The Cisco 1751 Voice over IP Quick Start Guide explains how to install WAN Interface Cards and Voice
Interface Cards, how to run configuration tools, and how to perform essential Voice-over-IP (VoIP)
configuration tasks.
This document contains the following sections:
• Audience, page 1
• Conventions, page 2
• Installing WAN and Voice Interface Cards, page 2
• Cisco Router and Security Device Manager, page 9
Audience
• Configuring Voice-over-IP, page 10
• List of Terms, page 30
• Obtaining Documentation, page 31
• Documentation Feedback, page 32
• Obtaining Technical Assistance, page 32
• Obtaining Additional Publications and Information, page 33
This guide is intended primarily for users who configure and maintain routers, but are not necessarily
familiar with tasks, the relationship between tasks, or the commands necessary to perform particular
tasks to configure VoIP. We assume that you are familiar with Cisco IOS software, IP and telephony
networks, and PBX operation for E&M installations.
List of Terms provides definitions for many of the technical terms used in this document.
Corporate Headquarters:
Cisco Systems, Inc., 170 West Tasman Drive, San Jose, CA 95134-1706 USA
• Optional keywords or arguments are in square brackets.
• Variables for which you supply values are in italic.
• Titles of publications and new words or concepts are in italic.
• Information the router displays on the console screen is in screen font.
• Information that you enter at the CLI is in boldface screen font.
Installing WAN and Voice Interface Cards
The Cisco 1751 router supports two slots that can be used to install either WAN Interface Cards (WICS)
or Voice Interface Cards (VICS), and one slot that can be used to install only a VIC.
If You Are Installing Voice Interface Cards
The router supports one to three Cisco voice interface cards (VICs). Each VIC provides two ports. You
need one VIC port for each voice connection.
NoteTo transmit voice calls over an IP WAN, you need (in addition to the VICs) at least one WAN interface
card (WIC) in the router to provide the connection to the WAN.
There are four types of VIC interfaces, as shown in Figure 3:
• FXS (foreign exchange station) interface connects directly to a standard telephone, fax machine, or
similar device. The FXS VIC interface supplies ringing voltage, dial tone, and similar signals to the
station. Ports on this VIC are color-coded gray.
• FXO (foreign exchange office) interface connects local calls to a PSTN central office or to a PBX
that does not support E&M signaling. This is the interface a standard telephone provides. Ports on
this VIC are color-coded pink.
• E&M is a signaling technique for two-wire and four-wire telephone and trunk interfaces. The E&M
VIC connects remote calls from an IP network to a PBX for local distribution. Ports on this VIC are
color-coded brown.
• The ISDN BRI NT/TE voice interface card provides a client-side (TE) ISDN S/T physical interface
for connection to an NT1 terminating an ISDN telephone network and can also be configured to
provide a network termination (NT) interface with phantom power. Each of its two ports can carry
two voice calls (one over each ISDN B channel), for a total of four calls per ISDN BRI card. At
present, the ISDN BRI VIC does not support ISDN devices such as digital telephone, fax, or modem.
Figure 1 shows a typical VIC.
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Figure 1Voice Interface Card
VIC ports
Installing WAN and Voice Interface Cards
VIC
E&M
IN USE
10
SEE MANUAL BEFORE INSTALLATION
You should install and cable VICs before performing the software configuration tasks explained later in
this guide.
Warning
CautionVICs do not support online insertion and removal (hot swap). To avoid equipment damage, turn off
Be sure to observe all warnings and safety precautions in the installation guide.
electrical power and disconnect network cables before you insert a VIC in the router.
WIC and VIC Installation Procedure
Follow these steps, and refer to Figure 2 to insert a WIC or a VIC in the router:
Step 1Make sure the router is turned off and is disconnected from the power supply.
Step 2Loosen the thumbscrews on the WIC or VIC slot cover on the rear panel.
You should be able to loosen the screws using your fingers; however, if the screws are very tight, you
might need to use a Phillips screwdriver.
IN USE
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Step 3Remove the metal plate that covers the card slot.
Step 4Hold the card by the edges on either side of the card front panel, and line up the card edges with the
guides inside the card slot, as shown in Figure 2.
