Dual-tone multi-frequency signaling (DTMF) is a telecommunication signaling system using the voice-frequency band over telephone lines between telephone equipment and other communications devices and switching centers. DTMF was first developed in the Bell System in the United States, and became known under the trademark Touch-Tone for use in push-button telephones supplied to telephone customers, starting in 1963. DTMF is standardized as ITU-T Recommendation Q.23. It is also known in the UK as MF4.
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).
Circuit switching is a method of implementing a telecommunications network in which two network nodes establish a dedicated communications channel (circuit) through the network before the nodes may communicate. The circuit guarantees the full bandwidth of the channel and remains connected for the duration of the communication session. The circuit functions as if the nodes were physically connected as with an electrical circuit.
Voice over Internet Protocol (VoIP), also called IP telephony, is a methodology 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 public Internet, rather than via the public switched telephone network (PSTN).
Signaling System No. 7 (SS7) is a set of telephony signaling protocols developed in 1975, which is used to set up and tear down telephone calls in most parts of the world-wide public switched telephone network (PSTN). The protocol also performs number translation, local number portability, prepaid billing, Short Message Service (SMS), and other services.
The public switched telephone network (PSTN) is the aggregate of the world's circuit-switched telephone networks that are operated by national, regional, or local telephony operators, providing infrastructure and services for public telecommunication. The PSTN consists of telephone lines, fiber optic cables, microwave transmission links, cellular networks, communications satellites, and undersea telephone cables, all interconnected by switching centers, thus allowing most telephones to communicate with each other. Originally a network of fixed-line analog telephone systems, the PSTN is now almost entirely digital in its core network and includes mobile and other networks, as well as fixed telephones.
A softswitch is a central device in a telecommunications network which connects telephone calls from one phone line to another, across a telecommunication network or the public Internet, entirely by means of software running on a general-purpose computer system. Most landline calls are routed by purpose-built electronic hardware; however, soft switches using general purpose servers and VoIP technology are becoming more popular.
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.
A business telephone system is a multiline telephone system typically used in business environments, encompassing systems ranging from the small key telephone system (KTS) to the large private branch exchange (PBX).
In telecommunications, a long-distance call (U.S.) or trunk call (U.K.) is a telephone call made to a location outside a defined local calling area. Long-distance calls are typically charged a higher billing rate than local calls. The term is not necessarily synonymous with placing calls to another telephone area code.
A dialing plan establishes the permitted sequences of digits dialed on subscriber or station lines with subscriber premises equipment, such as telephones and private branch exchange (PBX) systems. Dialing plans in the public switch telephone network (PSTN) have traditionally been more commonly referred to as dialing procedures. The dialing plan of a private telephone system or a customer premise equipment, such as an analog telephone adapter (ATA) or an IP phone, is sometimes also called dial plan. The digit sequences (numbers) permissible in a dialing plan may be as short as a single digit, e.g. for reaching an operator, or as long as a complete international telephone number, including trunk prefixes and international prefixes.
A digital subscriber line (DSL) modem is a device used to connect a computer or router to a telephone line which provides the digital subscriber line service for connectivity to the Internet, which is often called DSL broadband.
A computer network is a digital telecommunications network which allows nodes to share resources. In computer networks, computing devices exchange data with each other using connections between nodes. These data links are established over cable media such as wires or optic cables, or wireless media such as Wi-Fi.
FreeSWITCH is a free and open-source application server for real-time communication, WebRTC, telecommunications, video and Voice over Internet Protocol (VoIP). Multiplatform, it runs on Linux, Windows, macOS and FreeBSD. It is used to build PBX systems, IVR services, videoconferencing with chat and screen sharing, wholesale least-cost routing, Session Border Controller (SBC) and embedded communication appliances. It has full support for encryption, ZRTP, DTLS, SIPS. It can act as a gateway between PSTN, SIP, WebRTC, and many other communication protocols. Its core library, libfreeswitch, can be embedded into other projects. It is licensed under the Mozilla Public License (MPL), a free software license.
H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences.
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 on Internet Protocol (IP) networks connected to the public switched telephone network (PSTN). 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).
A telephone exchange is a telecommunications system used in the public switched telephone network or in large enterprises. An exchange consists of electronic components and in older systems also human operators that interconnect (switch) telephone subscriber lines or virtual circuits of digital systems to establish telephone calls between subscribers.
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).
A VoIP gateway is a gateway device that uses Internet Protocols to transmit and receive voice communications (VoIP). The general term is ambiguous and can mean many different things. There are many such devices. They are quickly becoming the most common type of voice phone service in many areas.
