Automatic Number Identification; A telephone function that transmits the billing number of the incoming call.
Application Service Provider (ASP)
An independent, third-party provider of software-based services delivered to customers across a wide area network (WAN).
Asynchronous Transfer Mode is a technology
for switched, connection-oriented transmission of voice, data and
video. It makes high-speed dedicated connections possible between a
theoretically unlimited number of network users and also to servers. As
a switching system ("Cell Relay") it is to be used in broadband ISDN
(B-ISDN) and also in the Switched Multimegabit Data Service (SMDS
networks). ATM is also becoming increasingly popular in the LAN area in
the form of ATM-LAN emulations. ATM is based on high-speed cell
switching (packets of fixed size: 48 5 bytes) that makes it possible to
vary bit rates (according to requirements). In connection with ATM one
speaks of message blocks or message cells rather than message packets.
Autonomous System (AS)
A group of networks
under mutual administration that share the same routing methodology. An
AS uses an internal gateway protocol and common metrics to route
packets within the AS, and uses an external gateway protocol to route
packets to other Autonomous Systems.
Average Hold Time (AHT)
The average length of time between the time a caller finishes dialing and the time the call is answered or terminated.
A very-high-speed network spanning
from one major metropolitan area to another. Such networks are
typically provided by national Internet service providers (ISPs). Local
ISPs connect to the backbone in order to transport data.
Bad Frame Interpolation
Bad Frame Interpolation
interpolates lost/corrupted packets by using the previously received
voice frames. It increases voice quality by making the voice
transmission more robust in bursty error environments.
The maximum data carrying capacity of
a transmission link. For networks, bandwidth is usually expressed in
bits per second (bps)
A call duration measurement unit, expressed in seconds.
Busy Lamp Indicator; A light or LED on a telephone that shows which line is in use.
Descriptive term for evolving digital
technology that provides consumers a single switch facility offering
integrated access to voice, high-speed data service, video demand
services, and interactive delivery services.
Establishment of (or an attempt to
establish) voice connection between two endpoints, or between two
points which provide a partial link (e.g. a trunk) between two
Call Detail Record (CDR)
a single call collected from the switch and available as an
automatically generated downloadable report for a requested time
period. The report contains information on the number of calls, call
duration, call origination and destination, and billed amount
Voice encoding/decoding mechanism. Codecs
are used to compress the voice signal into data packets. Each codec has
different bandwidth requirements. The most popular codecs are: G.729,
G.729A, G.723.1, G711A-Law, and G.711mU-Law.
Compression is used at anywhere
from 1:1 to 12:1 ratios in VOIP applications to consume less bandwidth
and leave more for data or other voice/fax communications. The voice
quality may decrease with increased compression ratios.
The condition in which the traffic on the network exceeds available network bandwidth/capacity.
Mode of communication in which a
connection (circuit or logical channel) does not need to be set up for
data transmission between the transmitter and receiver. It is the
underlying protocol for packet-switched transmission. The individual
data packets can go from the transmitter to the receiver via different
paths. A well-known connectionless protocol is UDP.
Mode of communication in
which a connection must be established between the transmitter and
receiver before transmission of user data. This can be done by
switching a circuit or by setting up a logical channel. A well-known
connection-oriented protocol is TCP. Connection-oriented is the
opposite of connectionless.
Addressable call endpoint -- a
software structure that binds a dialed digit string to a voice port or
IP address of the destination gateway. Several dial peers always exist
on each router in the network, and at least two will be involved in
making a call across the network, one on the originating end and one on
the terminating end. In Voice over IP, there are two kinds of dial
peers: POTS and VoIP. VoIP peers point to specific VoIP devices
Process when the originating
router tries to establish call on different dial peers if the
originating router receives a user-busy invalid number or an
unassigned-number disconnect cause code from a destination router.
Direct Inward Dialing; The ability to make
a telephone call directly into an internal extension without having to
go through the an attendant.
DiffServ (Differentiated Services) is
a quality of service protocol that prioritizes IP voice and data
traffic to help preserve voice quality even when network traffic is
Dialed Number Identification Service; A telephone function that sends the dialed telephone number to the answering service.
Dual-Tone Multi-Frequency; The type of audio signals generated when you press the buttons on a touch-tone telephone.
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.
(Ear and Mouth) a method of connecting
two pbx’s together via the PSTN dedicated trunking. Sometimes this is
the interface on a VOIP device that allows it to be connected to analog
PBX trunk ports (tie lines).
SIP or H.323 terminal or Gateway. An
endpoint can call and be called. It generates and terminates the
information stream. Typically, end point are thought of as telephones
or soft phones, but on an incomplete call it can be another device.