Step 5Insert the card in the slot and gently push it into the router until the front panel of the card is flush with
the rear panel of the router.
Step 6Tighten the screws.
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Installing WAN and Voice Interface Cards
Figure 2Inserting a WIC or VIC in the Router
Model Cisco 1750
SLOT 1
SLOT 2
V
IC
IN USE
2
F
X
O
C
O
N
S
O
1
LE
SEE MANUAL BEFORE INSTALLATION
AU
X
IN USE
THIS SLOT
ACCEPTS
ONLY VOICE
0
INTERFACE
CARDS
PVDM OK
MOD OK
SLOT 2 OK
+5, +12, -12 VDC
V
2
IC
F
X
S
IN USE
1
SEE MANUAL BEFORE INSTALLATION
SLO
T 1 OK
SLOT 0 OK
IN USE
0
Guides
SLOT 0
FDX
100
LINK
10/100 ETHERNET
Interface card
If you need more detailed installation instructions, refer to the Cisco WAN Interface Cards Hardware
Installation Guide.
Connecting the FXS, FXO, and E&M VICs to the Network
If you installed FXS, FXO, or E&M VICs, use the following cables, and refer to Figure 3 to connect
them to the network:
• Standard RJ-11 modular telephone cable to connect FXS VIC ports (color-coded gray) to a
telephone or fax machine.
• Standard RJ-11 modular telephone cable to connect FXO VIC ports (color-coded pink) to the PSTN,
or to a PBX that does not support E&M signaling, through a telephone wall outlet.
• Standard RJ-48S connector and cable to connect E&M VIC ports (color-coded brown) to a PBX line.
The cable wiring depends on the type of connection. For details, refer to the Cisco WAN Interface
Cards Hardware Installation Guide.
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Figure 3Connecting VICs to the Network
FXS VIC
VIC
IN USE
FXS
10
SEE MANUAL BEFORE INSTALLATION
FXO VIC
VIC
IN USE
FXO
10
SEE MANUAL BEFORE INSTALLATION
IN USE
IN USE
Installing WAN and Voice Interface Cards
E&M VIC
VIC
IN USE
E&M
10
SEE MANUAL BEFORE INSTALLATION
IN USE
PBX
10691
When you are finished, reinstall any network interface cables you removed and turn on power to the
router.
Connector Pinouts for FXS, FXO, and E&M VICs
Figure 4 shows the wiring patterns for FXS and FXO VIC connectors, and Table 1 lists the FXS and FXO
pinouts.
NotePins that are not used should not be connected.
Figure 4RJ-11 Wiring for FXS and FXO Connectors
1
6
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H11420
Table 1FXS and FXO Pinouts
PinSignal
1–
2–
3Ring
4Tip
5–
6–
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Installing WAN and Voice Interface Cards
Figure 5 shows the wiring pattern for an E&M connector, and Ta b l e 2 lists the E&M pinouts.
NoteThe E&M VIC pinout depends on the PBX type and connection. Pins that are not used should not be
connected.
Figure 5RJ-48S Wiring for E&M Connectors
1
Table 2E&M Pinouts
8
H11421
PinSignalDescription
1SB–48V signaling battery
2M-leadSignaling input
3RRing, audio input
4R or R1Ring, audio input/output, or
output
5T or T1Tip, audio input/output, or
output
6TTip, audio input
7E-leadSignaling output
8SGSignaling ground return
Checking FXS VIC Installation
If you installed an FXS VIC, connect a handset to the VIC port. When router power is on, you should be
able to hear the dial tone when you lift the handset. The dial tone should stop after you dial a digit. If
you have trouble, use the show voice port command to make sure that the VIC is installed correctly, or
try connecting a different handset to the VIC.
2-Port ISDN BRI Card
The ISDN BRI NT/TE voice interface card provides a client-side (TE) ISDN S/T physical interface for
connection to an NT1 terminating an ISDN telephone network. It can also be configured to provide a
network termination (NT) interface with phantom power. Each of its two ports can carry two voice calls
(one over each ISDN B channel), for a total of four calls per ISDN BRI card.
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The card is illustrated in Figure 6.
