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I want to add WebRTC to my Kotlin project, but when I add it, I have this problem and when I ask any AI, they say change the implementation. Either it doesn't exist any more, or the same error arises ...
yassine larbibi's user avatar
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43 views

I'm trying to test low-latency video streaming using ffmpeg's WHIP protocol to stream to an aiortc-based Python server. The ICE connection completes successfully, but the DTLS handshake fails with &...
jac's user avatar
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0 answers
33 views

I'm developing a VS Code extension that loads Yjs inside a WebView. VS Code WebViews cannot import ES modules due to CSP restrictions, so I cannot use: import * as Y from "https://esm.run/yjs&...
dont stalk's user avatar
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-5 votes
0 answers
71 views

I'm working on a screen sharing app, and I have an incoming 1920x1080px video stream via WebRTC. My problem is that when I resize the browser, the video becomes really blurry as the browser is ...
Daniel's user avatar
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-1 votes
1 answer
48 views

I’m building an SFU in C#, using the SIPSorcery library for handling WebRTC media streams. I need to calculate the exact time difference between an incoming audio RTP packet from client A and a video ...
Matin's user avatar
  • 13
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0 answers
72 views

I'm using the WebRTC example for Android. I find that audioTrack is always used in java layer. By checking the native code, I found that WebRTC native codes would call isLowLatencyInputSupported() by ...
Chandler's user avatar
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1 vote
0 answers
41 views

I am trying to build realtime voice translation react-native application using mediasoup. My doubt is how I can pass the audio stream of webrtc in mediasoup server to audio translation pipeline made ...
Nitin Sharma's user avatar
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0 votes
1 answer
64 views

I'm building a peer-to-peer chat app using WebRTC and a WebSocket signaling server. Everything works fine up to registration — I can connect to the server, and I see "registered" in the ...
kuu huu's user avatar
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1 vote
0 answers
44 views

Question: I'm implementing a layered SFU architecture using Pion WebRTC in Go. The setup is as follows: Architecture Master SFU: Receives a single upstream client stream (audio + video). Only ...
yternal's user avatar
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0 answers
62 views

I am working on an electron pet project of mine to send files over local network. For the actual sending part I choose to use wrtc via Simple-Peer, and it's on a backend(maybe weird I know) so I ...
Vlad Karelov's user avatar
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0 answers
48 views

I got the library to work ('react-native-webrtc'), and I can receive an audio stream. But on iOS, the mic permission is turned on and I can see the orange dot in the top right corner of the screen ...
20 Credi's user avatar
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34 views

How to debug audio output issues like hearing sound from both speakers when only one is desired in mobile WebRTC? I developing one to one audio calling with webrtc + react js. When connected each ...
VYSHNAV MV's user avatar
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0 votes
1 answer
62 views

I have an Android Webview that has a meeting running inside of it. When I'm in full-screen mode, everything works perfectly. However, when I enter the PiP mode, I lose the audio. Participants can hear ...
Mohammad Ihraiz's user avatar
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0 answers
20 views

I'm implementing a WebRTC conference call feature using JSIP, and I'm trying to record both the local audio stream and multiple remote audio streams using MediaRecorder. This works fine in 1:1 calls, ...
Bhumesh Deekonda's user avatar
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0 votes
0 answers
47 views

I built an ai voice agent with TTS->LLM->STT pipeline, it should make outbound calls and interact with customers. How do I utilize amazon contact center with kinesis video streams to manage this ...
Zaki's user avatar
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0 answers
82 views

I need help to figure out why I am unable to authenticate WebRTC clients of my TURN server. Server I installed coturn on freebsd via $ pkg install turnserver. I have the following settings written in ...
Duncan Britt's user avatar
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1 vote
1 answer
46 views

I'm building a Mac and iOS app and using Google's WebRTC (m137 using LiveKit's binary distribution). For a regular app it's very convenient that it just automatically starts playback/playout of all ...
nevyn's user avatar
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1 vote
1 answer
108 views

