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Online IP Telephony Dictionary - More than 10,937 Terms and Definitions
Rich Tehrani - Editor Industry leader and expert Rich Tehrani ensured this dictionary is unbiased and includes the latest information related the IP Telephony industry. IP Telephony Contributing Experts Althos recruited 15 voice and data telephony experts, each with specific knowledge of IP telephony technologies and business practices. This team, together with input from over 1,025 other online contributors gathered, added, and edited what are now the latest VoIP, telecom, and data network terms and acronyms in use today.
Lawrence Harte- Telecom Technologies
Sample Diagrams and Descriptions This figure shows how a SIP system uses relatively simple text messages to setup and control telephone calls. This diagram shows how a telephone has SIP capability that is controlled by a call server. This SIP based telephone is called a User Agent (UA). The User Agent (UA) is actually a gateway that converts audio (e.g. sound) and control information (e.g. dialed digits) into packets that can be routed through a data network (such as the Internet) to call servers and other User Agents (UAs.) The control packets are sent to and from the call server to request and receive calls. Call servers may communicate with other call servers to setup distant call connections. This diagram shows how a distant call server controls a User Agent (UA) gateway that allows calls to connect from the Internet to another telephone.
This figure shows the basic digital speech compression process. In this example, the word "HELLO" is digitized. The initial digitized bits represent every specific shape of the digitized word HELLO. This digital information is analyzed and it is determined that this entire word can be represented by three sounds: "HeH" + "LeL" + "OH." Each of these sounds only requires a few digital bits instead of the many bits required to recreate the entire analog waveform.
Voice Gatekeeper This figure shows how a gatekeeper sets up connections between Internet telephones and telephone gateways. The gatekeeper receives registration messages from an Internet telephone when it is first connected to the Internet. This registration message indicates the current Internet address (IP address) of the Internet telephone. When the Internet telephone desires to make a call, it sends a message to the ITSP that includes the destination telephone number it wants to talk to. The ITSP reviews the destination telephone number with a list of authorized gateways. This list identifies to the ITSP one or more gateways that are located near the destination number and that can deliver the call. The ITSP sends a setup message to the gateway that includes the destination telephone number, the parameters of the call (bandwidth and type of speech compression), along with the current Internet address of the calling Internet telephone. The gatekeeper then sends the address of the destination gateway to the calling Internet telephone. The Internet telephone then can send packets directly to the gateway and the gateway initiates a local call to the destination telephone. If the destination telephone answers, two audio paths between the gateway and the Internet telephone are created. One for each direction and the call operates as a telephone call. Top of page Internet Telephone Service Provider (ITSP) This figure shows that ITSPs are primarily made of computers that are connected to the Internet and software to operate call processing and other services. In this diagram, a computer keeps track of which customers are active (registration) and what features and services are authorized. When call requests are processed, the ITSP sends messages to gateways via the Internet allowing calls to be completed to telephones that are connected to the public telephone network. These gateways transfer their billing details to a clearinghouse so the ITSP can pay for the gateway's usage. The ITSP then can use this billing information to charge the customer for calls. Quick Start- Some of the most popular terms and 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. 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. Ethernet can be provided on twisted pair, coaxial cable, wireless, or fiber cable. In 2001, the common wired connections for Ethernet was 10 Mbps or 100 Mbps. 100 Mbps Ethernet (100 BaseT) systems are also called "Fast Ethernet." Ethernet systems that can transmit at 1 Gbps (1 Gbps = 1 thousand Mbps) or more, are called "Gigabit Ethernet (GE)." Wireless Ethernet have data transmission rates that are usually limited from 2 Mbps to 11 Mbps. Originally created by an alliance between Digital Equipment Corporation, Intel and Xerox, Ethernet DIX, is slightly different than IEEE 802.3. In Ethernet the packet header includes a type field and the length of the packet is determined by detection. In IEEE 802.3, the packet header includes a length field and the packet type is encapsulated in an IEEE 802.2 header. Most modern day "Ethernet" devices are capable of using both protocol variation, however, older equipment was not able to do this. 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. 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. IP PBX represents the evolution of enterprise telephony from circuit to packet. Traditional PBX systems are voice-based, whereas their successor is designed for converged applications. IP PBX supports both voice and data, and potentially a richer feature set. Current IP PBX offerings vary in their range of features and network configurations, but offer clear advantages over TDM-based PBX, mainly in terms of reduce Opex (operating expenses). 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. 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. There are two main types: Access gateways provide regular analog or primary rate (PRI) interfaces to a voice-over-packet (VoP) network. The inverse function is also available in VoB (voice over broadband) applications: calls are encoded digitally before entering the access network and are routed via conventional telephony once inside. Trunking gateways interface directly between the telephone network and a voice over packet (VoP) network in the core. Such gateways typically manage large numbers of digital virtual circuits. 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. 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. 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.
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