Figure 62-Port ISDN BRI Card Front Panel
RJ-45 ports
Installing WAN and Voice Interface Cards
VIC
2B-NT/TE
ISDN BRI S/T 1
Connecting the 2-Port ISDN BRI Card
Use the straight-through RJ-45 cable to connect the 2-port ISDN BRI card to the ISDN network through
a telephone wall outlet.
CautionTo prevent damage to the router, be sure to connect the BRI cable to the BRI connector only and not to
any other RJ-45 connector.
Step 1Confirm that the router is still turned off.
Step 2Connect one end of a straight-through RJ-45-to-RJ-45 cable to the card’s RJ-45 port. (See Figure 7.)
Figure 7Connecting the 2-Port ISDN BRI Card
RJ-45 ports
B1
B2
OK
ISDN BRI S/T 0
SEE
MANUAL
BEFORE
INSTALLATION
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VIC
2B-S/T TE
ISDN BRI S/T 1
Straight-through
RJ-45-to-RJ-45 cable
B1
B2
OK
ISDN BRI S/T 0
SEE
MANUAL
BEFORE
INTALLATION
51076
NT1 device
S/T port
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Installing WAN and Voice Interface Cards
NoteWhen the interface is configured as NT and is connecting to a TE device, the cable must have the transmit
and receive pins swapped (crossover cable). (See Tabl e 3.)
Table 3Interface Pin Numbers and Functions
ISDN BRI NT/TE
Pin 3/T+Pin 3/R+Pin 3/T+
Pin 4/R+Pin 4/R+Pin 4/R+
Pin 5/R-Pin 5/T-Pin 5/R-
Pin 6/T-Pin 6/R-Pin 6/T-
Step 3Connect the other end to the RJ-45 wall outlet.
ISDN BRI Card LEDs
ISDN BRI voice interface cards have three LEDs, as listed in Table 4.
Table 4ISDN BRI Voice Interface Card LEDs
LEDMeaning
B1Call active on B1 channel
B2Call active on B2 channel
OKInterface is connected to an
NT Interface (use
straight-through cable)
ISDN network
TE Interface (use
crossover cable)
If you have additional cards to install, proceed to the appropriate sections in this manual. When you have
finished all installations, see the Cisco 1751 Router Software Configuration Guide available on the
Documentation CD-ROM.
Determining Voice Port Numbering
Enter the show voice port command to identify the port numbers of voice interfaces installed in your
router:
If you install a VIC in both SLOT 0 and SLOT 1 of the router, the ports in each of these slots would be
numbered as follows:
SLOT 0—0/0 and 0/1
SLOT 1—1/0 and 1/1
Cisco Router and Security Device Manager
Cisco Router and Security Device Manager (SDM) is a web-based configuration tool that allows
you to configure LAN and WAN interfaces, routing, Network Address Translation (NAT),
firewalls, VPNs, and other features on your router. If SDM is installed on your router, configure
the router by following the instructions in the Cisco Router and Security Device Manager (SDM) Quick Start Guide. If this document was not shipped with your router, you can obtain SDM and
instructions for installing it on your router from the following
http://www.cisco.com/pcgi-bin/tablebuild.pl/sdm
location:
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Configuring Voice-over-IP
To obtain the SDM release notes, and other SDM documentation, go to http://www.cisco.com/go/sdm
and click the Technical Documentation link.
NoteIf you need to configure Voice-over-IP, use the Configuring Voice-over-IP section of this Quick Start
Guide.
Configuring Voice-over-IP
Voice-over-IP (VoIP) enables a Cisco 1751 router to carry voice traffic (for example, telephone calls and
faxes) over an IP network, simultaneously with data traffic. (See Figure 10.) This guide describes the
configuration tools and the major configuration tasks to configure the router.
Figure 10Voice and Data Traffic on an IP Network
Configuration Tools
You can configure VoIP by using one of the following tools:
• Cisco ConfigMaker application
• Cisco IOS command-line interface (CLI)
Cisco ConfigMaker Application
Use the Cisco ConfigMaker application to address and configure one or more devices or to create an
entire network. A series of wizards and dialog boxes can make your voice-configuration task simple and
fast.
Follow these steps to install the Cisco ConfigMaker application on your PC:
IP cloud
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Step 1Disable any virus-scanning software you have running.
