nginx-rtmp-module VS FirebaseRTC

Does anyone know if H264 to HLS conversion requires transcoding at all? I see that some web servers with plugins support the "conversion", but it's not clear if the conversion requires transcoding - and thus high processing power. Anyone have an idea?

Edit: Q1 yes it’s possible with auto push, I’ll share it Good info here : https://github.com/arut/nginx-rtmp-module

> video

Not personally, but I think that's what Peertube is supposed to help with.

> streaming

yep: https://github.com/arut/nginx-rtmp-module + something like https://github.com/Nesseref/html5-livestreaming for serving the video in a webpage instead of making your viewers fire up RTMP clients.

IME it is actually pretty simple to get a minimal setup going; the fun part is around how streaming to lots of people, surprise surprise, requires a lot of compute and bandwidth. In other words, you can totally roll your own Twitch, but you're going to be rolling both the user chrome and the infra aspects yourself.

I'm trying to self hosted videos with nginx using nginx-rtmp-module (VOD) similar to youtube.

To achieve this I used a module in nginx called nginx-rtmp-module . this module exactly does that. It introduces some sort of API to control this module from a web server like NodeJS.

Until then I will look into preStop and graceful terminations. I came across this article which mentions some specifics about Nginx graceful shutdowns, however I was unable to accomplish it and found this open issue for nginx-rtmp

So I found the answer. rtmp server have these things called directives which serve this exact purpose! https://github.com/arut/nginx-rtmp-module/wiki/Directives#on_publish

You should take a look at WebRTC. I can't recommend a resource specific to Flask, but this article might be helpful.

With the strike in session, I had a lot of time on my hands to learn what I needed to learn, and things went on smoothly till I hit a brick wall. I couldn’t find a Web RTC implementation in Rust. Web RTC is a framework that allows browsers and web-based applications to communicate in real-time. It is important to know that while Rust has made a lot of waves in recent times, it is still relatively new compared to other languages and development technology. There were still things that had not fully been abstracted to the level where regular developers could easily access the functionalities without having a deep understanding of how they worked. Web RTC was one of them. Now that I think of it, I found one, but it was basically just C embedded in Rust. While that sort of worked, It wasn’t what I wanted. I searched around a bit more, then I found someone, Rainliu working on a Web RTC implementation in Rust. It was in its early stages though, in fact, all that had been done was a collated list of frameworks that would allow Web RTC to work in Rust. I decided to help out, or at least try to help. There I was, a few weeks into learning Rust, and I wanted to help build a library.

“The WebRTC project is open-source and supported by Apple, Google, Microsoft and Mozilla, amongst others.” Source

If you want to deal with NAT traversal I recommend doing this with WebRTC ... The pion project is perfect for this.

You’re looking for webRTC, FTP is just File Transfer Protocol. You are going to have to setup a TURN server for your webRTC app if this is going further then your LAN and a learning project. You could also look at websockets(central server), VoIP. https://github.com/flutter-webrtc/flutter-webrtc https://webrtc.org/

However, with WebRTC these above comes, with worked in into the program. Here, this innovation doesn't need any modules or outsider programming. The WebRTC handles everything consequently. Besides, being an open-source all its source codes are accessible for nothing at https://webrtc.org/

Web Real-time Connection (RTC) is a standardized API for peer to peer communication. It is available on all major browsers and supports video and voice. Many video calling applications use Web RTC Under the hood.

Beyond the RFCs (there are ~50), MDN's guide, and webrtc.org's guide, I found this draft and this debugging guide useful.