Questions tagged [webrtc]

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Why we need crossbar.io (WAMP) when there is WebRTC?

As far as I understand, crossbar provides transparent connection between clients. So clients can communicate with each other. WebRTC does almost same thing, without a central server(apart from signalling server) with lower latency. So why we need crossbar in the first place? Am I missing something?
Ramazan Polat
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How does a TURN server send data to peers behind a NAT?

I understand in TURN there is a client, a TURN server, and peer(s). The client itself creates an Allocation on the TURN server, after which a relayed transport address is returned back to the client in order to send data to one or many peers. I understand how the client can get around its NAT, howev...
Kris
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Using Restcomm SIP & Webrtc without the platform

I am looking for a way to use SIP as signalling protocol for Webrtc in Android. I saw the RestComm open source code and I think it fits my needs. But I already have the infrastructure setup for TURN and SIP server. We use FreeSwitch for that purpose. My Question is, is it possible to use the restcom...
vivek1794
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Kurento Hello World, not returning loopback

I am trying to setup the basic hello world example. I am using a t2.micro instance with Ubuntu 14.04 LTS and I have the github code for the example on my local machine. The url I am using is: https://localhost:8443/index.html?ws_uri=wss://ec2INSTANCE:8888&ice_servers=[{"urls":"stun:stun.l.google.com...
MikeL
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WebRTC cannot record screen

I'm trying to make screen sharing app using WebRTC. I have code that can get and share video stream from camera. I need to modify it to instead get video via MediaProjection API. Based on this post I have modified my code to use org.webrtc.ScreenCapturerAndroid, but there is no video output shown. T...
Kubik
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WebRTC channel reliability

I'd like to check my understanding of WebRTC data channels is correct, in particular the different types of channels that can be achieved by varying the ordered & maxRetransmits or maxPacketLifeTime properties to the RTCDataChannelInit dictionary. Are my below assumptions correct: Creates a reliable...
dbotha
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Getting imagens in second peer using WebRTC (whithout a server)

I'm creating one direction chat, but I can't receive the imagens in second peer. My application have the following code: const video = document.querySelector('video') const caller = async () => { // --- // const localStream = await navigator.mediaDevices.getUserMedia({ video: true }) // --- // v...
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Kurento WebRTC Connection Fails in ~30% of Cases

I have spent days hunting down a connection problem without any luck. I'm trying to implement a relatively simple one2one Call with Kurento. Below you will find a debug log of Kurento of a case where the connection could be established and a case where the connection failed. If you need any more log...
omnibrain
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nodejitsu deploy of webrtc demo failing to server client-side javascript

I'm trying to deploy to a simple WebRTC.io demo app (https://github.com/dougnukem/webrtc.io-demo) to nodejitsu, but when it's deployed (http://dougnukemwebrtc.jit.su/) it fails to serve the client-side javascript: http://dougnukemwebrtc.jit.su/webrtcio.js Cannot GET /webrtcio.js It runs fine when I'...
Dougnukem
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480

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cannot enable RTCDataChannel on chrome

my chrome version is 26.0.1410.0 canary. I typed "chrome://flags/" in the browser, and try to find RTCDataChannel. this flag is not on the list. is this flag removed or I need to get newer version?
qinking126
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How to create online users list using webrtc and nodejs on the server end

I am using webrtc to make a audio, video and chat application where I need keep all the users in a user list in the serverside. Need help how to get this done. Also, how can I remove users from the list when they logout from the system. Need help to implement this. webRTC.rtc.on('connect', function(...
Jeet
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HTML5 Sound Recording in Firefox

I have been trying to record audio through browsers using HTML5 and was successful in doing that in chrome. I wanted to know if there was any such API that supports audio recording in firefox. I tried executing the code used in chrome in firefox but was not able to since firefox does not support Aud...
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Asterisk instantly terminates WebRTC (JSSIP) call

