FirebaseRTC VS janus-gateway

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.

OpenTalk is a young project for creating online meeting software similar to Jitsi or BigBlueButton. It is a completely new development, and while it is not a fork of an existing open-source project, it integrates with other projects such as the Janus WebRTC server, Redis for volatile state, RabbitMQ for communication between server instances, and PostreSQL for persistent state.

I've been looking into Janus WebRTC Server due to the ability for Uv4L to join Janus rooms (I'm building a RaspberyyPi doorbell)

A related answer to the above is to check out Janus. It's a general purpose WebRTC server that has RTMP and FTL ingest support. I think it's also batteries not included, but I think it's what Glimesh is based on.

I've been using Janus gateway for similar. Pretty easy to setup.

A number of open-source projects also exist, which give developers a great head start if they're looking to build their own infrastructure - the most popular of these include Jitsi, Mediasoup, Janus, and Pion. These projects provide a layer of abstraction and expose a number of helper functions to perform various tasks, such as creating transports, etc. They have helpful guides on how to get started, but you would still face the aforementioned issues regarding scaling, resources, etc.

Yes, there are many self-hosted options out there. https://github.com/meetecho/janus-gateway works well for multi-party video with up to about 15 users in a room assuming everyone has a reasonably reliable connection.

- Janus (C / C++)

You can also take a look at https://github.com/meetecho/janus-gateway which can help you implement the video call part (as well as more traditional rtc scenario)

You can try janus https://janus.conf.meetecho.com/