "wKAm8eeyI-mQ5dsslkhu", "msgType": "offer", "senderrtcid": "53wp_LP5CYDie3eIlkhw", "msgData": { "type": "offer", "sdp": "v=0\r\no=- 951920257545056255 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS OfkjcHABgxUkHlk8mfJ8ayYZdCHqdpQGFSTM\r\nm=audio 1 RTP/SAVPF 111 103 104 0 8 106 105 13 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:1 IN IP4 0.0.0.0\r\na=ice-ufrag:CF4q+RW54gQVPaz0\r\na=ice-pwd:hEIbgX4MME6cPkZKGih7bjQM\r\na=ice-options:google-ice\r\na=fingerprint:sha-256...

Using WebRTC with Socket.io

socket.io,webrtc
I'm trying to create an app for making audio calls in the browser. I found this tutorial and started using it as a basis: https://www.webrtc-experiment.com/docs/WebRTC-PeerConnection.html After some tweaking to fit my needs. I ended up with the following: var iceServers = [ { url: 'stun:stun1.l.google.com:19302' }, { url: 'turn:numb.viagenie.ca', credential:...

WebRTC streams freeze after first frame

javascript,google-chrome,video-streaming,webrtc,chromium
Here's the deal, I've got a WebRTC 1 on 1 conversation using: SimpleWebRTC library CoTurn server Signaling server Everything seems to work fine, but there is one problem: Chrom* browsers display only first frame of the video and then the video freezes, as well as audio. Looking at the Chromium...

Creating meta-data for binary chunks for sending via WebRTC datachannel

javascript,browser,blob,webrtc,rtcdatachannel
I have a datachannel connection between two browsers, and would like to break a file into chunks and send them to/from the clients. I can read the file and break it up into chunks just fine. However I need a way for the receiving client to know which file the...

How to save webRTC opus audio stream on server side using nodejs?

node.js,webrtc,opus
There are some solutions to save a raw usermedia audio stream on the server side but I want to save the webRTC encoded stream which has low channel bandwidth transmission. I think of a solution that I'm not sure about: Connect server and client using webRTC, the stream from the...

Capture image with webRTC + canvas via jQuery

jquery,html5-canvas,webrtc
When I execute this code, I get this error message: Uncaught TypeError: Failed to execute 'drawImage' on 'CanvasRenderingContext2D': The provided value is not of type '(HTMLImageElement or HTMLVideoElement or HTMLCanvasElement or ImageBitmap) $(document).ready(function () { window.URL = window.URL || window.webkitURL || window.mozURL || window.msURL; navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia ||...

PeerJS/WebRTC connection fails on rapid chunks transmittion

javascript,webrtc,file-sharing,chunks,peerjs
I'm using PeerJS, but thought that this problem can be about WebRTC in general, hope You can help me out: I'm trying to write a simple peer-to-peer file sharing. I'm using serialisation: "none" for PeerJS connection DataChannel, as I'm sending just pure ArrayBuffers. Everything is good with files around 10mb...

how to convert getUsermedia audio stream into a blob or buffer?

javascript,buffer,blob,webrtc,audio-streaming
I am getting audio stream from getUserMeda and then convert it into a blob or buffer and send it to server as audio is comming I am using socket.io to emit it to server how can i convert audio mediastream into buffer? Following is the code that i have written...

WebRTC Failed to Add a Third Peer: “Cannot set remote answer in state stable”

javascript,html5,websocket,webrtc
I am writing a multi-peer WebRTC video chat. Two peers have no trouble connecting, no error or warning in console, and video works well, but I cannot add a third party to the chat successfully. On the host (the first participant, Firefox), the error appear as "Cannot set remote answer...

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

javascript,node.js,socket.io,webrtc,node-static
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...

Recommended WebRTC Server Configuration for Native app (iOS/Android)?

webrtc,server
I tried to build a server for integrating the webrtc native APIs in an native app, but I am not sure about how the server should be configured, like the ICE/STUN/TURN, signaling, media server etc.. So far as I know is the open source project: https://github.com/priologic/easyrtc Can anybody give some...

