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Glossary of Terms

Analog audio signals:

Analog audio signals are used to transmit voice data over telephone lines. An analog representation is usually electrical; a voltage level represents the air pressure waveform of the sound.


Average success rate. The ratio of successfully connected calls to attempted calls (also called ‘Call Completion Rate’).

Audio encoding:

The ITU has defined multiple audio codecs for use with H.323. All of them are also compatible with SIP, which is codec-agnostic. G.711 is 3 kHz audio encoded at 64-kbps. G.711 is PCM audio, the format used for voice delivery over traditional telephone networks and exchanges. G.729 is a newer voice codec using 8-kbps streams and 15ms packet sizes. There are two variations, G.729 and G.729A, that differ only in their mathematical implementation. Call forwarding: In telephony, is a feature on some telephone networks that allows an incoming call to a called party, which would be otherwise unavailable, to be redirected to a mobile telephone or other telephone number where the desired called party is situated.

Asymmetric Digital Subscriber Line (ADSL):

One form of the Digital Subscriber Line technology, a data communications technology that enables faster data transmission over copper telephone lines than a conventional voiceband modem can provide. It does this by utilizing frequencies that are not used by a voice telephone call.


ATA, or the analog telephone adaptor is the hardware device that connects the conventional telephone to the Internet through a high speed bandwidth line, provides the interface to convert the analog voice signals into IP packets, delivers dial tone and manages the call setup.

Average Call Duration (ACD):

is a measure based on a call record (or CDR) sample to determine traffic demand and forecast call volumes, serving also as a tool for infrastructure monitoring (such as switches and cables).


Bandwidth is the volume of data that can be transmitted over a communication line in a fixed amount of time. It is expressed in bits per second (bps) or bytes per second for digital devices and in cycles per second, or Hertz (Hz) for analog devices. Bandwidth can also be defined as the difference between a band of frequencies or wavelengths.


It is a term used to define high speed Internet connection, generally provided by cable TV, DSL or dedicated telecom lines. The high speeds are achieved by the carrying capacity of the cable that can carry multiple messages simultaneously.

Cable modem:

The cable modem is a device that is used to connect a computer to the high speed coaxial cable run by cable TV companies to provide access to the Internet. The connection is made through an Ethernet port, which is a shared medium and can affect download speeds if too many users log on simultaneously to the Internet on that particular cable segment. However, despite this cable modems provide extremely fast access to the net.

Call duration:

The time interval between when the phone is taken off the hook for a test call and when it is put back on the hook.

Caller ID:

Also called calling line identification (CLID) or calling number identification (CNID), is a telephone service, available in analog and digital phone systems and most Voice over Internet Protocol (VoIP) applications, that transmits a caller’s number to the called party’s telephone equipment during the ringing signal, or when the call is being set up but before the call is answered. Where available, caller ID can also provide a name associated with the calling telephone number. The information made available to the called party may be displayed on a telephone’s display or on a separately attached device.

Circuit switched networks:

These networks have been used for making phone calls since 1878. They use a dedicated point-to-point connection for each call. This reduces their utility because no network traffic can move across the switches that are being used to transmit a call.

Client (Softphone client):

The software installed in the user’s computer to make calls over the Internet.


Abbreviation for Coder-Decoder. It’s an analog-to-digital (A/D) and digital-to-analog (D/A) converter for translating the signals from the outside world to digital, and back again


This is a term that is used to indicate the squeezing of data in a format that takes less space to store or less bandwidth to transmit. It is very useful in handling large graphics, audio and video files.

Data compression:

This is the process that is used to compress large data files into mall files so that they use less bandwidth during transmission and less disk space when stored. The compression depends upon the repeatable patterns of binary 0s and 1s. The higher the number of repeatable patters, the higher is the compression. The right compression codes can compress data files to 40% of their original size. The graphics files can be compressed even more – from 20% to 90%.

Direct dial-in (DDI):

Another term used for DID, is a feature offered by telephone companies for use with their customers’ private branch exchange (PBX) systems.


Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without going through an attendant or auto-attendant.

DSL modem:

A DSL modem is a device that is used to connect one or more computers to the high speed DSL line provided by a DSL operator to gain access to the Internet. The customers use these modems to log on the net to download or transmit data. Since the DSL lines have high bandwidth capacity the data transfer speeds are very high.


Dual Tone Multi Frequency. The standard tone-pairs used on telephone terminals for dialing using in-band signaling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals support only 12 of them (0-9, * and #).


The designation for the 2.048Mbps. ITU standard for Europe’s 30-channel digital telephone service. It is the European version of T-1 (DS-1). The bandwidth is divided into two signaling channels (channels 15 and 31 starting from 0) and thirty bearer (voice channels). A&B bit signaling (robbed bit signaling) is not used here. E-1 uses one of the control channels for signaling and the other for clock synchronization.


