The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE).
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).
Pidgin is a free and open-source multi-platform instant messaging client, based on a library named libpurple that has support for many instant messaging protocols, allowing the user to simultaneously log in to various services from a single application, with a single interface for both popular and obsolete protocols, thus avoiding the hassle of having to deal with a new software for each device and protocol.
A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks.
Skype for Business Server is real-time communications server software that provides the infrastructure for enterprise instant messaging, presence, VoIP, ad hoc and structured conferences and PSTN connectivity through a third-party gateway or SIP trunk. These features are available within an organization, between organizations and with external users on the public internet or standard phones.
A secure telephone is a telephone that provides voice security in the form of end-to-end encryption for the telephone call, and in some cases also the mutual authentication of the call parties, protecting them against a man-in-the-middle attack. Concerns about massive growth of telephone tapping incidents led to growing demand for secure telephones.
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.
Wengo was at the beginning of 2004 a subsidiary of French telecom service provider Neuf Cegetel. As of February 2012, Wengo employs 80 people in the Paris headquarters, and is a subsidiary of Vivendi. Wengo is now repositioned as an online personal and consulting services marketplace.
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.
This is a comparison of voice over IP (VoIP) software used to conduct telephone-like voice conversations across Internet Protocol (IP) based networks. For residential markets, voice over IP phone service is often cheaper than traditional public switched telephone network (PSTN) service and can remove geographic restrictions to telephone numbers, e.g., have a PSTN phone number in a New York area code ring in Tokyo.
Twinkle is a free and open-source app for voice communications over Voice over IP (VoIP) protocol. It is designed for Linux operating systems and uses the Qt toolkit for its graphical user interface. For call signaling it employs the Session Initiation Protocol (SIP). It also features direct IP-to-IP calls. Media streams are transmitted via the Real-time Transport Protocol (RTP) which may be encrypted with the Secure Real-time Transport Protocol (SRTP) and the ZRTP security protocols.
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.
Yahoo Voice was a Voice over Internet Protocol (VoIP), PC-PC, PC-Phone and Phone-to-PC telecommunications service. It was provided by Yahoo via its Yahoo Messenger instant messaging application.
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.
Empathy is an instant messaging (IM) and voice over IP (VoIP) client which supports text, voice, video, file transfers, and inter-application communication over various IM communication protocols.
A softphone is a software program for making telephone calls over the Internet using a general purpose computer rather than dedicated hardware. The softphone can be installed on a piece of equipment such as a desktop, mobile device, or other computer and allows the user to place and receive calls without requiring an actual telephone set. Often, a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a handset, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC or with a USB phone.
CSipSimple is a Voice over Internet Protocol (VoIP) application for Google Android operating system using the Session Initiation Protocol (SIP). It is open source and free software released under the GPL-3.0-or-later license.
Acrobits is a privately owned software development company creating VoIP Clients for mobile platforms, based in Prague, Czech Republic.
WebRTC Gateway connects between WebRTC and an established VoIP technology such as SIP. WebRTC is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and messaging without the need of either internal or external plugins.
