SIP (Session Initiation Protocol) Study Notes: +W - Technology Skills For Women Series

SIP (Session Initiation Protocol)
Configuring - Monitoring - Troubleshooting
Study Notes
+W - Technology Skills For Women Series1
http://SlideShare.net/OxfordCambridge
Men are allowed to read too, if they wish, as the language style and the document format are universal.
Study Notes
Table of Contents
About +W - Technology Skills For Women series ................................................................................ 4
Course Objectives .................................................................................................................................... 6
References ............................................................................................................................................... 7
A.
Introducing SIP......................................................................................................................................... 8
1. SIP and associated standards .............................................................................................................. 8
Question ................................................................................................................................................ 12
Question ................................................................................................................................................ 12
B.
2. SIP components ................................................................................................................................. 12

Question ................................................................................................................................................ 15
Question ................................................................................................................................................ 16
Summary................................................................................................................................................ 16
C.
SIP Messages and Addressing................................................................................................................ 18

1. SIP messages...................................................................................................................................... 18
Question ................................................................................................................................................ 22
Question ................................................................................................................................................ 22
2. SIP addressing .................................................................................................................................... 23
Question ................................................................................................................................................ 26
Question ................................................................................................................................................ 26
Summary................................................................................................................................................ 26
D.
SIP Call Setup Models and Fault Tolerance ........................................................................................... 28

1. Call setup models............................................................................................................................... 28
Note ....................................................................................................................................................... 30
Question ................................................................................................................................................ 34
Question ................................................................................................................................................ 34
2. Robust SIP design............................................................................................................................... 35
Question ................................................................................................................................................ 36
Question ................................................................................................................................................ 37
3. Cisco implementation of SIP .............................................................................................................. 37
Summary................................................................................................................................................ 38
E.
Configuring and Monitoring SIP ............................................................................................................ 39

1. Configuring SIP on a Cisco router ...................................................................................................... 39
Question ................................................................................................................................................ 41
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Study Notes
2. Monitoring and troubleshooting SIP ................................................................................................. 42

Question ................................................................................................................................................ 44
Question ................................................................................................................................................ 45
Question ................................................................................................................................................ 45
Summary................................................................................................................................................ 45
F.
Words and Definitions ........................................................................................................................... 47
G.
Answers to Quizzes................................................................................................................................ 83
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Study Notes
About +W - Technology Skills For Women series

Study Notes in the field of technology will be put together under this category for the following reasons:
to encourage ladies, who wish to do so, to stand up and look over the fence into technology related
topics;
with apprehension or fear;
and perhaps consider embracing a career move into this technological path;
or simply as to broaden their general knowledge; after all ICT is in most aspects of everyday life;
no matter the decision, their skills, professional strengths, and contribution can only be something
positive for technical and technological fields.
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Study Notes
SIP: Session Initiation Protocol

A. Introducing SIP
B. SIP Messages and Addressing
C. SIP Call Setup Models and Fault Tolerance
D. Configuring and Monitoring SIP
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Study Notes
Course Objectives
Topic
When you have completed this topic, you should be able to
Introducing SIP
recognize the functionality of SIP, and identify the types of

user agents and servers used by SIP.
SIP Messages and Addressing
distinguish between the types, use, and structure of SIP

messages, identify SIP address formats, and recognize how SIP
addresses are registered and resolved.
SIP Call Setup Models and Fault

Tolerance
distinguish between SIP interworking models for call setup,

recognize strategies for maintaining VoIP service, and identify
SIP components supported by Cisco.
Configuring and Monitoring SIP
identify the configuration commands used to implement SIP

call setup models, and the commands used to provide support
for monitoring and troubleshooting SIP.
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Checked?
Yes/No
Study Notes
References
SIP - An Introduction (PDF) 2011-01-11? James Wright. Konnetic
Integrating Voice and Data Networks 2000, Scott Keagy, Cisco Press, 1578701961
Troubleshooting Cisco IP Telephony 2002, Paul Giralt, Addis Hallmark, Anne Smith, Cisco Press, 1587050757
Voice over IP First-Step 2005, Kevin Wallace, Cisco Press, 1587201569
http://en.wikipedia.org/wiki/Media_Gateway_Control_Protocol 3 September 2013
http://en.wikipedia.org/wiki/Session_Initiation_Protocol 5 March 2014
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Study Notes
A. Introducing SIP
After completing this topic, you should be able to recognize the functionality of SIP, and identify the
types of user agents and servers used by SIP.
1. SIP and associated standards
2. SIP components
Summary
1. SIP and associated standards

Session Initiation Protocol (SIP) provides a framework for establishing and maintaining Voice over IP (VoIP)
calls.
SIP is a signaling and control protocol for the establishment, maintenance, and termination of multimedia
sessions with one or more participants. SIP multimedia sessions include Internet telephone calls, multimedia
conferences, and multimedia distribution. Session communications may be based on multicast, unicast, or
both.
SIP operates on the principle of session invitations. Through invitations, SIP initiates sessions or invites
participants into established sessions. Descriptions of these sessions are advertised by any one of several
means, including the Session Announcement Protocol (SAP) defined in RFC 2974, which incorporates a
session description according to the Session Description Protocol (SDP) defined in RFC 2327.
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Study Notes
SIP uses other Internet Engineering Task Force (IETF) protocols to define other aspects of VoIP and
multimedia sessions; for example, URLs for addressing, Domain Name System (DNS) for service location, and
Telephony Routing over IP (TRIP) for call routing.
SIP supports personal mobility and other Intelligent Network (IN) telephony subscriber services through name
mapping and redirection services. Personal mobility allows a potential participant in a session to be identified
by a unique personal number or name.
IN provides carriers with the ability to rapidly deploy new user services on platforms that are external to the
switching fabric. Access to the external platforms is by way of an independent vendor and standard user
interface. Calling-card services, toll-free number services, and local number portability are just three of these
services.
Multimedia sessions are established and terminated by these services:
user location services
user capabilities services
user availability services
call setup services
call handling services
user location services

User location services are employed to locate an end system.
user capabilities services
User capabilities services are used to select the media type and parameters for multimedia sessions.
user availability services
User availability services are employed to determine the availability and desire for a party to participate in a
session.
call setup services
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Call setup services are used to establish a session relationship between parties and to manage call progress.
call handling services
Call handling services are used to transfer and terminate calls.
Although the IETF has made great progress in defining extensions that allow SIP to work with legacy voice
networks, the primary motivation behind the protocol is to create an environment that supports nextgeneration communication models that use the Internet and Internet applications.
SIP is described in IETF RFC 3261 (published in June 2002), which renders RFC 2543 (published in March 1999)
obsolete.
The Cisco SIP-enabled product portfolio encompasses all components of a SIP network infrastructure, from IP
Phones and access devices to call control and public switched telephone network (PSTN) interworking.
The first Cisco SIP products were deployed with live traffic several years ago.
All of these Cisco SIP products are deployed in live networks spanning a variety of applications and
continents. The first four products are
Cisco IP Phones
Cisco Analog Telephone Adaptor (Cisco ATA 186)
Cisco packet voice gateways
Cisco SIP Proxy Server
Cisco IP Phones
The Cisco IP Phone series, including the Cisco IP Phone 7970, Cisco IP Phone 7960 and Cisco IP Phone 7940,
support SIP user agent (UA) functionality.
These IP Phones deliver functionality such as inline power support and dual Ethernet ports, and deliver
traditional desktop functionality such as call hold, transfer, conferencing, caller ID, call waiting, and a lighted
message waiting indicator.
Cisco Analog Telephone Adaptor (Cisco ATA 186)
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Study Notes
The Cisco ATA 186 supports SIP UA functionality. With two Foreign Exchange Station (FXS) ports and a single
Ethernet port, the ATA 186 provides a low-cost means to connect analog phones to a SIP network.
The ATA 186 also delivers traditional desktop functionality such as call hold, transfer, conferencing, caller ID,
and lighted call-waiting and message waiting indicators.
Cisco packet voice gateways
The Cisco Series 1700 Modular Access Routers that are voice-capable, Cisco 2600 Series multiservice
platforms, Cisco 3800 Series Integrated Services Routers, 3700 Series Integrated Services Routers, Cisco
AS5000 Series Universal Gateways, and Cisco 7200 Series voice gateways all support SIP UA functionality.
These products provide a means of connecting SIP networks to traditional time-division multiplexing (TDM)
networks via T1, E1, digital service level 3 (DS3), channel associated signaling (CAS), PRI or BRI, R2 signaling,
FXS, Foreign Exchange Office (FXO), or ear and mouth (E&M) interfaces.
Cisco packet voice gateways are used to build the largest packet telephony networks in the world.
Cisco SIP Proxy Server
The Cisco SIP Proxy Server provides the functionality of a SIP proxy, SIP redirect, SIP registrar, and SIP location
services server.
The Cisco SIP Proxy Server provides the foundation for call routing within SIP networks; it can interwork with
traditional SIP location services, such as DNS or telephone number mapping (E.164 number [ENUM]), with
feature servers via a SIP redirect message, and with H.323 location services using standard location request
(LRQ) messages.
The Cisco SIP Proxy Server runs on either Solaris or Linux operating systems.
The last three Cisco SIP products deployed in live networks spanning a variety of applications and continents
are
Cisco BTS 10200 Softswitch
Cisco PGW 2200 PSTN Gateway
Cisco PIX Security Appliance and Cisco Adaptive Security Appliance (ASA)
Cisco BTS 10200 Softswitch

The Cisco BTS 10200 Softswitch provides softswitch functionality to Class 4 and Class 5 networks, and provides
SIP-to-Signaling System 7 (SS7) gateway functionality for American National Standards Institute (ANSI)
standardized networks.
The BTS 10200 Softswitch supports SIP UA functionality in conjunction with a Cisco packet voice media
gateway, such as a Cisco AS5000 Series Universal Gateway or a Cisco MGX 8000 Series Voice Gateway.
Cisco PGW 2200 PSTN Gateway
The Cisco PGW 2200 PSTN Gateway provides softswitch functionality for Class 4 networks, as well as Internet
offload and SIP-to-SS7 gateway functionality for international networks.
The PGW 2200 PSTN Gateway supports ISDN User Part (ISUP) certification in over 130 countries.
The PGW 2200 PSTN Gateway supports SIP UA functionality in conjunction with a Cisco packet voice media
gateway such as an AS5000 Series Universal Gateway or MGX 8000 Series Voice Gateway.
Cisco PIX Security Appliance and Cisco Adaptive Security Appliance (ASA)
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Study Notes
The Cisco PIX Security Appliance and the Cisco ASA are SIP-aware networking devices that provide firewall and
Network Address Translation (NAT) functionality.
Because these devices are SIP-aware, they are able to dynamically allow SIP signaling to traverse network and
addressing boundaries without compromising overall network security.
When functioning in this capacity, the Cisco PIX Security Appliance and the Cisco ASA are called application
layer gateways (ALGs).
Questioni
Identify the Internet Engineering Task Force (IETF) protocol used by SIP for call routing.
Options:
1.
Border Gateway Protocol (BGP)
2.
Open Shortest Path First (OSPF)
3.
Routing Information Protocol (RIP)
4.
Telephony Routing over IP (TRIP)
Questionii
Identify the SIP services that select the media type and parameters.
Options:
1.
Call handling services
2.
Call setup services
3.
User availability services
4.
User capabilities services
5.
User location services
B. 2. SIP components
SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). SIP is modeled on the
interworking of UAs and network servers.
A UA consists of two functional components:
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Study Notes
user agent client (UAC)
user agent server (UAS)
user agent client (UAC)

A UAC is a client application that initiates a SIP request.
user agent server (UAS)
A UAS is a server application that contacts the user when a SIP invitation is received and then returns a
response on behalf of the user to the invitation originator.
Typically, a SIP UA can function as a UAC or a UAS during a session, but not both in the same session.
Whether the endpoint functions as a UAC or a UAS depends on the UA that initiated the request; the
initiating UA uses a UAC and the terminating UA uses a UAS.
From an architectural standpoint, the physical components of a SIP network are grouped into two categories:
UAs and SIP servers.
SIP UAs include these devices:
IP telephone
gateway
IP telephone
An IP telephone acts as a UAS or UAC on a session-by-session basis. Software telephones and Cisco SIP IP
Phones initiate SIP requests and respond to requests.
gateway
A gateway acts as a UAS or UAC and provides call control support. Gateways provide many services, the most
common being a translation function between SIP UAs and other terminal types. This function includes
translation between transmission formats and between communications procedures. A gateway translates
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Study Notes
between audio and video signals and performs call setup and clearing on both the IP side and the switched
circuit network (SCN) side.
SIP servers include these types:
proxy server
redirect server
registrar server
location server
proxy server
A proxy server is an intermediate component that receives SIP requests from a client, then forwards the
requests on behalf of the client to the next SIP server in the network. The next server can be another proxy
server or a UAS. Proxy servers can provide functions such as authentication, authorization, network access
control, routing, reliable request transmissions, and security.
redirect server
A redirect server provides a UA with information about the next server that the UA should contact. The server
can be another network server or a UA. The UA redirects the invitation to the server identified by the redirect
server.
registrar server
A registrar server makes requests from UACs for registration of their current location. Registrar servers are
often located near or even colocated with other network servers, most often a location server.
location server
A location server is an abstraction of a service providing address resolution services to SIP proxy or redirect
servers. A location server embodies mechanisms to resolve addresses. These mechanisms can include a
database of registrations or access to commonly used resolution tools such as Finger protocol, Referral Whois
(RWhois), Lightweight Directory Access Protocol (LDAP), or operating system-dependent mechanisms. A
registrar server can be modeled as one subcomponent of a location server; the registrar server is partly
responsible for populating a database associated with the location server.
Except for the voice register mode request, communication between SIP components and a location server is
not standardized.
Leaders in the communications industry are constantly developing new products and services that rely on SIP,
and they are offering attractive new communications services to their customers.
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Study Notes
Microsoft added, in the past, support for SIP clients in core product offerings - a step that proliferates SIP
clients on personal computers worldwide. SIP is gaining momentum in every market.
Cisco has be enabling the advance of new communications services with a complete SIP-enabled portfolio,
including proxy servers, packet voice gateways, call control and signalling, IP Phones, and firewalls. Cisco
solutions support a variety of call control and standard protocols, including H.323, Media Gateway Control
Protocol (MGCP), and SIP, which can coexist in the same customer network.
Questioniii
Identify which are SIP server types.
Options:
1.
Dynamic Host Configuration Protocol (DHCP)
2.
Gateway
3.
Location
4.
Proxy
5.
Redirect
6.
Registrar
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Study Notes
Questioniv
Which SIP server is often collocated with the location server?
Options:
1.
Gateway
2.
Proxy
3.
Redirect
4.
Registrar
Summary
SIP is a signaling and control protocol for the establishment, maintenance, and termination of multimedia
sessions with one or more participants. Such multimedia sessions are established and terminated by five
services: user location, user capabilities, user availability, call setup, and call handling. The Cisco SIP-enabled
product portfolio comprises all components of a SIP network infrastructure, from IP Phones and access
devices to call control and PSTN interworking.
SIP is modelled on the interworking of UAs and network servers. A UA consists of two functional components:
the UAC and the UAS. From an architectural viewpoint, the physical components of a SIP network are
grouped into two categories: UAs, including IP telephones and gateways, and SIP servers, which include
proxy, redirect, registrar, and location servers.
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Study Notes
C. SIP Messages and Addressing

After completing this topic, you should be able to distinguish between the types, use, and structure
of SIP messages, identify SIP address formats, and recognize how SIP addresses are registered and
resolved.
1. SIP messages
2. SIP addressing
Summary
1. SIP messages
Communication between SIP components uses a request and response message model.
INVITE sip:bob@biloxi.com SIP/2.0

Via: SIP/2.0/UDP
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Study Notes
pc33.atlanta.com;branch=z9hG4bk776asdhds
Max-Forwards: 70
To: Bob<sip:bob@biloxi.com>
From: Alice<sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66170@pc33.atlanta.com
CSeq: 314159 INVITE
Contac: <sip:alice@pc33.atlanta.com>
Content-Type: application/sdp
Content-Length: 142
SIP communication involves two messages:
request from a client to a server
response from a server
request from a client to a server

A request from a client to a server consists of a request line, header lines, and a message body.
response from a server
A response from a server to a client consists of a status line, header lines, and a message body.
All SIP messages are text-based and modeled on RFC 822, Standard for the Format of ARPA Internet Text
Messages, and RFC 2068, Hypertext Transfer Protocol - HTTP/1.1.
SIP defines four types of headers: a general header, an entity header, a request header, and a response
header. The first two types of headers appear on both message types. The latter two types of headers are
specific to request and response, respectively.
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Study Notes
INVITE sip:bob@biloxi.com SIP/2.0

