Glossary Terms Explanation

TERMS EXPLANATION
3-way-calling Talk to two people at once with 3-way calling.
411 In North America the number for directory assistance
A-Z International rate table listing destinations from around the world
ACD Average call duration. Average length of a call
ACD Automatic Call Distribution. A method of distributing calls to multiple users using queing
ALOC Average Length of Call (sometimes ACD — Average Call Duration)
ANI Usually Pronounced as Annie this is the billing telephone number (BTN) that the call originated from. This is not necessarily the same as the callerid (which can be set by the calling party). In VoIP many carriers do not send an ANI but it is generally available
ASR Answer Seizure Ratio — (successfull calls / total calls) * 100
ASR Automatic Speech Recognition
ATA Analog Telephone Adaptor connects a regular analog phone or PBX to the Internet converts the analog voice signals into IP packets delivers dial tone and manages the call setup.
Area code Defines the geographic area within a country for a telephone number. An area code could be assigned to a city region or state.
Auto attendant See Interactive Voice Response (IVR)
BLIF Basic Listing Information File. Database entry related to 411 service
BRI Basic Rate ISDN Circuit. Rarely seen these days this was a 2 Line digital circuit.BRI is an ISDN access method typically used for home and small offices. Consists of two 64Kps digital channels and one 16Kbps digital signalling and control channel. Often referred to as 2B+D
BTN Billing telephone number. The primary phone number on a telephone account
Broadband High speed Internet connection such as cable TV DSL or dedicated telecom lines(T1/E1).
CALLERID The Number (and in some cases name) of the calling party that is transmitted to the caller
CDR Call Detail Record. The database that contains details of inbound and outbound calls including calling extension caller id number dialed total call time (including ringing) talk time and outcome (connected busy unavailable)
CENTREX Centrex is a phone company service with PBX type features but instead of the equipment residing at the customers premise it sits at the phone company central office. Often used to refer to services when the equipment providing the service is located at a central location away from the user. in VoIP terms Hosted PBX is the equivalent of Centrex
CID Caller Identification (ID). See CALLERID
CLEC Competitive Local Exchange Carrier. These are the non-Incumbent carriers in a region. A carrier can be a CLEC in one region but an ILEC in others. For example Bell Canada is the ILEC in Ontario but a CLEC in BC (where Telus is the ILEC)
CPE Customer Premises Equipment for example a PBX or phone.
Channelbank A channelbank converts multiple lines into one higher speed line. For example 24 ordinary analog telephone lines into one digital T1 line.
Circuit switched network The traditional telephone network used for making phone calls since 1878.
CoS Class of Service.
Codec Compressor-Decompressor or enCOder/DECoder process.
Committed Access Rate (CAR) A QoS feature.
Compression The squeezing of data in a format that takes less space to store or less bandwidth to transmit. Compression is generally accompanied by some loss of quality.
Compression delay The delay caused by the compression of data.
Congestion The situation in which the traffic present on the network exceeds available network bandwidth/capacity.
DDI See DID. European term
DID Direct Inward Dialing (also called DDI in Europe) is a feature offered by telephone companies for use with their customers PBX system whereby the telephone company (telco) allocates a range of numbers all connected to their customers PBX. In VoIP a DID is typically used as the term for a telephone number
DSL modem A device used to connect computers to the DSL line provided by a DSL operator to gain access to the Internet.
DTMF Dual-Tone Multi Frequency. The tones you hear when you press a key used for dialing on analog networks and by IVR menus and voicemail on most PBXs.
DiffServ An architecture for implementing scalable service differentiation in the Internet for QoS.
Dynamic Jitter Buffer Collects voice packets stores them and shifts them to the voice processor in evenly spaced intervals to reduce any distortion in the sound.
E&M (Ear and Mouth) A type of supervisory line signaling.
E911 Enhanced 911; used for providing emergency service on cellular and Internet voice calls.
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.
Emergency 911 calls An emergency telephone number that handles all calls related to police fire or medical emergencies in North America.
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).
FoIP Fax over Internet Protocol.
Frame Relay Forum 12 (FRF.12) A Frame Relay specification of fragmenting Frame Relay frames into smaller frames.
