110-Mbps baseband Ethernet specification using two pairs of twisted-pair cabling (Categories 3, 4, or 5): one pair for transmitting data and the other for receiving data. 10BASE-T, which is part of the IEEE 802.3 specification, has a distance limit of approximately 328 feet (100 meters) per segment.
A
A-law
ITU-T companding standard used in the conversion between analog and digital signals in PCM systems. A-law is used primarily in European telephone networks and is similar to the North American u-law standard. See also companding and u-law.
B
BTXML
Basic Telephony Extensible Markup Language. See XML.
C
category x cable
One of five grades of UTP cabling described in the EIA/TIA-586 standard. Category 3 cabling is used in 10BASE-T networks and can transmit data at speeds up to 10 Mbps. Category 5 cabling can transmit data at speeds up to 100 Mbps.
CDP
Cisco Discovery Protocol. Used primarily to obtain protocol addresses of neighboring devices and discover the platform of those devices. CDP can also be used to show information about the interfaces your router uses. CDP is media- and protocol-independent, and runs on all Cisco-manufactured equipment including routers, bridges, access servers, and switches.
CED tone
CallED station identification. A 3-second, 2100-Hz tone generated by a fax machine while answering a call, which is used in the handshaking used to set the call; the response from a called fax machine to a CNG tone.
CELP
Code excited linear prediction compression. Compression algorithm used in low bit-rate voice encoding. Used in ITU-T Recommendations G.728, G.729, and G.723.1.
CLIP
Calling Line Identification Presentation. Shows your identity to callers with Caller ID.
CLIR
Calling Line Identification Restriction. Hides your identity from callers with Caller ID.
CNG
Comfort noise generation or calling tone. Distinctive, repeating 1100-Hz tone (on for 0.5 seconds, off for 3 seconds) generated by a fax machine when placing a call.
codec
Coder-decoder. In Voice over IP, Voice over Frame Relay, and Voice over ATM, a DSP software algorithm used to compress/decompress speech or audio signals.
companding
Contraction derived from the opposite processes of compression and expansion. Part of the PCM process whereby analog signal values are rounded logically to discrete scale-step values on a nonlinear scale. The decimal step number then is coded in its binary equivalent before transmission. The process is reversed at the receiving terminal using the same nonlinear scale. Compare with compression and expansion. See also A-law and u-law.
compression
The running of a data set through an algorithm that reduces the space required to store or the bandwidth required to transmit the data set. Compare with companding and expansion.
CoS
Class of service. An indication of how an upper-layer protocol requires a lower-layer protocol to treat its messages. In SNA subarea routing, CoS definitions are used by subarea nodes to determine the optimal route to establish a given session. A CoS definition comprises a virtual route number and a transmission priority field. Also called ToS.
D
DHCP
Dynamic Host Configuration Protocol. Provides a mechanism for allocating IP addresses dynamically so that addresses can be reused when hosts no longer need them.
dial peer
An addressable call endpoint. In Voice over IP (VoIP), there are two types of dial peers: POTS and VoIP.
DNS
Domain Name System. System used on the Internet for translating names of network nodes into IP addresses.
DSL
Digital subscriber line. Public network technology that delivers high bandwidth over conventional copper wiring at limited distances. There are four types of DSL: ADSL, HDSL, SDSL, and VDSL. All are provisioned via modem pairs, with one modem located at a central office and the other at the customer site. Because most DSL technologies do not use the whole bandwidth of the twisted pair, there is room remaining for a voice channel.
DSP
Digital signal processor. Specialized hardware and software algorithms that perform complex processing of digitized data that was originally analog data. Typically segments a voice signal into frames and stores the frames in voice packets.
DTMF
Dual tone multifrequency. A type of signaling that combines two distinct frequencies to generate a tone for each digit or character dialed, which is used by customers to signal the network. Sometimes referred to as "touchtone," because a customer generally touches keypad keys to generate the tones.
E
E.164
The international public telecommunications numbering plan. A standard set by the ITU-T that addresses telephone numbers.
endpoint
A source or sink of voice data, which may be physical (such as a trunk interface or a line interface in a media gateway) or logical (such as an announcement stored on a server).
expansion
The process of running a compressed data set through an algorithm that restores the data set to its original size. Compare with companding and compression.
F
firewall
Router or access server, or several routers or access servers, designated as a buffer between any connected public networks and a private network. A firewall router uses access lists and other methods to ensure the security of the private network.
