Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services over the Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS).
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 Protocol (VoIP) networks. Its name comes from the asterisk (*) symbol for a signal used in dual-tone multi-frequency (DTMF) dialing.
Digium, Inc. is a communications technology company based in Huntsville, Alabama, and since 2018, a subsidiary of Sangoma Technologies. In 1999, Digium's founder Mark Spencer created Asterisk, the open source software project that can be used to turn a personal computer into a communications server or Voice over IP (VoIP) phone system. Today, Digium's core business lines include Switchvox, the Asterisk-based VoIP business phone system, Digium IP phones, and Asterisk telephony software and hardware products. Digium continues to make Asterisk available to the global development community for free at Asterisk.org.
A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet. This is in contrast to a standard phone which uses the traditional public switched telephone network (PSTN).
sipXecs is a free software enterprise communications system. It was initially developed as a proprietary voice over IP telephony server in 2003 by Pingtel Corporation in Boston, MA, and later extended with additional collaboration capabilities in the SIPfoundry project. Its core feature is a software implementation of the Session Initiation Protocol (SIP), which makes it an IP based communications system.
Gizmo5 was a Voice over IP communications network and a proprietary freeware soft phone for that network. On November 12, 2009, Google announced that it had acquired Gizmo5. On March 4, 2011, Google announced that the service would be discontinued as of April 3, 2011.
QuteCom was a free-software SIP-compliant VoIP client developed by the QuteCom community under the GNU General Public License (GPL). It allows users to speak to other users of SIP-compliant VoIP software at no cost. It also allows users to call landlines and cell phones, send SMS and make video calls. None of these functions are tied to a particular provider, allowing users to choose among any SIP provider.
Ekiga is a VoIP and video conferencing application for GNOME and Microsoft Windows. It is distributed as free software under the terms of the GNU GPL-2.0-or-later. It was the default VoIP client in Ubuntu until October 2009, when it was replaced by Empathy. Ekiga supports both the SIP and H.323 protocols and is fully interoperable with any other SIP compliant application and with Microsoft NetMeeting. It supports many high-quality audio and video codecs.
Zfone is software for secure voice communication over the Internet (VoIP), using the ZRTP protocol. It is created by Phil Zimmermann, the creator of the PGP encryption software. Zfone works on top of existing SIP- and RTP-programs, but should work with any SIP- and RTP-compliant VoIP-program.
ZRTP is a cryptographic key-agreement protocol to negotiate the keys for encryption between two end points in a Voice over IP (VoIP) phone telephony call based on the Real-time Transport Protocol. It uses Diffie–Hellman key exchange and the Secure Real-time Transport Protocol (SRTP) for encryption. ZRTP was developed by Phil Zimmermann, with help from Bryce Wilcox-O'Hearn, Colin Plumb, Jon Callas and Alan Johnston and was submitted to the Internet Engineering Task Force (IETF) by Zimmermann, Callas and Johnston on March 5, 2006 and published on April 11, 2011 as RFC 6189.
Mobile VoIP or simply mVoIP is an extension of mobility to a Voice over IP network. Two types of communication are generally supported: cordless/DECT/PCS protocols for short range or campus communications where all base stations are linked into the same LAN, and wider area communications using 3G/4G protocols.
FreeSWITCH is a free and open-source application server for real-time communication, WebRTC, telecommunications, video and Voice over Internet Protocol (VoIP). Multiplatform, it runs on Linux, Windows, macOS and FreeBSD. It is used to build PBX systems, IVR services, videoconferencing with chat and screen sharing, wholesale least-cost routing, Session Border Controller (SBC) and embedded communication appliances. It has full support for encryption, ZRTP, DTLS, SIPS. It can act as a gateway between PSTN, SIP, WebRTC, and many other communication protocols. Its core library, libfreeswitch, can be embedded into other projects. It is licensed under the Mozilla Public License (MPL), a free software license.
Jitsi is a collection of free and open-source multiplatform voice (VoIP), video conferencing and instant messaging applications for the web platform, Windows, Linux, macOS, iOS and Android. The Jitsi project began with the Jitsi Desktop. With the growth of WebRTC, the project team focus shifted to the Jitsi Videobridge for allowing web-based multi-party video calling. Later the team added Jitsi Meet, a full video conferencing application that includes web, Android, and iOS clients. Jitsi also operates meet.jit.si, a version of Jitsi Meet hosted by Jitsi for free community use. Other projects include: Jigasi, lib-jitsi-meet, Jidesha, and Jitsi.
Linphone is a free voice over IP softphone, SIP client and service. It may be used for audio and video direct calls and calls through any VoIP softswitch or IP-PBX. Linphone also provides the possibility to exchange instant messages. It has a simple multilanguage interface based on Qt for GUI and can also be run as a console-mode application on Linux.
Elastix is an unified communications server software that brings together IP PBX, email, IM, faxing and collaboration functionality. It has a Web interface and includes capabilities such as a call center software with predictive dialing.
Jami is a SIP-compatible distributed peer-to-peer softphone and SIP-based instant messenger for Linux, Microsoft Windows, OS X, iOS, and Android. Jami was developed and maintained by the Canadian company Savoir-faire Linux, and with the help of a global community of users and contributors, Jami positions itself as a potential free Skype replacement.
Sipdroid is a voice over IP mobile app for the Android operating system using the Session Initiation Protocol.
GNU SIP Witch is a free SIP server software with Peer-to-peer capabilities from the GNU Project. It is the GNU implementation of the Session Initiation Protocol (SIP), which is being used for the routing of the calls.