G.711 G.711 is an ITU codec. Also known as Pulse Code Modulation (PCM) it is a very commonly used waveform codec. G.711 uses a sampling rate of 8000 samples per second with the tolerance on that rate 50 parts per million (ppm). Non-uniform (logarithmic) quantization with 8 bits is used to represent each sample resulting in a 64 kbit/s bit rate. in VoIp terms G711 is the most common codec and considered as uncompressed traffic for best quality
G.729 G.729 is a common ITU codec used in VoIP. It uses an audio data compression algorithm for voice that compresses digital voice in packets of 10 milliseconds duration. It is officially described as Coding of speech at 8 kbit/s Because of its low bandwidth requirements G.729 is popular in Voice over Internet Protocol (VoIP) applications where bandwidth must be conserved. Standard G.729 operates at a bit rate of 8 kbit/s. Voice quality using g729 is lower than with g711 the voice often sounding slightly tinny and detail can be lost or distortion on music on hold. If bandwidth isnt a consideration use g711 for best voice quality
Gatekeeper A device that translates network addresses and aliases to make connections via the H.323 protocol on a packet-switched network.
Gateway A device that acts as an interface between two or more networks to connect dissimilar communications systems.
H.323 An umbrella recommendation from the ITU-T that defines the protocols to provide audio-visual communication sessions on any packet network.
Hairpin To send a call back in the direction that it came from.
Hosted PBX PBX located in data centre providing services to business customers remotely located using VoIP technology. Similar to older Centrex solutions
ILEC Incumbent Local exchange Carrier. In Canada this is the original telephone company to cover an area (Bell Telus MTS)
IP Inernet Protocol.
IP Centrex Using IP-based network to provide centrex services such as call hold call transfer last number look-up and redial call forward three-way calling.
IP PBX IP Private Branch Exchange. A telephone data and video switching system usually located on customer premises and belonging to the user.
IP fragmentation IP datagrams to be fragmented into pieces small enough to pass over a link with a smaller MTU than the original datagram size.
IP phone A device that converts voice into digital packets and vice versa to make phone calls over Internet possible.
IP telephony Technologies and services for the two-way transmission of voice over IP network.
ISDN Integrated Services Digital Network – is a type of circuit switched telephone network system designed to allow digital (as opposed to analog) transmission of voice and data over ordinary telephone copper wires resulting in better quality and higher speeds than analog systems. There are two flavours of ISDN: BRI (Basic Rate Interface) which carries 2 voice channels and PRI (Primary Rate Interface) which carries 23 voice channels in N.America (or 30 in europe)
ITSP Internet Telephony Service Provider.
IVR Interactive Voice Response Interactive voice response (IVR) is a technology that allows a computer to interact with humans through the use of voice and DTMF keypad inputs. IVRs usually have multiple levels of menus.
Instant Messenging (IM) A software that allows users to exchange messages in real time. For example MSN Messenger Yahoo! Messenger etc.
Internet telephony Technologies and services of using the Internet for voice and multimedia communications.
Jitter A momentary fluctuation in the transmission signal. Jitter can result in degraded voice quality
LATA Local access and transport area (LATA) is a term used in U.S. telecommunications regulation determining billing of long distance calls. It represents a geographical area of the United States under the terms of the breakup of the original AT&T into the Baby Bells or created since that time for wireline regulation.
LDCELP Low-Delay Code Excited Linear Prediction.
LOA Letter of authorisation for a number port
LOL Expression of amusement
Lag The extra time taken by a packet of data to travel from the source computer to the destination computer and back again.
Latency The time that elapses between the initiation of a request for data and the start of the actual data transfer.
Local calling area In N.Americaarea where calls can be made to without incurring long distance charges. In the US these are typically less than 12 mile radius in Canada typically 25 miles
Long Distance Calling within a country charged on a per minute basis. The term wont be around for much longer!
MGCP Media Gateway Control Protocol.
MOH Music in Hold
MOS Mean Opinion Score a numerical indication of the perceived quality of received media after compression and/or transmission.
MTU Maximum Transmission Unit.
MoIP Modem over IP.
Modulation To carring information on a signal by varying one or more of the signals basic characteristics -- frequency amplitude and phase.
NANP North America Numbering Plan: defines the number and patterns of digits for telephone numbers in N.America
NGN Next Generation Network.
NPA Numbering Plan Area: See Area code
NPA-NXX First 6 digits of a N.American phone number
NXX Digits 4-6 of a North America phone number. Typically in combination with the first 3 digits (NPA or area code) these define the switch that handle the number.
