Welcome to Neko, a self-hosted virtual browser that runs in Docker and uses WebRTC technology. Neko is a powerful tool that allows you to run a fully-functional browser in a virtual environment, giving you the ability to access the internet securely and privately from anywhere. With Neko, you can browse the web, run applications, and perform other tasks just as you would on a regular browser, all within a secure and isolated environment. Whether you are a developer looking to test web applications, a privacy-conscious user seeking a secure browsing experience, or simply someone who wants to take advantage of the convenience and flexibility of a virtual browser, Neko is the perfect solution.

In addition to its security and privacy features, Neko offers the ability for multiple users to access it simultaneously. This makes it an ideal solution for teams or organizations that need to share access to a browser, as well as for individuals who want to use multiple devices to access the same virtual environment. With Neko, you can easily and securely share access to a browser with others, without having to worry about maintaining separate configurations or settings. Whether you need to collaborate on a project, access shared resources, or simply want to share access to a browser with friends or family, Neko makes it easy to do so.

Neko is also a great tool for hosting watch parties and interactive presentations. With its virtual browser capabilities, Neko allows you to host watch parties and presentations that are accessible from anywhere, without the need for in-person gatherings. This makes it easy to stay connected with friends and colleagues, even when you are unable to meet in person. With Neko, you can easily host a watch party or give an interactive presentation, whether it's for leisure or work. Simply invite your guests to join the virtual environment, and you can share the screen and interact with them in real-time.

A fully featured browser based WebRTC SIP phone for Asterisk.

This web application is designed to work with Asterisk PBX. Once loaded application will connect to Asterisk PBX on its web socket, and register an extension. Calls are made between contacts, and a full call detail is saved. Audio Calls can be recorded. Video Calls can be recorded, and can be saved with 5 different recording layouts and 3 different quality settings. This application does not use any cloud systems or services, and is designed to be stand-alone. Additional libraries will be downloaded at run time (but can also be saved to the web server for a complete off-line solution).

Galène (or sometimes Galene) is a videoconference server (an “SFU”) that is easy to deploy and that requires moderate server resources. It was originally designed for lectures and conferences (where a single speaker streams audio and video to hundreds or thousands of users), but later evolved to be useful for student practicals (where users are divided into many small groups), and meetings (where a few users interact with each other).

Galène's server side is implemented in Go, and uses the Pion implementation of WebRTC. The server is regularly tested on Linux/amd64 and Linux/arm64; it has been shown to run on Linux/armv7 and Linux/mips (OpenWRT), and even on Windows. It should in principle be portable to other systems, including Mac OS X. The client is implemented in Javascript, and works on recent versions of all major web browsers, both on desktop and mobile (but see below for caveats with specific browsers).

While traffic is encrypted and authenticated from sender to server and again from server to receiver, Galène does not perform end-to-end encryption: anyone who controls the server might, in principle, be able to access the data being exchanged. For best security, you should install your own server.

Github: https://github.com/jech/galene/

HTML5 SIP client. Written in Javascript, throw it onto a page and configure it to contact a SIP gateway of some kind. Uses webRTC built into all modern web browsers. Does audio as well as video. Can contact the PSTN if the gateway can.

Github repo: https://github.com/DoubangoTelecom/sipml5

Javascript SIP client for VoIP. NodeJS backend. Uses WebRTC built into every modern browser. Found in the NPM collection, so it's easy to install (in theory).