A system designed to prevent
unauthorized access to or from a private network. Firewalls can be
implemented as hardware, software, or a combination of both. All
messages entering or leaving the intranet pass through the firewall,
which examines each message and blocks those that do not meet the
security criteria specified on the firewall.
(foreign exchange office)Usually a central
office connection characterized by Dial tone. Often this is the
interface on a VOIP device for connecting to an analog PBX extension.
(foreign exchange station) On a PBX, this
is the ports that connect to the telephones (stations). On a PBX, these
can be either analog or digital. The analog station port is sometimes
used as an interface on a VOIP device for connecting directly to
phones, faxes, and CO ports on PBXs or key telephone systems.
An ITU-T PCM half-duplex codec that uses either A-law or U-law compression (64 kbps, high quality, minimum processor load).
An ITU-T double rate CELP codec (6.4/5.3 kbps, medium quality, high processor load).
An ITU-T ADPCM wave form codec (16/24/32/40 kbps, good quality, low processor load).
An ITU-T low delay CELP codec (16 kbps, medium quality, very high processor load).
An ITU-T ACELP codec (8 kbps, medium quality, high processor load).
The central control entity that
performs management functions in a Voice and Fax over IP network and
for multimedia applications such as video conferencing. Gatekeepers
provide intelligence for the network, including address resolution,
authorization, and authentication services, the logging of Call Detail
Records, and communications with network management systems.
Gatekeepers control bandwidth, provide interfaces to existing legacy
systems, and monitor the network for engineering purposes as well as
for real-time network management and load balancing.
A gateway is basically a protocol
converter, i.e. a network point that connects networks using different
protocols so that data can be exchanged seamlessly between endpoints.
For example, a POTS-to-VoIP Gateway connects PSTNs and packet-switched
networks, translating the media into IP packets, so that "legacy"
telephony becomes Voice-over-IP.
GKTMP (Cisco Gatekeeper Transaction Message Protocol)
A proprietary Cisco protocol used for communication between the Cisco IOS Gatekeeper and external applications.
Protocols (RAS, RTP/RTCP, Q.931 call signaling) and message formats for H.323.
A protocol for capability negotiation,
messages for opening and closing channels for media streams, etc. (i.e.
The ITU standard for real-time voice,
video and data communication over packet-based networks such as the
Internet. H.323 addresses problems inherent to packet-switched networks
such as packet delay and packet loss on LANs, corporate intranets, and
An ITU-T "umbrella" of standards for
Packet-based multimedia communications systems. This standard defines
the different multimedia entities that make up a multimedia system -
Endpoints, Gateways, Multipoint Conferencing Units (MCUs), and
Gatekeepers -- and their interaction. This standard is used for many
Voice-over-IP applications, and is heavily dependent on other
standards, mainly H.225 and H.245.
Comprised of 24 64Kbps channels,
T1 lines can be used for a diverse number of applications. Commonly
referred to as an integrated T1 or channelized T1, this highly flexible
circuit is designed for businesses that need to run multiple services
over the same line.
Communications services that
use the Internet to initiate, process and receive voice, fax and other
forms of information. Internet Telephony is an example of IP Telephony
and is differentiated only by the use of the Internet for transport of
the voice packets. When using the Internet to transport voice packets,
it is assumed that QOS is not guaranteed.
Internet Protocol. One of a large family of
specifications that define the transmission of information over data
networks. It tracks the Internet addresses of nodes, routes outgoing
messages, and recognizes incoming messages.
IP Centrex delivers such services as
call hold, call transfer, last number look-up and redial, call forward,
three-way calling utilizing a packet-based network.
IP PBX is a customer premises telephone
system that manages telephones in the enterprise and acts as the
gateway to external networks. Unlike a conventional PBX that requires
two separate networks, one each for data and voice, an IP PBX is based
on converged networks that enable true one-wire to the desktop
connection. An IP PBX can be used with IP phones, softphones and
traditional phones connected to Ethernet adapters or PCs.
(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. Since IP Telephony can exist on any network, the
term Internet Telephony encompasses the same basic technologies, but
uses the Internet for transporting the voice packets.
The variation in the amount of Latency among Packets being received
A local area network (LAN) is a group of
computers and associated devices that share a common communications
line or wireless link and typically share the resources of a single
processor or server within a small geographic area (for example, within
an office building).
(Also called Delay) The amount of time
it takes a Packet to travel from source to destination. Together,
Latency and Bandwidth define the speed and capacity of a network.
Distribution of calls among terminating Gateways.
