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Questions tagged [mediasoup]

64 questions
8
votes
2 answers

How do I generate fullchain.pem and privkey.pem?

I'm trying to install this project: https://github.com/versatica/mediasoup-demo It requires fullchain.pem and privkey.pem files. How do I generate these with openssl or something similar, on Ubuntu 20?
harry young
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6
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0 answers

Error On Build after updating mediasoup-client to v3.6.46 and Above Version

I was using mediasoup-client 3.6.16 and have updated to the latest version 3.6.57. After updating I am not able to build. Using webpack (v2.4.1) to build. Dependencies: Node: v16.13.2 NPM: 8.1.2 React: 16.8.4 React-Dom: 16.8.4 Webpack:…
Not A Bot
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6
votes
3 answers

What is the role of SFU., Janus, mediasoup or medooze. on a webRTC application

I'm using a webRTC application with a simple-peer npm package. I want to know what is the purpose of all these topics (SFU., Janus, mediasoup or medooze.) and how can I integrate them to make my application performance greater? PS: I'm using a…
CTMA
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5
votes
1 answer

WebRTC: How to enable hardware acceleration for the video encoder

I'm trying to send video of screen capture to mediasoup with the help of WebRTC. There is a class for it in the WebRTC library: ScreenCapturerAndroid. It works, but the performance on the some devices is really bad. Especially if I use HD or better…
Valelik
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4
votes
3 answers

mediasoup v3 with Docker

Im trying to run an 2 WebRTC example(using mediasoup) in docker I want to run two servers as I am working on video calling across a set of instances! My Error: Have you seen this Error: createProducerTransport null Error: port bind failed due to…
Brian
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4
votes
1 answer

Why does using more than two STUN/TURN servers slow down discovery?

I'm passing a handful of STUN and TURN servers for my WebRTC application (built on top of mediasoup). When I do this, I get a message in the console telling me: "Using more than two STUN/TURN servers slows down discovery" I can cut down the servers…
Eric
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3
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2 answers

Mediasoup error when running ObserveRTC example: `getaddrinfo ENOTFOUND host.docker.internal`

This is the project I want to setup: https://github.com/ObserveRTC/full-stack-examples I started the app using this command: SFU_ANNOUNCED_IP="192.168.60.79" docker-compose up And this error happened at MediaSoup: ObserveRTC::RestTransport Wed, 14…
rostamiani
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3
votes
1 answer

UnsupportedError: cannot produce video at Transport.produce

As soon as I want to create a producer on client side I get this error: UnsupportedError: cannot produce video at Transport.produce I created the device and I got the RtpCapabilities from the server. That all works fine. So why do I get this error…
Johann nefdt
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3
votes
0 answers

WebRTC mixing two audio streams with AudioContext

I'm using mediasoup and webrtc to create media streams. async consume(transport) { const { rtpCapabilities } = this.device; const data = await this.socket.request('consume', { rtpCapabilities }); const { …
thunder1221sa
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3
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1 answer

Why is video not displaying when user joins room?

So, I have an adaptation of https://github.com/Dirvann/mediasoup-sfu-webrtc-video-rooms working in vanilla JS that I am attempting to adapt to use React. Instead of every user being a broadcaster, in my version, only the room creator is the…
harry young
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3
votes
2 answers

WebRTC: do I need a TURN server? (Would it help?)

I have a webcam chat room application (so it's many-to-many video sharing) using WebRTC and a mediasoup server. I am having problems with SOME of my users not being able to get an incoming video feeds to work. It's a difficult problem because I…
Eric
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2
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0 answers

Filtering and mixing WebRTC sound in NodeJs

Few weeks ago I wrote WebRTC browser client using mediasoup library. Now I am in middle of rewriting it as a NodeJS client. I am stuck with one thing. I want to receive multiple WebRTC audio souroces, mix them into single track then apply some…
Michał Bogusz
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2
votes
1 answer

Stream static video file through webrtc

what I am trying to accomplish is to have on my page audio and video file. And then send them through webrtc. I managed to do this with audio using web audio api like this. HTML:
Michał Bogusz
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2
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1 answer

WebRTC/MediaSoup library conflict with Gradle build

I am trying to user both MediaSoup and WebRTC libraries in my app. Separately it works flawlessly, but when trying to add both in the same project I have some conflicts. It's either: I add both mediasoup…
Anael
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2
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2 answers

Gstreamer opusenc encoder produces distorted/choppy audio

I am using the following gstreamer pipeline to grab RTMP src and transcode it with opusenc encoder and sending it as rtp packet to Mediasoup (a webrtc library). gst-launch-1.0 \ -v \ rtpbin name=rtpbin rtp-profile=avpf do-retransmission=true \ …
HM Moniruzzaman
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