Via: SIP/2.0/UDP
pc33.atlanta.com;branch=z9hG4bk776asdhds
Max-Forwards: 70
To: Bob<sip:bob@biloxi.com>
From: Alice<sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66170@pc33.atlanta.com
CSeq: 314159 INVITE
Contac: <sip:alice@pc33.atlanta.com>
Content-Type: application/sdp
Content-Length: 142
In the request line, SIP uses a message to indicate the action to be taken by the responding component
(usually a server).
These six request messages indicate the action that the responding component should take:
INVITE
acknowledgment (ACK)
BYE
CANCEL
OPTIONS
REGISTER
INVITE
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Study Notes
The INVITE message is originated by a client to indicate that the server is invited to participate in a session. An
invitation includes a description of the session parameters.
acknowledgment (ACK)
The ACK message is originated by a client to indicate that the client has received a response to its earlier
invitation.
BYE
The BYE message is originated by a client or server to initiate call termination.
CANCEL
The CANCEL message is originated by a client or server to interrupt any request currently in progress. CANCEL
is not used to terminate active sessions.
OPTIONS
The OPTIONS message is used by a client to solicit capabilities information from a server. This method is used
to confirm cached information about a UA or to check the ability of a UA to message accept an incoming call.
REGISTER
The REGISTER message is used by a UA to provide information to a network server. Registrations have a finite
life and must be renewed periodically. This prevents the use of stale information when a UA moves.
SIP response messages are sent in response to a request and indicate the outcome of request interpretation
and execution. Responses take one of three basic positions: success, failure, or provisional. A status code
reflects the outcome of the request.
There are six response messages to indicate the status of a request.
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1xx (informational)
2xx (successful)
3xx (redirection)
4xx (client error)
5xx (server error)
6xx (global failure)
Study Notes
1xx (informational)
The 1xx (informational) status code is
a provisional response indicating that the request is still being processed.
2xx (successful)
The 2xx (successful) status code indicates that the requested action is complete and successful.
3xx (redirection)
The 3xx (redirection) status code indicates that the requestor requires further action; for example, when a
redirect server responds with "moved" to advise the client to redirect its invitation.
4xx (client error)
The 4xx (client error) status code is a fatal response indicating that the client request is flawed or impossible to
complete.
5xx (server error)
The 5xx (server error) status code is a fatal response indicating that the request is valid but the server failed to
complete it.
The 6xx (global failure) status code is a fatal response indicating that the request cannot be fulfilled by any
server.
Questionv
Identify the SIP message that is used to provide information to a network server.
Options:
1.
ACK
2.
INVITE
3.
OPTIONS
4.
REGISTER
Questionvi
Identify the SIP response message that is provisional.
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Study Notes
Options:
1.
1xx (informational)
2.
2xx (successful)
3.
3xx (redirection)
4.
4xx (client error)
5.
5xx (server error)
6.
2. SIP addressing
To obtain the IP address of a SIP UAS or a network server, a UAC performs address resolution of a user
identifier.
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Study Notes
Fully qualified domain names
sip:jdoe@cisco.com
E.164 addresses
sip:14805551234@gateway.com: user=phone
Mixed addresses
sip:14805551234: password=changeme@10.1.1.1
sip:jdoe@10.1.1.1
An address in SIP is defined in the syntax for a URL with "sip:" or "sips:" (for secure SIP connections) as the
URL type. SIP URLs are used in SIP messages to identify the originator, the current destination, the final
recipient, and any contact party.
When two UAs communicate directly with each other, the current destination and final recipient URLs are
the same. However, the current destination and the final recipient are different if a proxy or redirect server is
used.
An address consists of an optional user ID, a host description, and optional parameters to qualify the address
more precisely. The host description may be a domain name or an IP address. A password is associated with
the user ID, and a port number is associated with the host description.
This example shows instances of SIP addresses.
In the example, "sip:14085551234@gateway.com; user=phone", the "user=phone" parameter is required to
indicate that the user part of the address is a telephone number. Without the "user=phone" parameter, the
user ID is taken literally as a numeric string. The "14085559876" in the URL "sip:14085559876@10.1.1.1" is
an example of a numeric user ID. In the same example, the password "changem" is defined for the user.
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Study Notes
A SIP address is acquired in several ways: by interacting with a user, by caching information from an earlier
session, or by interacting with a network server. For a network server to assist, it must recognize the
endpoints in the network. This knowledge is abstracted to reside in a location server and is dynamically
acquired by its registrar server.
To contribute to this dynamic knowledge, an endpoint registers its user addresses with a registrar server. This
example shows a voice REGISTER mode request to a registrar server.
To resolve an address, a UA uses a variety of internal mechanisms such as a local host table, DNS lookup,
Finger protocol, rwhois, or LDAP, or it leaves that responsibility to a network server. A network server uses
any of the tools available to a UA or interacts through a nonstandard interface with a location server.
This example shows a SIP proxy server resolving the address by using the services of a location server.
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Study Notes
Questionvii
Identify the ways in which a SIP UA can resolve an address.
Options:
1.
Dynamic Host Configuration Protocol(DHCP)
2.
It lets the network server resolve it
3.
It relies on WINS
4.
It uses a local host table
5.
It uses rwhois
Questionviii
Identify the type of SIP address that is represented by:
"sip:12486593178@gateway.com;user=phone".
Options:
1.
An E.164 address
2.
A fully qualified domain name
3.
A mixed address
Summary
SIP employs a request/response messaging model for communication. All SIP messages are text-based and
modeled on the HTTP syntax. SIP uses six response codes to indicate the status of a request: 1xx
(informational), 2xx (successful), 3xx (redirection), 4xx (client error), 5xx (server error), and 6xx (global
failure).
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SIP addresses use the format and structure of a URL. Network components such as location and registrar
servers record addresses and carry out address resolution.
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Study Notes
D. SIP Call Setup Models and Fault Tolerance

After completing this topic, you should be able to distinguish between SIP interworking models for
call setup, recognize strategies for maintaining VoIP service, and identify SIP components supported
by Cisco.
1. Call setup models
2. Robust SIP design
3. Cisco implementation of SIP
Summary
1. Call setup models

If a UAC recognizes the destination UAS, the client communicates directly with the server.
In situations in which the client is unable to establish a direct relationship, the client solicits the assistance of
a network server. There are three interworking models for call setup: direct, using a proxy server, and using a
redirect server.
When a UA recognizes the address of a terminating endpoint from cached information, or has the capacity to
resolve it by some internal mechanism, the UAC may initiate direct (UAC-to-UAS) call setup procedures.
Direct setup is the fastest and most efficient of the call setup procedures. However, direct setup has some
disadvantages. It relies on cached information or internal mechanisms to resolve addresses, which can
become outdated if the destination is mobile.
In addition, if the UA must keep information on a large number of destinations, management of the data can
become prohibitive. This makes the direct method nonscalable.
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Study Notes
Direct call setup is a three-step process:
step 1
step 2
step 3
step 1
In step 1, the originating UAC sends an invitation (INVITE) to the UAS of the recipient. The message includes an
endpoint description of the UAC and SDP.
step 2
In step 2, if the UAS of the recipient determines that the call parameters are acceptable, it responds positively
to the originator UAC.
step 3
In step 3, the originating UAC issues an ACK.
After the final step of the direct call setup process, the UAC and UAS have all the information that is required
to establish Real-Time Transport Protocol (RTP) sessions between them.
The proxy server procedure is transparent to a UA. The proxy server intercepts and forwards an invitation to
the destination UA on behalf of the originator.
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Study Notes
A proxy server responds to the issues of the direct method by centralizing control and management of call
setup and providing a more dynamic and up-to-date address resolution capability. The benefit to the UA is
that it does not need to learn the coordinates of the destination UA, yet can still communicate with the
destination UA. The disadvantages of this method are that using a proxy server requires more messaging and
creates a dependency on the proxy server. If the proxy server fails, the UA is incapable of establishing its own
sessions.
Note
Although the proxy server acts on behalf of a UA for call setup, the UAs establish RTP sessions directly with each
other.
When a proxy server is used, call setup involves a seven-step procedure. These are the first four steps of the
procedure.
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step 1
step 2
step 3
step 4
Study Notes
step 1
In step 1, the originating UAC sends an invitation (INVITE) to the proxy server.
step 2
In step 2, the proxy server, if required, consults the location server to determine the path to the recipient and
its IP address.
step 3
In step 3, the proxy server sends the INVITE to the UAS of the recipient.
step 4
to the proxy server.
These are the last three steps of the call setup procedure for the proxy server method.
step 5
step 6
step 7
step 5
In step 5, the proxy server responds to the originating UAC.
step 6
step 7
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Study Notes
In step 7, the proxy server forwards the ACK to the recipient UAS.
After the final step of the proxy server call setup procedure, the UAC and UAS have all the information
required to establish RTP sessions.
A redirect server is programmed to discover a path to the destination. Instead of forwarding the INVITE to the
destination, the redirect server reports back to a UA with the destination coordinates that the UA should try
next.
A redirect server offers many of the advantages of the proxy server. However, the number of messages
involved in redirection is fewer than with the proxy server procedure. The UA has a heavier workload
because it must initiate the subsequent invitation.
When a redirect server is used, call setup involves a seven-step procedure. These are the first four steps of
this process.
step 1
step 2
step 3
step 4
step 1
In step 1, the originating UAC sends an invitation (INVITE) to the redirect server.
step 2
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Study Notes
In step 2, the redirect server, if required, consults the location server to determine the path to the recipient
and its IP address.
step 3
In step 3, the redirect server returns a "moved" response to the originating UAC with the IP address obtained
from the location server.
step 4
In step 4, the originating UAC acknowledges the redirection.
These are the last three steps of the call setup procedure for the redirect server method.
step 5
step 6
step 7
step 5
In step 5, the originating UAC sends an INVITE to the remote UAS.
step 6
to the UAC.
step 7
After the final step of the redirect server call setup procedure, the UAC and UAS have all the information that
is required to establish RTP sessions between them.
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Study Notes
Questionix
Identify the disadvantage of using the direct call setup method.
Options:
1.
It has to learn the coordinates of the destination UA
2.
It needs the assistance of a network server
3.
It relies on cached information that may be out of date
4.
It uses more bandwidth
Questionx
Which of these describes call setup using a proxy server?
Options:
1.
If the proxy server fails, the UA cannot establish its own sessions
2.
If the proxy server fails, the UA uses RTP to establish its sessions
3.
The proxy server sends fewer redirection messages than a redirect server
4.
The UAs establish RTP sessions through the proxy server
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Study Notes
2. Robust SIP design

Maintaining high availability of a SIP environment requires a design that accommodates the failure of a
network server. There are two strategies for maintaining VoIP service in such situations.
In a SIP environment, the failure of a network server cripples UAs that are dependent on that server. In SIP,
the network servers are the proxy server, the redirect server, and the location server.
The most obvious way to preserve access to the critical components is to implement multiple instances of
access.
For replication of a proxy or redirect server to be effective, a UA must have the ability to locate an active
server dynamically. You can achieve this using either of these methods:
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method 1
method 2
Study Notes
method 1
In method 1, you must preconfigure a UA with the address of at least two of the servers. If access to its first
choice fails, it shifts to the second.
method 2
In method 2, if all servers are configured with the same name, you must configure a UA to look up the name
using DNS. The DNS query returns the addresses of all the servers matching the name, and the UA proceeds
down the list until it finds one that works.
In this example, SIP servers have been replicated to ensure survival of the SIP environment in the event of the
failure of a network server.
Questionxi
Identify the SIP components that need to be replicated in order to provide fault tolerance.
Options:
1.
Gateway server
2.
Location server
3.
Proxy server
4.
Redirect server
5.
Registrar server
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Study Notes
Questionxii
Identify the method that can be used to replicate a proxy server.
Options:
1.
Configuring a redirect server to act as a proxy server
2.
Configuring two location servers on the network
3.
Configuring two replication servers on the network
4.
Enabling the UA to dynamically locate an active server
3. Cisco implementation of SIP

Cisco implements SIP by providing support for SIP components. Therefore, Cisco provides support for these
three SIP components:
SIP UAs
network servers
other support
SIP UAs
Cisco provides support for SIP UAs in Cisco IP Phone. Cisco implements SIP UA (gateway) support in four
devices:
Cisco voice-enabled routers (first available in Cisco IOS Release 12.1), Cisco PGW 2200 PSTN Gateways, Voiceenabled Cisco AS5xx0 universal access servers, and Cisco BTS 10200 Softswitch.
network servers
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Study Notes
Cisco implements SIP proxy and redirect server support in the Cisco SIP Proxy Server. The server is an
application designed for a Red Hat Linux 7.3 or Solaris 8 operating environment.
other support
Other support refers to Cisco PIX Security Appliance and Cisco ASA monitoring the SIP handshaking to
dynamically open conduits for the RTP sessions.
Summary
Although call setup between UAs is possible, a proxy or redirect server may be employed for scalability or to
simplify UA configuration.
Maintaining high availability of a SIP environment requires a design that accommodates the failure of a
network server. Using multiple SIP proxy or redirect servers enhances survivability in such a situation.
Cisco supports standalone and gateway clients. The Cisco SIP Proxy Server supports SIP proxy or redirect
services.
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Study Notes
E. Configuring and Monitoring SIP

After completing this topic, you should be able to identify the configuration commands used to
implement SIP call setup models, and the commands used to provide support for monitoring and
troubleshooting SIP.
1. Configuring SIP on a Cisco router
2. Monitoring and troubleshooting SIP
Summary
1. Configuring SIP on a Cisco router

A SIPA SIP configuration consists of two parts: the SIP UA and the VoIP dial peers that select SIP as the session
protocol.
You need to use configuration commands to implement SIP call setup models.
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Study Notes
The SIP UA is one part of the SIP configuration. This example displays a sample SIP UA configuration.
The UA is enabled with the sip-ua command. Subcommands are optional. This example shows how you can
change the value of four retry counters. The configuration also specifies the name of a SIP proxy or redirect
server.
!
sip-ua
retry invite 2
retry response 2
retry bye 2
retry cancel 2
sip-server dns:server
!
SIP is selected as the call control protocol from inside a dial peer. SIP is requested by the session protocol
sipv2 dial-peer subcommand. This example displays two dial-peer variations.
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Study Notes
!
dial-peer voice 444 voip
destination-pattern 2339000
session protocol sipv2
session target ipv4:172.18.192.205
!
dial-peer voice 111 voip
destination-pattern 111
session protocol sipv2
session target sip-server
!
In this example, both dial peers include the session protocol sipv2 subcommand, and SIP is used when the
destination pattern matches either dial peer. The session target distinguishes one session from the other.
dial-peer 444
dial-peer 111
dial-peer 444
In dial-peer 444, the IP address of the server is provided as the session target. The address can be the address
of a UA, proxy server, or redirect server.
dial-peer 111
In dial-peer 111, the session target is the sip-server parameter. When the sip-server parameter is the target,
the IP address of the actual server is taken from the sip-server subcommand in the SIP UA configuration. This
means that from global configuration mode, the network administrator has entered the sip-ua command and
the sip-server dns:server subcommand. The address represents the location of a proxy server or redirect
server. In this example, the name of the SIP server is "server".
Questionxiii
Identify the show command that displays SIP UA response and retry information.
Options:
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Study Notes
1.
show call active voice
2.
show sip-ua retry
3.
show sip-ua statistics
4.
show sip-ua status
2. Monitoring and troubleshooting SIP

You can use the show and debug commands to provide support for monitoring and troubleshooting SIP.
There are six show commands that are valuable when examining the status of SIP components and
troubleshooting:
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show call active voice [brief]
show call history voice [last n | record | brief]
show sip-ua retry
show sip-ua status
show sip-ua timers
Study Notes
show call active voice [brief]

The show call active voice [brief] command displays the status, statistics, and parameters for all active voice
calls.
show call history voice [last n | record | brief]
The show call history voice [last n | record | brief] command displays call records from the history buffer.
show sip-ua retry
The show sip-ua retry command displays the SIP protocol retry counts. High counts should be investigated.
The show sip-ua statistics command displays the SIP UA response, traffic, and retry statistics.
show sip-ua status
The show sip-ua status command displays the SIP UA listener status, which should be enabled.
show sip-ua timers
The show sip-ua timers command displays the current value of the SIP UA timers (shown in the figure).
There are seven debug commands that are valuable when examining the status of SIP components and
troubleshooting. Here are the first four commands:
debug voip ccapi inout
debug ccsip all
debug ccsip calls
debug ccsip errors

The debug voip ccapi inout command shows every interaction with the call control application programming
interface (API) on both the telephone interface and on the VoIP side. By monitoring the output, you can follow
the progress of a call from the inbound interface or VoIP peer to the outbound side of the call. This debug
command is very active, so you must use it sparingly in a live network.
debug ccsip all
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Study Notes
The debug ccsip all command enables all ccsip-type debugging. This debug command is very active, so you
must use it sparingly in a live network.
debug ccsip calls
The debug ccsip calls command displays all SIP call details as they are updated in the SIP call control block. You
must use this debug command to monitor call records for suspicious clearing causes.
debug ccsip errors
The debug ccsip errors command traces all errors encountered by the SIP subsystem.
The last three debug commands are displayed.
debug ccsip events
debug ccsip messages
debug ccsip states
debug ccsip events