FQDN
Fully qualified domain name. FQDN is the full name of a system, including the domain name and not just the host name. For example, aldebaran is a host name, and aldebaran.interop.com is an FQDN.
FSK
Frequency shift key.
FXO
Foreign exchange office. An FXO interface connects to the public switched telephone network (PSTN) central office and is the interface offered on a standard telephone. The Cisco FXO interface is an RJ-11 connector that allows an analog connection at the PSTN central office or to a station interface on a PBX.
FXS
Foreign exchange station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. The Cisco FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and PBXs.
G
G.711
The 64-kbps PCM voice coding technique. In G.711, encoded voice is already in the correct format for digital voice delivery in the PSTN or through PBXs. Described in the ITU-T standard in its G-series recommendations.
G.723.1
Compression technique that can be used for compressing speech or audio signal components at a very low bit rate as part of the H.324 family of standards. This codec has two bit rates associated with it: 5.3 and 6.3 kbps. The higher bit rate is based on ML-MLQ technology and provides a somewhat higher quality of sound. The lower bit rate is based on CELP and provides system designers with additional flexibility. Described in the ITU-T standard in its G-series recommendations.
G.729A
CELP compression where voice is coded into 8-kbps streams. There are two variations of this standard (G.729 and G.729 Annex A) that differ mainly in computational complexity; both provide speech quality similar to 32-kbps ADPCM. Described in the ITU-T standard in its G-series recommendations.
gatekeeper
H.323 entity on a LAN that provides address translation and controls access to the LAN for H.323 terminals and gateways. The gatekeeper can provide other services to the H.323 terminals and gateways, such as managing bandwidth and locating gateways. A gatekeeper maintains a registry of devices in the multimedia network. The devices register with the gatekeeper at startup and request admission to a call from the gatekeeper.
gatekeeper zone
The set of H.323 nodes controlled by a single gatekeeper. Gatekeepers coexisting on a network can be configured so that they register endpoints from different subnets. Endpoints attempt to discover their controlling gatekeeper, and consequently what zone they are members of, using the RAS message protocol.
gateway
A gateway allows SIP or H.323 terminals to communicate with terminals configured to other protocols by converting protocols. A gateway is the point where a circuit-switched call is encoded and repackaged into IP packets.
GMT
Greenwich Mean Time.
H
H.245
An ITU standard that governs H.245 endpoint control.
H.323
H.323 allows dissimilar communication devices to communicate with each other by using a standardized communication protocol. H.323 defines a common set of codecs, call setup and negotiating procedures, and basic data transport methods.
I
ICMP
Internet Control Message Protocol. Network-layer Internet protocol that reports errors and provides other information relevant to IP packet processing. Documented in RFC 792.
IETF
Internet Engineering Task Force. Task force consisting of over 80 working groups responsible for developing Internet standards.
IP
Internet Protocol. Network-layer protocol in the TCP/IP stack offering a connectionless internetwork service. IP provides features for addressing, type-of-service specification, fragmentation and reassembly, and security. Defined in RFC 791.
ITU
International Telecommunication Union. An organization established by the United Nations to set international telecommunications standards and to allocate frequencies for specific uses.
IVR
Interactive voice response. Describes systems that provide information in the form of recorded messages over telephone lines in response to user input in the form of spoken words or, more commonly, DTMF signaling. Examples include banks that allow you to check your balance from any telephone and automated stock quote systems.
L
LDAP
Lightweight Directory Access Protocol. Protocol that provides access for management and browser applications that provide read/write interactive access to the X.500 Directory.
LEC
Local exchange carrier. A local telephone company or communications common carrier that provides ordinary local voice-grade telecommunications service under regulation within a specified service area.
location server
A SIP redirect or proxy server uses a location server to get information about a caller's location. Location services are offered by location servers.
M
MCU
Multipoint control unit. A bridging or switching device that supports multipoint videoconferencing.
MGCP
Media Gateway Control Protocol.
MWI
Message waiting indication.
mu-law
Commonly written u-law. North American companding standard used in conversion between analog and digital signals in PCM systems. Similar to the European A-law. See also A-law and companding.
N
NAT
Network Address Translation. Mechanism for reducing the need for globally unique IP addresses. NAT allows an organization with addresses that are not globally unique to connect to the Internet by translating those addresses into address space with global routing. Also known as Network Address Translator.
NTP
Network Time Protocol. Protocol built on top of TCP that ensures accurate local time-keeping with reference to radio and atomic clocks located on the Internet. This protocol is capable of synchronizing distributed clocks within milliseconds over long time periods.
