FirebaseRTC VS mediasoup

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.

This release ships lots of new features to the already comprehensive set of them. Currently, we offer several working tutorials on how to set up STUNner with widely used WebRTC media servers and other applications that use WebRTC in Kubernetes, such as: - LiveKit - Jitsi - mediasoup - n.eko - Kurento

Architecture WebRTC SFU (server with Selective Forwarding Unit). Can handle unlimited rooms without limits of time, each having 8+ users, potentially many as it is scalable. Routing is a multiparty topology, where each participant sends its media to the MiroTalk media server mediasoup and receives all other’s media from it. This version is Ideally suited for large group video conferences.

I started the MiroTalk P2P & MiroTalk SFU projects during the pandemic period (about 1+ year ago), not knowing anything about the WebRTC. Making often the video conferences with my colleagues and not wanting to depend on Zoom, Teams, Google Meet... I decided to do some research about how it works and from there MiroTalk was born :) I Giving to everyone the chance to have its own instance of MiroTalk, which can be customized as you like and run in any cloud, vps, server. If you're just starting out, I suggest you take a look at the MiroTalk C2C (New) code, which can be a good starting point to understand how the architecture WebRTC Mesh (P2P) works. Later you can also study how the WebRTC SFU (Selective Forwarding Units - I recommend mediasoup which I personally love) or MCU (Multipoint Control Unit) architecture works. I wish you all the best!

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!

Indeed, maddening, especially as the wonderful https://mediasoup.org/ is developed here. Europe will never have great tech companies when the answer seems to be throwing €€€ away instead of investing locally

Hello thedominux, Thanks for your interest in MiroTalk ;) MiroTalk SFU code is: - JAVA-SCRIPT: 85.2% - HTML: 10.0% - CSS: 4.5% And has built in mediasoup server, more details about: https://mediasoup.org/

Mediasoup

The frontend is built using Next.js + Pixi.js and the backend audio and video is powered mostly by Mediasoup with physics done with Matter.js.