I have my own server with ubuntu 19. NOTE: I need a stun/turn server for a video chat web application. It works already if both devices are in the same network. I have successfully installed and ...

I have read from here that how i can mute/unmute mic for a localstream in webrtc:WebRTC Tips & Tricks When i start my localstream mic is enable at that time by default so when i set audioTrac...

We are trying to make a conference call with multiple users, So by using Kurento server we have achieved this and it's working on safari browser. But when it comes to implementation in WebView / WK...

Now I am working on a peer-to-peer chatting system based on WebRTC. This system can make a pair with any person who is listening on the peer list at the same time and I have finished the basic func...

I have a MediaStreamSource sourced from navigator.getUserMedia, which is connected to a GainNode, which in turn is connected to a ScriptProcessorNode, like so: [getUserMediaStream] -> [MediaStr...

I generate webm files in two different ways. One using Chrome WebRTC MediaRecorder, the other one is using a js library which generates the webm video frame by frame (webm-writer-js). The file size...

Friends, I want to develop an application for Flutter using webrtc, but unfortunately I am facing problems that I am not able to solve. errorCode: AddStream is not available with Unified Plan SdpSe...

I am trying to figure out how to test whether a STUN/TURN server is alive and properly responding to connections. Ideally this test would be performed from an external machine, just in case the STU...

I'm trying to create a Chrome extension that captures the audio from the active tab and either sends it to another server or makes it accessible via a URL. I'm using the chrome.tabCapture.capture ...

I wish to play audio from remote stream of webrtc connection, in Flutter. The examples of flutter webrtc use RTCVideoRenderer, but there is no video in my case. The remote stream only consists of a...

I'm trying to get highest video resolution as possible through JS navigator.getUserMedia. I know about constraints, but don't know how to choose right in my case. The problem is looks like there i...

What's the most efficient way to acquire the webcam stream directly into WebGL as a texture? The standard process (as documented in Three.js) is to use the video tag, then manipulate it in a canva...

I newly wrote a simple chat application, but I didn't really understand the background of ICE Candidates. When the peer create a connection they get ICE Candidates and they exchange them and...

I am encountering an issue with my WebRTC implementation in Django using Django Channels. The error I am facing is: "Uncaught (in promise) DOMException: Failed to execute 'setRemoteDescription...

I'm currently trying to get a clean as possible audio channel via webrtc. Via the getUserMedia mediaconstraints object, I've set the following options: constraints: { audio: { mandatory: { echo...

we are fiddling around with WebRTC in our company. And i stumbled upon a weird thing, which i'm not sure is by design in WebRTC or an implementaiton error on our side. We currently have a simple W...

I am using the navigator function - navigator.mediaDevices.enumerateDevices() to get the list of media devices, in my mobile browser application. I am using the below code: navigator.mediaDevices....

I created a .net core chat application using SignalR and I used WebRTC for video calls. As I need to send the SDP to the receiver using server method so I created a hub method call "SendOffer". Whe...

I'm developing a FileShare application with webRTC. I want to implement the client in JavaScript/HTML. The code should be run on the clients browser. I need to save them when downloaded via webRTC....

I am building a react native application using Expo and i need to add audio calling and video calling functionalities, however it seems that i cant use react-native-webrtc (Link here) without eject...

I need some help. What is the best way to set up LIVE STREAMING over the web from my WEBCAM to the server and back to multiple users? Essentially I'm trying to create a group video chat application...

I am trying to use WebRTC to build a web application that needs to pause/resume the video/audio stream when some events trigger. I have tried the getTracks()[0].stop() but I am not sure how to resu...

I have been trying to get the webrtc logs for my web application running over safari browser, similar to what we get in firefox by going to the page about:webrtc and on chrome by using chrome://web...

all I'm in peer to peer communication using webRTC , we have media stream object from the getUserMedia which is given as input stream to peerconnection. Here I need video stream from the selected ...

According to RTCPeerConnection.ontrack documentation, "ontrack" event suppose to fire for each incoming streams. I have a PeerConnection with two video streams, after connection, "on...