The debug ccsip events command traces events, such as call setups, connections, and disconnections. An
events version of a debug command is often the best place to start, because detailed debugs provide a great
deal of useful information.
The debug ccsip messages command shows the headers of SIP messages that are exchanged between a client
and a server.
debug ccsip states
The debug ccsip states command displays the SIP states and state changes for sessions within the SIP
subsystem.
Questionxiv
Which debug command would you use to trace call setups, connections, and disconnections?
Options:
1.
debug ccsip calls
2.
debug ccsip events
3.
4.
debug ccsip states
5.
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Study Notes
Questionxv
Which debug command must you use to monitor call records for suspicious clearing causes?
Options:
1.
debug ccsip all
2.
debug ccsip calls
3.
debug ccsip errors
4.
debug ccsip states
Questionxvi
Identify the show command that displays the SIP UA listener status, which should be enabled.
Options:
1.
show call active voice
2.
show sip-ua retry
3.
4.
show sip-ua status
Summary
A SIP configuration comprises two elements: the SIP UA and the VoIP dial peers that select SIP as the session
protocol. You enable the SIP UA using the sip-ua command. SIP is selected as the call control protocol from
inside a dial peer using the session protocol sipv2 dial-peer subcommand.
You can use the six show and seven debug commands to provide support for monitoring and troubleshooting
SIP.
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Study Notes
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Study Notes
F. Words and Definitions

AAA
Acronym for authentication, authorization, and accounting. Systems implemented to securely determine the identity and privileges
of a user and track that user's activities.
access class
The class of service a customer chooses when subscribing to DS-3 based Switched MultiMegabit Data Services (SMDS). The access
class is defined as 4, 10, 16, 25, or 34 Mbps. For users who subscribe to an access class lower than 34 Mbps, a 34 Mbps bandwidth is
available for burst transmissions; However, the duration of user bursts is limited so that the average throughput does not exceed the
specified access class. See also SMDS.
access code
A sequence of dialed digits that allows a user to gain access to a facility, service, feature, or function of a network or system.
access coordination
The design, ordering, installation, testing, and maintenance of local access services.
access delay
The time interval from the last digit of a dialed number until the call is delivered by the local exchange carrier (LEC) to the
appropriate interexchange carrier (IXC). Also known as call setup time. See also IXC.
access device
The hardware component used in a signaling controller system, access server, or multiplexer.
access digit
On a PBX, an outside line is normally accessed by dialing an access digit, such as 9.
access gateway
A gateway that allows the IP PBX to communicate with the PSTN or traditional PBX systems. See also IP, PBX, and PSTN.
access layer
Part of ISO-OSI layered protocol model.
access line
A transmission line that provides access to a larger system or network.
access link
The local access connection between a customer's premises and a carrier's point of presence (POP), which is the carrier's central
switching office or closest point of local termination. See also POP.
access method
The technique for moving data, voice, or video between storage and input/output devices. Also, the technique and/or program code
used in local area networks (LANs) to grant selective access to individual stations.
access node
See AN.
access port
Connects a network device to an IP device. For example, a computer can be connected to an IP phone through an access port.
access protocol
A set of specific procedures that enable a user to obtain services from a telephone company or network.
access server
Communications processor that connects asynchronous devices to a LAN or WAN through network and terminal emulation software.
Performs both synchronous and asynchronous routing of supported protocols. Sometimes called a network access server. Access
servers for the Cisco signaling controller are the Cisco AS5200, Cisco AS5300, and Cisco AS5800.
account code
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Study Notes
A numeric code that identifies the calling party for internal billing or accounting purposes. Account codes are often used by service
companies such as accountants and lawyers to bill specific clients for telephone expenses. Also known as a project code or bill-back
code.
ACD
Acronym for automatic call distributor. A device that handles a large number or incoming calls. An ACD performs four functions: first,
it recognizes and answers incoming calls; second, it looks in a database to decide how to route the call; third, based on these
instructions, it sends the call to an answering position based on a pre-determined, logical answering pattern. (Or, if all positions are
busy, the ACD plays a recorded message and places the call in a queue until an answering position becomes available); finally, the
ACD connects the call to an agent, once that agent has completed the previous call.
ACL
Acronym for access control list. A roster of users and groups of users, along with their access rights.
ACP
Acronym for automatic call processing. A system in which calls are processed entirely by computer.
additional call offering
An Integrated Services Digital Network (ISDN) feature that allows multiple calls to be placed simultaneously to the same telephone
number. A serving switch is programmed with the number of lines on the receiving telephone equipment. The switch will offer an
additional call if there is a line available to accept it. See also ISDN.
address
In a communications network, the identifying designation of an entity that is physically and/or logically distinct. Also, the destination
of a message. Also, in software, any location that can be specifically referred to in a program storage location, terminal, peripheral
device, cursor location or any other component.
Ad-Hoc conference
A Cisco CallManager feature that allows a conference controller to build a conference that has not been previously arranged. In an
Ad-Hoc conference, the conference controller individually calls and adds each participant to the conference. Compare to Meet-Me
Conference.
Administrative Reporting Tool
See ART.
Administrative VLAN
Used in non-Cisco switched networks in conjunction with Cisco IP Phones to indicate the virtual local area network (VLAN) of which
the phone is a member. Assigns the phone to an auxiliary VLAN. See also Operational VLAN.
ADPCM
Acronym for adaptive differential pulse code modulation. A speech coding method that uses fewer bits than the traditional pulse
code modulation (PCM).
ADU
Acronym for automatic dialing unit. A device that automatically generates a predetermined telephone number when a specific
button is pressed.
AEC
Acronym for automatic echo cancellation.
agent
Individuals or companies that market the services of a carrier, but are not directly employed by the carrier.
AIM
Advanced Interface Module. The data compression AIM provides hardware-based compression and decompression of packet data
transmitted and received on the serial network interfaces of the Cisco 2600 series router without occupying the Port Module Slot
that might otherwise be used for additional customer network ports. Designed to plug directly into a header on the Cisco 2600 series
router motherboard.
a-law
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Study Notes
ITU-T companding standard used in the conversion between analog and digital signals in pulse code modulation (PCM) systems. In
contrast to the North American mu-law, a-law provides a constant signal-to-distortion ratio over a broader dynamic range of analog
input signals at the expense of a poorer signal-to-distortion ratio for low-level signals. See also companding and -law.
ALB
Acronym for analog loop back. A method of testing modems in which the telephone line is disconnected and the transmitted signal is
looped back to the receiver.
alerting
The process a switch uses to inform customer premises equipment (CPE) that an incoming call is present and waiting for an answer.
For analog lines, alerting consists of applying a ringing voltage; for hybrid telephones, alerting consists of sending signaling bits; and
for digital telephones, cellular telephones, or personal communications service (PCS) handsets, it consists of sending a message to
the CPE that alerts the user. Alerting of the end user is a function of the CPE (e.g., audible ring, flashing lamp, voice announcement).
On some CPE, additional incoming calls for busy lines may be indicated via messages, lamps or call waiting tones. See also CPE and
PCS.
ambient noise
The background noise that is present on a non-digital communications line at all times.
AMIS
Acronym for Audio Messaging Interchange Specification. A series of standards aimed at addressing the problem of how voice
messaging systems produced by different vendors can network or inter-network. Before AMIS, systems from different vendors could
not exchange voice messages. AMIS deals only with the interaction between two systems for the purpose of exchanging voice
messages. It does not describe the user interface to a voice messaging system, specify how to implement AMIS in a particular
system, or limit the features a vendor may implement. See also AMIS-A.
AMIS-A
Acronym for Audio Messaging Interchange Specification-Analog. See also AMIS.
amplifier
An electronic device used to increase the amplitude or power level of a signal. Amplifiers are used in telecommunications on analog
transmission lines to offset the signal loss that occurs as the signal is propagated along the line.
AN
Acronym for access node. A broadband Integrated Services Digital Network (ISDN) remote switch that performs grooming,
concentration, and switching functions.
analog bridge
A device for connecting multiple analog circuits to a common circuit.
analog channel compression
A technique for fitting more than one program into a single channel using analog processes.
analog loop back
See ALB.
analog signal
A continuous signal that is infinitely and continuously variable in amplitude and/or frequency.
analog transmission
The transmission of continuously variable (analog) signals. As a signal is transmitted along an analog network, the signal strength
eventually weakens or attenuates. Amplifiers may be installed in the network to amplify the signal, but because there is no way to
differentiate between an analog signal and noise, both are amplified. Therefore, noise tends to accumulate in an analog network.
ANC
Acronym for Answer, Charge.
ANI
Acronym for automatic number identification. A PSTN system that transmits the billing number of the calling party for accounting
and billing purposes.
ANM
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Study Notes
Acronym for answer message. An off-hook signal sent in the reverse direction that indicates when the called party answers. Billing
starts when the answer message is received.
ANN
Acronym for Answer, No Charge.
ANSI
Acronym for American National Standards Institute. A U.S. organization chartered to accredit standards developed by a wide variety
of industry groups while avoiding improper influence from any one company or organization. ANSI does not develop standards, but
reviews and implements those developed by other organizations. For example, ANSI accredits standards for telephony developed by
the Alliance for Telecommunications Industry Standards (ATIS) under the auspices of the T1 Committee, and standards for cellular
radio developed by the Electronics Industry Association (EIA) and the Telecommunications Industry Association (TIA). ANSI is a
member of the International Organization for Standardization (ISO). See also ATIS, EIA, and ISO.
answerback
A signal sent by a data receiver to a data transmitter indicating that it is ready to receive data or to acknowledge the receipt of data.
answering machine
A CPE device that, in the absence of the called party, automatically answers incoming calls with a prerecorded message and records
messages from callers.
ANU
Acronym for Answer, Unqualified.
a-number
A cellular term referring to the number of the calling party. The originating switch analyzes the a-number in order to route a call to
the b-number, the number of the called party. The a-number can be analyzed by configuring dial plans created with the dial plan
provisioning (DPP) utility. See also dial plan and b-number.
AOS
Acronym for alternative operator service. A non-telephone company operator service. Users of AOS include hotels and non-telco
public telephones where a commission is paid to the establishment for allowing the AOS to bill for the call. Many AOS operations
have billing agreements with local exchange companies (LECs) which will pass the billed charges back to the customer's hotel room
or home telephone number.
API
Acronym for application programming interface. Software that an application program uses to request and carry out lower-level
services.
application
A software program that performs a function directly for a user. Examples include the Cisco CallManager administrative reporting
tool (ART) and Bulk Administration Tool (BAT), as well as Microsoft Word. A web browser is a network application.
application sharing
A form of data collaboration that allows a participant to select one or more of the applications resident on his/her PC and make it
available to the other participants. All participants may then manipulate the application as if it were executing on their PCs.
area code
The first three digits of a 10-digit telephone number in the North American Numbering Plan. See also NANP.
ARP
Acronym for Address Resolution Protocol. Internet protocol used to map an IP address to a MAC address. Defined in RFC 826. Allows
host computers and routers to determine the data link layer address corresponding to the IP address in a packet routed through the
LAN. Although the packet is addressed to an IP address, the LAN hardware responds only to data link layer addresses. The host or
router with the destination IP address replies with its own data link layer address in an ARP response, which the forwarding host or
router will use to construct a data link layer frame. The result is stored in cache memory so subsequent packets addressed to the
same destination can be routed without an explicit ARP process.
ARPA
Acronym for Advanced Research Projects Agency of the U.S. Department of Defense. ARPA funded research and experimentation
with ARPANET, the predecessor to the Internet. See also TCP/IP.
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Study Notes
ARQ
Acronym for automatic retransmission request (ARQ). A method of checking transmitted data on high speed data communications
systems in which the sender encodes an error detection field based on the contents of the message. The receiver recalculates the
check field and compares it with the received field. If the fields match, a positive acknowledgment ("ACK" or "PAK") is returned to
the sender. If the fields do not match, a negative acknowledgment (NAK) is returned to the sender.
ART
An acronym for audible ringing tone. A signal sent back to the calling party to indicate the called number is ringing. Also, an acronym
for administrative reporting tool. A web-based application for Cisco CallManager that generates reports on performance and service
details. See also CDR and CMR.
ARU
Acronym for audio response unit. An output device that provides a spoken response to digital inquiries from a telephone or other
device (For example, "Press 1 to hear this information again; Press 2 to hear more options.") Also known by the generic name
audiotex.
ASIC
Acronym for application specific integrated circuit. Circuit designs used by manufacturers to consolidate many chips into a single
package, reducing board size and power consumption.
AST
Acronym for automatic spanning tree. Function that supports the automatic resolution of spanning trees in source-route bridging
networks, providing a single path for spanning explorer frames to traverse from a given node in the network to another. AST is based
on the IEEE 802.1 standard. See also SRB.
AT
Acronym for Analog Access Trunk. Expressed as AT-2, AT-4, or AT-8 to correspond to 2-, 4-, and 8-port gateways.
ATB
Acronym for all trunks busy. A single tone repeated at a 120 impulse per minute (ipm) rate to indicate that all trunks in a routing
group are in use.
ATIS
Acronym for Alliance for Telecommunications Industry Standards, a Washington D.C. trade group heavily involved in standards
issues, including interconnection and interoperability issues.
ATM
Acronym for Asynchronous Transfer Mode. International standard for cell relay in which multiple service types (such as voice, video,
or data) are conveyed in fixed-length (53-byte) cells. Fixed-length cells allow cell processing to occur in hardware, thereby reducing
transit delays. ATM is designed to take advantage of high-speed transmission media such as E3, SONET, and T3.
attendant console
A large, specialized telephone set used by the operator to answer incoming calls and send those calls to the proper extension.
audio stream RTP packets
Capable of conducting real-time voice data over connectionless networks such as TCP/IP. See also RTP.
audio switch
A remote controlled device for switching conference room audio circuits that are used to deliver compressed video transmission
service. An audio switch can switch room audio connections to either a coder/decoder or a separate return required for multipoint
conferences. See also codec.
audiotex
Generic term for interactive voice response equipment and services. See also ARU.
authentication
The process of determining the identity of a user attempting to access a system.
authorization
The process of granting a user access to a system.
auto registration
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Study Notes
Process by which Cisco CallManager automatically detects and adds new IP telephony devices to its database, such as Cisco IP
Phones and Cisco DPA 7630 devices. Auto registration assigns the next available directory number designated for the device type at
the time that each new device is plugged into the network.
automatic callback
A feature of a telecommunications system or an IP telephony device that records, and can dial, the originating phone number of the
last incoming call.
availability
The degree to which a system or resource is operable and not in a state of congestion or failure at any given point in time.
AVD
Acronym for alternate voice data. A single transmission facility used for either voice or data.
AVVID
See Cisco AVVID.
back end
Functions and procedures of a database server, such as a node or software application, designed to manipulate data on a network.
See also client, FRF.11, and server.
back haul
A method of call routing in which the call is taken beyond its destination and then back to that destination. Usually used to attain
cheaper rates.
backup
The logical or physical provisioning of facilities to speed the process of restart and recovery following network failures. Also,
redundant facilities, including duplicated transaction files, duplicated processors, storage devices, terminal, telecommunications
hardware or switches.
band
The range of frequencies between two defined limits. Also, one of the six specific wide-area telephone service (WATS) geographic
service areas.
bandwidth
Difference between the highest and lowest frequencies available for network signals.
Amount of data that can be transmitted in a fixed amount of time, or the rated throughput capacity of a given network medium or
protocol.
baseband
A network technology in which only one carrier frequency is used (for example, Ethernet).
bastion server
A server that is accessible from a public network (such as the Internet) without protection from a firewall.
BAT
Acronym for bulk administration tool. A web-based application for Cisco CallManager that enables bulk system modifications,
including adding and deleting phones, modifying phones, and adding users and mailboxes.
BGP
Acronym for Border Gateway Protocol. The routing protocol used between separate administrative domains (for example, between
an enterprise corporation and its ISP).
BH
Acronym for busy hour. The peak 60-minute period during a business day when the largest volume of traffic is handled by a network.
B-ISDN
Acronym for Broadband Integrated Services Digital Network. A network that employs switching techniques independent of
transmission speeds, and that allows a network to expand its capacity without major equipment overhauls. B-lSDNs support gigabit
speed circuits in the public network and high speed switching of all traffic types in public and private networks. B-lSDNs also provide
bandwidth-on-demand capabilities. Contrast with N-ISDN. See also BRI, ISDN, and PRI.
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Study Notes
blind transfer
Passing a call without notifying the recipient. Also known as unsupervised transfer or cold transfer.
blocked call
An attempted call that cannot be connected. The two most common reasons for blocked calls are all lines or trunks to the central
office are in use, or all paths through a private branch exchange (PBX) or switch are in use. Also, a service offered by 900 providers
that permits users to request that their local carrier blocks all 900 calls in order to avoid incurring charges.
blocking
The inability to establish a new call because of restrictions or inaccessibility of facilities in the system being called.
b-number
A cellular term for the number of the called party. The originating switch analyzes the a-number, the number of the calling party, in
order to route the call to the b-number. See also a-number.
BOOTP
Acronym for Bootstrap Protocol. A TCP/IP protocol that enables a network device to discover certain startup information, such as its
IP address.
BPDU
Acronym for Bridge Protocol Data Unit. Spanning-Tree Protocol hello packet that is sent out at configurable intervals to exchange
information among bridges in the network. See also PDU.
break
To interrupt the sending of a message and take control of the circuit at the receiving end. Also, an interruption of a transmission or
process.
BRF
Acronym for Bridge Relay Function. As defined by the IEEE, an internal bridge function on a Token Ring switch that is responsible for
forwarding frames between port groupings with the same logical ring number. Within a BRF, source-route bridging or source-route
transparent bridging can be used to forward frames. See also CRF.
BRI
Acronym for Basic Rate Interface. ISDN interface composed of two B-channels and one D-channel for circuit-switched
communication of voice, video, and data. Compare with PRI. See also B-ISDN, ISDN, and N-ISDN.
bridge
A device that passes information between two network segments. Operates at layer 2 of the Open Systems Interconnection (OSI)
reference model (the data link layer). See OSI. Also, a device used to match circuits to each other to ensure minimum transmission
impairment. Bridging is normally required on multipoint data channels where several local loops or channels are interconnected.
Bridged Telnet
Offers Cisco Service Engineers (CSEs) transparent firewall access to the Cisco CallManager server on a customer site for diagnostic
and troubleshooting purposes. It enables a telnet client inside the Cisco Systems firewall to connect to a telnet process behind a
customer firewall.
broadband
A type of communications channel capable of carrying a large portion of the electromagnetic spectrum. A broadband channel can
accommodate the following media: audio, digital, and television. Also, a transmission facility having a bandwidth greater than 20 kHz
capable of high speed data transmission. Also, an analog transmission technique used with data and video transmissions that
provides multiple user channels through frequency-division multiplexing (FDM). See FDM.
broadcast
Data packet that is sent to all nodes on a network. Broadcasts are identified by a broadcast address. Compare with multicast and
unicast. See also broadcast address.
broadcast address
Special address reserved for sending a message to all stations. Generally, a broadcast address is a MAC destination address of all
ones. Compare with multicast address and unicast address. See also broadcast.
broadcast packet
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A data packet transmitted simultaneously to all network devices.