If the installer does not start after a few seconds, select Start>Run, and enter d:\setup where d is your
CD-ROM drive.
Step 5Follow the instructions on the screen to install the Cisco ConfigMaker application.
To learn about the Cisco ConfigMaker application, view the online movie, or use Cisco ConfigMaker
Help Topics from the help menu.
You need administrative privileges to install the Cisco ConfigMaker application on Windows NT.
Cisco IOS Command-Line Interface
To configure your router from the command-line interface (CLI), refer to this guide, the Cisco 1700
Series Software Configuration Guide, or the Cisco 1751 Router Software Configuration Guide available
on the Documentation CD-ROM.
Major Voice over IP Configuration Tasks
Follow these major steps to configure Voice over IP (VoIP) on your router:
Configuring Voice-over-IP
Step 1Configure your IP network to support real-time voice traffic. Before you move to the next step, ping the
remote router to make sure you have an IP connection.
Step 2Configure your router to support voice ports. Follow these steps for your specific requirements:
• Foreign exchange station (FXS) and foreign exchange office (FXO) voice ports
The default voice-port values are usually adequate for both FXS and FXO voice ports. However, if
you need to change the default values, refer to the Cisco 1751 Router Software Configuration Guide.
• E&M voice port
The default values for an E&M voice port are not sufficient to enable voice and data transmission
over the IP network. E&M values must match those of the connected PBX device. Refer to the
“Connecting the FXS, FXO, and E&M VICs to the Network” section on page 4.
• ISDN BRI port
The default values for a basic rate interface (BRI) port are not sufficient to enable voice and data
transmission over the IP network. BRI values must match those of the connected PBX device. Refer
to the “Connecting the FXS, FXO, and E&M VICs to the Network” section on page 4.
Step 3Enter configuration mode to configure your router for VoIP traffic. Refer to the “Configuring
Voice-over-IP” section on page 10.
Step 4Save the configuration. Refer to the “Saving the Configuration” section on page 13.
Step 5Establish a dial plan to determine how the call can be routed. Use the dial-peer voice command to define
dial peers and to implement the dial plan. Refer to the “Dial Plan” section on page 14.
Step 6Fine-tune your network to adequately support VoIP by configuring a series of protocols and features for
quality of service (QoS). Refer to the “Tuning Parameters for Real-Time Voice Traffic” section on
page 25.
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Configuring Voice-over-IP
Information for International Users
International users must set the values of certain voice-port commands, like cptone, that are specific to
their country. Refer to the Cisco 1751 Router Software Configuration Guide for details.
Entering Configuration Mode
You can configure your router for VoIP by entering commands at the CLI. This method of entering
commands is called configuration mode.
NoteVoIP requires one of the following IOS Release 11.3(1)T (or later) feature sets: IP Plus, Desktop Plus,
or Enterprise Plus.
Follow these steps to enter configuration mode:
Step 1Connect a terminal or PC to the router.
To configure the router with a PC, the PC must have some type of terminal emulation software installed.
The software should be configured with the following parameters: 9600 baud, 8 data bits, no parity, 1
stop bit.
Step 2Power up the router.
Step 3If the current configuration is no longer valid (for instance, because you added an interface), after about
1 minute you will see the following prompt:
Would you like to enter the initial configuration dialog? [yes]:
Do not use the yes default. You need to enter no to enter the normal operating mode of the router.
NoteIf the current configuration is valid, you enter the normal operating mode automatically.
Step 4At the user EXEC prompt (Router>), enter enable and the password to enter enable mode:
Router> enable
Password:
Configuration changes can be made only in enable mode. The prompt changes to the privileged EXEC
(enable) prompt (
Router#
Step 5Enter the following command to enter global configuration mode:
Router# config terminal
Router(config)#
Step 6If you have not configured the router before or want to change the configuration, you need to configure
Router#):
global parameters, passwords, network management, and routing protocols. In this example, IP routing,
AppleTalk routing, and IPX routing are all enabled:
Router(config)# ip routing
Router(config)# appletalk routing
Router(config)# ipx routing
12
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For complete information about the global configuration commands and about configuring LAN and
WAN interfaces on your router, refer to the Cisco IOS configuration guides and command references.
The rest of this guide explains how to configure your router for VoIP traffic.
