FirebaseRTC
janus-gateway
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FirebaseRTC | janus-gateway | |
---|---|---|
40 | 8 | |
351 | 6,528 | |
4.8% | 1.8% | |
0.0 | 9.4 | |
20 days ago | 4 days ago | |
JavaScript | C | |
- | GNU General Public License v3.0 only |
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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Writing a Game in Typescript
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.
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Video Live Streaming: Notes on RTMP, HLS, and WebRTC
WebRTC is used by phenixrts as the delivery from server to client. The promise of WebRTC was P2P direct connections for video/data transport, and server/client for coordination and fallback.
https://phenixrts.com/en-us/faqs.html
> The scalability of Phenix’s platform does not come from the protocol itself, but from the systems built and deployed to accept WebRTC connections and deliver content through them. Our platform is built to scale out horizontally. In order to serve millions of concurrent users subscribing to the same stream in a short period of time, resources need to be provisioned timely or be available upfront.
> With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. It supports video, voice, and generic data to be sent between peers...
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How to make a live streaming from client's camera
You should take a look at WebRTC. I can't recommend a resource specific to Flask, but this article might be helpful.
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Rust Time Communication.
With the strike in session, I had a lot of time on my hands to learn what I needed to learn, and things went on smoothly till I hit a brick wall. I couldn’t find a Web RTC implementation in Rust. Web RTC is a framework that allows browsers and web-based applications to communicate in real-time. It is important to know that while Rust has made a lot of waves in recent times, it is still relatively new compared to other languages and development technology. There were still things that had not fully been abstracted to the level where regular developers could easily access the functionalities without having a deep understanding of how they worked. Web RTC was one of them. Now that I think of it, I found one, but it was basically just C embedded in Rust. While that sort of worked, It wasn’t what I wanted. I searched around a bit more, then I found someone, Rainliu working on a Web RTC implementation in Rust. It was in its early stages though, in fact, all that had been done was a collated list of frameworks that would allow Web RTC to work in Rust. I decided to help out, or at least try to help. There I was, a few weeks into learning Rust, and I wanted to help build a library.
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Does this happen to everyone or just me…the check for WebRTC leaks always takes a while longer than the other checks?
“The WebRTC project is open-source and supported by Apple, Google, Microsoft and Mozilla, amongst others.” Source
- My first full-stack web app
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Video streaming in go?
If you want to deal with NAT traversal I recommend doing this with WebRTC ... The pion project is perfect for this.
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How to make a FTP video call in flutter
You’re looking for webRTC, FTP is just File Transfer Protocol. You are going to have to setup a TURN server for your webRTC app if this is going further then your LAN and a learning project. You could also look at websockets(central server), VoIP. https://github.com/flutter-webrtc/flutter-webrtc https://webrtc.org/
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Webrtc Video Chatting App
However, with WebRTC these above comes, with worked in into the program. Here, this innovation doesn't need any modules or outsider programming. The WebRTC handles everything consequently. Besides, being an open-source all its source codes are accessible for nothing at https://webrtc.org/
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Connection Types In Web Apps
Web Real-time Connection (RTC) is a standardized API for peer to peer communication. It is available on all major browsers and supports video and voice. Many video calling applications use Web RTC Under the hood.
janus-gateway
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Low-latency audio streaming (local network)
I've been using Janus gateway for similar. Pretty easy to setup.
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Live video calling - the Dyte way
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.
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Casey Muratori: refterm and the philosophy of non-pessimization (how you can make programs run 100x faster without optimizations)
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.
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Stop using Zoom, Hamburg’s data protection agency warns state government
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.
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WebRTC over Asp.Net Core - Any examples?
- Janus (C / C++)
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Building a customer support solution focused on video calls
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)
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need suggestions for options for media server (WebRTC preferred)
You can try janus https://janus.conf.meetecho.com/
- Show HN: WebRTC-Echoes: Interop for C#, C++, Python, TypeScript, Go and Servers
What are some alternatives?
mediasoup - Cutting Edge WebRTC Video Conferencing
Pion WebRTC - Pure Go implementation of the WebRTC API
jitsi - Jitsi is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features.
media-server-node - WebRTC Media Server for Node.js
simple-peer - 📡 Simple WebRTC video, voice, and data channels
aiortc - WebRTC and ORTC implementation for Python using asyncio
libdatachannel - C/C++ WebRTC network library featuring Data Channels, Media Transport, and WebSockets
galene - The Galène videoconference server
flutter-webrtc-demo - Demo for flutter-webrtc
NodePlayer.js - Pure JavaScrip HTML5 live stream player
kms-core - Core library of Kurento Media Server