H.450 refers to a set of standards created by the International Telecommunication Union (ITU) to define several Supplementary Services of the packet based telecommunication protocol known as H.323. It parallels another set of standards known as QSIG which define similar services for ISDN based networks.
H.450.1, Supplementary Services Framework
The general mechanism for delivering supplementary services is explained in this paper. Supplementary services messages are exchanged by means of ROSE (Remote Operations Service Extension).
H.450.2, Call Transfer Supplementary Service
E.g. explains how a party-B can turn an active call between party-A and party-B into a call between party-A and a new party-C.
H.450.3, Call Diversion Supplementary Service
E.g. explains how an IP phone can activate a diversion to e.g. a cell phone. E.g. explains how an IP phone can interrogate whether it has any active diversion.
H.450.4, Call Hold Supplementary Service
E.g. explains how a call can be put on hold and be fed with a music on hold.
H.450.5, Call Park and Pickup Supplementary Service
Think of big warehouse, where a call is coming in at the front-desk for Mrs. Smith. The front-desk parks the call and broadcasts via the intercom: "Mrs. Smith, please 223". Mrs. Smith proceeds soon after to the next wall phone, dials 223 and gets the call.
H.450.6, Call Waiting Supplementary Service
Explains how to signal a second call to an IP phone already engaged in an active call.
H.450.7, Message Waiting Indication Supplementary Service
Explains elements related to voicebox systems and how these can be implemented by means of H.323.
H.450.8, Name Identification Supplementary Service
Explains how names are displayed or how to intentionally call incognito.
H.450.9, Call Completion Supplementary Service
Explains how to schedule an automatic call-back request in case of a remote party being busy in a call or being absent for a while and becoming available later on.
H.450.10, Call Offer Supplementary Service
A variation of Call Waiting. Also known as "Camp-On".
H.450.11, Call Intrusion Supplementary Service
Explains how e.g. the secretary of a CEO can intentionally and legally intrude into a call of her boss, in order to communicate urgent information.
H.450.12, Common Information Additional Network Feature for H.323
A means to communicate additional miscellaneous information between endpoints. E.g. whether certain features are available, and/or allowed.
Related Research Articles
QSIG is an ISDN based signaling protocol for signaling between private branch exchanges (PBXs) in a private integrated services network (PISN). It makes use of the connection-level Q.931 protocol and the application-level ROSE protocol. ISDN "proper" functions as the physical link layer.
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 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).
Caller identification is a telephone service, available in analog and digital telephone systems, including voice over IP (VoIP), that transmits a caller's telephone number to the called party's telephone equipment when the call is being set up. The caller ID service may include the transmission of a name associated with the calling telephone number, in a service called Calling Name Presentation (CNAM). The service was first defined in 1993 in International Telecommunication Union—Telecommunication Standardization Sector (ITU-T) Recommendation Q.731.3.
Technological convergence, also known as digital convergence, is the tendency for technologies that were originally unrelated to become more closely integrated and even unified as they develop and advance. For example, watches, telephones, television, computers, and social media platforms began as separate and mostly unrelated technologies, but have converged in many ways into interrelated parts of a telecommunication and media industry, sharing common elements of digital electronics and software.
Call forwarding, or call diversion, is a telephony feature of some telephone switching systems which redirects a telephone call to another destination, which may be, for example, a mobile or another 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 forwarding is disabled by dialing *73. This feature requires a subscription from the telephone company. Also available in some areas is Remote Access to call forwarding, which permit the control over call forwarding from telephones other than the subscriber's telephone. VOIP and cable telephone systems also often allow call forwarding to be set up and directed via their web portals.
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 session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks.
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.
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.
UNIStim is a deprecated Telecommunications protocol developed by Nortel for IP Phone and IP PBX communications.
Fixed–mobile convergence (FMC) is a change in telecommunications that removes differences between fixed and mobile networks.
Web-based VoIP is the integration of voice over IP technologies into the facilities and methodologies of the World-Wide Web. It enables digital communication sessions between Web users, or to users of traditional telecommunication services.
A softphone is a software program for making telephone calls over the Internet using a general purpose computer rather than dedicated hardware. The softphone can be installed on a piece of equipment such as a desktop, mobile device, or other computer and allows the user to place and receive calls without requiring an actual telephone set. Often, a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a handset, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC or with a USB phone.
A communication protocol is a system of rules that allows two or more entities of a communications system to transmit information via any kind of variation of a physical quantity. The protocol defines the rules, syntax, semantics and synchronization of communication and possible error recovery methods. Protocols may be implemented by hardware, software, or a combination of both.
Avaya 9600-series IP deskphones are 15 different desk handset devices that are used for unified communications. The phones are compatible with the Avaya Aura platform of products and IP office systems. The systems add high-quality voice codecs like the G.722 codec and new menus over older IP phone series. The 9620 includes 16 MB of flash memory and the 9630 includes 32 MB of flash memory.
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).
Assistive Technology for the Deaf and Hard of Hearing is special technology made to assist them including Hearing aids, Video relay services, tactile devices, alerting devices and technology for supporting communication.
This page is based on this Wikipedia article Text is available under the CC BY-SA 4.0 license; additional terms may apply. Images, videos and audio are available under their respective licenses.