P
POTS
Plain Old Telephone Service. Basic telephone service that supplies standard single-line telephones, telephone lines, and access to the PSTN.
proxy server
An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets and, if necessary, rewrites a request message before forwarding it.
PSTN
Public Switched Telephone Network. PSTN refers to the local telephone company.
Q
QoS
Quality of service. The ability of a network, whether the network is a complex network, small corporate network, Internet service provider (ISP), or enterprise network, to provide better service to selected network traffic over various technologies, including Frame Relay, ATM, Ethernet and 802.1 networks, and SONET, as well as IP-routed networks that may use any or all of these underlying technologies.
R
RAS
Registration, Admission, and Status Protocol. Protocol that is used between endpoints and a gatekeeper to perform management functions. RAS signalling function performs registration, admissions, bandwidth changes, status, and disengage procedures between the VoIP gateway and the gatekeeper.
redirect server
A server that accepts a SIP request, maps the address into zero or more new addresses, and returns these addresses to the client. It does not initiate its own SIP request nor accept calls.
registrar server
A server that accepts REGISTER requests. A registrar is typically colocated with a proxy or redirect server and may offer location services.
RFC 2833
IETF standard that describes RTP payload for DTMF digits, telephony tones, and telephony signals.
router
Network layer device that uses one or more metrics to determine the optimal path along which network traffic should be forwarded. Routers forward packets from one network to another based on network layer information. Occasionally called a gateway (although this definition of gateway is becoming increasingly outdated). Compare with gateway.
RSVP
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 that they want to receive. RSVP depends on IPv6. Also known as Resource Reservation Setup Protocol.
RTP
Real-Time Transport Protocol. One of the IPv6 protocols. RTP is designed to provide end-to-end network transport functions for applications that transmit real-time data, such as audio, video, or simulation data, over multicast or unicast network services. RTP provides services such as payload type identification, sequence numbering, time stamping, and delivery monitoring to real-time applications.
S
SCCP
Skinny Call Control Protocol. A VoIP protocol utilized by a CallManager server.
SDP
Session Definition Protocol. An IETF protocol for the definition of multimedia services. SDP messages can be part of SIP, MGCP, and SCCP and messages.
SIP
Session Initiation Protocol. Protocol developed by the IETF MMUSIC Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999. SIP equips platforms to signal the setup of voice and multimedia calls over IP networks.
SIP endpoint
A terminal or gateway that acts as a source or sink of SIP voice data. An endpoint can call or be called, and it generates or terminates the information stream.
SLIC
Subscriber Line Interface Circuit. An integrated circuit providing central office-like telephone interface functionality.
SNA
Systems Network Architecture. Large, complex, feature-rich network architecture developed in the 1970s by IBM. Similar in some respects to the OSI reference model but with a number of differences. SNA essentially is composed of seven layers.
SOHO
Small office, home office. Networking solutions and access technologies for offices that are not directly connected to large corporate networks.
T
TCP
Transmission Control Protocol. Connection-oriented transport layer protocol that provides reliable full-duplex data transmission. TCP is part of the TCP/IP protocol stack.
TFTP
Trivial File Transfer Protocol. Simplified version of FTP that allows files to be transferred from one computer to another over a network, usually without the use of client authentication (for example, username and password).
ToS
Type of service. See CoS.
U
UAC
User agent client. A client application that initiates the SIP request.
UAS
User agent server (or user agent). A server application that contacts the user when a SIP request is received, and then returns a response on behalf of the user. The response accepts, rejects, or redirects the request.
UDP
User Datagram Protocol. Connectionless transport layer protocol in the TCP/IP protocol stack. UDP is a simple protocol that exchanges datagrams without acknowledgments or guaranteed delivery, requiring that error processing and retransmission be handled by other protocols. UDP is defined in RFC 768.
u-law
See mu-law.
V
VAD
Voice activity detection. When enabled on a voice port or a dial peer, silence is not transmitted over the network, only audible speech. When VAD is enabled, the sound quality is slightly degraded but the connection monopolizes much less bandwidth.
VoIP
Voice over IP. The capability to carry normal telephony-style voice over an IP-based Internet 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. In VoIP, the DSP segments the voice signal into frames, which then are coupled in groups of two and stored in voice packets. VoIP is a blanket term, which generally refers to Cisco's standard-based (for example H.323) approach to IP voice traffic.
X
XML
eXtensible Markup Language. XML is an open standard for defining data elements on web pages and in business documents.
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