9,070 questions
-3
votes
0
answers
27
views
React chrome extension and React webapp with outgoing WebRTC stream [closed]
I have chrome extension made with react, and the same webapp on a website.
each app should send and receive stream over webrtc. With chrome extension it works well, it sends and receives. on the ...
- 53
0
votes
0
answers
44
views
DTLS handshake fails with "no SRTP profile negotiated" when streaming from ffmpeg (schannel) to aiortc WHIP server
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 &...
- 1
0
votes
0
answers
33
views
Where is the yjs.js UMD bundle in the latest Yjs releases? VS Code Webview cannot load ESM [closed]
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&...
-1
votes
1
answer
48
views
How to accurately measure time difference between audio and video RTP packets coming from different clients?
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 ...
- 13
0
votes
0
answers
72
views
How to make WebRTC use opensles for playing in Android?
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 ...
- 63
1
vote
0
answers
41
views
Translation using webrtc
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 ...
0
votes
1
answer
64
views
Why does my P2P chat app always redirect back to the login page after registration?
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 ...
- 11
1
vote
0
answers
44
views
Pion WebRTC: Video not received in Sub SFU when using ReplaceTrack in layered SFU architecture
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 ...
- 11
0
votes
0
answers
62
views
WRTC connection failure on sending binary
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 ...
0
votes
0
answers
48
views
react-native-webrtc IOS: Mic is enabled even if only consuming
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 ...
- 1
0
votes
0
answers
34
views
How to Troubleshoot Audio Routing Problems in Mobile WebRTC web Apps
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 ...
0
votes
1
answer
63
views
No audio in WebRTC over WebView in Android PiP mode
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 ...
0
votes
0
answers
20
views
Auto call recording: Issue with merging remote and local streams in conference call (JSIP WebRTC)
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, ...
0
votes
0
answers
47
views
How to stitch together Amazon Connect, Kinesis Video Stream, and In-house ai agent pipeline (TTS-STT-LLM)
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 ...
- 107
0
votes
0
answers
82
views
Unable to authenticate WebRTC clients of my TURN server
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 ...
- 340