NoteVoice configuration uses a number of Cisco IOS commands. For complete information about these
commands, refer to the Cisco 1751 Router Software Configuration Guide. You can also enter a question
mark after a command or partial command at the
help.
NoteAt any point, you can see the operating configuration, including changes you just made, by entering the
show running-config command. To enter this command, you have to exit from configuration mode by
pressing Ctrl-Z or entering the end global configuration command.
Saving the Configuration
Configuring Voice-over-IP
Router(config)# prompt to get syntax and argument
You must save the new configuration to nonvolatile RAM (NVRAM) periodically during the
configuration process and also when you finish configuring the router.
Cisco IOS software uses two configuration files—startup config and running config. The startup-config
file is a backup file that has all the information you specified about the router interfaces and is used by
the router on the next restart. The running-config file has the current operating configuration with the
changes you have just made. You can make any changes to the running-config file; however, these
changes will be lost when the router powers down. To make these changes permanent, copy the
running-config file to the startup-config file stored in NVRAM.
Follow these steps for each router to write the new configuration to NVRAM:
Step 1Exit configuration mode, and return to the enable prompt by pressing Ctrl-Z or entering the end global
configuration command.
Step 2To see the startup configuration stored in NVRAM, enter the following command at the enable prompt:
Router# show startup-config
Step 3To see the operating configuration, including any changes you have just made, enter the following
command:
Router# show running-config
Step 4To write your changes to NVRAM and make them permanent, enter the following command:
Router# copy running-config startup-config
Building configuration. . .
[OK]
Router#
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The router is now configured to boot in the new configuration.
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Configuring Voice-over-IP
Dial Plan
Use a dial plan to map the destination telephone numbers with the voice ports on the router. In North
America, the North American Numbering Plan (NANP) is used, which consists of an area code, an office
code, and a station code. Area codes are assigned geographically, office codes are assigned to specific
switches, and station codes identify a specific port on that switch. The format in North America is
1Nxx-Nxx-xxxx, with N = digits 2 through 9 and x = digits 0 through 9. Internationally, each country is
assigned a one- to three-digit country code; the country’s dialing plan follows the country code.
In corporate environments, VoIP can be configured to recognize extension numbers and expand them
into their full E.164 dialed number by using two commands in tandem: destination-pattern and num-exp. Before you configure these two commands, map individual telephone extensions with their
full E.164 dialed numbers. This can be done easily by creating a number expansion table.
For Cisco voice implementations, two types of dial peers are used to match a dialed number to either a
local telephony port or a remote IP address:
• POTS dial peer associates a physical voice port with a local telephone device, and the key commands
you need to configure are the port and destination-pattern commands as shown in the following
example:
The destination-pattern command defines the telephone number associated with the POTS dial
peer. The port command associates the POTS dial peer with a specific logical dial interface,
normally the voice port connecting your router to the local POTS network.
• VoIP dial peer associates a telephone number with an IP address, and the key commands you need
to configure are the destination-pattern and session target commands as shown in the following
example:
The destination-pattern command defines the telephone number associated with the VoIP dial peer.
The session target command specifies a destination IP address for the VoIP dial peer.
Use the dial-peer voice command to define dial peers and change to dial peer configuration mode. To
see an example, refer to the individual sections on configuring FXS, FXO, and E&M interfaces later in
this guide.
Wildcards and Number Expansion
Office PBXs are configured so that a user can dial a local call (within the same PBX) by dialing the
extension only—for instance, the four-digit extension 3737 or the five-digit extension 53737 instead of
the full telephone number, 1 408 555-3737.
You can provide the same shortcut on a VoIP network by using the number-expansion (num-exp)
command.
East(config)# num-exp 5.... 1408555....
14
This command tells the router to expand a particular sequence of dialed numbers into a complete
telephone number (destination pattern) as shown in Tabl e 5.
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Configuring Voice-over-IP
Table 5Sample Number Expansion Table
Destination
Extension
373714085553737num-exp 3737 14085553737 To expand 3737 to 14085553737
....1408555....num-exp .... 1408555....To expand any dialed sequence
5....1408555....num-exp 5.... 1408555....To expand a five-digit extension
You can use a period (.) as a wildcard character representing a single digit in a telephone number.
