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 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).
The Telephony Application Programming Interface (TAPI) is a Microsoft Windows API, which provides computer telephony integration and enables PCs running Microsoft Windows to use telephone services. Different versions of TAPI are available on different versions of Windows. TAPI allows applications to control telephony functions between a computer and telephone network for data, fax, and voice calls. It includes basic functions, such as dialing, answering, and hanging up a call. It also supports supplementary functions, such as hold, transfer, conference, and call park found in PBX, ISDN, and other telephone systems.
Voice over Internet Protocol (VoIP), also known as IP telephony, refers to a set of technologies used for voice communication sessions over Internet Protocol (IP) networks, such as the Internet. VoIP enables voice calls to be transmitted as data packets, facilitating various methods of voice communication, including traditional applications like Skype, Microsoft Teams, Google Voice, and VoIP phones. Regular telephones can also be used for VoIP by connecting them to the Internet via analog telephone adapters (ATAs), which convert traditional telephone signals into digital data packets that can be transmitted over IP networks.
Telephone number mapping is a system of unifying the international telephone number system of the public switched telephone network with the Internet addressing and identification name spaces. Internationally, telephone numbers are systematically organized by the E.164 standard, while the Internet uses the Domain Name System (DNS) for linking domain names to IP addresses and other resource information. Telephone number mapping systems provide facilities to determine applicable Internet communications servers responsible for servicing a given telephone number using DNS queries.
A telephone numbering plan is a type of numbering scheme used in telecommunication to assign telephone numbers to subscriber telephones or other telephony endpoints. Telephone numbers are the addresses of participants in a telephone network, reachable by a system of destination code routing. Telephone numbering plans are defined in each of the administrative regions of the public switched telephone network (PSTN) and in private telephone networks.
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.
VoIP spam or SPIT is unsolicited, automatically dialed telephone calls, typically using voice over Internet Protocol (VoIP) technology.
A business telephone system is a telephone system typically used in business environments, encompassing the range of 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. Various voice over IP technologies are available on smartphones; IMS provides a standard protocol across vendors.
A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks.
A Name Authority Pointer (NAPTR) is a type of resource record in the Domain Name System of the Internet.
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 private branch exchange (PBX) systems. 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).
In the traditional telephony context, third party call control allows one entity to set up and manage a communications relationship or telephone call between two or more other parties. Third party call control is often used for operator services and conferencing.
The SIP URI scheme is a Uniform Resource Identifier (URI) scheme for the Session Initiation Protocol (SIP) multimedia communications protocol. A SIP address is a URI that addresses a specific telephone extension on a voice over IP system. Such a number could be a private branch exchange or an E.164 telephone number dialled through a specific gateway. The scheme was defined in RFC 3261.
A telephone number is a sequence of digits assigned to a landline telephone subscriber station connected to a telephone line or to a wireless electronic telephony device, such as a radio telephone or a mobile telephone, or to other devices for data transmission via the public switched telephone network (PSTN), or other public and private networks. Modern smart phones have added a built-in layer of abstraction whereby individuals or businesses are saved into a contacts application and the numbers no longer have to be written down or memorized.
Voxbone S.A. is a global communication as a service (CaaS) company that is a wholly owned subsidiary of Bandwidth, Inc., with offices in locations including Brussels, London, San Francisco, Austin, Simi Valley, Dublin, Singapore and Iași. Voxbone became a part of Bandwidth on November 2, 2020.
The Session Initiation Protocol (SIP) is the signaling protocol selected by the 3rd Generation Partnership Project (3GPP) to create and control multimedia sessions with multiple participants in the IP Multimedia Subsystem (IMS). It is therefore a key element in the IMS framework.
STIR/SHAKEN, or SHAKEN/STIR, is a suite of protocols and procedures intended to combat caller ID spoofing on public telephone networks. Caller ID spoofing is used by robocallers to mask their identity or to make it appear the call is from a legitimate source, often a nearby phone number with the same area code and exchange, or from well-known agencies like the Internal Revenue Service or Ontario Provincial Police. This sort of spoofing is common for calls originating from voice-over-IP (VoIP) systems, which can be located anywhere in the world.
