IP Telephony Dictionary - More than 10,937 Terms and Definitions

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	Tehrani's IP Telephony Dictionary. Over 10,000 of the latest IP Telephony and VoIP Terms and Definitions along with            + 400 diagrams and photographs!!!
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Sample Definitions

Analog Telephone Adapter (ATA)-Analog Telephone Adapter (ATA) is a device that converts analog telephone signals into another format (such as digital Internet protocol). These adapter boxes may provide a single function such as providing Internet telephone service or they may convert digital signals into several different forms such as audio, data, and video. When adapter boxes convert into multiple information forms, they may be called multimedia terminal adapters (MTAs) or integrated access devices (IADs).
Analog telephone adapters (ATA) must convert both the audio signals (voice) and control signals (such as touch tone or hold requests) into forms that can be sent and received via the Internet.
Call Server-A call server is a particular form of application server that manages the setup or connection of telephone calls. The call server will receive call setup request messages, determine the status of destination devices, check the authorization of users to originate and/or receive calls, and create and send the necessary messages to process the call requests.
Data Network-A data network is a system that transfers data between network access points (nodes) through data switching, system control and interconnection transmission lines.
Echo Canceling-Echo cancellation is a process of extracting an original transmitted signal from the received signal that contains one or more delayed signals (copies of the original signal). Echoes may be created in a baseband or broadband signal. When echoes occur on an audio baseband signal, it is usually through acoustic feedback where some of the audio signal transferring from a speaker into a microphone. When echoes occur on a broadband signal, it is usually the result of the same signal (such as a radio signal) that travels on different paths to reach its destination. In either case, echoed signals cause distortion and may be removed by performing via advanced signal analysis and filtering.
Ethernet-Ethernet is a packet based transmission protocol that is primarily used in LANs. Ethernet is the common name for the IEEE 802.3 industry specification and it is often characterized by its data transmission rate and type of transmission medium (e.g., twisted pair is T and fiber is F).
Ethernet systems in 1972 operated at 1 Mbps. In 1992, Ethernet progressed to 10 Mbps data transfer speed (called 10 Base T). In 2001, Ethernet data transfer rates included 100 Mbps (100 BaseT) and 1 Gbps (1000 Base T). In the year 2000, 10 Gigabit fiber Ethernet prototypes had been demonstrated.
Forking Proxy Server-A proxy server that forwards a communication session request to more than one device on behalf of the communication connection request.
G.723-An International Telecommunication Union (ITU) standard for audio codecs that provides for compressed digital audio over standard analog telephone lines.
G.729-G.729 is a low bit rate speech coder that was developed in 1995. It has low delay due to a small frame size of 10 msec and look ahead of 5 msec. It has a relatively high voice quality level for the low 8 kbps data transmission rate. There are two versions of G.729: G.729 and G.729 A.
H.323-H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that may not provide a guaranteed Quality of Service (QoS). H.323 specifies techniques for compressing and transmitting real-time voice, video, and data between a pair of videoconferencing workstations. It also describes signaling protocols for managing audio and video streams, as well as procedures for breaking data into packets and synchronizing transmissions across communications channels.
Integrated Access Device (IAD)- A device that converts multiple types of input signals into a common communications format. IADs are commonly used in PBX systems to integrate different types of telephone devices (e.g. analog phone, digital phone and fax) onto a common digital medium (e.g. T1 or E1 line).
Internet Protocol Private Branch Exchange (IPBX) or (IP PBX)-A private local telephone system that uses Internet protocol (IP) to provide telephone service within a building or group of buildings in a small geographic area. IPBX systems are often local area network (LAN) systems that interconnect IP telephones. IPBX systems use a IP telephone server to provide for call processing functions and to control gateways access that allows the IPBX to communicate with the public switched telephone network and other IPBX's that are part of its network. IPBX systems can provide advanced call processing features such as speed dialing, call transfer, and voice mail along with integrating computer telephony applications. Some of the IPBX standards include H.323, MGCP, MEGACO, and SIP.
Internet Protocol Telephony (IP Telephony)-IP telephone systems provide voice or multimedia communication services through the use Internet protocol (IP) networks. These IP networks initiate, process, and receive voice or multimedia communications using IP protocol. These IP systems may be public IP systems (e.g. the Internet), private data systems (e.g. LAN based), or a hybrid of public and private systems.