E911 is the short form of the term Enhanced 911, and is used for providing emergency service on cellular and Internet voice calls.

Emergency 911 calls:

This is an emergency telephone number that handles all calls related to police, fire or medical emergencies. The number, which is allotted under the North American Numbering Plan (NANP), is answered by either a telephone operator or an emergency service dispatcher, who, in turn, alerts the appropriate emergency service.

Fax Server:

A computer based fax machine. Fax servers are “shared use” devices, typically installed on a LAN. Clients on the LAN can use the fax server from their PCs in much the same way they share a network-based (shared) printer. Faxes can be generated by users at their workstations and “printed” to the fax server for transmission. Likewise, fax servers can route incoming faxes to printers, file server directories or to individual users. Fax servers save users from having to print documents, carry them to the fax machine and subsequently wait for them to be transmitted after creating a cover page.


A feature that allows calls to find you wherever you are, ringing multiple phones (such as your cell phone, home phone, and work phone) all at once.

Frame mutes:

The duration and number of prolonged clipping events during a call, where the degraded surface of the signal falls close to zero. The ratio of frame mutes to total clipping events is displayed by the Frame Muting Ratio (%) indicator.

Frame Relay:

In data communications, a packet switching method that uses available bandwidth only when it is needed. This fast packet switching method is efficient enough to transmit voice communications with the proper network management.

Full Duplex:

In telephony and data communications, the ability for both ends of a communication to simultaneously send and receive information without degrading the quality or intelligibility of the content.


Foreign eXchange Office. An FXO device can be an analog phone, answering machine, fax, or anything that handles a call from the telephone company like AT&T. They should also operate the same way when connected to an FXS interface.

. An FXO interface will accept calls from FXS or PSTN interfaces. All countries and regions have their own standards.
. FXO is complimentary to FXS (and the PSTN).


Foreign eXchange Station. An FXS device has hardware to generate the ring signal to the FXO extension (usually an analog phone).

. An FXS device will allow any FXO device to operate as if it were connected to the phone company. This makes your PBX the POTS+PSTN for the phone.

. The FXS Interface connects to FXO devices (by an FXO interface, of course). 

Gateway In VoIP systems:

A network device that converts voice and fax calls in real time from the public switched telephone network (PSTN) to an IP network.


An ITU standard that lays down guidelines for real time voice and videoconferencing utilities on the Internet. The H.323 standard supports voice, video, data, application sharing and whiteboarding and defines media gateways for conversion to packets.


Refers to devices or deployment strategies designed to provide access to fully functioning systems at all times. One such strategy is to cluster devices so that the primary device can fail over to the secondary one if necessary. International Direct Dialling (IDD): is an international telephone call dialled by the caller rather than going via an operator. LD: Long distance call is a telephone call made outside a certain area, usually characterized by an area code outside of a local call area, Long-distance calls usually carry long-distance charges which, within certain nations, vary between phone companies and are the subject of much competition.


IM, which stands for Instant Messenging, is a software that allows users to exchange messages in real time. However, to do so both the users must be logged on to the instant messaging service at the same time. Some of the popular IM services are: MSN Messenger, AOL Instant Messenger, Yahoo! Messenger, Google Talk and ICQ.

Internet congestion:

Internet congestion occurs when a large volume of data is being routed on low bandwidth lines or across networks that have high latency and cannot handle large volumes. The result is slowing down of packet movement, packet loss and drop in service quality.

IP address:

An IP address, also known as Internet Protocol address, is the machine number used to identify all devices that are connected to the net. Each device has its own unique number which it uses to communicate. This number is fixed in the case of those computing devices that have a fixed IP address. The rest are allotted a dynamic IP address, which is valid for the period they are connected to the net. The numbers range from to

IP mapping:

IP mapping is the process of identifying IP addresses on the basis of their geographical locations. The mapping enables web administrators to pinpoint the location of any computing device connected to the Internet.

IP Phone:

An IP phone is one that converts voice into digital packets and vice versa to make phone calls over Internet possible. It has built-in IP signaling protocols such as H.323 that ensure that the voice is routed to the right destination over the net. The IP phones come with several value added services like voicemail, e-mail, call number blocking etc.

IP telephony:

(Internet Protocol telephony, also known as Voice over IP Telephony) A general term for the technologies that use the Internet Protocol’s packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). The basic steps involved in originating an IP Telephony call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet or other packet-switched networks; the process is reversed at the receiving end. The terms IP Telephony and Internet Telephony are often used to mean the same; however, they are not 100 per cent interchangeable, since Internet is only a subcase of packet-switched networks. For users who have free or fixed-price Internet access, IP Telephony software essentially provides free telephone calls anywhere in the world. However, the challenge of IP Telephony is maintaining the quality of service expected by subscribers. Session border controllers resolve this issue by providing quality assurance comparable to legacy telephone systems.


