GNU SIP Witch
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GNU SIP Witch
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. Availability SIP Witch is released as free software under the terms of version 3 or later of the GNU General Public License (GPL). It is designed for Linux, macOS, BSD and Windows and planned support for Android. In the popular Linux distributions ''Ubuntu'' and ''Fedora'' it may be installed directly from the standard package sources. Technology SIP Witch is written in the programming language ''C++'' and uses the uCommon, eXosip and GNU oSIP libraries. Features The software enables Voice over IP as part of a self-organising Peer-to-peer telephone network. It supports features like call forwarding, call distribution, call hold, presence information and (text) messages, supports encrypted calls and also enables NAT traversal to establish the peer-to-peer co ...
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GNU Project
The GNU Project () is a free software, mass collaboration project announced by Richard Stallman on September 27, 1983. Its goal is to give computer users freedom and control in their use of their computers and computing devices by collaboratively developing and publishing software that gives everyone the rights to freely run the software, copy and distribute it, study it, and modify it. GNU software grants these rights in its license. In order to ensure that the ''entire'' software of a computer grants its users all freedom rights (use, share, study, modify), even the most fundamental and important part, the operating system (including all its numerous utility programs) needed to be free software. According to its manifesto, the founding goal of the project was to build a free operating system, and if possible, "everything useful that normally comes with a Unix system so that one could get along without any software that is not free." Stallman decided to call this operating sy ...
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OSIP
Osip (Russian language, Russian ''О́сип'') is a Russian male given name, a variant of the name Joseph (name), Joseph. Notable people with the name include: * Osip Abdulov (1900–1953), Soviet actor * Osip Aptekman, Russian revolutionary * Ossip Bernstein (1882-1962), Russian-French chess player * Osip Bilchansky (1858-1879), Russian terrorist hanged for using a gun to resist arrest * Osip Bodyansky (1808-1877), Russian Imperial Slavist of Ukrainian Cossack descent * Osip Braz (1873-1936), Russian-Jewish realist painter * Osip Brik, Russian writer and literary critic, a futurist * Osip Dymov (writer), pseudonym for Yosif (Osip) Isidorovich Perelman (1878-1959), Russian writer * Osip Gelfond (1868-1942), Russian physician and Marxist philosopher * Osip Komissarov, hatter's apprentice famous for thwarting the assassination of Alexander II of Russia * Osip Kozodavlev (1754–1819), Russian statesman, politician and Minister of the Interior * Osip Mikhailovich Lerner (1847–1907), ...
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FreeSWITCH
FreeSWITCH is free and open-source server software for real-time communication applications, including WebRTC, video, and voice over Internet Protocol (VoIP). It runs on Linux, Windows, macOS, and FreeBSD. FreeSWITCH is used to build private branch exchange (PBX) telecommunication systems, IVR services, videoconferencing with chat and screen sharing, wholesale least-cost routing, Session Border Controller (SBC) and embedded communication appliances. It has support for encryption, ZRTP, DTLS, and Session Initiation Protocol (SIP). It is implemented as a core library, libfreeswitch, which can be embedded into other projects. It is distributed under the Mozilla Public License (MPL), a free software license. History The FreeSWITCH project was announced in January 2006 at O'Reilly Media's ETEL Conference. In June 2007, FreeSWITCH was selected by Truphone for use, and in August 2007, Gaboogie announced that it selected FreeSWITCH as its conferencing platform. The official release 1. ...
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List Of SIP Software
This list of SIP software documents notable software applications which use Session Initiation Protocol (SIP) as a voice over IP (VoIP) protocol. Servers Free and open-source license 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. * Asterisk * Cipango SipServlets 1.1 application server * ejabberd * FreeSWITCH * FreePBX * GNU SIP Witch * Issabel, fork of Elastix * Kamailio, formerly OpenSER * Mobicents Platform (JSLEE 1.0 compliant and SIP Servlets 1.1 compliant application server) * Mysipswitch * OpenSIPS, fork of OpenSER * SailFin * SIP Express Router (SER) * Enterprise Communications System sipXecs * Yate Proprietary license * 3Com VCX IP telephony module: back-to-back user agent SIP PBX * 3CX Phone System, for Windows, Debian 8 GNU/Linux * Aastra 5000, 800, MX-ONE * Alcatel-Lucent 5060 IP Call server * Aric ...
