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FreePBX is a web-based open-source graphical user interface (GUI) that manages Asterisk, a voice over IP (VoIP) and telephony server. [2]FreePBX is licensed under the GNU General Public License version 3, [3] with commercial modules available under their own licenses.
Jami, with GTK/Qt GUI, also supports IAX2 protocol, for Linux, OS X, Windows GPL; Jitsi, a Java VoIP and Instant Messaging client with ZRTP encryption, for FreeBSD, Linux, OS X, Windows; LGPL; Linphone, with a core/UI separation, the GUI is using Qt libraries, for Linux, OS X, Windows, and mobile phones (Android, iPhone, Windows Phone, BlackBerry)
Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints such as customary telephone sets, destinations on the public switched telephone network (PSTN) and devices or services on voice over Internet ...
The FreePBX Distro was released in 2011 as an turnkey solution for building a PBX using Asterisk, CentOS and FreePBX. [9] [10] [11] FreePBX has over 1 million active production PBXs and over 20,000 new systems added each month. [12] The core telephony engine is Asterisk (PBX), as configured by the Open Source FreePBX GUI. [13]
Blink is based on the Session Initiation Protocol, and beyond voice over IP, its features also include video, instant messaging, file transfer and multi-party conferencing sessions based on MSRP protocol, remote desktop sharing using RFB protocol (VNC), and SIMPLE presence using XCAP protocol.
Yate ( Yet Another Telephony Engine) is a free and open source communications software with support for video, voice and instant messaging. It is an extensible IP PBX under the GPLv2 with linking exception license. [3]
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G.722 is an ITU standard codec that provides 7 kHz wideband audio at data rates from 48, 56 and 64 kbit/s. This is useful for voice over IP applications, such as on a local area network where network bandwidth is readily available, and offers a significant improvement in speech quality over older narrowband codecs such as G.711, without an excessive increase in implementation complexity.