FirebaseRTC
webrtc-unreliable
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FirebaseRTC | webrtc-unreliable | |
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58 | 8 | |
473 | 385 | |
2.7% | 0.8% | |
0.0 | 5.7 | |
11 months ago | 18 days ago | |
JavaScript | Rust | |
- | GNU General Public License v3.0 or later |
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
Activity is a relative number indicating how actively a project is being developed. Recent commits have higher weight than older ones.
For example, an activity of 9.0 indicates that a project is amongst the top 10% of the most actively developed projects that we are tracking.
FirebaseRTC
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WebSocket vs. HTTP communication protocols
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.
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Wishing Upon A Star with Web AR for Disney’s Wish
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.
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Create a SwiftUI Video Streaming App With Fun Emoji Reactions
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.
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Develop a Video Chat App with WebRTC, Socket.IO, Express and React.
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.
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Live video streaming app
Possibly you what to look into WebRTC: https://webrtc.org/
- Chat protokoli
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Use JS suited for Online Games?
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.
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Send data to specific client from another client with a server in middle[C#][TCP][UDP]
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.
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Taking the Power Back with Web Meshes
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?
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I made a website sending file P2P
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-unreliable
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could not find tokio in webrtc_unreliable
The Cargo.toml on GitHub has the optional dependency specified correctly.
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How are rust devs doing?
I use WebRTC for multiplayer, and Rust has a great library (webrtc-unreliable) To be fair, C++ has libdatachannel which I like using too.
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What are the differences between UDP and WebSockets, and which type of games benefit from which?
You can, in theory, get UDP-like characteristics out of a different web API - with a WebRTC datachannel configured to be unreliable and unordered. I haven't seen anybody use that API just because it's a pretty big pain to use. The developer of Agar IO called it too hard to use in a HackerNews comment, but nowadays there's simpler libraries than the ones he had in 2016 like webrtc-unreliable and libdatachannel. You still need to run a STUN/TURN server and the integration is easier than it was back then, but still much harder than raw UDP sockets though.
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Shattersong Online - New Rust MMO Browser platformer in development by former Starbound and Wargroove devs
webrtc-unreliable: Provides an async API to a UDP-like transport over WebRTC
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WebRTC.rs
You can do this today with https://github.com/kyren/webrtc-unreliable
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A pure Rust implementation of WebRTC API.
Take a look at webrtc-unreliable. I haven't tried it, but it claims to implement just a simple subset of webrtc to get udp-like functionality.
What are some alternatives?
flutter-webrtc-demo - Demo for flutter-webrtc
datachannel-rs - Rust wrappers for libdatachannel
mediasoup - Cutting Edge WebRTC Video Conferencing
ozz-animation - Open source c++ skeletal animation library and toolset
NodePlayer.js - Pure JavaScrip HTML5 live stream player
webrtc - A pure Rust implementation of WebRTC
open-easyrtc - Open-EasyRTC - EasyRTC Free of Priologic
turbulence - Networking library for games, multiplex reliable and unreliable streams over unreliable datagrams.
webrtc-sdk - WebRTC Simple Calling API + Mobile SDK - A simplified approach to RTCPeerConnection for mobile and web video calling apps.
hashlink - An updated version of linked-hash-map and friends
janus-gateway - Janus WebRTC Server
Pion WebRTC - Pure Go implementation of the WebRTC API