trickle | userland bandwidth shaper for Unix-like systems

I'm trying to figure out how super really works in JavaScript. I have an idea but I'm not sure of it's exactitude so I need some help.

I am trying to understand how to use WebRTC to establish P2P connections following the example here: https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Simple_RTCDataChannel_sample. In my mind, it seems like when I create a new RTCPeerConnection, that connection should contain information about my public ip and subnet, so that when I create an offer and pass it to the remote computer, the remote would then have the details about where to send the response offer. However, whenever I try creating an RTCPeerConnection, it has 0.0.0.0 in it and no mention of my ip (which is not 0.0.0.0). Do you know why this could be? What am I doing wrong? How do I get it to display my public ip?

I have a c++ application that gets the video in RTSP and H264 format from a camera using gstreamer an re-sends the videos using webrtcbin. I have followed the example from this link and I can see the video trough firefox (with the tips suggested in this post), when use VP9 encoding.

The pipeline I have used is:

First, I want to mention that I am very new to WebRTC, so any advice would be very helpful.

Currently I am using aiortc library to build my own WebRTC app. Here is what I am trying to do.

I have 2 peers, one is web browser, which is written in javascript, and another one is python script, which is working as signaling server and peer at the same time. So If you access to my web page, you will send video frame to server and then the server will make modification of that then send it back.

So I finished testing my app on LAN environment and everything worked as I expected. But once I deployed my app to remote server (Google cloud run) , I encountered Ice connection state failing issue. And gets this log on remote server.

enter image description here

(I think it is due to disconnection between peers, not low memory problem. I tried with 16GB RAM and 4 cpus and still didn't work)

Then, I dig into more information, and found that TURN/STUN server is necessary to build WebRTC app over Internet. So I added google STUN server to my RTCPeerConnection like this. [{'urls': 'stun:stun.l.google.com:19302'}, {'urls': 'stun:stun1.l.google.com:19302'}, {'urls': 'stun:stun2.l.google.com:19302'}] (I added both side on javascript and python because both side is working as peer) Unfortunately, it still didn't work.

Now, I am planning to build my own TURN server, but I am afraid if TURN server wouldn't solve this problem. So I would like to have any advice from you since I am quite stuck within my situation.

p.s I have done SSL encryption.(So GetUserMedia is working fine)

Sdp details(Offer/Answer):

SDP

Offer

I would like to devise a scalable error handling strategy for my Observable-based Firebase requests. To do so, I am trying to create a custom RxJS operator for each type of error (e.g. unauthenticated, internal, you name it). The operators would be stacked after the request as follows:

I have implemented yt-dlp as part of my Python script, it works well, but I am unable to get the rate-limit feature to work. If you run the same command from the CLI the rate is limited correctly, is anyone able to tell me the correct syntax?

I have tried several combinations such as rate-limit, limit-rate 0.5m, 500k, 500KiB, 500, and none seem to work

Hi there, I am attempting to build a WebRTC client in Android that subscribes to a video feed that is being broadcast using NodeJS and JavaScript.

The broadcaster code can be viewed in its entirety in this lovely article by Gabriel Tanner.

It works beautifully when running it in localhost under the http://localhost:4000/broadcaster.html in Chrome and then visiting my IP Address from another device on the network. I can see the video and it is near real time.

I have tried this using two different webcam devices, both a built-in and a USB webcam but the Android client does not work even though the JavaScript Broadcaster and Client works fine.

After following the tutorial and getting the example to work I decided to try and implement my own Android application for which the entire source code can be viewed right here on my GitHub.

I have followed various tutorials around the place and the issue always stems from attempting to set the remote description which is done with the following bit of code:

I am trying to create a video chat and using express and socket.io on the backend and react and simple-peer on the frontend.

This is my server code:

I have two different WebRTC clients: an Android device and an angular application. I set up a turn and stun server and both seems to work with the trickle ice tester and the webrtc tester.

As you can see here:

But all ice candidates fail in Firefox when I am watching the candidates in about:webrtc.

Does anyone have an explanation for this?
More info:

• A similar error occurs in Google Chrome, but the logs are from Firefox because Firefox has better logging.
• The devices are on different networks, so the host request should fail and the turn server is needed.
• The turn server is a coturn turn server, but I also tried it with a Pion turn server and the same result occurs.
• Firfox logs: https://gist.github.com/Nick-v-L/365b7da10039d28a6a23a27fea15df52
• Coturn logs: https://gist.github.com/Nick-v-L/04c3cfc677847e3cdcb7f6b5ca15c743