broadcast storm
An undesirable network event in which many broadcasts are sent at once. Broadcast storms use substantial network bandwidth and
may cause network time-outs.
browser
GUI-based hypertext client application, such as Internet Explorer, Mosaic, and Netscape Navigator, used to access hypertext
documents and other services located on innumerable remote servers throughout the World Wide Web and Internet. See also GUI.
BSI
Acronym for Basic Rate Interface. ISDN interface composed of two B-channels and one D-channel for circuit-switched
communication of voice, video, and data.
Bulk Administration Tool
See BAT.
busy
A call condition in which transmission facilities are already in use. A line is considered busy when the caller goes off-hook.
busy tone
A single tone that is repeated at a 60 impulse per minute (ipm) rate to indicate that a call's terminating location is already in use.
CAC
Acronym for call admission control. In Cisco CallManager, CAC maintains a desired level of voice quality over a WAN link by
regulating bandwidth consumption used by calls over the link. Limits the number of simultaneous active calls over the link. See also
locations and gatekeeper.
call admission control
See CAC.
call control
Telephone industry term used to describe the setting up, monitoring, and tearing down of phone calls.
call detail recording
See CDR.
call forward all calls
Configurable feature that re-routes all incoming calls destined for one telephony device to another phone or device.
call forward busy
Configurable feature that re-routes incoming calls to an alternate line when the first line is in use.
call forward no answer
Configurable feature that re-routes incoming calls from one phone to another phone when the first phone is not answered after a
certain number of rings.
call forwarding
Configurable feature that sends incoming calls routed to a particular directory number to another number.
Call Management Record
See CMR.
call park
Configurable feature that allows the user to deposit a stable call at a specified directory number, then go to another phone and dial
the park number to retrieve the call. (Call park differs from a "hold" feature by allowing the user to retrieve the call from any phone
on the same system. A system administrator must configure a call park number, or range of numbers, for this feature to work).
call pickup
Configurable feature that allows a user to redirect an incoming call that routed to another destination in order to retrieve the call on
the user's own phone or directory number. See also group call pickup.
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call processing
See distributed call processing and centralized call processing.
call waiting
Feature of telephony systems that notifies a caller when another call is coming in during an active call.
callback
Callback allows remote clients to dial into a central site, and then have the central site immediately call back the remote site.
caller ID
A display, available to the called party before the party answers a telephone call, that identifies the originating telephone number
and the subscriber's name associated with that number. See also CLID.
Calling Line Identification
See CLID.
calling party transformation settings
Allows the user to manipulate the appearance of the calling party's number for outgoing calls.
calling search space
Determines which partitions a calling device searches when attempting to complete a call.
camp on
A technique in which an incoming call is stored on hold until an attendant, trunk, trunk group, or station is available to accept it, at
which time the call is completed.
CAS
Acronym for centralized attendant service. One group of switchboard operators answers all incoming calls for several telephone
systems located throughout one city or region. Also, acronym for channel associated signaling. In-band signaling used to provide
emergency signaling information along with a wireless 911 call to the Public Safety Answering Point (PSAP).
CBQ
Acronym for class-based queuing. A queuing algorithm used in routers to manage congestion. Through user-definable class
definitions, incoming packet traffic is divided into classes. These divisions might fall along the lines of traffic from a given interface,
associated with a particular application, intended for a particular network or device destination, and all traffic of a specific priority
classification. Each class of traffic is assigned to a specific first-in-first-out (FIFO) queue, each of which is guaranteed some portion of
the total bandwidth of the router. See also WFQ.
CBWFQ
Acronym for class-based weighted fair queuing. Allows you to define traffic classes that are based on certain match criteria, such as
access control lists, input interface names, protocols, and Quality of Service (QoS) labels.
CC
Acronym for common carrier. A government regulated private company that furnishes the general public with telecommunications
services and facilities. Also, acronym for country code. Part of a numbering plan.
CCAPI
Acronym for Call Control Applications Programming Interface.
CCIS
Acronym for Common Channel Interoffice Signaling. A way of carrying telephone signaling information along a path different from
the path used to carry voice. CCIS occurs over a separate packet-switched digital network. Accelerates the setting up and tearing
down of phone calls and increases the amount of information compared to what can be carried by in-band signaling. See SS7.
CCITT
Acronym for Consultative Committee for International Telegraph and Telephone. A telecommunications organization that
recommended worldwide standards for common carrier communications services. This organization was superseded by the
International Telecommunications Union, now called the ITU-T. See ITU-T.
CCS
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Acronym for Common Channel Signaling. Signaling system used in telephone networks that separates signaling information from
user data. A specified channel is exclusively designated to carry signaling information for all other channels in the system. See CCIS.
CDB
Acronym for call detail blocking.
CDP
Acronym for Cisco Discovery Protocol. A device-discovery protocol that runs on all Cisco-manufactured equipment. Enables a device
to advertise its existence to other devices and receive information about other devices in the network. Cisco IP devices use CDP to
communicate information such as auxiliary VLAN ID, per port power management details, and quality of service (QoS) configuration
information with the Cisco Catalyst switch. See also VLAN and QoS.
CDR
Acronym for call detail recording. A stored database record containing data about a specific call. Processed as a unit and used to
create billing records, a CDR contains details such as the called and calling parties, originating switch, terminating switch, call length,
and time of day.
cell
Basic data unit for ATM switching and multiplexing. Cells contain identifiers that specify the data stream to which they belong. Each
cell consists of a 5-byte header and 48 bytes of payload. See also cell relay and ATM.
cell relay
Network technology based on the use of small, fixed-size packets, or cells. Because cells are fixed-length, they can be processed and
switched in hardware at high speeds. Cell relay is the basis for many high-speed network protocols including ATM, 802.6, and SMDS.
See also cell, ATM, and SMDS.
central office
See CO.
centralized call processing
Refers to a processing construct where all call processing is performed at a central site, or hub, and no call processing is performed at
branch sites.
channel service unit
See CSU.
CHAP
Acronym for Challenge/Handshake Authentication Protocol. A system for determining if a user has the correct password without
openly revealing that password. CHAP does not itself prevent unauthorized access, it merely identifies the remote end. The router or
access server then determines whether that user is allowed access.
checksum
Method for checking the integrity of transmitted data. A checksum is an integer value computed from a sequence of octets taken
through a series of arithmetic operations. The value is recomputed at the receiving end and compared for verification.
CIR
Acronym for committed information rate. A Frame Relay term identifying a certain average maximum data transmission rate.
circuit switching
Switching system in which a dedicated physical circuit path must exist between sender and receiver for the duration of the "call."
Used heavily in the public switched telephone network.
circuit-switched gateways
Gateways that require an open circuit for communications until the connection is released. See also gateway.
Cisco AVVID
Acronym for Cisco Architecture for Voice, Video, and Integrated Data.
Cisco CallManager
Software-based call processing component of the Cisco IP telephony solution, which 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. See also Cisco CallManager Administration.
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Study Notes
Cisco CallManager Administration

