FirebaseRTC VS janus-gateway

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

> despite WebRTC mostly being about client/client communication

This is actually kind of a misconception, though it’s an understandable one given that WebRTC is almost always pitched as a peer-to-peer protocol.

In practice, most people using WebRTC for video are sending their video to a server, not directly to another client. It’s pretty safe to assume that most people who use your app are going to need TURN, and at that point, you’re not really doing peer-to-peer at all, so you might as well just have your browser-based app talk to a server that’s pretending to be another browser.

These servers (called Selective Forwarding Units or SFUs) can operate like a TURN server in the case of a one-on-one call, but they can also multiplex everyone’s feeds in the case of a larger conference (peer-to-peer 5 person calls would require each participant to send 4 copies of their video) and often have extra features like the ability to record calls, transcode streams or convert to other protocols.

The one I’ve used a lot is called Janus[0], it’s open source and has good docs, I recommend people check it out if they’re interested in getting deeper into WebRTC or other video streaming tech.

[0] https://janus.conf.meetecho.com

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.

This all changes when you are actually a domain expert: You can treat the various components as a "white box" because you see the forest for the trees and can make cross-cutting assumptions which will inherently make the code faster. I've noticed a lot of projects written by domain experts are often these giant clusterfucks of C that violate pretty much every guideline there are so many Medium blogs about, and yet they're very stable and widely used. See: https://github.com/meetecho/janus-gateway for example.

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++)