MGCP (Media Gateway Control Protocol)
protocol complementary to H.323 and SIP, designed to control media
gateways from external call control elements in decomposed gateway
architectures. Working in conjunction with the Gateway Location
Protocol (GLP), MGCP enables a caller with a PSTN phone number to
locate the destination device and establish a session. It provides the
gateway-to-gateway interface for the Session Initialization Protocol
(SIP). MGCP is meant to simplify standards for the new Voice over
Packet technology by eliminating the need for complex,
processor-intense IP telephony devices, thus simplifying and lowering
the cost of these terminals
In data communication, the basic logical unit of information transferred.
Private branch exchange; An in-house
telephone switching system that interconnects telephone extensions to
each other as well as to the outside telephone network.
Plain Old Telephone Service. The term
refers to the standard telephone service that most homes and businesses
tradionally use. The POTS network is also called the PSTN.
Primary Rate Interface; An ISDN service
that provides 23 64-Kbps B (Bearer) channels and one 64-Kbps D (Data)
channel (23 B and D).
Public Switched Telephone Network. The
worldwide voice telephone network that traditionally routes voice calls
from one location to another.
Quality of Service. Measure of performance
for a transmission system that reflects its transmission quality and
service availability. Standards based QOS for VoIP usually involves the
implementation of Ethernet standards 802.1p and 802.1q at layer 2
across an Ethernet. At layer 3, the IP standard DiffServ defines bits
settings in the TOS (type of service) in the IP header which will
identify packets as being associated with a specific service.
RAS (Registration, Admission, Status)
A management protocol between terminals and Gatekeepers.
Redundant describes computer or
network system components, such as fans, hard disk drives, servers,
operating systems, switches, and telecommunication links that are
installed to back up primary resources in case they fail.
Resource Reservation Protocol. Protocol
that supports the reservation of resources across an IP network.
Applications running on IP end systems can use RSVP to indicate to
other nodes the nature (bandwidth, jitter, maximum burst, and so on) of
the packet streams they want to receive. RSVP depends on IPv6. Also
known as Resource Reservation Setup Protocol.
Real-Time Transport Protocol. Commonly used
with IP networks. RTP is designed to provide end-to-end network
transport functions for applications transmitting real-time data, such
as audio, video, or simulation data, over multicast or unicast network
services. RTP provides such services as payload type identification,
sequence numbering, timestamping, and delivery monitoring to real-time
Session border controller (SBC)
A category of
network equipment that enables interactive communications across IP
network borders. SBCs closely integrate signaling and media control and
serve as a transit point for all signaling and media streams going
through the network.
Session Initiation Protocol. An ASCII-based
protocol that provides telephony services similar to older protocols
like h.323, but is less complex and uses less resources. It creates,
modifies, and terminates sessions with one or more participants. Such
sessions include Internet telephony and multimedia conferences. SIP is
a request-response protocol, dealing with requests from clients and
responses from servers.
(Also called a Proxy Gatekeeper,
Call Server, Call Agent, Media Gateway Controller, or Switch
Controller) Software used to bridge a public switched telephone network
and voice over Internet by separating the call control functions of a
phone call from the media gateway (transport layer). Softswitch
performs call control functions such as protocol conversion,
authorization, accounting and administration operations.
1.544-Mbps point-to-point dedicated digital circuit provided by the telephone companies consisting of 24 channels.
TCP (Transmission Control Protocol)
transport layer protocol that provides reliable full-duplex data
transmission. TCP is part of the TCP/IP protocol stack.
A communications channel between two
points. Trunks can consist of POTS lines, T-1, PRI or any other logical
voice or data transport.
Voice over IP. The capability to carry
normal telephony-style voice over an IP-based Internet or data links
with POTS-like functionality, reliability, and voice quality. VoIP
enables a router to carry voice traffic (for example, telephone calls
and faxes) over an IP network.
Virtual Private Network. Enables IP packets
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.
DHCP: Dynamic Host Configuration Protocol
FTP: File Transfer Protocol
IP: Internet Protocol
IP PBX: IP-based Private Branch Exchange
MGCP: Media Gateway Control Protocol
NAT: Network Address Translation
PBX: Private Branch exchange
PSTN: Public Switched Telephone Network
QoS: Quality of Service
RAS: Registration, Admission, and Status protocol.
RTCP: Real-time Transport Control Protocol
RTP: Real-time Transport Protocol
SCP: Secure Copy
SIP: SIP Session Initiation Protocol
SNMP: Simple Network Management Protocol
TCP/IP: Transmission Control Protocol/Internet Protocol
VoIP: Voice Over Internet Protocol
VPN: Virtual Private Network