Graphical user interface used to configure and maintain the Cisco CallManager.
Cisco CallManager Directory
An LDAP directory that stores authentication and authorization information about telephony application users. See also LDAP.
Cisco CallManager group
A set of Cisco CallManagers that provide a failover scheme for the devices associated with that group. Defined and maintained
through Cisco CallManager Administration.
Cisco CallManager server
Cisco's high-availability server platform on which Cisco CallManager software comes preinstalled.
Cisco Discovery Protocol
See CDP.
Cisco IP Phone
A full-feature telephone that provides voice communication over an IP network while functioning much like a traditional analog
phone. Allows you to place and receive telephone calls, and supports features such as call forwarding, redial, speed dialing, call
transfer, and conference calling. Also allows you to access voice mail, providing connectivity to Cisco IP Telephony Solutions.
Cisco IP Telephony Solutions
A software and hardware product suite offering an IP alternative to traditional PBXs. Includes Cisco IP Phones, H.323-compatible
gateway clients, and server software enabling voice and data over an existing LAN or WAN infrastructure. See also Cisco IP Phone,
Cisco CallManager.
Cisco Media Convergence Servers
The Cisco MCS-7800 series server family, which includes the high-availability MCS-7830 and the Cisco AVVID IP telephony starter kits.
Comes with Cisco CallManager preloaded.
CLID
Acronym for caller line ID. Information about the billing telephone number from which a call originated. The CLID value might be the
entire phone number, the area code, or the area code plus local exchange. Also known as called Caller ID.
client
Node or software program (front-end device) that requests services from a server. The Cisco IP Phone is an example of a client.
client/server computing
Term used to describe distributed computing (processing) network systems in which transaction responsibilities are divided into two
parts: client (front end) and server (back end). Both terms (client and server) can be applied to software programs or actual
computing devices. Also called distributed computing (processing). See also back end and client.
client-server model
The process of workload sharing between the client, the server, and the network. Examples include the nameserver/nameresolver
paradigm of the domain name system (DNS), as well as fileserver/file-client relationships, such as network file system (NFS) and
diskless hosts. See also DNS and NFS.
closest-match routing
The process of matching the narrowest range of numbers in a given route pattern.
CMR
Acronym for call management record. Contains the count of bytes sent, packets sent, jitter, latency, dropped packets, etc. Also called
diagnostic records.
CNF
Acronym for a configuration file.
CO
Acronym for central office. Local telephone company office to which all local loops in a given area connect and in which circuit
switching of subscriber lines occurs. Central office can also refer to a single telephone switch, or what is known as a "public
exchange" in Europe.
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codec
Acronym for coder-decoder.
A device that typically uses pulse code modulation to transform analog signals into a digital bit stream, and digital signals back to
analog. See also <a href="#n."><span class="crossref">G .</span></a> , , and . Also, in Voice over IP, Voice over Frame Relay, and
Voice over ATM, a software algorithm used to compress/decompress speech or audio signals.
community strings
Passwords used by Simple Network Management Protocol to remotely manage network devices. See also SNMP.
companding
Contraction derived from the opposite processes of compression and expansion. Part of the pulse code modulation process,
whereby analog signal values are logically rounded to discrete scale-step values on a nonlinear scale. The decimal step number is
then coded in its binary equivalent prior to transmission. The process is reversed at the receiving terminal using the same nonlinear
scale. Compare with compression and expansion. See also a-law and -law.
companding law
See a-law and -law.
compression
Reducing the representation of the information, but not the information itself. Compression is accomplished by running a data set
through an algorithm that reduces the space required to store or the bandwidth required to transmit the data set. See also
expansion.
compression types
One of the key factors that determines the amount of bandwidth used per call. Compression types available in Cisco CallManager are
G.711 (default), G.723, and G.729.
conference bridge
A network used to interconnect three or more lines or trunks to allow simultaneous conversations.
conference call
A connection established between three or more stations that allows each station to communicate with all others simultaneously.
configuration file
An unformatted ASCII file that stores initialization information for an application. For Cisco CallManager, files in the .cnf format that
define the parameters for Cisco IP Phone connection.
CoS
Acronym for class of service. Also, an indication of how an upper-layer protocol requires a lower-layer protocol to treat its messages.
In SNA sub-area routing, CoS definitions are used by sub-area nodes to determine the optimal route to establish a given session. A
CoS definition contains a virtual route number and a transmission priority field. Also called type of service (ToS). Also, a collection of
features, privileges, and services that are easily assignable to a group or "class" of telephones. Class of service is used to simplify
administration and maintenance tasks in complex telephony networks.
country code
A one-, two-, or three-digit number used to specify the destination country for international calls. See also route filter tags.
CPE
Acronym for customer premises equipment. Telephone equipment, such as key systems, PBXs, answering machines, etc., that reside
on the customer's premises (e.g., office building, home office, or factory). Also called customer provided equipment.
CPU
Acronym for Central Processing Unit. The computing part, or "brain," of the computer. The CPU manipulates data and processes
instructions coming from software or manual operations.
CRF
Acronym for cell relay function. The basic function that an ATM network performs in order to provide a cell relay service to ATM endstations. Also, acronym for connection related function. A term used by Traffic Management to reference a point in a network or a
network element where per connection functions are occurring. This is the point where policing at the virtual channel connection or
virtual path connection level may occur.
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cRTP
Acronym for compressed Real-time Transport Protocol. See RTP.
CSU
Acronym for channel service unit. Digital interface device that connects end-user equipment to the local digital telephone loop.
Often referred to together with DSU as CSU/DSU. See also DSU. Also, acronym for channel status unit. A device used in conjunction
with a T-1 multiplexor that monitors each channel of the T-1 to ensure it is functioning properly.
CTI ports
Acronym for Computer Telephony Interface ports. Virtual devices that are used by Cisco CallManager applications such as Cisco
SoftPhone, Cisco IP AutoAttendent, and Cisco IP Interactive Voice Response System (IVR) to create virtual lines. CTI ports are
configured through the same Cisco CallManager Administration area as phones, but require different configuration settings.
CTI route point
Acronym for Computer Telephony Interface route point. Virtual device that can receive multiple simultaneous calls for the purpose
of application-controlled redirection. Once a CTI route point has been created, lines (directory numbers) can be added and
configured. Applications that use CTI route points include Cisco IP Auto Attendant, Cisco IP Interactive Voice Response System (IVR),
and Cisco TAPI/JTAPI.
data channel
See D-channel.
data service unit
See DSU.
database redundancy
See redundancy.
datagram
Logical grouping of information sent as a network layer unit over a transmission medium without prior establishment of a virtual
circuit. IP datagrams are the primary information units in the Internet. The terms cell, frame, message, packet, and segment are also
used to describe logical information groupings at various layers of the OSI reference model and in various technology circles.
D-channel
Acronym for data channel. Full-duplex, 16-kbps (BRI) or 64-kbps (PRI) ISDN channel used to carry control signals and customer call
data in a packet switched mode. Provides the signaling information for each of the voice channels (or B-channels).
DCOM
Acronym for Distributed Component Object Model. Protocol that enables software components to communicate directly over a
network. Developed by Microsoft and previously called Network OLE, DCOM is designed for use across multiple network transports,
including Internet protocols such as HTTP.
DDI
Acronym for discard digits instruction. Removes a portion of the dialed digit string before passing the number on to the adjacent
system. For example, DDI can remove an external access code from the dialed digit string for calls placed to a PSTN.
default router
For IP devices, identifies the default gateway used by the device. Also called default gateway.
device loads
Files that contain updated application software for phones or gateways. Provided automatically during installation or upgrades. See
also patch.
device pool
In Cisco CallManager, a collection of commonly configured devices (such as phones, computers and gateways,) that belong to a
common database, cluster and group. Use device pools to define common characteristics for devices, including region, Date/Time
Group, Cisco CallManager group, and calling search space for auto-registration.
DHCP
Acronym for Dynamic Host Configuration Protocol. A TCP/IP protocol that enables PCs and workstations to get temporary or
permanent IP addresses out of a pool from centrally-administered servers. Like its predecessor, BOOTP, DHCP provides a mechanism
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Study Notes
for allocating IP addresses manually, automatically and dynamically, so that addresses can be reused when hosts no longer need
them. Also, for Cisco CallManager, a DHC server is queried by a telephone or gateway device upon booting to determine network
configuration information. The DHCP server provides the device with an IP address, subnet mask, default gateway, DNS server, and a
TFTP server name or address. With Cisco IP Phones, DHCP is enabled by default. If disabled, you must manually enter the IP address
and other specifications manually on each phone locally.
dial pad
Buttons on a phone that are used to dial a phone number. The dial pad on a Cisco IP Phone operates like the dial pad on a traditional
telephone.
dial plan
A system that allows one telephone or Cisco IP device to connect to another telephone or Cisco IP device by using dialed digits. In
North America and many Caribbean nations, the dial plan is called the North American Numbering Plan. See also NANP.
dial tone
An audible signal that indicates automatic switching equipment is ready to receive DTMF or dial pulse signals required for a
connection. See also DTMF.
dialing sequence
Used to enable and disable the message waiting indicator. See also MWI.
Dialogic voice board
Printed circuit board containing digital signal processor (DSP) chips to digitize voice.
dial-up
The use of a rotary or dual tone multiple frequency (DTMF) telephone to initiate a call over the public switched telephone network.
See also DTMF.
dial-up line
A communications circuit established by a switched connection. Also, any circuit available over the public switched telephone
network. See also PSTN.
DID
Acronym for direct inward dialing. A method of directly dialing the directory number of a Cisco IP Phone or a telephone attached to a
PBX without routing calls through an attendant or an automated attendant console, such as Cisco WebAttendant. Compare to DOD.
direct inward dialing
See DID.
direct outward dialing
See DOD.
directory number
See DN.
directory services
A service that provides information about network objects and helps network devices locate service providers.
distributed call processing
A processing construct in which each central site and branch office contains its own call processing resources. In terms of the Cisco
CallManager, distributed call processing means that each central site and branch site contains its own Cisco CallManager or Cisco
CallManager cluster.
DN
Acronym for directory number. The telephone number or internal extension assigned to a Cisco IP Phone. The directory number is
assigned to the phone itself, not a location or a user, so if the phone is moved, it still retains the same directory number. Also called
subscriber number.
DNIS
Acronym for dialed number identification service.
DNS
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Acronym for domain name system. System used in the Internet for translating names of network nodes into IP addresses.
DOD
Acronym for direct outward dialing. The ability to dial directly from Cisco CallManager or PBX extension without routing calls through
an operator, attendant or automated attendant functions. Compare to DID.
domain name
In the Internet, a World Wide Web site consisting of a hierarchical sequence of names (labels) separated by periods (dots) that you
can visit with your browser. See also browser.
domain name server
Server that maintains a database for resolving host names and IP addresses. Network devices query the DNS server to specify remote
computers by host names rather than IP addresses.
domain name system
See DNS.
DoS
Acronym for Denial of Service. Type of attack that prevents access to or operation of a device or network.
DSCP
Acronym for differentiated services code point, or DiffServe CodePoint. A marker in the header of each IP packet that prompts
network routers to apply differentiated grades of service to various packet streams, forwarding them according to different Per-Hop
Behaviors (PHBs). Part of DiffServe, a set of technologies proposed by the IETF that allows Internet and other IP-based network
service providers to offer differentiated levels of service to customers and their information streams.
DSP
Acronym for digital signal processor. Specialized computer chip designed to perform speedy and complex operations on digitized
waveforms. Useful in processing sound, such as voice phone calls, and video.
DSU
Acronym for data service unit. Device used in digital transmission that adapts the physical interface on a DTE device to a transmission
facility such as T1 or E1. The DSU is also responsible for such functions as signal timing. Often referred to together with CSU, as
CSU/DSU. See also CSU.
DTMF
Acronym for Dual Tone Multi-Frequency. 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.
Dynamic Host Configuration Protocol
See DHCP.
E&M
Acronym for recEieve and transMit (or ear and mouth). Trunking arrangement generally used for two-way switch-to-switch or
switch-to-network connections. Cisco's analog E&M interface is an RJ-48 connector that allows connections to PBX trunk lines (tie
lines). E&M is also available on E1 and T1 digital interfaces.
E1
Wide-area digital transmission scheme used predominantly in Europe that carries data at a rate of 2.048 Mbps. E1 lines can be
leased for private use from common carriers. E1 is the European equivalent of a T1 line. See also T1.
echo
A type of distortion that occurs when a signal is reflected or otherwise returned with sufficient magnitude and delay to be perceived
by the speaker.
echo canceller
A device or system that reduces or eliminates echoes in voice transmission systems.
EEPROM
Acronym for electrically erasable programmable read-only memory. Basically, EPROM that can be erased using electrical signals
applied to specific pins. See also EPROM.
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EIA
Acronym for Electronics Industries Alliance. Group that specifies electrical transmission standards.
EIGRP
Acronym for Enhanced Interior Gateway Routing Protocol. Advanced version of IGRP developed by Cisco. Provides superior
convergence properties and operating efficiency, and combines the advantages of link state protocols with those of distance vector
protocols.
end-of-dialing character
A single character used to identify the end of the dialed digit string. For international numbers dialed within the NANP, this is usually
the # character. See also route filter tags.
endpoint
Device at which a virtual circuit or virtual path begins or ends.
enterprise parameters
Default settings that apply to all devices and services in the same cluster.
EPROM
Acronym for erasable programmable read-only memory. Nonvolatile memory chips that are programmed after they are
manufactured, and, if necessary, can be erased by some means and reprogrammed. Compare with EEPROM and PROM.
Ethernet
Baseband LAN specification invented by Xerox Corporation and developed jointly by Xerox, Intel, and Digital Equipment Corporation.
Used to connect computers, workstations, terminals, printers, and other devices located in the same building or campus.
event type
In error traces, specifies one or more of the following types of error events: debug, information, notice, warning, error, critical alert,
and emergency.
Event Viewer
A Windows NT server application that graphically displays a log of NT server events.
expansion
The switching of a number of input channels, such as telephone lines, onto a larger number of output channels. Compare to
compression.
Fast Ethernet
Any of a number of 100-Mbps Ethernet specifications. Fast Ethernet offers a speed increase 10 times that of the 10BaseT Ethernet
specification, while preserving such qualities as frame format, MAC mechanisms, and MTU. Such similarities allow the use of existing
10BaseT applications and network management tools on Fast Ethernet networks. Based on an extension to the IEEE 802.3
specification. Compare with Ethernet. See also 100BASE-T.
fax relay
Also known as demod/remod. One of the methods for IP fax transmission as defined by ITU-T. Fax relay defines the specification for
the demodulation of standard analog fax transmission from originating machines equipped with modems, and their remodulation for
presentation to a matching destination device, with the long-haul portion of the transmission being supported over an IP-based
network.
FCC
Acronym for Federal Communications Commission. U.S. government agency that supervises, licenses, and controls electronic and
electromagnetic transmission standards.
FDM
Acronym for frequency-division multiplexing. Technique in which the available transmission bandwidth of a circuit is divided by
frequency into narrower bands, each used for a separate voice or data transmission channel. FDM means many conversations can be
carried on one circuit. Compare to TDM.
FIFO
Acronym for first-in, first-out.
File Transfer Protocol
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Study Notes
See FTP.
Flash memory
A special kind of EEPROM that can be erased and reprogrammed in blocks instead of one byte at a time. Provides nonvolatile storage
so that software images can be stored, booted, and rewritten as necessary. Flash memory resides in a chip when the power is turned
off. See also EEPROM.
frame
Logical grouping of information sent as a data link layer unit over a transmission medium. Often refers to the header and trailer, used
for synchronization and error control, that surround the user data contained in the unit. The terms cell, datagram, message, packet,
and segment are also used to describe logical information groupings at various layers of the OSI reference model and in various
technology circles.
Frame Relay
ITU-T-defined access standard. Frame Relay services, as delivered by the telecommunications carriers, employ a form of packet
switching analogous to a streamlined version of X.25 networks. Packets are in the form of frames that are variable in length with the
payload being anywhere between zero and 4,096 octets. Frame Relay networks are able to accommodate data packets of various
sizes associated with virtually any native data protocol.
FRF.11
Acronym for Frame Relay Forum implementation agreement for Voice over Frame Relay (v1.0 May 1997). This specification defines
multiplexed data, voice, fax, DTMF digit-relay, and CAS/Robbed-bit signaling frame formats, but does not include call setup, routing,
or administration facilities. See also VoFR.
FTP
Acronym for File Transfer Protocol. Application protocol, part of the TCP/IP protocol stack, used for transferring files between
network nodes. Defined in RFC 959.
full-duplex
Capability for simultaneous data transmission between a sending station and a receiving station. Compare to half-duplex.
FXO
foreign exchange office. A connection between a POTS telephone and a digital telephony switching system.
FXS
Acronym for foreign exchange station. A connection between a digital telephony switching system and a POTS telephone.
G.711
An audio compression standard used for digital telephones on a digital PBX/ISDN. In G.711, encoded voice is already in the correct
format for digital voice delivery in the PSTN or through PBXes. G.711 uses a bandwidth of 64 Kbps. G.711-compliant devices can
communicate with other G.711 devices, but not with G.723 devices. Described in the ITU-T standard in its G-series
recommendations.
G.723.1
Describes a 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 allows dissimilar communication devices to communicate with each other using a
standardized communications protocol. Used for digital telephones on a digital PBX/ISDN that produces digital audio at either 6.4 or
5.3 Kbps. The higher bit rate provides a somewhat higher quality of sound. The lower bit rate provides system designers with
additional flexibility. Described in the ITU-T standard in its G-series recommendations.
G.729
ITU-T's standard voice algorithm. Describes the coding of encoding/decoding of speech at 8 kbps using CS-ACELP methods.
gatekeeper
Component of an H.323 conferencing system that performs call address resolution, admission control, and subnet bandwidth
management. Also, telecommunications: H.323 entity on a LAN that provides address translation and control 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 bandwidth
management 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. Also, in the Cisco CallManager, for example, the
gatekeeper is a device that supports the H.225 RAS message set used for call admission control (CAC), bandwidth allocation, and dial
pattern resolution. There is one gatekeeper device per Cisco CallManager cluster.
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gateway
The point at which a circuit-switched call is encoded and repackaged into IP packets. A gateway is an optional element in an H.323
conference and bridge H.323 conferences to other networks, communications protocols, and multimedia formats.
gateway loads
See device loads.
group call pickup
A feature that allows users to pick up incoming calls within their own group or within other call pickup groups by dialing the group
call pickup number for that group. See also call pickup.
GUI
Acronym for graphical user interface. User environment that uses pictorial as well as textual representations of the input and output
of applications and the hierarchical or other data structure in which information is stored. Conventions such as buttons, icons, and
windows are typical, and many actions are performed using a pointing device (such as a mouse).
H.320
Suite of ITU-T standard specifications for video conferencing over circuit-switched media such as ISDN, fractional T-1, and switched56 lines.
H.323
ITU-T standard that describes packet-based video, audio, and data conferencing. Allows dissimilar communication devices to
communicate with each other using a standardized communications protocol. H.323 is an umbrella standard that describes the
architecture of the conferencing system, and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its actual
protocol. For example, the Cisco IOS integrated router gateways use H.323 to communicate with Cisco CallManager. See also
gateway.
H.323 clients
Conferencing and collaboration tools designed for the Internet or intranet, including Microsoft NetMeeting devices and symbol
phones. See also Microsoft NetMeeting.
H.323 RAS
registration, admission, and status. The RAS signaling protocol performs registration, admissions, bandwidth changes, and status and
disengage procedures between the VoIP gateway and the gatekeeper. See also VoIP and gatekeeper.
half-duplex
A method of alternating the direction of signals between two terminals but not transmitting in both directions simultaneously.
Compare to full-duplex
handset
The portion of a telephone set containing the transmitter and receiver, usually designed to be hand-held when the telephone is in
use. For example, lift the handset of a Cisco IP Phone to press the dial pad numbers to place a call, review voice mail messages,
answer a call, and so on.
hookflash
A form of telecommunications signalling that works by briefly depressing the hookswitch on a telephone. Commonly used for call
waiting. Some phones have a "flash" button for this purpose.
hookflash duration
The hookflash interval. A configurable setting used to determine the length of the hookflash generated by pressing the flash button
on a phone.
hookswitch
The switch in a telephone that activates or deactivates the device when the handset is picked up or replaced.
host name
Name that identifies network devices on the network, enabling you to access the device using this name rather than the IP address.
HSRP
Acronym for Hot Standby Routing Protocol. A Cisco proprietary protocol used to increase availability of default gateways used by end
hosts.
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HTTP
Acronym for HyperText Transfer Protocol. Used by the web server and the client browser to communicate over the internet.
hub
A device that serves as the center of a star topology network. Also, a device that contains a number of independent interconnected
network modules. Hubs may be active (repeaters) or passive (splitters). Also, a common connection point for devices in a network.
hunt group
A series of directory numbers organized to share the load in such a way that if the first line is busy or unavailable, the next line is
"hunted" until an available number is found. In Cisco CallManager, for example, the hunt group is a list of destinations that
determine the call forwarding order of a call once it has arrived at a pilot point. Hunt groups and pilot points must be established for
call routing by the Cisco Telephony Call Dispatcher (TCD).
hunting
The automatic routing of calls to an idle circuit in a prearranged group when the circuit being called is busy or unavailable. Also, the
movement of a call as it progresses through a group of lines. The call will try to connect to the first line of the group. If that line is
busy or unavailable, it will try the second line, and then the third line, etc.
IEEE
Acronym for Institute of Electrical and Electronics Engineers. Professional organization whose activities include the development of
communications and network standards. IEEE LAN standards are the predominant LAN standards today.
IETF
Acronym for Internet Engineering Task Force. Task force consisting of over 80 working groups responsible for developing Internet
standards. The IETF operates under the auspices of ISOC.
IMAP
Acronym for Internet Message Access Protocol. Method of accessing e-mail or bulletin board messages kept on a mail server that can
be shared. IMAP permits client electronic mail applications to access remote message stores as if they were local without actually
transferring the message.
information (i) button
On a Cisco IP Phone, provides online help for selected keys or features and network statistics about the active call.
inter-LATA
Services, traffic or facilities that originate in one local access and transport area (LATA), crossing over and terminating in another
LATA, both interstate and intrastate. See also LATA.
internal extension
See DN.
international-access
A two-digit code necessary for international dialing. For calls originating in the U.S., the international-access code is 01. See also
route filter tags.
Internet address
See IP address.
Intra-LATA
Services, traffic or facilities that originate and terminate in the same LATA, both interstate and intrastate. See also LATA.
IP
Acronym for Internet Protocol. Messaging protocol that addresses and sends packets across the network in the TCP/IP stack, offering
a connectionless internetwork service. To communicate using IP, network devices must have an IP address, subnet, and gateway
assigned to them. IP provides features for addressing, type-of-service specification, fragmentation and reassembly, and security.
Standardized in RFC 791.
IP address
Internet protocol address. A 32-bit address assigned to hosts using TCP/IP. An IP address belongs to one of five classes (A, B, C, D, or
E) and is written as 4 octets separated by periods (dotted decimal format). Each address consists of a network number, an optional
subnetwork number, and a host number. The network and subnetwork numbers together are used for routing, while the host
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number is used to address an individual host within the network or subnetwork. A subnet mask is used to extract network and
subnetwork information from the IP address. Also known as an Internet address. See also subnet mask.
IP phone
IP telephone. A phone that transports voice over a network using data packets instead of circuit switched connections over voice
only networks. Full-featured IP phones can be plugged directly into an IP network and used very much like a standard private branch
exchange (PBX) telephone.
IPv6
IP version 6. Replacement for the current version of IP (version 4). IPv6 includes support for flow ID in the packet header, which can
be used to identify flows. Formerly called IPng (next generation). See also RSVP.
IPX
Acronym for Internetwork Packet Exchange. NetWare network layer (Layer 3) protocol used for transferring data from servers to
workstations. IPX is similar to IP and XNS.
ISDN
Acronym for Integrated Services Digital Network. Communication protocol, offered by telephone companies, that permits telephone
networks to carry data, voice, and other source traffic. See also B-ISDN, BRI, N-ISDN, and PRI.
IS-IS
Acronym for Intermediate System to Intermediate System Protocol. A standards-based routing protocol used mainly in large ISP
networks.
ISO
Acronym for International Organization for Standardization. International organization that is responsible for a wide range of
standards, including those relevant to networking. ISO developed the OSI reference model, a popular networking reference model.
See also OSI.
ISP
Acronym for Internet service provider. Company that provides Internet access to other companies and individuals.
ITU
Acronym for International Telecommunication Union. The telecommunications agency of the United Nations established to provide
worldwide standard communications practices and procedures. Formerly known as the Comite Consultatif Internationale de
Telegraphique et Telephonique (CCITT).
ITU-T
Acronym for Telecommunication standardization sector of ITU. International body that develops worldwide standards for
telecommunications technologies. See also ITU.
IVR
Acronym for interactive voice response. Term used to describe 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. Also known as interactive voice
response.
IXC
Acronym for Inter exchange carrier. Also known as IEC and IC. Long-haul long distance carriers, including all facilities-based interLATA carriers. The term generally applies to voice and data carriers, but not to Internet carriers. Although large IXCs can provide
intraLATA toll service and may also operate as competitive local exchange carriers in several states, an IXC is understood to be in
contrast to a LEC (local exchange carrier) in terms of scope and service.
Java
Programming language from Sun Microsystems designed primarily for writing software to leave on World Wide Web sites.
Downloadable over the Internet to a PC. It has the ability to bring motion to static Web pages.
jitter
A type of distortion caused by the variation of a signal from its reference that can cause data transmission errors, particularly at high
speeds.
JTAPI
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Acronym for Java Telephony Application Programming Interface. See also API and TAPI.
keepalive message
A message sent by one network device to another that the circuit between the two is still active.
keypad template
See phone button template.
kill message
A message played at the beginning of a call to a 900 or other pay-per-call number that warns callers of the charges and allows the
caller hang up before costs are incurred.
KTS
Acronym for key telephone system. A small telephone system in which the telephones have multiple buttons requiring the user to
directly select central office phone lines and intercom lines. Key telephone systems are similar to PBX systems, but differ in that they
do not provide their own switching capabilities, routing and trunking capabilities, dial plans, or feature sets. Most key telephone
systems support from 10 to 50 telephones.
LAN
Acronym for Local-area network. High-speed, low-error data network covering a relatively small geographic area (up to a few
thousand meters). LANs connect workstations, peripherals, terminals, and other devices in a single building or other geographically
limited area. LAN standards specify cabling and signaling at the physical and data link layers of the OSI model. Ethernet, FDDI, and
Token Ring are widely used LAN technologies. Compare with MAN, VLAN and WAN.
LATA
Acronym for local access and transport area.
LBR
Acronym for low bit rate.
LDAP
Acronym for Lightweight Directory Access Protocol. Emerging standard offered as an Internet solution to the intricacies of DAP for
disparate legacy e-mail directories, network operating system directories, and databases.
LDIF
Acronym for LDAP Interchange Format.
LFI
Acronym for ink fragmentation and interleaving.
Lightweight Directory Access Protocol
See LDAP.
line
Any communications path between two or more points, including satellite or microwave channels.
line button
A button on a telephone set that is used to access the associated line. On Cisco IP Phones, a button you press to access a new line.
line conditioning
The adjustment and control of the properties of a leased line to bring its characteristics within specified tariff limits. Line conditioning
generally improves the frequency response and delay characteristics of the line.
line driver
A device to amplify signals and reshape distorted pulses. Also, an alternative to a modem when transmitting via cable over short
distances (up to several hundred feet).
line loading
The use of electrical components to improve the response characteristics of a communications line.
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line side
A connection that extends from an end office (EO), central office (CO), or private branch exchange (PBX) to the subscriber's
telephone or extension.
LLQ
Acronym for low latency queuing.
local dial peer
A software object that ties together a voice port and the telephone number of a device attached to the port. Also called POTS dial
peer.
local loop
The communication line between a telephone subscriber and the local exchange carrier (LEC) switching center. Also, a local
connection between an end user and a central office (CO) or end office (EO).
locations
A feature that regulates voice quality by limiting bandwidth availability over shared links. Also, for example, Cisco CallManager uses
locations in conjunction with a single, primary Cisco CallManager in a centralized (non-distributed) call processing system that
includes multiple remote devices, such as phones or gateways. Under this scheme, locations are created with a geographical
correspondence, such as a branch office. A maximum bandwidth to be used by inter-location voice calls is then specified for the
location and devices within that location are designated as belonging to that location. See also distributed call processing and CAC.
logical connection
A connection between two or more end points in which no contiguous, physical connection exists. The opposite of a physical
connection.
loop
A closed circuit. Also, a single connection from a switching center to an individual communications device.
loop back
A method of performing transmission tests on a circuit that does not require the assistance of personnel at the far end.
loop start
A method of calling a central office (CO) by applying a closed direct current loop across the line.
loop-start trunk
A two-wire central-office trunk or dial-tone link that recognizes an off-hook situation by putting a 1000-ohm short across the tip and
ring leads when the handset is lifted. The most common type of line, also called a POTS line. See also off-hook.
MAC
Acronym for media access control. Lower of the two sublayers of the data link layer defined by the IEEE. The MAC sublayer handles
access to shared media. See also MAC address.
MAC address
Standardized data link layer address that is required for every port or device that connects to a LAN. Other devices in the network
use these addresses to locate specific ports in the network and to create and update routing tables and data structures. MAC
addresses are 6 bytes long and are controlled by the IEEE. Also known as a hardware address, MAC-layer address, and physical
address. Compare with network address.
MAN
Acronym for metropolitan-area network. Network that spans a metropolitan area. Generally, a MAN spans a larger geographic area
than a LAN, but a smaller geographic area than a WAN. Compare with LAN and WAN.
MAPI
Acronym for Messaging Application Programming Interface. A system built into Microsoft Windows that enables different e-mail
applications to work together to distribute mail. As long as both applications are MAPI-compliant, they can share mail messages with
each other.
mapping
The logical association of one set of values (e.g., addresses in one network) with other quantities or values (e.g., devices on a second
network).
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MCS
Acronym for Media Convergence Server. Refers to the Cisco MCS-7800 series server family, which includes the Cisco AVVID IP
telephony starter kits. Comes with Cisco CallManager preloaded.
MCU
Acronym for multipoint control unit. The combination of a multipoint controller and a multipoint processor.
MD5
Acronym for Message Digest 5. A cryptographic algorithm that can be used to securely verify who sent a specific packet.
Media Access Control
See MAC.
media stream
The information content carried on a call. Refers to what is actually transmitted and received over the line, and can be read or
written by a media stream API.
media termination point
See MTP.
Meet-Me Conference
A Cisco CallManager feature that allows users to dial in to specific conference directory number. Requires allocation of exclusive
range of directory numbers. When a Meet-Me conference is set up, the conference controller selects a directory number and
advertises it to members of the group. The users call the directory number to join the conference. Anyone who calls the directory
number while the conference is active, joins the conference. Compare to Ad-Hoc conference.
message layer
Application layer (Layer 7 of the OSI model). A logical grouping of information, often composed of a number of lower-layer logical
groupings such as packets. The terms datagram, frame, packet, and segment are also used to describe logical information groupings
at various layers of the OSI reference model and in various technology circles.
message waiting indicator
See MWI.
MGCP
Acronym for Media Gateway Control Protocol. Enables external control and management of data communications equipment
