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).
Extensible Messaging and Presence Protocol is an open communication protocol designed for instant messaging (IM), presence information, and contact list maintenance. Based on XML, it enables the near-real-time exchange of structured data between two or more network entities. Designed to be extensible, the protocol offers a multitude of applications beyond traditional IM in the broader realm of message-oriented middleware, including signalling for VoIP, video, file transfer, gaming and other uses.
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 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 by Pingtel Corporation in 2003 as a voice over IP telephony server located in Boston, MA. The server was later extended with additional collaboration capabilities as part of the SIPfoundry project. Since its extension, sipXecs now acts as a software implementation of the Session Initiation Protocol (SIP), making it a full IP-based communications system.
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.
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.
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 application for voice communications over Voice over IP (VoIP) protocol.
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.
Sipdroid is a voice over IP mobile app for the Android operating system using the Session Initiation Protocol.
Phoner and PhonerLite are softphone applications for Windows operating systems available as freeware. Phoner is a multiprotocol telephony application supporting telephony via CAPI, TAPI and VoIP, while PhonerLite provides a specialized and optimized user interface for VoIP only. Beside the different user interface focus both programs share the same code base.
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.