Network convergence The integration of all traffic types - voice data and video solutions - onto a single IP network.
PBX Private Branch Exchange. A business private phone switch
PCM Pulse Code Modulationis a method used to digitally represent sampled analog signals.
PDD Post Dialing Delay. Time between the start of a call and the time the phone starts ringing.
POTS Plain Old Telephone Service. Traditional analog phone service
PRI PRI is the standard for providing telecommunication services to offices. It is based on the T-carrier (T1) line in the US and Canada and the E-carrier (E1) line in Europe. The T1 line consists of 24 channels in N.America while an E1 has 31.
PSAP A Public Safety Answering Point (PSAP) sometimes called Public Safety Access Point is a call center responsible for answering calls to an emergency telephone number for police firefighting and ambulance services.
PSTN The public switched telephone network – is the concentration of the worlds public circuit-switched telephone networks in much the same way that the Internet is the concentration of the worlds public IP-based packet-switched networks.
Packet loss The loss of data packets during transmission over a computer network. Packet loss is bad in VoIP networks
Packet switched network Networks that break messages into small packets and route them across different channels to their destination where they are reassembled in their proper sequence.
Peer-to-Peer (P2P) A form of computing where two or more than two users can communicate directly without a central control point.
Policy-based Routing A technique used to make routing decisions based on policies set by the network administrator.
Port A number transfer between providers
Proxy server Performs routing of a session invitations according to invitees current location authentication accounting etc.
QSIG Signaling standard for PBX.
QoS Quality of Service. Then ability to prioritise different types of traffic can be used to enhance QOS for VoIP
RADIUS Remote Authenticaion Dial-In User Service.
RTCP Real Time Control Protocol.
RTP Real Time Transport Protocol.
Rate Centre Typically a city or geographic area. Calls within a rate centre are not charges in N.America. A grouping of rate centres make up a Local Calling Area.
SGCP Simple Gateway Control Protocol.
SIP Session Initiation Protocol (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.
SIP phone A telephone that uses the SIP (Session Initiation Protocol) standard to make a voice call over the Internet.
SS7 Signalling System No. 7 (SS7) is a set of telephony signaling protocols which are used to to inteconnect carriers (ILECs and CLECs)
Sampling A methodology used to measure the value of an analog signal at regular intervals and encoding it into a digital format for phone services.
Sampling rate The number of samples per second taken from a continuous (analog) signal to make a discrete(digital) signal.
Service Level Agreement (SLA) A contract between a network service provider and a customer that specifies what services and quality the service provider will furnish.
Signaling gateway A network component responsible for translating signaling messages between one medium (usually IP) and another (PSTN).
Skype Rhymes with Hype.
Soft switch A software application that is used to keep track of monitor or regulate connections at the junction point between circuit and packet networks.
Softphone A software application that is installed in the user’s PC enables voice calls over the Internet.
T.38 ITU-T specification for Facsimile over IP. Although the success rate of Fax using T38 is higher than using standard g711 protocol it still falls well short of direct TDM fax.
T1 1.54mb data connection. When divided into 24 8bit channels (digitised voice calls) of 64kbits this is called a PRI or DS1 in voice terms
TAPI Telephony API.
TDM Time-division multiplexing (TDM) is a type of digital multiplexing in which two or more bit streams or signals are transferred apparently simultaneously as sub-channels in one communication channel but are physically taking turns on the channel. PRIs are an example of TDM
TLA Three letter acronym.
Traffic shaping To control network traffic in order to optimize or guarantee performance low latency and/or bandwidth.
Unified Messaging (UM) The integration of different streams of messages (e-mail Fax voice video etc.) into a single in-box accessible from a variety of different devices.
User Agents A software program installed in a user’s terminal or an IP phone to initiate and terminate phone calls.
VOIP Voice over Internet Protocol (also called IP Telephony Internet telephony and Digital Phone) – is the routing of voice conversations over the Internet or any other IP-based network.
VOIP Gateway A device provides the conversion interface between the PSTN and an IP network for voice and fax calls.
Voicemail A telephone messaging system that digitizes the analog voice signals and stores them on disk or flash memory in a central computer.
WMV Windows Media Video.
WTF Expression of disbelief
WiFi phone A device that enables users to make phone calls from WiFi network environments.
iLBC Internet Low Bitrate Vocoder.

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