operating at the edge of multi-service packet networks (known as media gateways) by software programs, which are known as "call
agents" or "media gateway controllers."
MIB
Acronym for Management Information Base. Database of network management information that is used and maintained by a
network management protocol such as SNMP or CMIP. The value of a MIB object can be changed or retrieved using SNMP or CMIP
commands, usually through a GUI network management system. MIB objects are organized in a tree structure that includes public
(standard) and private (proprietary) branches.
Microsoft NetMeeting
A virtual meeting application from Microsoft. NetMeeting allows you to share applications and a virtual whiteboard, transfer files,
and chat with other NetMeeting users through real-time, point-to-point audio conferencing over the Internet or corporate intranet.
MIME
Acronym for Multipurpose Internet Mail Extension. The standard format for including non-text information in Internet mail, thereby
supporting a transmission of mixed-media messages across TCP/IP networks.
MLPPP
Acronym for Multilink Point-to-Point Protocol. Method of splitting, recombining, and sequencing datagrams across multiple, logical
data links.
modem
Acronym for modulator-demodulator. Device that converts digital and analog signals. At the source, a modem converts digital signals
to a form suitable for transmission over analog communication facilities. At the destination, the analog signals are returned to their
digital form. Modems allow data to be transmitted over voice-grade telephone lines.
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MPPC
Acronym for Microsoft Point-to-Point Compression (PPC) compression algorithm, used to exchange compressed information with a
Microsoft NT remote access server.
MTP
Acronym for media termination point. A virtual device that allows transfer, forward, conference, and hold features on any G.711 law call between an IP Phone and any H.323 gateway, gatekeeper, or client. A call using MTP will automatically convert A-law to law (and vice versa), if required. As a Cisco software application, MTP installs on a server during the software installation process.
Also, acronym for Message Transfer Part. Part of SS7 protocol that provides functions for basic routing of signaling messages
between signaling points.
multicast
Single packets copied by the network and sent to a specific subset of network addresses. A process of transmitting messages from
one source to many destinations. Compare with broadcast and unicast.
multicast address
Single address that refers to multiple network devices. Synonymous with group address. Compare with broadcast address and
unicast address. See also multicast.
multilink PPP
See MLPPP.
multipoint control unit
See MCU.
multipoint controller
The component of a conferencing engine that manages the participants' access into a conference and the multipoint processors.
multipoint processor
The component of a conferencing engine that redistributes the groups shared media to other participants outside the group.
multipoint-unicast
A process of transferring PDUs (Protocol Data Units) where an end point sends more than one copy of a media stream to different
end points. This may be necessary in networks which do not support multicast.
MWI
Acronym for message waiting indicator. Method of indicating that a voice mail message was left for a particular directory number.
For example, Cisco IP Phones indicate this by lighting an LED on the handset. The Cisco 7630 Digital PBX Adapter works with Cisco
CallManager, Octel, and Lucent systems to ensure that the MWI is set properly.
name server
A user directory in a local or wide area network.
NANP
Acronym for North American Numbering Plan. The North American Numbering Plan (NANP) was invented in 1947 by AT&T and Bell
Laboratories. It conforms to the International Telecommunications Union Recommendation E.164, the international standard for
numbering plans. The NANP is the numbering plan for the Public Switched Telephone Network (PSTN) in the United States and its
territories, Canada, Bermuda, and many Caribbean nations. NANP numbers are 10 digits in length, and they are in the format: NXXNXX-XXXX, where N is any digit 2-9 and X is any digit 0-9. The first three digits are called the numbering plan area (NPA) code, often
called simply the area code. The second three digits are called the central office code or prefix. The final four digits are called the line
number.
NAT
Acronym for network address translation. Changing the IP address of a packet in transit to allow an enterprise to appear to use fewer
addresses than are actually necessary.
network
Collection of computers, printers, routers, switches, and other devices that are able to communicate with each other over some
transmission medium. Examples include LANs and WANs.
network address
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Network layer address referring to a logical, rather than a physical, network device. Also called a protocol address. Compare with
MAC address.
network port
Connects an IP device to the network.
NFS
Acronym for Network File System. As commonly used, a distributed file system protocol suite developed by Sun Microsystems that
allows remote file access across a network. In actuality, NFS is simply one protocol in the suite. NFS protocols include NFS, RPC, XDR,
and others. These protocols are part of a larger architecture that Sun refers to as ONC.
NIC
Acronym for network interface card. Board that provides network communication capabilities to and from a computer system. Also
called an adapter. Also, acronym for network interface controller. An intelligent device that connects a workstation to a network.
N-ISDN
Acronym for Narrowband Integrated Services Digital Network. Communication standards developed by the ITU-T for baseband
networks. Based on 64-kbps B channels and 16- or 64-kbps D channels. See also BRI, ISDN, and PRI.
NMS
Acronym for network management system. System responsible for managing at least part of a network. Generally, a reasonably
powerful and well-equipped computer, such as an engineering workstation. NMSs communicate with agents to help keep track of
network statistics and resources. See also agent.
node
Computers on a network, or any endpoint of a network connection or a junction common to two or more lines in a network. Nodes
can be processors, controllers, or workstations. Nodes, which vary in routing and other functional capabilities, can be interconnected
by links, and serve as control points in the network. Sometimes used generically to refer to any entity that can access a network.
Used interchangeably with device. Also, H.323 entity that uses RAS to communicate with the gatekeeper (for example, an endpoint
such as a terminal, proxy, or gateway). Also, in SNA, the basic component of a network and the point at which one or more
functional units connect channels or data circuits. See also SNA.
North American Numbering Plan
See NANP.
NTP
Acronym for Network Time Protocol. Protocol that ensures that device clocks are set to the same time, relative to Greenwich Mean
Time.
NTP server
Used by network devices to synchronize date and time settings to ensure proper recording in log files. See also NTP.
object code
The output obtained by processing a source program through an assembler or compiler.
object program
A fully compiled software program that is ready to be loaded into a computer.
off-hook
A change in line voltage caused when the receiver or handset is lifted from the hookswitch. A traditional PBX or local telephone
company recognizes this line voltage change as a request for dial tone. Also, a call condition in which transmission facilities are
already in use. Also known as busy.
office code
The first three digits of your seven-digit local telephone number. See also route filter tags.
off-line
A device that is not permanently connected to a network.
on-hook
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The condition that exists when a receiver or handset is resting on the hookswitch. Also, the idle state (open loop) of a single
telephone or private branch exchange (PBX) line loop.
online
A signal that indicates a communications link has been established and transmission can begin.
on-screen mode keys
On a Cisco IP Phone, retrieves information about current settings, recent calls, available services, and voice mail messages.
Operational VLAN
A type of VLAN that is obtained through Cisco Discovery Protocol (CDP). Used to indicate the VLAN of which a Cisco IP Phone is a
member. Cannot be configured locally. Compare to Administrative VLAN.
OPX
Acronym for off premises extension. A peripheral private branch exchange (PBX) device located in a building other than the one
housing the PBX system itself. See also PBX.
OSI
Acronym for Open Systems Interconnection. The only internationally accepted framework of standards for communication between
different systems made by different vendors. Developed by the International Organization for Standardization, OSI is a model, not an
active protocol. OSI organizes the communication process into seven different categories and places these in a layered sequence
based on their relation to the user. The seven layers are: physical, data link, network, transport, session, presentation and
applications.
OSPF
Acronym for open shortest path first.
packet
Logical grouping of information that includes a header containing control information and (usually) user data. Packets are most often
used to refer to network layer units of data. The terms datagram, frame, message, and segment are also used to describe logical
information groupings at various layers of the OSI reference model and in various technology circles. See also PDU.
PAD
Acronym for packet assembler/disassembler.
partitions
Divides a route plan into subsets. Partitions include organization, location, and type of call.
password
A word or string of characters recognized by automatic means that permits a user access to a place or to protected storage, files, or
input/output devices.
patch
A small addition to the original software code, written to bypass or correct a problem, and also provided between software releases.
PBX
Acronym for private branch exchange. Digital or analog telephone switchboard located on the subscriber premises, typically with an
attendant console, and used to connect private and public telephone networks. A PBX is a small, privately owned version of the
phone company's larger central switching office. It is connected to one or more central offices by trunks, and provides service to a
number of individual phones, such as in a hotel, business, or government office. On a PBX, an outside line is normally accessed by
dialing an access digit, such as 9.
PCM
Acronym for pulse code modulation. Transmission of analog information in digital form through sampling and encoding the samples
with a fixed number of bits.
PCS
Acronym for Personal Communications Service. A lower-powered, higher-frequency competitive technology to cellular. Whereas
cellular typically operates in the 800-900 MHz range, PCS operates in the 1.5-1.8 Ghz range.
PDU
Acronym for protocol data unit. OSI term for packet. See also BPDU, OSI and packet.
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performance monitor
a Windows NT server application that displays NT server and CCN activities in real time.
PGP
Acronym for Pretty Good Privacy. Powerful public-key encryption application that allows secure file and message exchanges. There is
some controversy over the development and use of this application due, in part, to U.S. national security concerns.
phone button template
Defines which keys on a phone or IP device perform which functions. Use templates to customize individual IP phones and to assign
common button configurations to a large number of phones. Cisco CallManager includes several default phone button templates, all
of which can be modified.
phone loads
See device loads.
pilot point
Directory number that receives and forwards calls based on a list of hunt group members. In Cisco Call Manager, a directory number
necessary for call routing by the Cisco Telephony Call Dispatcher (TCD). See also hunt group.
PIN
Acronym for personal identification number. A multiple digit number, generally known only to the user, that allows access to
networks or other systems.
POP
Acronym for point of presence. The IXC equivalent of a local phone company's central office. In other words, a long distance carrier's
office in the local community (defined as the LATA). Also refers to the point of presence at which Internet service providers exchange
traffic and roots at Layer 2 (Link Layer) of the OSI model. Also,short for population. One pop equals one person. Also, acronym for
Post Office Protocol. An e-mail server protocol used in the Internet.
port
An input/output connection for a computer or for communications equipment.
POTS
Acronym for plain old telephone service. Standard telephone service used by most residential locations. For example, POTS line
connections are used to join a Cisco Analog Station gateway and an SMDI-compliant voice mail system. See PSTN, SMDI.
PPP
Acronym for Point-to-Point Protocol. A link-layer encapsulation method for dialup or dedicated circuits. Successor to SLIP that
provides router-to-router and host-to-network connections over synchronous and asynchronous circuits. Whereas SLIP was designed
to work with IP, PPP was designed to work with several network layer protocols, such as IP, IPX, and ARA.
PQ
Acronym for priority queueing.
PRI
Acronym for Primary Rate Interface. A type of ISDN service designed for large organizations. Includes B-channels (bearer channels)
for voice or data, and one D-channel (data channel). PRI comprises 23 B-channels in North America and 30 B-channels in Europe.
Compare with BRI. See also B-ISDN, ISDN, and N-ISDN.
PROM
Acronym for programmable read only memory. A type of nonvolatile memory that is electrically programmed by an equipment
manufacturer and can only be changed with special equipment that erases the previous program. Compare with EPROM and
EEPROM.
protocol
A set of rules or conventions that govern the format and relative timing of data in a communications network. There are three basic
types of protocols: character-oriented, byte-oriented, and bit-oriented. The protocols for data communications cover such things as
framing, error handling, transparency, and line control. Ethernet is an example of a LAN protocol.
proxy
A device that relays network connections for other devices that usually lack their own network access.
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PSTN
Acronym for Public Switched Telephone Network. General term referring to the variety of telephone networks and services in place
worldwide.
PTM
Acronym for point-to-multipoint. A network configuration that connects one point to multiple points on the network. A main source
to many destination connections. Also called PTMPT. Also,acronym for pulse time modulation. A type of modulation in which the
duration of the modulating pulse varies according to some characteristic of the original analog signal while the pulse amplitude
remains constant. More commonly known as pulse duration modulation.
Public Switched Telephone Network
See PSTN.
pulse code modulation
See PCM.
PVID
Acronym for port VLAN ID.
QoS
Acronym for quality of service. Measure of performance for a transmission system that reflects its transmission quality and service
availability.
query
A request from a master station asking a slave station to identify itself and indicate its status (e.g., busy, alive, waiting, etc...).
queue
A temporary delay in service caused by the inability of a particular system to handle the number of calls attempted. For example, a
call may be queued (essentially, waiting in line) for the least expensive route.
queuing
A technique in which incoming calls are stored on hold until an attendant, trunk, trunk group, or station is available to accept them.
Also known as camp on.
RADIUS
Acronym for Remote Authentication Dial In User Service. A standards-based protocol for AAA. See also AAA.
RAS
Acronym for Registration, Admission, and Status Protocol. Used in the H.323 protocol suite for discovering and interacting with a
gatekeeper. See also H.323 and gatekeeper.
Real-Time Transport
See RTP.
redial
A button on many modern phones used to redial the most recently dialed number.
redialer
Interface hardware device that interconnects between a fax device and a Public Switched Telephone Network (PSTN) network. A
redialer is used to forward a dialed number to another destination. Redialers contain a database of referral telephone numbers.
When the user dials a specific number, the redialer collects the dialed digits and matches them to a listing in its database. If there is a
match, the redialer dials the referral number (transparent to the user) and forwards the call to the referral number.
redundancy
Having one or more back up systems available in case of failure of the main system. Also, backup Cisco CallManagers that handle call
processing for a disabled Cisco CallManager within the same group. Also known as call processing redundancy. Also, backup copies of
a database shared by a cluster of Cisco CallManagers. Also known as database redundancy.
relay
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OSI terminology for a device that connects two or more networks or network systems. A data link layer (Layer 2) relay is a bridge; a
network layer (Layer 3) relay is a router. See also bridge and router.
relay server
A node on the public internet that is configured as a multiuser system. The relay server may be a dedicated device or, during a pilot
period, it may be attached on an ad hoc basis when the need arises. The relay server is located outside of the Cisco Systems firewall,
but it is a secure and controlled system. Although it is a node on the public internet, it is owned, managed and operated by Cisco
Systems.
repeater
Device that regenerates and propagates electrical signals between two network segments. Used by transmission systems to
regenerate analog or digital signals that were distorted by transmission loss. See also segment and amplifier.
ringback
The tone heard at the calling party's end when the called party's phone rings. Also, a signal used by an operator at the receiving end
of an established connection to recall an operator at the originating end.
ringdown
A signaling method in which the incoming signal is actuated by alternating current (AC) over the circuit.
RJ-45 port
The 8-pin connector used for data transmission over standard telephone wire. RJ-45 connectors come in two varieties: keyed and
non-keyed and accommodate flat or twisted wire.
route
The process of directing a call to the appropriate destination based on the dialed digits, translation patterns, transformation masks,
and other route plan considerations. Also, the process of directing a message to the appropriate line and terminal based on
information contained in the message header.
route filter
Allows or restricts access to specified routing patterns, such as 1+900, etc. Only applicable in conjunction with routing patterns that
use the North American Numbering Plan (NANP).
route filter tags
Applies a name to a subset of the dailed digit string. For example, the phone number 972-555-1234 contains three route filter tags:
the local-area-code (972), the office-code (555), and the subscriber (1234). Other route filter tags include the country code, end-ofdialing character, and international-access code.
route group
A route group determines the order of preference for gateway and port usage. All members of a route group must have the same
route pattern. Route groups are optional. For example, if two Cisco Access Digital Gateways accept only long distance calls and one
carrier is priced below the other, a route group could be created so that calls are first routed to the least expensive carrier. In this
case, calls would route to the more expensive carrier only if the first trunk is unavailable.
route list
Determines the order of preference for route group usage. If a route list is configured, at least one route group must be configured.
See also route group.
route pattern
Route patterns range from the very simple to the very complex. For example, a routing pattern of "0" assigned to a gateway would
route all calls to the operator through that gateway. Route patterns are used by Cisco CallManager Administration to route inbound
and outbound calls. For a Cisco IP Phone, the assigned directory number. Cisco Access Analog Trunk Gateways, Cisco Access Digital
Trunk Gateways, and H.323-compliant gateways also use route patterns.
route plan report
In Cisco CallManager, a listing of all call park, call pickup and conference numbers, plus route patterns and translation patterns in the
system.
router
An interface device between two networks that selects the best route even if there are several networks between the originating
network and the destination. Also, a device that provides network management capabilities (e.g., load balancing, network
partitioning, usage statistics, communications priority and troubleshooting tools) that allow network managers to detect and correct
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problems. Also, an intelligent device that forwards data packets from one local area network (LAN) to another and that selects the
most expedient route based on traffic load, line speeds, costs, or network failures.
routing
The process of finding a path to the destination host. Routing can be very complex in large networks because of the many potential
intermediate destinations a packet might traverse before reaching its destination host.
routing bridge
A device that uses network layer methods to determine a network's topology.
routing table
A database stored in a router or other internetworking device that keeps track of routes (and in some cases, the metrics associated
with those routes) to particular network destinations.
RSVP
Acronym for 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 forth) of
the packet streams they want to receive. RSVP depends on IPv6. Also known as Resource Reservation Setup Protocol. See also IPv6.
RTCP
Acronym for Real-Time Control Protocol. Monitors the QoS of an IPv6 RTP connection and conveys information about the on-going
session. See also RTP, IPv6 and QoS.
RTP
Acronym for Real-Time Transport Protocol. A network protocol used to carry packetized audio and video traffic over an IP network.
SA/DA
Acronym for sending address/destination address.
scalable
Indicates that a software application or a hardware device has the ability to migrate from small operations to large operations.
segment
Section of a network that is bounded by bridges, routers, or switches. Also, in a local area network (LAN) that uses a bus topology, a
continuous electrical circuit that is often connected to other such segments with repeaters. Also, term used in the TCP specification
to describe a single transport layer unit of information. The terms datagram, frame, message, and packet are also used to describe
logical information groupings at various layers of the OSI reference model and in various technology circles.
server
Node or software program that provides services to clients. See also back end, client, and FRF.11. Also, in network addressing, a
concentrator, data switch, or host computer being accessed. Also, in a synchronous packet assembler/disassembler (PAD), a device
that assigns remote devices to a logical multipoint host line.
signal transfer point
See STP.
Simple Network Management Protocol
See SNMP.
Skinny Station Protocol
See SSP.
SLIP
Acronym for Serial Line Internet Protocol. Standard protocol for point-to-point serial connections using a variation of TCP/IP.
Predecessor of PPP.
SMDI
Acronym for simplified message desk interface. Analog data line from the central office containing information and instructions to
your on-premises voice mail box. A required interface for voice mail systems used with Cisco CallManager. SMDI was designed to
enable voice mail integration services to multiple clients. However, to use SMDI, the voice mail system must meet several
qualifications, including providing database support for two PBX systems simultaneously and IP network connectivity to the voice
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messaging system while maintaining the existing link to the PBX. SMDI-compliant voice mail systems must be accessible with a nullmodem RS-232 cable and available serial port.
SMDS
Acronym for Switched Multimegabit Data Service. A connectionless high-speed data transmission service intended for application in
a Metropolitan Area Network (MAN) environment. A public network service designed primarily for LAN-to-LAN interconnection. See
also cell relay.
SMTP
Acronym for Simple Mail Transfer Protocol. Internet protocol providing e-mail services.
SNA
Acronym for 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 is essentially composed of seven layers.
SNMP
Acronym for Simple Network Management Protocol. The protocol governing network management and monitoring of network
devices and their functions.
soft keys
On a Cisco IP Phone, buttons that activates features described by a text message. The text message is displayed directly above the
soft key button on the LCD screen.
SoftPhone
Application that enables you to use a desktop PC to place and receive software telephone calls and to control an IP telephone. Also
allows for audio, video, and desktop collaboration with multiple parties on a call. Cisco IP SoftPhone can be used as a standalone
application or as a computer telephony integration (CTI) control device for a physical Cisco IP phone. All features are functional in
both modes of operation. See also Cisco IP phone.
SPAN
Acronym for Switched Port Analyzer. Feature of the Cisco Catalyst 5000 switch that extends the monitoring abilities of existing
network analyzers into a switched Ethernet environment. SPAN mirrors the traffic at one switched segment onto a predefined SPAN
port. A network analyzer attached to the SPAN port can monitor traffic from any of the other Catalyst switched ports.
speakerphone
A telephone equipped with a speaker and a microphone that allows hands-free conversation.
speed dial number
A one to four-digit number that replaces a seven- or ten-digit number for speed dialing.
speed dialing
A system that allows a telephone user to reach frequently called numbers by dialing less than seven digits. Also known as
Abbreviated Dialing.
SRB
Acronym for source-route bridging. Method of bridging originated by IBM and popular in Token Ring networks. In an SRB network,
the entire route to a destination is predetermined, in real time, prior to the sending of data to the destination.
SS7
Acronym for Signaling System 7. A telephone signaling system with three basic functions: supervising (monitoring the status of a line
or circuit to see if it is busy, idle, or requesting service); alerting (indicating the arrival of an incoming call); addressing (transmission
of routing and destination signals over the network). See also CCIS.
SSP
Acronym for Skinny Station Protocol. A Cisco protocol using low bandwidth messages that communicate between IP devices and the
Cisco CallManager.
STP
Acronym for shielded twisted-pair. Two-pair wiring medium used in a variety of network implementations. STP cabling has a layer of
shielded insulation to reduce EMI. Also, acronym for Spanning Tree Protocol. Inactivation of links between networks so that
information packets are channeled along one route and will not search endlessly for a destination. Also, acronym for signal transfer
point. The packet switch in the Common Channel Interoffice Signaling (CCIS) system. See CCIS and SS7.
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subnet mask
A 32-bit address mask used in IP to indicate the bits of an IP address that are being used for the subnet address. See also IP address.
switch
Network device that filters, forwards, and floods pieces of a message (packets) based on the destination address of each frame.
Switches operate at the data link layer of the OSI model. See also OSI.
SYN attack
A particular type of DoS attack that exploits a common flaw in host TCP implementations. See also TCP, DoS.
T1
Trunk Level 1. Digital transmission link with a total signaling speed of 1.544 Mbps. Transmits through the telephone-switching
network using AMI or B8ZS coding. The standard in North America, a T1 device combines the output of up to 24 regular telephone
lines for transmission over a digital network. Also known as T-1.
TACACS+
Acronym for Terminal Access Controller Access Control System Plus. A Cisco proprietary protocol for authentication, authorization,
and accounting. See also AAA.
TAPI
Acronym for Telephony Application Programming Interface TCP/IP. A set of functions that allow Windows applications to program
telephone line-based devices such as single and multi-line phones (including Cisco IP Phones), modems, and fax machines in a
device-independent manner. See also JTAPI.
TCAP
Acronym for Transaction Capabilities Application Part. An ISDN application protocol that provides the signaling function for network
data bases.
TCD
Acronym for Cisco Telephony Call Dispatcher. A Cisco CallManager service that handles requests by the Cisco WebAttendant for call
control, call dispatching, line status, and user directory information.
TCP
Acronym for Transmission Control Protocol. Connection-oriented transport layer protocol that provides reliable full-duplex data
transmission. TCP is part of the TCP/IP protocol stack. See also TCP/IP.
TCP/IP
Acronym for Transmission Control Protocol/Internet Protocol. The two best-known internet protocols, often erroneously thought of
as one protocol. The transmission control protocol (TCP), which corresponds to Layer 4 (the transport layer) of the open systems
interconnection (OSI) reference model, provides reliable transmission of data. The internet protocol (IP) corresponds to Layer 3 (the
network layer) of the OSI model and provides connectionless datagram service. TCP/IP was developed by the U.S. Department of
Defense in the 1970s to support the construction of worldwide internetworks. See also ARPA, IP and TCAP.
TDM
Acronym for time division multiplexing. A technique for transmitting a number of separate data, voice, and video signals
simultaneously over one communications medium by quickly interleaving a piece of each signal one after the other.
telco connector
In LAN terms, a 25-pair polarized connector used to consolidate multiple voice or data lines.
teleconference
A system in which three or more people can be connected by telephone and maintain a continuous connection and conversation.
See also Ad-Hoc conference and Meet-Me conference.
telephone
A device that converts acoustic energy into electrical energy for transmission to a distant point.
telephony
Science of converting sound to electrical signals and transmitting it between widely removed points.
Telephony Call Dispatcher
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Study Notes
See TCD.
Telnet
A program that lets you connect to other computers on the Internet. The terminal-remote host protocol developed for ARPA that
allows you to work from your PC as if it were a terminal attached to another machine by a hardwire line. See ARPA.
Telnet proxy
Provides a connection from the customer side to the external application residing on the relay server. When initiating this connector
program, the customer is required to specify certain command line parameters, while others are optional.
Telnet proxy program tndconnect
Links the Telnet server at the customer site to the relay server. When started by the customer, it initiates a "Telnet tunnel",
establishing a TCP connection from inside the customer firewall out to the relay server on the public internet. Then it establishes
another connection to the local Telnet server, creating a two-way link between the entities.
TFTP
Acronym for Trivial File Transfer Protocol. Builds and serves files consistent with the TFTP protocol for transfer over the network. A
simplified version of the FTP protocol, TFTP requires a TFTP server in your network, which can be automatically identified from the
DHCP server. Cisco TFTP serves both Embedded Component Executable and configuration (.cnf) files. See also DHCP.
third party call control
If an audio stream terminates at some location or physical device other than your application or device, you have third party call
control. For example, the Cisco SoftPhone can control the Cisco IP Phones. Used in TAPI development.
toll bypass
A toll-free telephony call in which the relative locations of the two ends of the connection would cause toll charges to be applied if
the call was made over the PSTN.
ToS
Acronym for type of service.
trace log
Collects and stores trace information, according to specifications set in the trace log components.
traffic
The load on a communications device or system.
transcoder
A transcoder is a device that takes the output stream of one codec and transcodes (converts) it from one compression type to
another compression type. A transcoder also provides MTP capabilities. See also codec.
transit network
A three or four-digit number that identifies a long distance carrier.
Trivial File Transfer Protocol
See TFTP.
trunk
Physical and logical connection between two switches across which network traffic travels. A trunk is a voice and data path that
simultaneously handles multiple voice and data connections between switches. A backbone is composed of a number of trunks. See
also CO.
trunk group
A group of essentially alike trunks (shared electronic characteristics) that go between the same two geographical points. A trunk
group performs the same function as a single trunk, but carries multiple conversations.
TSP
Acronym for Telecommunications Service Priority. The regulatory, administrative, and operational system that authorizes and
provides priority treatment for initiating and restoring telecommunication services.
UDP
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Acronym for User Datagram Protocol. A connectionless messaging protocol for delivery of data packets. A simple protocol that
exchanges datagrams without acknowledgements or guaranteed delivery, requiring that error processing and retransmission be
handled by other protocols.
unattended operation
A transmission that is controlled automatically and does not require operator intervention.
unicast
A process of transmitting messages from one source to one destination. Compare with broadcast and multicast.
unicast address
Address specifying a single network device. See also unicast.
unicode
Standard 16-bit system for encoding letters and characters of all the world's languages. In contrast, ASCII uses 8 bits to represent a
character.
UPS
Acronym for uninterruptible power supply. A continuous on-line UPS is one in which the load is continually drawing power through
the batteries, battery charger, and inverter, and not directly from the AC supply.
User Datagram Protocol
See UDP.
user mask
A series of flags, or bits, that enable and disable specific types of trace information.
VAC
Acronym for voice activity compression. A method of conserving transmission capacity by not transmitting pauses in speech.
VAD
Acronym for voice activity detection. When enabled on 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.
VIC
Acronym for voice interface card.
virtual connection
A communications channel between two stations in which information or data transmitted by one station is automatically routed
through the network via the most expeditious path to the other station. No long-haul circuit capacity is preassigned to a virtual
connection, but capacity is made available as data is transmitted by the stations.
VLAN
Acronym for virtual LAN. Group of devices on one or more LANs that are configured (using management software) so that they can
communicate as if they were attached to the same wire, when in fact they are located on a number of different LAN segments.
Because VLANs are based on logical instead of physical connections, they are extremely flexible. See also LAN.
VoFR
Acronym for Voice Over Frame Relay. Enables a router to carry voice traffic (for example, telephone calls and faxes) over a Frame
Relay network.When sending voice traffic over Frame Relay, the voice traffic is segmented and encapsulated for transit across the
Frame Relay network using FRF.12 encapsulation.
voice compression
See compression types.
voice mail (or messaging) system
A device to record, store, and retrieve voice messages. A stand alone system is similar to a collection of individual answering
machines; an integrated version provides a higher level of call processing services and features.
Voice Over Frame Relay
See VoFR.
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voiceband
A transmission service with a bandwidth considered suitable for transmission of audio signals. Generally 300 or 500 (hertz) to 3,400
(hertz).
VoIP
Acronym for Voice over IP. Enables users to transfer voice communications over a data network using the Internet Protocol (IP).
WAN
Acronym for wide-area network. Data communications network that serves users across a broad geographic area and often uses
transmission devices provided by common carriers. Frame Relay, SMDS, and X.25 are examples of WANs. Compare with LAN and
MAN.
WATS
Acronym for wide area telecommunications service. A discounted toll service provided by long distance and local phone companies
in which the owner of the WATS line is charged a flat-rate monthly fee for long distance services.
wave devices
Any device that uses a wave driver, such as a voice modem.
wave driver
Software that plays a proprietary file by Microsoft Windows called a Wave File. Wave files are often used to encode music, rather
than voice.
Web interface
A software application that runs on the World Wide Web, and is usually accessed by entering an address starting with www. The
Cisco CallManager Administration uses a Web interface.
WebAttendant
Client-server application that provides attendant console capabilities and uses a web-based GUI interface to answer and handle
inbound and outbound calls that are not serviced by direct inward dialing (DID).
WFQ
Acronym for weighted fair queuing. A variation on the class-based queuing (CBQ) technique used in routers. Like CBQ, WFQ divides
queues traffic according to traffic class definition, guaranteeing each queue some portion of the total available bandwidth. WFQ
goes further to portion out available bandwidth on the basis of individual information flows according to message parameters. See
also CBQ.
whiteboarding
A form of data collaboration that recreates the effect of sharing a common drawing surface viewable by all participants.
wide area telecommunications service
See WATS.
wink start
A short off-hook signal.
WOSA
Acronym for Windows Open Service Architecture. Microsoft's single system level interface for connecting front-end applications with
back-end services. Windows Telephony is part of WOSA.
WRED
Acronym for weighted random early detection. A congestion-avoidance and QoS mechanism for IP-based networks.
WRR
Acronym for weighted round-robin.
XNS
Acronym for Xerox Network System. A five-layer architecture of protocols that served as the foundation of the OSI seven-layer
model.
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-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.
100BASE-T
A 100-Mbps Ethernet specification defined by IEEE 802.3 that uses Category 3 or Category 5 twisted pair wiring. Designed to
integrate with existing networks with minimal disruption. Generically called Fast Ethernet.
10BaseT
A 10-Mbps Ethernet specification defined by IEEE 802.3 that uses Category 3 or Category 5 twisted pair wiring.
802.1 P
Networking protocol and IEEE specification for the prioritization of traffic.
802.1 Q
Networking protocol and IEEE specification for the implementation of VLANs in Layer 2 LAN switches.
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Study Notes
G. Answers to Quizzes
Answer
The IETF protocol used by SIP for call routing is TRIP.

