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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 ...
20 Credi's user avatar
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0 answers
28 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 ...
VYSHNAV MV's user avatar
0 votes
1 answer
48 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 ...
Mohammad Ihraiz's user avatar
0 votes
0 answers
16 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, ...
Bhumesh Deekonda's user avatar
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0 answers
33 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 ...
Zaki's user avatar
  • 107
0 votes
0 answers
51 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 ...
Duncan Britt's user avatar
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0 answers
27 views

Client's firewall seems to be blocking Agora WebRTC in my webapp? [closed]

I'll start this post by saying that I'm no expert on the matter, as I've just recently started working on these things. At my job we have an Angular webapp that's really simple, it creates a videocall ...
Lorenzo Bertolaccini's user avatar
1 vote
1 answer
39 views

How can I disable playout in in Google's WebRTC Swift/ObjC framework?

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
  • 7,143
1 vote
1 answer
76 views

Audio Output Device Getting and Selection

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
58 views

Node.js WebRTC server on Azure App Service fails STUN discovery (srflx candidates) despite open NSG ports

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
1 vote
0 answers
79 views

Max throughput in Python WebRTC data channel capped at ~110 Mb/s for large file transfer

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
  • 63
2 votes
1 answer
115 views

What are the downsides of ICE trickling (signaling) over TURN (data channel)? [closed]

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
  • 786
4 votes
1 answer
139 views

SCTP over UDP: Stop retransmission message

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
  • 795
-2 votes
1 answer
63 views

How do I create a data channel in webrtc (JS, Firefox 115.6.esr)? [closed]

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
131 views

How to include WebRTC VAD in my C project

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

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