webrtc-nuts-and-bolts VS FirebaseRTC

I recommend WebRTC For The Curious too, as another comment. Personally, I didn't use WebRTC in C#, but the WebRTC concepts are nearly the same in every different language/platform. Also, you can check out my project and documentation on internals of WebRTC, written in Go and JavaScript: https://github.com/adalkiran/webrtc-nuts-and-bolts . It's not related directly, but another example of usage at my other project, written in Go, Python, and JavaScript: https://github.com/adalkiran/distributed-inference .

I am heavily biased toward WebRTC. Here is my take on it though!

> It's incredibly complex as a specification

What is complex about it? I can go and read the IETF drafts, webrtcforthecurious.com, https://github.com/adalkiran/webrtc-nuts-and-bolts and multiple implementations.

QUIC/WebTransport seems simple because it doesn't address all the things WebRTC does.

> has limitations and numerous issues that set limits in how scalable it can be

https://phenixrts.com/en-us/ does 500k viewers. I don't think anything about WebRTC makes it unscalable.

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IMO the future is WebRTC.

* Diverse users makes the ecosystem rich. WebRTC supports Conferencing, embedded, P2P/NAT Traversal, remote control... Every group of users has the made the ecosystem a little better.

* Client code is minimal. For most users they just need to exchange Session Descriptions and they are done. You then additional APIs if you need to change behaviors.

* Lots of implementations. C, C++, Python, Go, Typescript

Show HN: WebRTC Nuts and Bolts, A holistic way of understanding how WebRTC runs\ (0 comments)

You can find it at: https://github.com/adalkiran/webrtc-nuts-and-bolts

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