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A softswitch (software switch) is a call-switching node in a telecommunications network, based not on the specialized switching hardware of the traditional telephone exchange, but implemented in software running on a general-purpose computing platform. Like its traditional counterparts it connects telephone calls between subscribers or other switching systems across a telecommunication network. Often a softswitch is implemented to switch calls using voice over IP (VoIP) technologies, [1] but hybrid systems exist.
Although the term softswitch technically refers to any such device, it is conventionally applied to a device that handles IP-to-IP phone calls, while the phrase access server or "media gateway" is used to refer to devices that either originate or terminate traditional land line phone calls. In practice, such devices can often do both. An access server might take a mobile call or a call originating from a traditional telephone line, convert it to IP traffic, then send it over an IP network to another such device, which terminates the call by reversing the process and converting the voice over IP call to circuit-switched digital systems using traditional digital time-division multiplexing (TDM) or analog POTS protocols.
The call agents are the software switching elements of the softswitch. Other components handle functions for billing, directory services, network signaling. The network elements that convert voice streams between VoIP links and traditional media technologies, such as analog telephone lines, pair-gain devices, carrier systems, are called media gateways. A call agent may control many different media gateways in geographically dispersed areas via an IP network.
The softswitch generally resides in a building owned by a telephone company, called a telephone exchange or central office, or in a data center. Such locations have high capacity connections to carry telephone calls or digital communication to other switching centers.
Access devices to the services of a softswitch range from large media gateways with high port density to integrated access devices (IAD) at office locations, to small analog telephone adaptors (ATA) which provide just one RJ11 telephone jack to a residence. Embedded multimedia terminal adapters (eMTAs) are also built into cable television modems.
A softswitch routes telephone calls using the Signalling System No. 7 (SS7) network. SS7 modules may be implemented directly in the softswitch, or accessed from standalone signaling servers.
At the turn of the 21st century with IP Multimedia Subsystem (IMS), the softswitch element is represented by the media gateway controller (MGC) element, while the term softswitch is rarely used in the IMS context, where it is called an access gateway control function (AGCF).
VoIP softswitches are subdivided into Class 4 and Class 5 systems, in analogy to the traditional functions in the public switched telephone network.
Softswitches used for transit VoIP traffic between carriers are usually called Class 4 softswitches. Analogous with other Class 4 telephone switches, the main function of the Class 4 softswitch is the routing of large volumes of long-distance VoIP calls. The most important characteristics of Class 4 softswitch are protocol support and conversion, transcoding, calls per second rate, average time of one call routing, number of concurrent calls.
Class 5 softswitches are intended to serve subscribers. Class 5 softswitches are characterized by additional services for end-users and corporate clients such as IP PBX features, call center services, calling card platform, types of authorization, QoS, Business Groups and other features similar to other Class 5 telephone switches.
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE).
Telephony is the field of technology involving the development, application, and deployment of telecommunication services for the purpose of electronic transmission of voice, fax, or data, between distant parties. The history of telephony is intimately linked to the invention and development of the telephone.
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).
The Skinny Client Control Protocol (SCCP) is a proprietary network terminal control protocol originally developed by Selsius Systems, which was acquired by Cisco Systems in 1998.
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.
An analog telephone adapter (ATA) is a device for connecting traditional analog telephones, fax machines, and similar customer-premises devices to a digital telephone system or a voice over IP telephony network.
A business telephone system is a multiline telephone system typically used in business environments, encompassing systems ranging in technology from the key telephone system (KTS) to the private branch exchange (PBX).
The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem (IMS) is a standardised architectural framework for delivering IP multimedia services. Historically, mobile phones have provided voice call services over a circuit-switched-style network, rather than strictly over an IP packet-switched network. Alternative methods of delivering voice (VoIP) or other multimedia services have become available on smartphones, but they have not become standardized across the industry. IMS is an architectural framework that provides such standardization.
A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks.
Cable telephony is a form of digital telephony over cable TV networks. A telephone interface installed at the customer's premises converts analog signals from the customer's in-home wiring to a digital signal, which is then sent over the cable connection to the company's switching center. The signal is then sent on to the public switched telephone network (PSTN). Cable telephone provides another revenue stream for cable television system operators and gives the consumer the convenience of a single bill for combined television, internet and telephone services.
The 21st Century Network (21CN) programme is the data and voice network transformation project, under way since 2004, of the UK telecommunications company BT Group plc. It was intended to move BT's telephone network from the AXE/System X Public Switched Telephone Network (PSTN) to an Internet Protocol (IP) system. As well as switching over the PSTN, BT planned to deliver many additional services over their new data network, such as on-demand interactive TV services.
The next-generation network (NGN) is a body of key architectural changes in telecommunication core and access networks. The general idea behind the NGN is that one network transports all information and services by encapsulating these into IP packets, similar to those used on the Internet. NGNs are commonly built around the Internet Protocol, and therefore the term all IP is also sometimes used to describe the transformation of formerly telephone-centric networks toward NGN.
Direct inward dialing (DID), also called direct dial-in (DDI) in Europe and Oceania, is a telecommunication service offered by telephone companies to subscribers who operate a private branch exchange (PBX) system. The feature provides service for multiple telephone numbers over one or more analog or digital physical circuits to the PBX, and transmits the dialed telephone number to the PBX so that a PBX extension is directly accessible for an outside caller, possibly by-passing an auto-attendant.
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).
Network-based Call Signaling (NCS) is a profile of the Media Gateway Control Protocol (MGCP) for use in PacketCable applications for voice-over-IP.
A media server is a computer appliance or an application software that stores digital media and makes it available over a network.
T.38 is an ITU recommendation for allowing transmission of fax over IP networks (FoIP) in real time.
The Media Gateway Control Protocol (MGCP) is a signaling and call control communications protocol used in voice over IP (VoIP) telecommunication systems. It implements the media gateway control protocol architecture for controlling media gateways connected to the public switched telephone network (PSTN). The media gateways provide conversion of traditional electronic media to the Internet Protocol (IP) network. The protocol is a successor to the Simple Gateway Control Protocol (SGCP), which was developed by Bellcore and Cisco, and the Internet Protocol Device Control (IPDC).
The media gateway control protocol architecture is a methodology of providing telecommunication services using decomposed multimedia gateways for transmitting telephone calls between an Internet Protocol network and traditional analog facilities of the public switched telephone network (PSTN). The architecture was originally defined in RFC 2805 and has been used in several prominent voice over IP (VoIP) protocol implementations, such as the Media Gateway Control Protocol (MGCP) and Megaco (H.248), both successors to the obsolete Simple Gateway Control Protocol (SGCP).