I am working on a video conferencing solutions web app and I am having issues while fetching the audio output devices on mobile browsers. I am able to get audio output devices on laptop browsers ...
Mayank Kumar's user avatar
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0 answers
60 views

I am building a Node.js WebRTC media bridge (a Back-to-Back User Agent or B2BUA) that connects a browser-based client to the WhatsApp Calling API. The backend is hosted on an Azure App Service for ...
Akshay Pagare's user avatar
  • 164
1 vote
0 answers
82 views

I’m building a peer-to-peer file transfer tool in Python using aiortc and WebRTC data channels. It’s designed to handle large files efficiently by reading 64 MB blocks from disk and sending them in ...
Daniel's user avatar
  • 53
2 votes
1 answer
126 views

I need to connect two peers with WebRTC, as one would. However, in my case the constraint is that each peer is able to pass only one message to the other peer over the initial signaling channel. One ...
WofWca's user avatar
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4 votes
1 answer
150 views

I'm trying to send SCTP messages over UDP. The setup appears to work, but I'm getting a retransmission message every time. Here's the process: Create a new UDP socket() bind() the socket to a local ...
brandav's user avatar
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-2 votes
1 answer
66 views

I have noticed if I create a webrtc data channel before creating and sending a local description (SDP) to a remote, then datachannel event is fired on the remote. If I create the channel when webrtc-...
LUN2's user avatar
  • 93
1 vote
2 answers
159 views

I am trying to include WebRTC VAD into my project, specifically I want the feature for distinguishing audio between voiced and unvoiced but having troubles including it. I am using gcc Built by MinGW-...
Muhammad Ali khalid's user avatar
  • 21
0 votes
0 answers
38 views

I am trying to render a WebRTC video on a Surface instead of using SurfaceViewRenderer. On some devices this works fine, but on others, it shows a black screen. If I use SurfaceViewRenderer, it works ...
Andelic47 Aaron's user avatar
  • 9
1 vote
2 answers
735 views

Following this documentation https://developers.facebook.com/docs/whatsapp/cloud-api/calling/user-initiated-calls When user initiates call, i'm receiving following webhook event "object": &...
Akshay Pagare's user avatar
  • 164
2 votes
1 answer
66 views

So i was building a Video calling app with nodejs and react and WebRTC, and wanted to test it at some point but whenever on user accepts the call the other user gets this error message ...
Günther's user avatar
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0 votes
1 answer
230 views

Our cloud contact center platform is being negatively impacted by the chromium flag "allow webrtc to adjust the input volume." ( Manually disabling it works, but we have ~550 users and I ...
Stephanie Wilson's user avatar
  • 1
0 votes
0 answers
98 views

'm building a mobile app using Flutter and integrating LiveKit for audio/video calls. On the web platform, everything works perfectly — media connects, video streams, etc. However, on mobile devices , ...
Sunil Kumar's user avatar
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0 votes
0 answers
34 views

Im making a mutli user video chatting app using webrtc but the remote streams that are recieved with the peer connections dont play . this is my peerManager class : import { v4 as uuidv4 } from "...
user27159012's user avatar
  • 1
0 votes
0 answers
34 views

I've a web application using WebRTC capabilities to handle phone calls. My customers would like be able to pickup calls using headset button like they use to do with a desktop application. I didn't ...
Pichou Bichou's user avatar
  • 11
0 votes
0 answers
73 views

I have a setup where the client (Angular) connects to a Server (Go + PionV4). Flow explained: The client sends an HTTP request to the server and then the server stores some data required for later ...
Noel Maróti's user avatar
  • 1
1 vote
1 answer
100 views

I'm using Azure Communication Services (ACS) in a React app to implement video calling and screen sharing. Calling and camera video functionality is working fine. But when I try to start screen ...
suraj karosia's user avatar
  • 73
0 votes
0 answers
29 views

I have implement the WebRTC in my flutter project and it's working completely but only issue is when user can rejoin the existing broadcasting so getting blank screen. Here is the my code: enter link ...
Alpit Panchal's user avatar
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152 views