I'm running Asterisk 11.2.2 with SRTP and STUN support under Calculate Linux (Gentoo-based distribution). When I try to call from one WebRTC instance to another, using JSSIP, the call passes, but if i answer it on another instance, the call suddenly terminates. Using Asterisk debug mode, i can catc...
JustLogin
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126

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Controlling the size of uiVideoPanel

I'm using the vLine JavaScript API with the uiVideoPanel widget to make WebRTC video calls. My code looks like this: window.vlineClient = client_ = vline.Client.create({"serviceId": serviceId, "ui": true, 'uiVideoPanel':'showvideo' }); and HTML: However when one person calls another, the video take...
whitebear
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TB.Socket error with OpenTok WebRTC on Meteor

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 toke...
AbigailW
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multiple webRTC connections

Can we technically open multiple webRTC connections from a browser to a server? Or each instance of a browser can make only one single webRTC connections to a specific server? I am thinking to use webworkers to create multiple thread from each of which establish a separate webRTC connection to a sp...
C graphics
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Web Audio API and WebRTC

Is it possible to getUserMedia... take the stream and process it with Web Audio API to run it through some filters and then send the processed audio to a peer using WebRTC? I have been scouring the web for such examples. I found plenty that use Web Audio API after receiving a peer's stream but none...
Sophie Säschön
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How to determine a webrtc mediastream only has audio?

When I receive an onaddstream event from the other side, how can I determine the MediaStream only has audio, no video? In other words, can I know a MediaStream object only has audio, no video?
Gisway
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Build Talk [libjingle] from WebRTC source for Android

I've the code checkeout from https://code.google.com/p/webrtc/source/browse/trunk/talk I can't seem to find how to build native libraries for ARM (android). I need these libraries in (jni/.so) in the Android example project: https://code.google.com/p/webrtc/source/browse/trunk/talk#talk%2Fexamples%2...
Taranfx
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DTLS Support for WebRTC iOS Libraries

I am trying to create an iOS application using the compiled WebRTC iOS libraries from http://webrtc.googlecode.com/svn/trunk/ and currently my backend server supports only DTLS. Now when I try to set the remote description it returns the following error Warning(webrtcsession.cc:146): Session descri...
Rajaraman
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1.4k

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webrtc works in local network and but on remote network?

I have created a quick video chat with webrtc and nodejs. Problem is : it works great on local network. But not on remote network. Any clue where I should have a look ? regards Source code: 'use strict'; var sendChannel; ///////////////////////////////////////////// String.prototype.replaceAll = fun...
yarek
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3

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6.8k

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Dummy video device - v4l2loopback - webRTC

I need to play multiple video for test a video server. I'm using lubuntu 14.04 and have installed V4l2loopback to make the device file ( /dev/videoN ) I am using mplayer to play video from this device as described mplayer cam I have done the modify to the source code and successfully played the vide...
Al3
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Is RTCDataChannel send() a synchronous/blocking call?

I'm trying to send a file in chunks over WebRTC, and I'm wondering if I can create a callback function to be called after RTCDataChannel.send() finishes sending each chunk of the file. Is RTCDataChannel.send() a synchronous/blocking call? If so, my callback can be executed on the line after .send()....
Hristo
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Your experience with LibSourcey

Just came across libsourcey when Googling libuv. It looks very featureful, but I haven't been able to find any solid reviews or benchmarks (in fact, this seems to be the first post here on stack overflow concerning LibSourcey.) Has anybody yet to have any experience with this framework?
JSON
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XirSys When do I request new Ice servers?

So I completely understand how to use getIceServers via your demo, but what's the best practice for implementing on the server side / compiled client-side? "This token should only be implemented in a secure environment, such as a server-side application or a compiled client-side application." Do the...
GreenEggs
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Resolution constraints to getUserMedia() not working properly (WebRTC)

I'm new to WebRTC and testing a simple application for video streaming in Chrome. I have three different types of constraints with the following resolutions: qvga: 320 x 240, vga: 640 x 480, hdVga: 1280 x 720. When I capture media it runs fine. But when I start with the qvga resolution and click on...
Faisal Naseer
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Can I use web rtc to write a web server in Javascript?