IP, which is the acronym for Internet Protocol, defines the way data packets, also called datagrams, should be moved between the destination and the source. More technically, it can be defined as the network layer protocol in the TCP/IP communications protocol suite.


IP-based Private Branch Exchange or key system.  A PBX system allows you to configure call rules, host voice mail boxes and host user extensions.


ITU, which is the acronym of International Telecommunication Union, is a telecommunications standards body based in Geneva. It works under the aegis of the United Nations and makes recommendations on standards in telecommunications, information technology, consumer electronics, broadcasting and multimedia communications.


IVR is a software application that accepts a combination of voice telephone input and touch-tone keypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail and perhaps other media.


It is a term used to indicate a momentary fluctuation in the transmission signal. This happens in computing when a data packet arrives either ahead or behind a standard clock cycle. In telecommunication, it may result from an abrupt variation in signal characteristics, such as the interval between successive pulses.


Kbps is the acronym for kilobits per second and is used to indicate the data transfer speed. If the modem speed, for instance, is 1 Kbps then it means that the modem can route data at the speed of one thousand bits per second.


Lag is the term used to indicate the extra time taken by a packet of data to travel from the source computer to the destination computer and back again. The lag may be caused by poor networking or by inefficient or excessive processing.


Latency is the time that elapses between the initiation of a request for data and the start of the actual data transfer. This delay may be in nanoseconds but it is still used to judge the efficiency of networks.


The process of identifying all related data fields or data streams and putting them in an easily identifiable context. For example, IP mapping enables users to pinpoint the geographical location of any computing device on the Internet.


Acronym of Media Gateway Control Protocol. Used for a Voice over IP system. It consists of a Call Agent and a set of gateways, of which at least one works as the “media gateway” and performs the conversions.


A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets (eight-bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximum for Ethernet is 1500 byte.


Stands for North American Numbering Plan. A telephone numbering system that has evolved the way area codes and numbers are allotted. The system was established in 1947 and covers the United States, Canada and a few neighboring areas. It uses a three-digit area code and seven-digit telephone numbers. Its fiat is, however, limited to the public switched telephone networks only.


Network Address Translation translates the internal IP addresses computers use on the local network to public IP addresses used on the Internet.  Private IP addresses always fall within 3 ranges.


Net Phone:

A net phone uses the Voice over IP technology to make voice calls. These calls are made by converting analog sound signals into digital data packets, and then moving the packets to their destination over the net.


Outbound Proxy or another name Session Border Controller. A device used in VoIP networks. OBP/SBCs are put into the signaling and media path between calling and called party. The OBP/SBC acts as if it was the called VoIP phone and places a second call to the called party. The effect of this behavior is that not only the signaling traffic, but also the media traffic (voice, video etc) crosses the OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP phones. Private OBP/SBCs are used along with firewalls to enable VoIP calls to and from a protected enterprise network. Public VoIP service providers use OBP/SBCs to allow the use of VoIP protocols from private networks with internet connections using NAT. PBX: Private branch exchange (PBX) is a telephone exchange that serves a particular business or office, as opposed to one that a common carrier or telephone company operates for many businesses or for the general public

Packet loss:

Packet loss is the term used to indicate the loss of data packets during transmission over a computer network. This may happen on account of high network latency or on account of overloading of switches or routers that are unable to process or route all the incoming data.

Packet switched networks:

These are networks that break messages into small digital packets, stamp each packet with the destination IP address, and route them across different channels to their destination where they are reassembled in their proper sequence. This is done to avoid network congestion and speed up data movement from multiple sources.


A packet is a unit of data transmitted over the network in a packet-switched system. It consists of a header that stores the destination address, a data area which carries the information that is being transmitted, and a trailer which contains information to prevent errors during transmission.

Peer-to-Peer (P2P):

The term peer-to-peer is used to indicate a form of computing where two or more than two users can share files or CPU power. They can even transmit real time data such as telephony traffic on their highly ad hoc networks. Interestingly, the peer-to-peer network does not work on the traditional client-server model but on equal peer nodes that work both as “clients” and “servers” to other nodes on the network.


POTS is the short form of plain old telephone service. It transmits voice as analog data on communication lines that are much slower when compared to today’s ISDN or FDDI lines. However, not long ago POTS, which is also known as the public switched telephone network, was the standard telephone system across the world.

Power over Ethernet or PoE:

Technology describes a system to safely pass electrical power, along with data, on Ethernet cabling

Processor drain:

This is a term used to indicate a drop in the quality of VoIP phone service when a user opens several applications on his computer simultaneously.


