trystero VS FirebaseRTC

This looks exciting and I'm pleased to see more and more frictionless ways of making p2p apps. I've been building a somewhat similar hobby project [1] that aims to connect peers in the browser by piggybacking on open protocols out on the net (BitTorrent, MQTT, Nostr, IPFS, etc).

This project seems to be using Hyperswarm which I've looked at for use as a peering medium but it seems like it's not supported in the browser. I'd love to implement it if that story changes since it's so easy to distribute apps on the web.

[1] https://github.com/dmotz/trystero/

I was going to mention WebRTC! It seems designed for video calling, but there are lots of cool use cases - I recently ran across https://github.com/dmotz/trystero , a dead simple WebRTC library for peer-to-peer multiplayer browser games.

I strongly endorse Trystero (https://github.com/dmotz/trystero) for enabling P2P communication in web apps. It’s open source and leverages public infrastructure for matchmaking.

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https://github.com/dmotz/trystero

Nice work! Any reason one might use this over https://github.com/dmotz/trystero, you think?

Yes, this is a major issue that I haven't found a real solution for. There seems to be a mixture of iOS Safari bugs and intentional design limitations at play, and I don't know if a fully P2P web app like Chitchatter is practical on that platform. There's an open issue to improve this in Trystero (the networking library that Chitchatter uses), but there may be a limit to how stable iOS will be with WebRTC apps. 😕

You can use Trystero (https://github.com/dmotz/trystero) to cut server costs to zero. That’s what I used to build https://chitchatter.im/, which supports P2P audio and video calls.

WebTorrent is obviously well suited for p2p file distribution, but using a minimal subset of the protocol also provides a nice hack for easily bootstrapping peer connections between web app users. Piggybacking on public mediums already designed to do peer exchange can let you rapidly prototype a WebRTC project without the hassle of running your own server anywhere.

I built a library that explores this idea: https://github.com/dmotz/trystero

You might also consider assessing complementary or alternative technologies; WebSocket and HTTP aren’t the only options when it comes to real-time communication, after all. WebRTC is similar to WebSocket, with the key difference being that it’s used to implement peer-to-peer connections without relying on a server. That can be especially helpful for video calls, allowing participants to communicate directly without introducing load to your server.

We use WebRTC to gain access to a user’s camera and microphone using the getUserMedia method. Typically, I would gain access to both of these from the same call. However, our experience requires the camera to flip from facing the environment to facing the user and I noticed that the small period of time the flip occurred (and microphone wasn’t available) contributed to a bit of audio lagging in the final recorded video. This was one of the nastier bugs I faced in development. So, we’ll just access each of these on their own media streams so that the camera can flip independently from the microphone.

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/

Use the language you're comfortable with. Sounds like you're interested in creating a blockchain game. Writing your own simple game engine isn't simple. I would recommend utilizing an existing one for whatever language you want. If you still choose to write your own it can be a valuable lesson in graphical programming which I personally find fun. It's easier to cheat a webpage embedded game written in Javascript than one ported to WebASM in my experience and I've heard good things about WebRTC for embedded multiplayer games.

Have you looked into WebRTC? https://webrtc.org Seems like it supports exactly what you're looking for. SignalR is more for real-time messages, not really for streaming.

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