Internet Telephone (IP Telephone)-A telephone device that is specifically designed to communicate through the Internet without the need for a voice gateway. Internet telephones contain embedded software that allows them to initiate and receive calls through the Internet using standard protocols such as H.323 or SIP.
Internet Telephony Service Provider (ITSP)-Internet Telephony Service Providers (ITSPs) are companies that provide telephone service using the Internet. ITSPs setup and manage calls between Internet telephones and other telephone type devices.
An ITSP coordinates Internet telephone devices so they can use the Internet as a connection path between other telephones. ITSPs are commonly used to connect Internet telephones or PC telephones to telephones that are connected to the public telephone network. This is accomplished by using gateways. Gateways convert packets of audio data from the Internet into standard telephone signals.
IP Centrex System- A system that provides Centrex services to customers using Internet protocol (IP) connections. IP Centrex allows customers to have and use features that are typically associated with a private branch exchange (PBX) without the purchase of PBX switching systems.
IP Multimedia Subsystem (IMS)- IP multimedia subsystem is service based architecture that uses Internet protocol (IP) based systems to provide enhanced multimedia services. IMS evolved from the evolution of the 3rd generation mobile telephone standards that enabled users to access multimedia services using any type of access network that could use Internet protocols.
Local Area Network (LAN)-Local area networks (LANs) are private data communication networks that use high-speed digital communications channels for the interconnection of computers and related equipment in a limited geographic area. LANs can use fiber optic, coaxial, twisted-pair cables, or radio transceivers to transmit and receive data signals. LAN's are networks of computers, normally personal computers, connected together in close proximity (office setting) to each other in order to share information and resources. The two predominant LAN architectures are token ring and Ethernet. Other LAN technologies are ArcNet, AppleTalk, and fiber distributed data interface (FDDI).
Media Gateway (MG)-A network component which converts one media stream to another. In IP telephony this most commonly refers to a device which converts IP streams (such as audio) to the TDM or analog equivalent. A media gateway may interact with call controllers, proxies, and softswitches via proprietary or standard protocols such as MGCP, Megaco (H.248) , and SIP.
Media Gateway Control Protocol (MGCP)-MGCP is a control protocol that uses text or binary format messages to setup, manage, and terminate multimedia communication sessions in a centralized communications system. This differs from other multimedia control protocol systems (such as H.323 or SIP) that allow the end points in the network to control the communication session. MGCP is specified in RFC 2705 and it was first drafted in 1998. MGCP forms the basis of the PacketCable NCS protocol.
Mobile Video-Mobile video is the transferring of signals that carry moving picture information to mobile devices. Mobile video is commonly associated with supplying video signals to mobile telephones.
Packet Buffer-Memory space set aside for storing a packet awaiting transmission or for storing a received packet. The memory may be located in the network interface controller or in the computer to which the controller is connected.
Proxy Server -Proxy servers are computing devices (typically a server) that interface between data processing devices (e.g. computers) and other devices within a communications network. These devices may be located on the same local area network or an external network (e.g. the Internet). A proxy server usually has access to at least two communication interfaces. One interface communicates with a device requesting services (e.g. a client) and a device that is being requested for a service (the server).
Q.931-A telecom call processing signaling protocol that is used in telephone communication systems. The Q.931 protocol defines the messages and formats are control messages that are created by the end communication device. Some of the common information contained in Q.931 messages include call setup and tear down messages, called and calling party telephone numbers, and other access control signaling messages.
Session Initiation Protocol (SIP)-SIP is an application layer protocol that uses text format messages to setup, manage, and terminate multimedia communication sessions. SIP is a simplified version of the ITU H.323 packet multimedia system. SIP is defined in RFC 2543.
Signaling Transport (SIGTRAN)-A set of standards that were defined by the Internet engineering task force (IETF) that contain a set of protocols that are suitable to provide signaling control messages (such as SS7 message) over an Internet Protocol (IP) network.
Stream Control Transport Protocol (SCTP)-A protocol that is used to coordinate the sending of signaling information over real time communication sessions. SCTP is defined in RFC 2960.
Voice over Internet Protocol (VoIP)-Voice Over Internet Protocol (VoIP) is a process of sending voice telephone signals over the Internet or other data network. If the telephone signal is in analog form (voice or fax) the signal is first converted to a digital form. Packet routing information is then added to the digital voice signal so it can be routed through the Internet or data network.

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