You can use the show num-exp command to verify the number expansion information.
East# show num-exp [dialed-number]
After you have configured dial peers and assigned destination patterns to them, you can use the show
dialplan number command to see how a telephone number maps to a dial peer.
East# show dialplan number [dialed-number]
PatternNum-Exp Command EntryDescription
of four digits (represented by
four periods) by prefixing
1408555 to it
beginning with the numeral 5 by
prefixing 140855 to it
NoteYou must still configure each telephone number in full on a local dial peer so that the router can find the
voice port it belongs to.
Configuring FXS Interfaces
This section explains how to configure ports on FXS VICs that connect directly to a standard telephone,
fax machine, or similar device.
Figure 11 shows a basic voice network. A small business uses a Cisco 1751 router (named Wes t) to
provide telephone and fax connections among employees in its office. Two of these telephones are
connected to an FXS VIC port in the We st router.
Figure 11Basic Voice Network (West Router)
408 555-3737
408 555-4141
FXS VIC
0/0
West
FXS VIC
0/1
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Configuring Voice-over-IP
NoteYou can name your router by using the global configuration hostname command.
(For information about port numbering, refer to the section on “Determining Voice Port Numbering”
earlier in this guide.)
Table 6West Router Telephone Numbers and Voice Ports
Telephone NumberVoice Port
408 555-37370/0
408 555-41410/1
NoteIf your router is configured with three VICs, a total of six telephones and fax machines can be connected
to it. As the router has only three slots, you need to replace one VIC with a WIC to provide an interface
for IP connectivity to the WAN and for data traffic. To accommodate more than six voice devices, you
need to add more routers or use an E&M VIC and a local PBX, rather than connecting every telephone
to its own FXS VIC.
Local Dial Peers
To route a received voice call to the right destination, the router needs to know which telephone number
belongs to each voice port. For instance, if a call comes in for 408 555-3737, the router needs to know
that this telephone is connected to voice port 0/0 (as shown in Figure 11.) In other words, the router
needs to know the information in Table 6.
To hold this information, Cisco IOS software uses objects called dial peers. A telephone number, a voice
port, and other call parameters are tied together by associating them all with the same dial peer.
Configuring dial peers is similar to configuring static IP routes—you are telling the router what path to
follow to route the call. All voice technologies use dial peers to define the characteristics associated with
a call leg. A call leg is a segment of a call path, for instance, between a telephone and a router, a router
and a network, a router and a PBX, or a router and the PSTN. Each call leg corresponds to a dial peer.
Dial peers are identified by numbers, but they are usually referred to as tags to avoid confusion with
telephone numbers. Dial-peer tags are arbitrary integers that can range from 1 to 2
31
–1(2147483647).
Within the allowed range, you can choose any dial-peer tag that is convenient or makes sense to you.
Dial peers on the same router must have unique tags, but you can reuse the tags on other routers.
Table 7 assigns a dial-peer tag to each telephone number and its associated voice port on the Wes t router.
This type of dial peer is called a POTS dial peer or a local dial peer. The term “POTS” (plain old
telephone service) means that the dial peer associates a physical voice port with a local telephone device.
Table 7West Router Local Dial Peers
Telephone NumberVoice PortDial-Peer Tag
408 555-37370/0401
408 555-41410/1402
16
You should construct a table similar to Tab le 7 for your own routers, assigning your own telephone
numbers and dial-peer tags.
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Configuring Voice-over-IP
NoteThe telephone numbers used in this guide are only examples and are invalid for public use in the United
States. When you configure your network, be sure to substitute your own telephone numbers.
To configure the router with the dial-peer information in Tab le 7, enter the following global
configuration commands:
Figure 12West Router Configured for Local Dial Peers
dial-peer voice 401 pots
destination-pattern 14085553737
port 0/0
FXS VIC
0/0
West
FXS VIC
0/1
dial-peer voice 402 pots
destination-pattern 14085554141
port 0/1
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The dial-peer command always takes the argument voice. The number following it is the dial-peer tag,
and pots is the type of dial peer.
