FirebaseRTC VS webrtc

Low latency streaming (<500ms): The Video SDK's infrastructure is built with WebbRTC, which helps to deliver secure and ultra-low latency video streams to all audiences at different bandwidths.

Web Real-Time Communication (WebRTC) is a technology developed by Google in 2013 for peer-to-peer communication. WebRTC enables web browsers to capture audio, video, exchange data, and teleconferencing without plugins or intermediaries. WebRTC achieve these through APIs and protocols that interact with one another. WebRTC media streaming when used with SocKet.IO will produce an application that streams media and exchange data instantly. Socket.IO is a library that provides low latency bi-directional communication between client and server. Socket.IO was built on websocket, a communication protocol that provides a full-duplex and low latency communication between server and browser. In this article, readers will learn how to build a video chat application using WebRTC and Socket.IO. This article is for web developers who wish to develop web applications that can stream media between two peers of computers in real-time without installing any plugins.

Possibly you what to look into WebRTC: https://webrtc.org/

P2P is nothing new. It is a long-established means of connecting two or more people directly over a network. Web browsers are very capable of a wide range of P2P connections. Many apps use WebRTC to enhance realtime apps, but it is still an underutilized technology. Even with WebRTC, many apps are designed around the dependence on a central app server with WebRTC performing a user experience enhancement. Web meshes turn this idea on its head: Instead of using P2P connections to enhance the user experience, what if P2P connections were the foundation of the user experience? In other words, what if there was no central server?

The good news is that after reading all I have a better understanding of the Web Realtime Communicate and the big view, not just about small things like sending files. You can read all about WebRTC here

WebRTC is an open-source project that delivers video streams to viewers with real-time latency. Initially developed for text-based chat apps and VoIP usage, its popularity has grown among video chat and conference app developers after being purchased by Google. The WebRTC protocol is a low-latency streaming solution that relies on peer-to-peer streaming (P2P); it is utilized by some of the most used apps like Google Meet, Discord, Houseparty, WhatsApp, and Facebook Messenger.

You may wanna look at WebRTC (https://webrtc.org) as thats the technological foundation of what youre looking to do.

And now there's a game, fully written in typescript and rendered in a , even fluently on all major browsers, and packaged inside a PWA. My future plans for it include more themes, more players, and remote multiplayer support, as an excuse to learn some WebRTC.

I have opened an issue on GitHub [1], we can continue there.

[1] https://github.com/webrtc-rs/webrtc/issues/426

webrtc

webrtc.rs is a port of Pion (which we also use). It's a better fit for server-side use

I want to use peer-to-peer communication and data transfer for my next side project (client-server web app). I've been doing some research and WebRTC seems to be the only option for this on the client. There are a ton of libraries and product offering for facilitating STUN/TURN servers as a service, but I'm quite interested in learning more about these protocols. That being said, I'm not the best rust programmer (part of the reason of using Rust as the server is so that I can learn more), and the signalling protocols seem rather complicated. I've looked at https://github.com/webrtc-rs/webrtc and it seems promising.

Regarding webrtc-rs. I'm a contributor to that project. The future of str0m could very well be to roll it in under that umbrella. However in these early days, I think it's better to keep str0m separate, until such time that it covers most of webrtc-rs functionality. Discussion here: https://github.com/webrtc-rs/webrtc/issues/230

This project was really fun for me is that I got to use https://webrtc.rs webrtc.rs Rust implementation of WebRTC that is based on Pion. The performance is fantastic and it is so easy to fetch + build. I had a great experience with it. It is really impressive how far the project has come!

I'm confused, you're saying as far as you can tell webrtc-rs does everything pion can do, but also saying "when the project is ready"? Can you explain what you mean? Also it seems webrtc.rs website says it's not production ready yet? How true is that?

The "Mediasoup" project provides a high level JavaScript/TypeScript interface to the WebRTC APIs. The core logic of this project is implemented in C++/Rust. Consider taking a look at the project if you want an easy-to-use library instead of the low-level libWebRTC APIs. A notable project to mention is the Pion/webrtc project which has a Golang implementation of the WebRTC API. Of course, we should mention the rust port WebRTC.rs. Let’s keep all the rustaceans happy too!

Have you considered using webrtc? It's a well-established protocol supported by all major browsers, and there's already a pure-Rust implementation you might be able to use.

If you ever did decide to add the complexity of RTC, my personal favorite SFU is MediaSoup with Janus a second. But if you want to be adventurous the Rust WebRTC project would be your go to. In any case cool project!!