Unsubscribe my own video in opentok

webrtc,opentok,tokbox
Hi I am using opentok. When I say publisher = OT.initPublisher(); session.publish(publisher); My own video is visible to myself. I want to see only other participants video but not my own. I want my video to be visible to everyone else in the session except me. How to make this...

How to debug Mobicents Media Server

debugging,server,media,webrtc,mobicents
I'm using RestComm Client iOS SDK to make a call to my RestComm instance deployed locally, but I'm having issues setting up a WebRTC call (something seems to be wrong with the STUN pings as part of ICE). How can I attach the eclipse debugger on the RestComm Media Server?...

Get file type after sending the file from WebRTC

javascript,download,webrtc,filereader
I am using WebRTC to get two clients communicated using peer.js var peer = new Peer( { key: "XXX", config: {"XXX": [{ url: "XXXXXXX" }]} }); My main aim is to send file from one client to another. For that I am using following code: $("#box").on("drop", function(e) { e.originalEvent.preventDefault(); var...

peerconnection_client does not show the list of peers

sockets,webrtc,peer-connection
Here I'm trying to run some of samples available in the WebRTC codes... I run the peerconnection_server.exe on my machine (laptop, running Windows 8.1) it successfully runs. I also run 2 clients via the peerconnection_client.exe. On the peerconnection_server I see the following: Server listening on port 8888 New connection... New...

Firefox 37 throwing error when trying to add microphone volume control for WebRTC audio context

firefox,webrtc,html5-audio,audiocontext
Since firefox 37 I cannot add volume control to the input(microphone), i get the error : IndexSizeError: Index or size is negative or greater than the allowed amount It works fine on Chrome. Here is the code sample : var audioContext = new (window.AudioContext || window.webkitAudioContext)(); // define audio context...

Running PhoneRTC Demo

android,webrtc,cordova-plugins,phonertc
I am trying to run the phoneRTC demo, I hae build the android demo app and have a signalling server running but when I run the client app on android all I get is a blank screen. these are the steps I have taken: npm install -g cordova bower grunt-cli...

Security issue with exposing TURN server credentials in WebRTC

javascript,security,google-chrome,webrtc,easyrtc
We are using google public stun server in one of our application in the test environment. And, we are also installed Turn server. The issue is - When we run the app, in the javascript file, we have put username, password and server address of turn server in order to...

WebRTC: Adding Video To Audio Call With No BUNDLE Lines in SDP

javascript,html5,video,html5-video,webrtc
I'm hoping that one of you WebRTC experts can point me in the right direction or tell me if what I am trying to do is feasible/supported in the current Chrome browser or even if Chrome will be supporting this use case in the future. I have searched for an...

WebRTC onicecandidate event

javascript,html5,video,webrtc
I have the following callback for the onicecandidate event of RTCPeerConnection: function iceCallback(event) { if (event.candidate) { var candidate = event.candidate; socSend("candidate", candidate. event.target.id); } } I am able to read the candidate from the event.candidate. But when I try to read the event.target.id, I get an exception, cannot read...

Google TURN server for WebRTC with REST API authentication

javascript,rest,authentication,webrtc,turn
I'm trying to set up the Google TURN server for webRTC from here. I was able to successfully relay my video through this TURN server using a turnuserdb.conf file where I have my username and password (my_user_name:my_password). And on the web client side I used: "iceServers":{[ "url": "turn:my_user_name,@turn_server_ip", "credential":"my_password" }]...

Is it possible to maintain WebRTC connection after disconnecting from the internet after signalling phase?

webrtc
If WebRTC connection is established between two peers in local network, can we maintain it, after losing connection to the internet? It seems possible, as it's peer-to-peer.

WebRTC bandwidth requirements

webrtc,bandwidth
Does anyone know what are WebRTC bandwidth minimal requirements? I'm interested in what are the values with or without video and for different video resolutions. I'm especially interested in a two party conference, but if you know the values per party it's also good. If you have actual metrics is...

Is screen sharing possible in cordova?

android,cordova,webrtc,opentok,screensharing
I am using opentok for video chat and screen sharing. It works perfectly for me on a browser. However my application also run natively using cordova. There is a cordova-plugin for opentok which supports video calls but not screen sharing. I want screen sharing to be implemented in my cordova...