It is a convention or standard that defines the procedures to be adopted regarding the transmission of data between two computing end points. These procedures include the way the sending device should sign off a message or how the receiving device should indicate the receipt of a message. Similarly, the protocols also lay down guidelines for error checking, data compression, and other relevant operational details.


Public Switched Telephone Network. The phone service we use for every ordinary phone call, or called POT (Plain Old Telephone), or circuit switched network.

QoS (Quality of Service):

The ability of a network (including applications, hosts, and infrastructure devices) to deliver traffic with minimum delay and maximum availability.


A router is a network device that that handles message transfer between computers that form part of the Internet. The messages, which are in the form of data packets, are forwarded to their respective IP destinations by the router. A router can also be called the junction box that routes data packets between computer networks.


Real-time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889


This is a methodology used to measure the value of an analog signal at regular intervals, and encoding it into a digital format for VoIP phone services.

Service provider:

A service provider is a business entity that provides a communication, storage or processing service for a fee. Some of the service providers in the digital world are the Internet service provider (ISP), application service provider (ASP), storage service provider, mobile phone service provider, web hosting provider, and of course, VOIP service provider.

SIP phone:

A SIP phone is a telephone that uses the SIP (Session Initiation Protocol) standard to make a voice call over the Internet. The SIP phones come with several value added services like voicemail, e-mail, call number blocking etc. There are no charges for making calls from one SIP phone to another, and negligible charges for routing the call from a SIP phone to a PSTN phone.


Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261). SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice transmission and uses fewer resources and is considerably less complex than H.323. All Grandstream products are SIP based.

Soft switch:

It is a software application that is used to keep track of, monitor or regulate connections at the junction point between circuit and packet networks. This software is loaded in computers and is now replacing hardware switches on most telecom networks.


This is a software application that is installed in the user’s PC. It uses the Voice over IP technology to route voice calls over the net and provides several value added features, such as call forwarding, conference calling, and integration with applications such as Outlook for automatic dialing The audio is provided through a microphone and speakers plugged into the sound card. The only limitation of a Softphone is that the phone call has to made through a PC. Many softphone are free VOIP software downloads.


Allows a single telephone connection to be used for both ADSL service and voice calls at the same time


Simple Traversal of UDP over NATs is a network protocol allowing clients behind NAT (or multiple NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by the NAT with a particular local port. This information is used to set up UDP communication between two hosts that are both behind NAT routers. The protocol is defined in RFC 3489. STUN will usually work well with non-symmetric NAT routers.


1.544-Mbps point-to-point dedicated digital circuit provided by the telephone companies consisting of 24 channels.


North American standard for DS-3. Operates at a signaling rate of 44.736 Mbps, or the equivalent of 28 T-1s.


A communications channel between two points, typically referring to large-bandwidth telephone channels between switching centers, that handle many simultaneous voice and data signals. UM: Unified Messaging is the integration of different electronic messaging and communications media (e-mail, SMS, Fax, voicemail, video messaging, etc.) technologies into a single interface, accessible from a variety of different devices

Voice chat:

This is an application that enables two or more than two individuals to carry on a verbal conversation over the Internet. Voice chat is also known as audio-conferencing or telephone conferencing on the net.

Voice over IP (VOIP):

VoIP or Voice over IP is the technology that is used to transmit voice over the Internet. The voice is first converted into digital data which is then organized into small packets. These packets are stamped with the destination IP address and routed over the Internet. At the receiving end the digital data is reconverted into voice and fed into the user’s phone.


VoIP PBX, which stands for Voice over Internet Protocol Private Branch eXchange, is a telephone switch that converts IP phone calls into traditional circuit-switched TDM connections. It also supports traditional analog and digital telephones.

VOIP Phone:

A VoIP phone is one that uses the Internet to route voice calls by converting the voice data into IP packets and vice versa. The phones come with built-in IP signaling protocols such as H.323 or SIP that help in the routing of data to the right destination. A VoIP phone can also be a software application that is installed in the user’s PC. In this case it is known as the Softphone. Also, the calls in this case have to be made from the PC, and not through a telephone instrument.

VOIP services:

The VoIP services are packet-based services that use the Internet to move voice data. These services are much cheaper than the traditional PSTN services because the investment in infrastructure is low. They also come with several value added features which make them more lucrative than the conventional landline phone services.


Virtual Private Network. Enables IP traffic to travel securely over a public TCP/IP network by encrypting all traffic from one network to another. A VPN uses ¡°tunneling¡± to encrypt all information at the IP level.

WiFi phone:

A WiFI phone is one that enables users to make phone calls from public WiFi hotspots or residential WiFI network environments. Besides voice calls, these phones can be used to send e-mails wirelessly.

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