Can I use webrtc to write a web server in Javascript so that someone could turn their web browser into a webserver by just visiting a URL like becomeawebserver.com (example name I just made up)? I am just trying to learn web rtc so am trying to find out what it is capable of Update I have my answer...
Zubair
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Is Google Chromecast mirroring done through WebRTC?

Read from here: http://giovanni.bajo.it/post/60352255570/airplay-vs-chromecast-breakdown-and-differences it says: in fact, the basic video support is just a simple html with a tag, while mirroring is being done through WebRTC. is this true?
Dave Xu
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How to save webRTC opus audio stream on server side using nodejs?

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 client is encoded the...
Sina Sharafzadeh
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4

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1.6k

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How can I transport external video stream, instead of the webcam on webrtc session?

I'm working over RTCPeerConnection js. I want to broadcast ip cam stream(M3U/HLS) or any video stream on android platform with webrtc. I dont know how to attach external stream to webrtc session. Is it possible with webrtc or another realtime protocol api?
Gnosis00
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can access webcam but cannot display it into div

This is a simple script I use: #myVideo { background-color:#CCC; position:absolute; width:320px; height:240px; border:1px solid #000; } navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia; navigator.getUserMedia({audio: true, video: true},...
yarek
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Button to Toggle Mute or Unmute of Microphone

I am only starting to learn WebRTC, but I can't get to mute the microphone. This is what I want: First, it will try to access the microphone. Then, there will be a button to toggle the mute or unmute of the microphone. This is what I had tried: Code 1: new MediaStream(); var constraints = track.get...
Lloyd Dominic
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WebRtc on slow connection

i have already used webrtc and socketIo on my project. Normally they are working well, however when the connection is slow down, seems two browsers can connect, but the webcam or voice does not show! I don't know how to detect that bugs and tell my users that "your connection is slow"! I used speedt...
songpham
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what are disadvantages of having two PeerConnections for one call?

I am thinking of changing my application from using a single PeerConnection for transferring media both ways to one PeerConnection for upstream and one for downstream for a single call between two peer. The advantages I foresee: Less worry about signalling state of PeerConnection when changing offer...
mido
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create application WebRTC [closed]

how create a WebRTC (video, audio & message) chat via socket.io on this application you can: talked to specify user. talked with several users at the same time (notion of room). the user can accept or reject the call.
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What is the role of IceConnectionState and IceGatheringState in webrtc

How to make use of IceConnectionState and IceGatheringState while sending the icecandiates to the peer in webrtc ?
Vinay
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3

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1.5k

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How does SDP in webrtc know candidate's ip?

Candidate attribute in a SDP provides connection address of the candidate. Which looks like this: a=candidate:4022866446 1 udp 2113937151 192.168.0.197 36768 typ host I want to know how does sdp collects information about my local ip. It'd be great help if there's any specific webrtc code to look at...
Anirban
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Webrtc without websockets

I am trying to test sip capabilities of firewalls using webrtc. However I noticed using the servers needed for webrtc (stun turn websockets etc.) will give me a false positive in that it won’t catch nuanced issues with the ALGs. For reference this is being done from a chrome app so I can’t just...
johnnyD97
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How to remove audio video call icon from firefox and chrome in getusermedia

How to remove audio video call icon from firefox and chrome in getusermedia. var constraints = { video: { mandatory: {} }, audio: { mandatory: {} } }; navigator.getUserMedia_ = (navigator.getUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia); console.log(constraints); navigator.getU...
Mustafa bw
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Kurento example error

I am trying the tutorial of kureno with http://doc-kurento.readthedocs.org/en/stable/tutorials/java/tutorial-1-helloworld.html I follow the steps as in this example,like bellow git clone https://github.com/Kurento/kurento-tutorial-java.git cd kurento-tutorial-java/kurento-hello-world git checkout 6...
Shijin

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