Using provided examples and the webrtc docs I have tried to implement webrtc reading from a raspberry pi camera streaming RTP to a webpage hosted by an app running on the same pi. Currently just a ...
Shovel_jockey's user avatar
  • 3
1 vote
0 answers
89 views

I'm trying to make web camera with latency 50-100ms for real-time control purposes. The server is a python script, the client is WebRTC application running in Android Google Chrome, directly connected ...
Vitaliy Tsirkunov's user avatar
  • 570
0 votes
0 answers
67 views

I'm building an open-source Android assistant app that uses a Node.js server as a bridge to Google's Gemini API. The app uses WebRTC for real-time audio streaming between the Android client and the ...
Bhaskar-kumar-arya's user avatar
  • 1
0 votes
0 answers
50 views

our team built a React Native calling app using sip.js and WebRTC. Everything works fine when the app is active. However, when a call is ongoing and the user swipes the app from recents (kill), the ...
mit rudani's user avatar
  • 1
0 votes
0 answers
82 views

I use libwebrtc in my C++ application. I'm trying to get a list of supported codecs. Something like this: auto factory = webrtc::CreateBuiltinVideoEncoderFactory(); auto supList = factory->...
Александр Малыгин's user avatar
  • 1
1 vote
0 answers
142 views

I am developing two applications, a Next.js and a QtPython application. The goal is that the Next.js application will generate a WebRTC offer, post it to a Firebase document, and begin polling for an ...
Medhansh Garg's user avatar
  • 21
2 votes
0 answers
61 views

I'm building a simple peer-to-peer file transfer app using WebRTC in a React application. The goal is to allow direct file transfer between two devices without always relying on a TURN server. However,...
fl0wl3tz's user avatar
  • 21
0 votes
2 answers
137 views

I have a problem with my web source code. I have made sure that the site is accessed via https and camera access permission is granted. However, the barcode scan display does not appear and only ...
Ari's user avatar
  • 11
0 votes
0 answers
33 views

I'm building a calling feature on our Flutter app and I'm having a different behaviour on iOS compared to Android. When a call is accepted, we navigate to the CallScreen widget using GoRouter and when ...
td2thinh's user avatar
  • 29
2 votes
3 answers
130 views

I'm making an app that connects to OpenAI's Realtime API using WebRTC. My Mute Microphone mutes the microphone, correctly. But what if I want to also Mute the output from the Realtime API? The whole ...
Sorry's user avatar
  • 630
0 votes
0 answers
59 views

I'm having trouble setting up Kamailio with NAT traversal for WebRTC endpoints. Calls drop after 30 seconds. Any insight into proper RTPEngine and STUN setup?
Jack-Morris_VoIP's user avatar
  • 1
0 votes
0 answers
80 views

I am creating a voice bot project where I need to setup Voice activity Detection with barge-in feature. So, when the bot speaks the output sound of the bot is picked up by the mic as input (this is so ...
Shivansh Kumar's user avatar
  • 1
0 votes
1 answer
408 views

I'm trying to write simple python server to browser video streamer using aiortc? For simplicity the server and the browser are in one local network. The python code: import asyncio from aiortc import ...
Vitaliy Tsirkunov's user avatar
  • 570
2 votes
2 answers
219 views

I have the next scenario in my webrtc video chat app where i use react js and nest js. const [localStream, setLocalStream] = useState<MediaStream | null>(null); const [remoteStreams, ...
Asking's user avatar
  • 4,276
0 votes
0 answers
47 views

I'm using RecordRTC to capture audio, I have echo cancellation turned on. This works perfectly in Edge and Chrome, but in Firefox it's picking up whatever sound is being output from the speakers. I'm ...
Mark Stapleton's user avatar
  • 1
1 vote
0 answers
60 views

I am developing a voice chat application. After the server receives an RTP packet from a client, it sends this packet unmodified to all the other clients. Each client has a different SSRC. The ...
Inexspectatus somnium's user avatar
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