Cisco Cisco IOS Software Release 12.2(4)T
Cisco 2600 and 3600 Routers MGCP Voice Gateway Interoperability with Cisco CallManager
Glossary
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Cisco IOS Release 12.2(2)XA and 12.2(4)T
Glossary
call leg—A discrete segment of a call connection that lies between two points in the connection. An
end-to-end call consists of four call legs, two from the perspective of the source access server, and two
from the perspective of the destination access server.
end-to-end call consists of four call legs, two from the perspective of the source access server, and two
from the perspective of the destination access server.
Cisco CallManager—The Cisco CallManager serves as the software-based call-processing component
of the Cisco IP telephony solution. The Cisco CallManager system extends enterprise telephony features
and functions to packet telephony network devices such as IP phones, media processing devices,
Voice-over-IP (VoIP) gateways, and multimedia applications. The Cisco CallManager system includes a
suite of integrated voice applications that perform voice conferencing and manual attendant console
functions.
of the Cisco IP telephony solution. The Cisco CallManager system extends enterprise telephony features
and functions to packet telephony network devices such as IP phones, media processing devices,
Voice-over-IP (VoIP) gateways, and multimedia applications. The Cisco CallManager system includes a
suite of integrated voice applications that perform voice conferencing and manual attendant console
functions.
Cisco CallManager server—Cisco's high-availability server platform on which Cisco CallManager
software comes preinstalled.
software comes preinstalled.
cluster—Set of Cisco CallManagers that share the same database.
codec—A DSP software algorithm that compresses/decompresses speech or audio signals.
dial peer—Defines the characteristics associated with a call leg. Dial peers are used to apply attributes
to call legs and to identify the call origin and destination. In Voice over IP, there are two types of dial
peers: POTS and VoIP. Use the dial-peer voice command to define dial peers and to switch to dial-peer
configuration mode.
to call legs and to identify the call origin and destination. In Voice over IP, there are two types of dial
peers: POTS and VoIP. Use the dial-peer voice command to define dial peers and to switch to dial-peer
configuration mode.
digital signal processor—See DSP.
DNS—Domain Name System. A system used in the Internet for translating names of network nodes into
IP addresses.
IP addresses.
domain name system—See DNS.
DSP—Digital signal processor. A specialized computer chip designed to perform speedy and complex
operations on digitized waveforms. It is useful in processing sound, such as voice phone calls, and video.
A DSP segments the voice signal into frames and stores them in voice packets.
operations on digitized waveforms. It is useful in processing sound, such as voice phone calls, and video.
A DSP segments the voice signal into frames and stores them in voice packets.
DTMF— Dual tone multifrequency. A system used by touch tone telephones where one high and one
low frequency, or tone, is assigned to each touch tone button on a phone. DTMF digits can be detected
by the voice ports after the call setup is complete and are also trapped by the session application at either
end of the connection and carried over the IP network encapsulated in Real Time Conferencing Protocol
(RTCP) by using the RTCP APP extension mechanism.
low frequency, or tone, is assigned to each touch tone button on a phone. DTMF digits can be detected
by the voice ports after the call setup is complete and are also trapped by the session application at either
end of the connection and carried over the IP network encapsulated in Real Time Conferencing Protocol
(RTCP) by using the RTCP APP extension mechanism.
dual tone multi-frequency—See DTMF.
E&M—The “ear and mouth” interface (also called the “earth and magnet” interface, or the “recEive and
transMit” interface). Trunk circuits connect telephone switches to one another; they do not connect
end-user equipment to the network. The most common form of analog trunk circuit is the E&M interface,
which uses special signaling paths that are separate from the trunk's audio path to convey information
about calls. The signalling paths are known as the E-lead and the M-lead. E&M connections from routers
to telephone switches or to PBXs are preferable to FXS/FXO connections, because E&M provides better
answer and disconnect supervision.
transMit” interface). Trunk circuits connect telephone switches to one another; they do not connect
end-user equipment to the network. The most common form of analog trunk circuit is the E&M interface,
which uses special signaling paths that are separate from the trunk's audio path to convey information
about calls. The signalling paths are known as the E-lead and the M-lead. E&M connections from routers
to telephone switches or to PBXs are preferable to FXS/FXO connections, because E&M provides better
answer and disconnect supervision.
FXO—Foreign Exchange Office interface. A connection between a POTS telephone and a digital
telephony switching system.
telephony switching system.
FXS—Foreign Exchange Station interface. A connection between a digital telephony switching system
and a POTS telephone.
and a POTS telephone.
gateway—A special purpose device that performs an application-layer conversion of information from
one protocol stack to another. To connect an IP telephony device to the Public Switched Telephone
Network (PSTN), you must use an intermediary device, called a gateway. A VoIP gateway allows users
one protocol stack to another. To connect an IP telephony device to the Public Switched Telephone
Network (PSTN), you must use an intermediary device, called a gateway. A VoIP gateway allows users