Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for voice calls for the delivery of voice communication sessions over Internet Protocol (IP) networks, such as the Internet.
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.
Microsoft NetMeeting is a discontinued VoIP and multi-point videoconferencing program offered by Microsoft. NetMeeting allows multiple clients to host and join a call that includes video and audio, text chat, application and desktop sharing, and file sharing. It was originally bundled with Internet Explorer 3 and then with Windows versions from Windows 95 to Windows Server 2003.
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).
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 GPL-2.0-or-later license. 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.
The following tables compare general and technical information for a variety of audio coding formats.
Ekiga was a VoIP and video conferencing application for GNOME and Microsoft Windows. It was 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.
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.
Mobile VoIP or simply mVoIP is an extension of mobility to a voice over IP network. Two types of communication are generally supported: cordless telephones using DECT or PCS protocols for short range or campus communications where all base stations are linked into the same LAN, and wider area communications using 3G or 4G 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.
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 was an instant messaging (IM) and voice over IP (VoIP) client which supported text, voice, video, file transfers, and inter-application communication over various IM communication protocols.
Jami is a SIP-compatible distributed peer-to-peer softphone and SIP-based instant messenger for Linux, Microsoft Windows, macOS, 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.
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.
WebRTC is a free and open-source project providing web browsers and mobile applications with real-time communication (RTC) via application programming interfaces (APIs). It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps.
Acrobits is a privately owned software development company creating VoIP Clients for mobile platforms, based in Prague, Czech Republic.
oSIP is a free software library for VoIP applications implementing lower layers of Session Initiation Protocol (SIP). The library includes the minimal codebase required by any SIP application and offers enough flexibility to implement any SIP extension or behavior. Started in September 2000 and published in April 2001, oSIP is among the oldest SIP open source stack still being developed and maintained. The project was made part of the GNU Project as GNU oSIP in 2002.
Web Call Server is unified intermedia server software developed by Flashphoner. It is a server-side platform, implemented in Java, dedicated for streaming video over wide range of communication protocols, including:
