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).