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Skype
Skype () is a proprietary telecommunications application operated by Skype Technologies, a division of Microsoft, best known for VoIP-based videotelephony, videoconferencing and voice calls. It also has instant messaging, file transfer, debit-based calls to landline and mobile telephones (over traditional telephone networks), and other features. Skype is available on various desktop, mobile, and video game console platforms. Skype was created by Niklas Zennström, Janus Friis, and four Estonian developers and first released in August 2003. In September 2005, eBay acquired Skype for $2.6 billion. In September 2009, Silver Lake, Andreessen Horowitz, and the Canada Pension Plan Investment Board bought 65% of Skype for $1.9 billion from eBay, valuing the business at $2.92 billion. In May 2011, Microsoft bought Skype for $8.5 billion and used it to replace their Windows Live Messenger. As of 2011, most of the development team and 44% of all the division' ...
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NAT Traversal
Network address translation traversal is a computer networking technique of establishing and maintaining Internet protocol connections across gateways that implement network address translation (NAT). NAT traversal techniques are required for many network applications, such as peer-to-peer file sharing and Voice over IP. Network address translation NAT devices allow the use of private IP addresses on private networks behind routers with a single public IP address facing the Internet. The internal network devices communicate with hosts on the external network by changing the source address of outgoing requests to that of the NAT device and relaying replies back to the originating device. This leaves the internal network ill-suited for hosting servers, as the NAT device has no automatic method of determining the internal host for which incoming packets are destined. This is not a problem for general web access and email. However, applications such as peer-to-peer file sharing, ...
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ZRTP
ZRTP (composed of Z and Real-time Transport Protocol) 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 . Overview ZRTP ("Z" is a reference to its inventor, Zimmermann; "RTP" stands for Real-time Transport Protocol) is described in the Internet Draft as a ''"key agreement protocol which performs Diffie–Hellman key exchange during call setup in-band in the Real-time Transport Protocol (RTP) media stream which has been established using some other signaling protocol such as S ...
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Presence Information
In computer and telecommunications networks, presence information is a status indicator that conveys ability and willingness of a potential communication partner—for example a user—to communicate. A user's client provides presence information (presence state) via a network connection to a presence service, which is stored in what constitutes his personal availability record (called a presentity) and can be made available for distribution to other users (called ''watchers'') to convey their availability for communication. Presence information has wide application in many communication services and is one of the innovations driving the popularity of instant messaging or recent implementations of voice over IP clients. Presence state A user client may publish a presence state to indicate its current communication status. This published state informs others that wish to contact the user of his availability and willingness to communicate. The most common use of presence today is to di ...
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Hold (telephone)
In telephony, a call may be placed on hold, in which case the connection is not terminated but no verbal communication is possible until the call is removed from hold by the same or another extension on the key telephone system. Music on hold or on hold messaging may be played for the caller while the call is on hold, especially if the call has been placed to a customer service Customer service is the assistance and advice provided by a company to those people who buy or use its products or services. Each industry requires different levels of customer service, but in the end, the idea of a well-performed service is that ... center. Alternatives to placing a caller on hold include virtual hold or virtual queuing solutions that allow scheduled or queue-based callbacks to be made to the caller. Telephony {{telephony-stub ...
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Automatic Call Distributor
An automated call distribution system, commonly known as automatic call distributor (ACD), is a telephony device that answers and distributes incoming calls to a specific group of terminals or agents within an organization. ACDs direct calls based on parameters that may include the caller's telephone number, the number they dialed, the time of day or a response to an automated voice prompt. Advanced ACD systems may use digital technologies such as Computer telephony integration (CTI), computer-supported telecommunications applications (CSTA) or IVR as input to determine the route to a person or voice announcement that will serve the caller. Experts claim that "the invention of ACD technology made the concept of a call centre possible." Background A Private Branch Exchange (PBX) is a telephone exchange device that acts as a switchboard to route phone calls within an organisation. This technology developed into Automated Call Distribution systems using computer technology to automat ...
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Call Forwarding
Call forwarding, or call diversion, is a telephony feature of all telephone switching systems which redirects a telephone call to another destination, which may be, for example, a mobile or another telephone number where the desired called party is available. Call forwarding was invented by Ernest J. Bonanno. In North America, the forwarded line usually rings once to remind the customer using call forwarding that the call is being redirected. More consistently, the forwarded line indicates its condition by stutter dial tone. Call forwarding typically can redirect incoming calls to any other domestic telephone number, but the owner of the forwarded line must pay any toll charges for forwarded calls. Call forwarding is often enabled by dialing *72 followed by the telephone number to which calls should be forwarded. Once someone answers, call forwarding is in effect. If no one answers or the line is busy, the dialing sequence must be repeated to effect call forwarding. Call forwardin ...
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Voice Over IP
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of speech, 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 (voice, fax, Short Message Service, SMS, voice-messaging) over the Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS). Overview The steps and principles involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched network, the digital information is packetized and transmission occurs as IP packets over a packet-switched network. They transport media streams using spec ...
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