NS_ERROR_UNEXPECTED in FireFox on mozRTCPeerConnection()

firefox,webrtc
I am using adapter.js in my webrtc 1-1 video call application. It works fine on Google Chrome and both peers see each other's video and can hear audio. However, when I run the same application on FireFox, I get the following error on console which comes from adapter.js. NS_ERROR_UNEXPECTED This...

TURN-Server with easyrtc doesn't work

node.js,webrtc,turn,easyrtc
I use easyrtc with node.js. The *****:8080/demos/demo_audio_video_simple.html work correct on the same network. But if i try it from 2 different networks i get only a black screen. After some research I found out, i need a TURN Server, but it doesn't work. // Load required modules var http =...

OpenTok and File Sharing

webrtc,file-sharing,opentok,videochat,tokbox
I am building a video chat website using OpenTok. I have the video and text chat working, (still working on the screen sharing), but I was wondering if anyone could point me in the right direction regarding file sharing? I would like both parties to be able to send files...

How to modify the content of WebRTC MediaStream video track?

javascript,html5,video,browser,webrtc
I use WebRTC in a scenario in which the client video stream is recorded on a third-party server https://tokbox.com/. I would like to put some kind of watermark in the recorded video. Investigation brought me to this page http://w3c.github.io/webrtc-pc/#mediastreamtrack and it seems that it is technically possible since it says...

Full quality MP3 streaming via webRTC

mp3,html5-audio,webrtc,audio-streaming,p2p
I'm interested in webRTC's ability to P2P livestream an mp3 audio from user's machine. Only example, that I found is this: https://webrtc-mp3-stream.herokuapp.com/ from this article http://servicelab.org/2013/07/24/streaming-audio-between-browsers-with-webrtc-and-webaudio/ But, as you can see, the audio quality on receiving side is pretty poor (45kb\sec), is there any way to get a full quality...

Getting Null pointer exception sometimes when creating QBRTCSession with opponents while making calls

android,webrtc,quickblox
Tried to implement webrtc audio/video call in my application using Quickblox SDK version 2.2.1. Implemented QBRTCClientCallback interface on an Android Service Class. Able to make audio/video calls, but consistency is the issue. I'm able to make audio/video calls, but consistency is the issue . Getting NullPointerException sometimes when creating session...

Sending HTML5 audio element content to ajax POST

jquery,ajax,html5,webrtc
I'm trying to give my users a way to record audio files and post them to my server. I'm using Recorder.js for the recording part, and I can have an element populated by the user's (webrtc) recording as a blob generated by the javascript library. I am able to record...

How to use kurento-media-server for audio only stream?

webrtc,kurento
I want to have only audio stream communication between peers , I changed the parts of kurento.utils.js to get only audio stream via getusermedia but it's not working I used this example node-hello-world example WebRtcPeer.prototype.userMediaConstraints = { audio : true, video : { mandatory : { maxWidth : 640, maxFrameRate...

Record webRTC session on server

cordova,webrtc,turn,rfc5766turnserver,kurento
I recently found out about a plugin for cordova called phoneRTC that allows for webrtc implementation. I build and run the demo provided and I am quite happy with the result. Now I want to know how I can record the webrtc sessions on a web server. curently the infrastructure...

Kurento Media WebRTC to RTP

webrtc,rtp,webm,kurento
I am using kurento's master git to make a WebRTC to RTP bridge. MediaPipeline pipeline = kurento.createMediaPipeline(); WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build(); HttpGetEndpoint httpEndpoint=new HttpGetEndpoint.Builder(pipeline).build(); org.kurento.client.Fraction fr= new org.kurento.client.Fraction(1, 30); VideoCaps vc= new VideoCaps(VideoCodec.H264,fr); httpEndpoint.setVideoFormat(vc); AudioCaps ac= new AudioCaps(AudioCodec.PCMU, 65536); httpEndpoint.setAudioFormat(ac);...

iOS Webrtc - Crash in capturing local Video Stream

ios,objective-c,swift,webrtc
I am trying to use webrtc libs from Google's repo. I followed the steps and created an individual project with instructions and code similar to APPRTC and I was able to get it working. I was able to conference between 2 devices. But when I try to integrate with an...