In telephony, call control refers to the software within a telephone switch that supplies its central function. Call control decodes addressing information and routes telephone calls from one end point to another. It also creates the features that can be used to adapt standard switch operation to the needs of users. These are called supplementary services and are commonly invoked by a Vertical service code. Examples include "Call waiting", "Call forward on busy" etc.
Call control software, because of its central place in the operation of the telephone network, is marked by both complexity and reliability. Call control systems will typically require many thousands of person years in development. They will contain millions of lines of high level code. However they must and do meet reliability requirements that specify switch down time of only a few minutes in forty years.
The required functionality and reliability of call control is a major challenge for Voice over IP (VoIP) systems. VoIP systems are based on Internet standards and technology which have not previously attempted to satisfy such complex and demanding requirements as those that specify call control.
An alternative name often used is call processing.[ citation needed ]
In a VoIP network, call control is one of three major categories of communications traffic, the other two being call signaling and media communications. Call control uses Q.931, a connection protocol for digital networks, especially VoIP systems. Messages are transmitted as octets as specified in ITU H.245, which resolves the type of call media to be used (for example, conventional call, videoconferencing, or VoIP), and then manages the connection after it has been established. Call control functions include, but are not limited to, the determination of master/slave status for the endpoints, monitoring of the status of the endpoints, modification of the parameters of a connection, termination of a connection, and restarting a terminated or failed connection. [1]
The Open Systems Interconnection model is a conceptual model that characterises and standardises the communication functions of a telecommunication or computing system without regard to its underlying internal structure and technology. Its goal is the interoperability of diverse communication systems with standard communication protocols.
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.
Plain old telephone service (POTS), or plain ordinary telephone system, is a retronym for voice-grade telephone service employing analog signal transmission over copper loops. POTS was the standard service offering from telephone companies from 1876 until 1988 in the United States when the Integrated Services Digital Network (ISDN) Basic Rate Interface (BRI) was introduced, followed by cellular telephone systems, and voice over IP (VoIP). POTS remains the basic form of residential and small business service connection to the telephone network in many parts of the world. The term reflects the technology that has been available since the introduction of the public telephone system in the late 19th century, in a form mostly unchanged despite the introduction of Touch-Tone dialing, electronic telephone exchanges and fiber-optic communication into the public switched telephone network (PSTN).
The end-to-end principle is a design framework in computer networking. In networks designed according to this principle, guaranteeing certain application-specific features, such as reliability and security, requires that they reside in the communicating end nodes of the network. Intermediary nodes, such as gateways and routers, that exist to establish the network, may implement these to improve efficiency but cannot guarantee end-to-end correctness.
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.
The Communications Assistance for Law Enforcement Act (CALEA), also known as the "Digital Telephony Act," is a United States wiretapping law passed in 1994, during the presidency of Bill Clinton.
A softswitch 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, but hybrid systems exist.
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 in technology from the key telephone system (KTS) to the private branch exchange (PBX).
The Gateway Control Protocol is an implementation of the media gateway control protocol architecture for providing telecommunication services across a converged internetwork consisting of the traditional public switched telephone network (PSTN) and modern packet networks, such as the Internet. H.248 is the designation of the recommendations developed by the ITU Telecommunication Standardization Sector (ITU-T) and Megaco is a contraction of media gateway control protocol used by the earliest specifications by the Internet Engineering Task Force (IETF). The standard published in March 2013 by ITU-T is entitled H.248.1: Gateway control protocol: Version 3.
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).
Mobile VoIP or simply mVoIP is an extension of mobility to a Voice over IP network. Two types of communication are generally supported: cordless/DECT/PCS protocols for short range or campus communications where all base stations are linked into the same LAN, and wider area communications using 3G/4G protocols.
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 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).
In digital communications networks, packet processing refers to the wide variety of algorithms that are applied to a packet of data or information as it moves through the various network elements of a communications network. With the increased performance of network interfaces, there is a corresponding need for faster packet processing.
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).
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.