Option 1 is incorrect. BGP is an Inter-domain routing protocol and is used as part of the SIP process for call routing.
Option 2 is incorrect. OSPF is an internal gateway protocol and also a link-state routing protocol and is not used as
part of the SIP process for call routing.
Option 3 is incorrect. RIP is a classful routing protocol that is also known as a distance vector routing protocol. It is
not used as part of the SIP process for call routing.
Option 4 is correct. SIP uses IETF protocols to define aspects of VoIP and multimedia sessions; for example, URLs
for addressing, Domain Name System (DNS) for service location, and TRIP for call routing.
ii
Answer
User capabilities services are used to select the media type and parameters for multimedia sessions.
Option 1 is incorrect. SIP is a signaling and control protocol for establishing, maintaining, and terminating
multimedia sessions with one or more participants. Call handling services are used to transfer and terminate calls.
Option 2 is incorrect. SIP operates on the basis of session invitations. Through invitations, SIP initiates sessions or
invites participants into established sessions. Call setup services are used to establish a session relationship
between parties and manage call progress.
Option 3 is incorrect. SIP uses invitations to initiate sessions or invite participants into established sessions. User
availability services are used to determine the availability and desire for a party to participate.
Option 4 is correct. SIP is a signaling and control protocol for the establishment, maintenance, and termination of
multimedia sessions with one or more participants. User capabilities services are used to select the media type and
parameters for the sessions.
Option 5 is incorrect. SIP creates, modifies, and terminates multimedia sessions with one or more participants.
User location services are used to locate an end system.
iii
Answer
Location, proxy, redirect and registrar are SIP server types.