Cisco IOS software refers to a telephone number as a destination pattern because it is the destination for
an incoming or outgoing call. Enter these numbers with the destination-pattern command. A
destination pattern can include asterisks (*) and pound signs (#) from the telephone keypad, and commas
(,) and periods (.), which have special meanings. Parentheses ( () ), hyphens (-), slashes (/), and spaces
( ), which are often used to make telephone numbers easier for humans to read, are not allowed.
Notice that the commands in the examples puts the prefix 1 (used in the United States to indicate a
long-distance number) and an area code in front of the remaining numbers to complete the destination
pattern. You need to include similar codes for your country if the VoIP equipment needs to establish a
connection to the PSTN.
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NoteThe Cisco IOS software does not check the validity of the telephone number. It accepts any string of
permitted characters as a valid number.
The business that owns the Wes t router also has a branch office in the East. Figure 13 shows the East
office network, and Table 8 lists the phone numbers, voice ports, and dial-peer tags for this office.
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Configuring Voice-over-IP
Figure 13Basic Voice Network (East Router)
919 555-8282
FXS VIC
1/0
IP cloud
East
FXS VIC
1/1
919 555-9595
22333
Table 8East Router Local Dial Peers
Telephone NumberDestination PatternVoice PortDial-Peer Tag
919 555-8282191955582821/0901
919 555-9595191955595951/1902
Enter the following commands to configure the local ports on the East Router with the dial-peer
information in Tab l e 8:
Figure 14East Router Configured for Local Dial Peers
dial-peer voice 901 pots
destination-pattern 19195558282
port 1/0
IP cloud
dial-peer voice 902 pots
destination-pattern 19195559595
port 1/1
Cisco 1751 Voice-over-IP Quick Start Guide
FXS VIC
1/0
East
FXS VIC
1/1
22334
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Checking Dial Peer Configuration
If you configured POTS dial peers on your router by following these examples, you can place calls
between telephones connected to the same router. You can also use the show dial-peer voice command
to verify that the data you configured is correct.
NoteIf the voice port is offline, use the interface configuration no shutdown command to enable it.
Calling Between Routers
To enable the We st and East offices to send voice traffic to each other over the same IP network they use
for data traffic, use a WIC on each router to provide a connection to the IP network, as shown in
Figure 15.
Figure 15IP Connection Between Routers
Configuring Voice-over-IP
408 555-3737
192.168.19.27
408 555-4141
FXS VIC
0/0
West
FXS VIC
0/1
IP cloud
FXS VIC
1/0
East
FXS VIC
1/1
192.168.11.3
919 555-8282
919 555-9595
22335
Look at the connection between the We st router and the IP network. This connection does not include a
voice port or an attached telephone—it leads from a WAN interface to a remote destination somewhere
on the IP network. IP routers know how to locate IP addresses on the network, but they do not know how
to locate telephone numbers. To route an outgoing voice call over this connection, the Wes t router has to
associate a telephone number in the East office with the IP address of the East router.
Table 9 assigns a dial-peer tag to each telephone number and its associated IP address on the We st router.
This type of dial peer is called a remote dial peer or VoIP dial peer. (Remember, the dial-peer tags are
arbitrary.) The term VoI P means that the dial peer associates a telephone number with an IP address.
Table 9West Router Remote Dial Peers
Remote LocationTelephone NumberDestination PatternIP AddressDial-Peer Tag
East919 555-828219195558282192.168.11.3501
East919 555-959519195559595192.168.11.3502
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Create a VoIP dial peer on the We st router for every telephone on the East router, all associated with the
same IP address. But it is much easier to use periods as wildcards, as shown in Table 10 .
Cisco 1751 Voice-over-IP Quick Start Guide
19
Configuring Voice-over-IP
Table 10West Router Remote Dial Peers with Wildcards
Remote LocationTelephone NumberDestination PatternIP AddressDial-Peer Tag
East919 555-xxxx1919555....192.168.11.3501
Construct a table similar to Tab le 10 for your own routers, assigning your own telephone numbers,
IP addresses, and dial-peer tags.
NoteThe IP addresses shown in this guide are meant only as examples. When you configure your network, be
sure to substitute your own IP addresses.
