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This list of SIP software documents notable software applications which use Session Initiation Protocol (SIP) as a voice over IP (VoIP) protocol.
| | This article needs additional citations for verification .(April 2012) |
This list of SIP software documents notable software applications which use Session Initiation Protocol (SIP) as a voice over IP (VoIP) protocol.
A SIP server, also known as a SIP proxy, manages all SIP calls within a network and takes responsibility for receiving requests from user agents for the purpose of placing and terminating calls.
HCL Sametime Premium is a client–server application and middleware platform that provides real-time, unified communications and collaboration for enterprises. Those capabilities include presence information, enterprise instant messaging, web conferencing, community collaboration, and telephony capabilities and integration. Currently it is developed and sold by HCL Software, a division of Indian company HCL Technologies, until 2019 by the Lotus Software division of IBM.
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.
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.
Kamailio, formerly OpenSER (and sharing some common history with SIP Express Router (SER)), is a SIP server licensed under the GPL-2.0-or-later license. It can be configured to act as a SIP registrar, proxy or redirect server, and features presence support, RADIUS / syslog accounting and authorization, XML-RPC and JSON-RPC-based remote control, SQL and NoSQL backends, IMS / VoLTE extensions and others.
Kamailio is a Hawaiian word. Kama'ilio means talk, to converse. "It was chosen for its special flavour."
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.
FreeSWITCH is a free and open-source telephony software for real-time communication protocols using audio, video, text and other forms of media. The software has applications in WebRTC, voice over Internet Protocol (VoIP), video transcoding, Multipoint Control Unit (MCU) functionality and supports Session Initiation Protocol (SIP) features.
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.
An IP PBX is a system that connects telephone extensions to the public switched telephone network (PSTN) and provides internal communication for a business. An IP PBX is a PBX system with IP connectivity and may provide additional audio, video, or instant messaging communication utilizing the TCP/IP protocol stack.
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.
The 3CXPhone System is the software-based private branch exchange (PBX) Phone system developed and marketed by the company, 3CX. The 3CX Phone System is based on the SIP standard and enables extensions to make calls via the public switched telephone network (PSTN) or via Voice over Internet Protocol (VoIP) services on premises, in the cloud, or via a cloud service owned and operated by the 3CX company. The 3CX Phone System is available for Windows, Linux, Raspberry Pi and supports standard SIP soft/hard phones, VoIP services, faxing, voice and web meetings, as well as traditional PSTN phone lines.
SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. Most unified communications applications provide voice, video, and other streaming media applications such as desktop sharing, web conferencing, and shared whiteboard.
VaxTele SIP Server SDK is a complete development toolkit, which allows software vendors and Internet telephony service providers (ITSP) to develop SIP Server and (SIP) Session Initiation Protocol based VoIP systems for Microsoft Windows to install computer to computer voice chat, chat rooms, IVR systems, call center services, calling card services, dial/receive computer to PSTN and mobile phone calling services.
The FreePBX Distro is a freeware unified communications software system that consists of a graphical user interface (GUI) for configuring, controlling, and managing Asterisk PBX software. The FreePBX Distro includes packages that offer VoIP, PBX, Fax, IVR, voice-mail and email functions.
Dialexia Communications, Inc. is a privately held Canadian corporation headquartered in Montréal, Quebec, that develops, manufactures, and sells VoIP-based Telecommunication products and services.
HERO Hosted PBX is a SIP-based hosted IP-PBX business phone system, first released in 2008 by Canadian telecommunications software provider Dialexia. The HERO software allows users to connect multiple phones, share lines among several phones and implement business PBX telephone phone features such as voicemail, caller ID, call forwarding & call recording into their virtual PBX. The software is also suitable for multi-office connections, connecting branches which are geographically distant from each other. Dialexia Communications, Inc. released the latest version of HERO Hosted PBX (4.3) in 2013.