Option 1 is incorrect. A DHCP server is used to assign IP addresses to hosts on a network. It is not considered to be
one of the SIP servers.
Option 2 is incorrect. A gateway acts as a UAS or UAC and provides call control support. Gateways provide many
services, the most common being a translation function between SIP UAs and other terminal types. This function
includes translation between transmission formats and between communications procedures.
Option 3 is correct. A location server is an abstraction of a service providing address resolution services to SIP
proxy or redirect servers. A location server embodies mechanisms to resolve addresses.
Option 4 is correct. A proxy server is an intermediate component that receives SIP requests from a client, then
forwards the requests on behalf of the client to the next SIP server in the network. The next server can be another
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Study Notes
proxy server or a UAS. Proxy servers can provide functions such as authentication, authorization, network access
control, routing, reliable request transmissions, and security.
Option 5 is correct. A redirect server provides a UA with information about the next server that the UA should
contact. The server can be another network server or a UA. The UA redirects the invitation to the server identified
by the redirect server.
Option 6 is correct. A registrar server makes requests from UACs for registration of their current location. Registrar
servers are often located near or even colocated with other network servers, most often a location server.
iv
Answer
A registrar server is often colocated with the location server.

Option 1 is incorrect. A gateway acts as a UAS or UAC and provides call control support. Gateways provide many
services, the most common being a translation function between SIP UAs and other terminal types. This function
includes translation between transmission formats and between communications procedures.
Option 2 is incorrect. A proxy server is an intermediate component that receives SIP requests from a client, and
then forwards the requests on behalf of the client to the next SIP server in the network. The next server can be
another proxy server or a UAS. Proxy servers can provide functions such as authentication, authorization, network
access control, routing, reliable request transmissions, and security.
Option 3 is incorrect. A redirect server provides a UA with information about the next server that the UA should
contact. The server can be another network server or a UA. The UA redirects the invitation to the server identified
by the redirect server.
Option 4 is correct. A registrar server makes requests from UACs for registration of their current location. Registrar
servers are often located near or even collocated with other network servers, most often a location server.
Answer
The REGISTER SIP message is used to provide information to a network server.

Option 1 is incorrect. A client originates the ACK message to indicate that the client has received a response to its
earlier invitation.
Option 2 is incorrect. A client originates the INVITE message to indicate that the server is invited to participate in a
session. An invitation includes a description of the session parameters.
Option 3 is incorrect. A client uses the OPTIONS message to solicit capabilities information from a server. This
method is used to confirm cached information about a UA or to check the ability of a UA to message accept an
incoming call.
Option 4 is correct. A UA uses the REGISTER message to provide information to a network server. Registrations
have a finite life and must be renewed periodically. This prevents the use of stale information when a UA moves.
vi
Answer
1xx (informational) is a provisional SIP response message.

Option 1 is correct. 1xx (informational) is a provisional response. It indicates that the request is still being
processed.
Option 2 is incorrect. 2xx (successful) indicates that the requested action is complete and successful.
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Option 3 is incorrect. 3xx (redirection) indicates that the requestor requires further action for example, a redirect
server responds with "moved" to advise the client to redirect its invitation.
Option 4 is incorrect. 4xx (client error) is a fatal response. It indicates that the client request is flawed or
impossible to complete.
Option 5 is incorrect. 5xx (server error) is a fatal response that indicates that the request is valid but the server
failed to complete it.
Option 6 is incorrect. 6xx (global failure) is a fatal response. It indicates that the request cannot be fulfilled by any
server.
vii
Answer
A SIP UA can resolve an address by letting the network server resolve it, using a local host table, or using rwhois.
Option 1 is incorrect. A DHCP server is used to assign IP addresses to hosts on a network. It is not considered to be
one of the SIP servers. It cannot be used by a SIP UA to resolve an address.
Option 2 is correct. To resolve an address, a UA uses a variety of internal mechanisms, one of which is to leave
that responsibility to a network server. The network server uses any of the tools available to a UA or interacts
through a nonstandard interface with a location server.
Option 3 is incorrect. SIP AU never uses WINS to resolve an address. To resolve an address, a UA uses a variety of
internal mechanisms such as a local host table, DNS lookup, Finger protocol, rwhois, or LDAP, or it leaves that
responsibility to a network server.
Option 4 is correct. To resolve an address, a UA uses a variety of internal mechanisms one of which is a local host
table. It can also use DNS lookup, Finger protocol, rwhois, LDAP, or leave that responsibility to a network server.
Option 5 is correct. To resolve an address, a UA uses a variety of internal mechanisms one of which is rwhois. It
can also use a local host table, DNS lookup, Finger protocol, LDAP, or leave that responsibility to a network server.
viii
Answer
The SIP address "sip:12486593178@gateway.com;user=phone" is an E.164 address.

Option 1 is correct. SIP addresses that contain "@gateway.com" are always E.164 addresses. An address consists
of an optional user ID, a host description, and optional parameters to qualify the address more precisely. The host
description may be a domain name or an IP address.
Option 2 is incorrect. Fully qualified domain names look something like this: "sip:jdoe@cisco.com".
Moreover, they do not have a user optional ID to further qualify the address more precisely.
Option 3 is incorrect. A mixed address contains information about the password and the host description; for
example, "sip:15085551234; password=changeme@10.1.1.1 sip:jdoe@10.1.1.1".
ix
Answer
The disadvantage of using the direct call setup method is that it relies on cached information that may be out of
date.
Option 1 is incorrect. Direct setup does not need to learn the coordinates of the destination UA. When a UA
recognizes the address of a terminating endpoint from cached information, or has the capacity to resolve it by
some internal mechanism, the UAC may initiate direct (UAC-to-UAS) call setup procedures.
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Study Notes
Option 2 is incorrect. Direct setup does not use a network server. Direct setup is the fastest and most efficient of
the call setup procedures. After the three step call setup has finished, the UAC and UAS have all the information
that is required to establish Real-Time Transport Protocol (RTP) sessions between them.
Option 3 is correct. Direct setup relies on cached information or internal mechanisms to resolve addresses, which
can become outdated if the destination is mobile. In addition, if the UA must keep information on a large number
of destinations, management of the data can become prohibitive. This makes the direct method nonscalable.
Option 4 is incorrect. Direct setup does not require more bandwidth for messaging, as there is only a three step
communication process and it is completed with minimal overhead.
Answer
In call setup using a proxy server, if the proxy server fails, the UA cannot establish its own sessions.
Option 1 is correct. The disadvantages of call setup using a proxy server are that using a proxy server requires
more messaging and creates a dependency on the proxy server.
Option 2 is incorrect. If the proxy server fails, the UA is incapable of establishing its own sessions.
Option 3 is incorrect. The number of messages involved in redirection is fewer than with the proxy server
procedure. The UA has a heavier workload because it must initiate the subsequent invitation.
Option 4 is incorrect. Although the proxy server acts on behalf of a UA for call setup, the UAs establish RTP
sessions directly with each other.
xi
Answer
The location server, proxy server, and redirect server are the SIP components that need to be replicated in order
to provide fault tolerance.
Option 1 is incorrect. A gateway server is a SIP network component. A gateway acts as a UAS or UAC and provides
call control support. Gateways provide many services, the most common being a translation function between SIP
UAs and other terminal types.
Option 2 is correct. A location server is an abstraction of a service providing address resolution services to SIP
proxy or redirect servers. A location server embodies mechanisms to resolve addresses.
Option 3 is correct. In SIP, the network servers are the proxy server, the redirect server, and the location server. For
replication of a proxy or redirect server to be effective, a UA must have the ability to locate an active server
dynamically.
Option 4 is correct. In a SIP environment, the failure of a network server cripples UAs that are dependent on that
server. In SIP, the network servers are the proxy server, the redirect server, and the location server.
Option 5 is incorrect. The registrar server is not one of the SIP components. A registrar server makes requests from
UACs for registration of their current location. Registrar servers are often located near or even colocated with
other network servers, most often a location server.
xii
Answer
You can replicate a proxy server by enabling the UA to dynamically locate an active server.
Option 1 is incorrect. The redirect server cannot act as a proxy server on its own. The proxy server would have to
be installed on the same computer, which would not allow for replication.
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Study Notes
Option 2 is incorrect. Having two location servers would not allow for replication of the proxy server. A location
server is an abstraction of a service providing address resolution services to SIP proxy or redirect servers. It
embodies mechanisms to resolve addresses.
Option 3 is incorrect. Configuring multiple replication servers would not replicate the proxy server. It would be
necessary to create multiple proxy servers in order to recreate replication.
Option 4 is correct. For replication of a proxy or redirect server to be effective, a UA must have the ability to locate
an active server dynamically.
xiii
Answer
The show sip-ua statistics command displays SIP UA response and retry information.
Option 1 is incorrect. The show call active voice command displays the status, statistics, and parameters for all
active voice calls.
Option 2 is incorrect. The show sip-ua retry command displays the SIP protocol retry counts. High counts should be
investigated.
Option 3 is correct. The show sip-ua statistics command displays the SIP UA response, traffic, and retry statistics.
Option 4 is incorrect. The show sip-ua status command displays the SIP UA listener status, which should be
enabled.
xiv
Answer
You can use the debug ccsip events command to trace call setups, connections, and disconnections.
Option 1 is incorrect. The debug ccsip calls command displays all SIP call details as they are updated in the SIP call
control block.
Option 2 is correct. The debug ccsip events command traces events, such as call setups, connections, and
disconnections. An events version of a debug command is often the best place to start, because detailed debugs
provide a great deal of useful information.
Option 3 is incorrect. The debug ccsip messages command shows the headers of SIP messages that are exchanged
between a client and a server.
Option 4 is incorrect. The debug ccsip states command displays the SIP states and state changes for sessions
within the SIP subsystem.
Option 5 is incorrect. The debug voip ccapi inout command shows every interaction with the call control
application programming interface (API) on both the telephone interface and on the VoIP side. By monitoring the
output, you can follow the progress of a call from the inbound interface or VoIP peer to the outbound side of the
call.
xv
Answer
You use the debug ccsip calls command to monitor call records for suspicious clearing causes.
Option 1 is incorrect. The debug ccsip all command enables all ccsip-type debugging. This command is very active,
so you need to use it sparingly in a live network.
Option 2 is correct. The debug ccsip calls command displays all SIP call details as they are updated in the SIP call
control block. You need to use this command to monitor call records for suspicious clearing causes.
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Study Notes
Option 3 is incorrect. You would use the debug ccsip errors command to trace all errors encountered by the SIP
subsystem.
Option 4 is incorrect. You would use the debug ccsip states command to display the SIP states and state changes
for sessions in the SIP subsystem.
xvi
Answer
The show sip-ua status command displays the SIP UA listener status, which should be enabled.
Option 1 is incorrect. The show call active voice command displays the status, statistics, and parameters for all
active voice calls.
Option 2 is incorrect. The show sip-ua retry command displays the SIP protocol retry counts. High counts should be
investigated.
Option 3 is incorrect. The show sip-ua statistics command displays the SIP UA response, traffic, and retry statistics.
Option 4 is correct. The show sip-ua status command displays the SIP UA listener status, which should be enabled.
88 | P a g e

SIP (Session Initiation Protocol) Study Notes: +W - Technology Skills For Women Series

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SIP (Session Initiation Protocol)

Configuring - Monitoring - Troubleshooting

2. SIP components ................................................................................................................................. 12

SIP Messages and Addressing................................................................................................................ 18

SIP Call Setup Models and Fault Tolerance ........................................................................................... 28

Configuring and Monitoring SIP ............................................................................................................ 39