Enter the following information on the Wes t router to create the dial-peer configuration given in
Cisco IOS software describes the remote network as the session target. This command is followed by
the IP address of the remote router. The prefix ipv4 means IP version 4. Alternatively, you can use the
prefix dns followed by the Domain Name System (DNS) name as follows:
Figure 17East Router Configured for Remote Dial Peers
192.168.19.27
408 555-xxxx
DestinationSource
Other Routers on the Network
If the path between endpoints of a voice call travels through intermediate routers, configure those routers
for VoIP traffic, as described in the section “Configuring FXS Interfaces” earlier in this guide.
You need to configure POTS or VoIP dial peers on an intermediate router only if that router also has voice
devices attached to it.
Checking FXS Configuration
If you configured VoIP dial peers on your router by following these examples, you can place calls from
that router to telephones on the remote routers (using just the extension if you configured number
expansion). If you have trouble placing calls, ping the remote router to make sure you have IP
connectivity, or use the show dial-peer voice command to verify that the data you configured is correct.
IP cloud
EastWest
dial-peer voice 801 voip
destination-pattern 1408555….
session target ipv4:192.168.19.27
num-exp 5.... 1408555....
22639
NoteRefer to the section “Tuning Parameters for Real-Time Voice Traffic” if you need to improve the quality
of voice connections.
Configuring FXO Interfaces
FXO interfaces provide a gateway from the VoIP network to the analog PSTN or to a PBX that does not
support E&M signaling so that users can reach telephones and fax machines outside the VoIP network.
Figure 18 shows a typical FXO gateway attached to the West router.
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21
Configuring Voice-over-IP
Figure 18FXO Gateway to PSTN
To create a POTS dial peer for an FXS interface as explained earlier, you enter the complete telephone
number of the attached telephone as the destination pattern for incoming calls. However, to create a
POTS dial peer for an FXO interface, the destination pattern refers to outgoing calls, and you can include
wildcards in it because the PSTN performs the switching.
The VoIP feature can also remove digits that you do not want to send to the PSTN. For instance, to dial
9 to reach an outside line (that is, the analog PSTN), enter the following commands:
When you dial 9, the router makes a connection to the PSTN through voice port 0/0. The PSTN then
provides a dial tone, and any digits you enter on the telephone thereafter are interpreted on the PSTN.
To enab l e East router users to make calls over the We st router local PSTN, enter the following
commands:
When you dial 7 on the East router, the call is connected to the PSTN on the We s t router. The PSTN then
provides a dial tone, and any digits you enter on the telephone thereafter are interpreted on the PSTN.
NoteIn this example, Wes t router voice port 0/0 has two separate POTS dial peers associated with it. Dial peer
201 matches calls beginning with the digit 9 and handles PSTN calls originating from the Wes t router.
Dial peer 601 matches calls beginning with the digit 7 and handles calls to the PSTN originating from
the East router.
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Checking FXO Configuration
If you configured your FXO interface according to this example, you can place outgoing calls over the
PSTN. If you have trouble placing calls, use the show voice port command to make sure that the VIC is
installed correctly. Use the show dial-peer voice command to make sure that the data you configured is
correct, and test the PSTN by connecting a handset directly to the PSTN outlet and placing a call.
NoteRefer to the section “Tuning Parameters for Real-Time Voice Traffic” if you need to improve the quality
of voice connections.
Configuring E&M Interfaces
If you have more than a few voice users at each location, the cost of voice ports and routers and the effort
needed to configure dial peers for all the combinations of origins and destinations increases rapidly. In
this situation, it might be more efficient to use a PBX at each location to switch local traffic and direct
incoming calls and then use E&M VICs to connect the PBXs over an IP network.
Figure 19 shows a company with two offices, Wes t and East. Each office has a PBX to operate its internal
telephone network, and the IP network carries voice traffic between the offices. Each PBX connects to
an E&M VIC port in the router.
Configuring Voice-over-IP
Figure 19Linking PBXs over the IP Network (Local Dial Peers)
408 555-xxxx
PBX
Destination
E&M VIC
0/0
WestEast
dial-peer voice 111 pots
destination-pattern 1408555….
port 0/0
IP cloud
192.168.11.3
PBX
919 555-xxxx
Source
22342
To configure E&M voice ports, you need to use the following commands beginning in privileged EXEC
mode:
CommandTask
dial-type {dtmf | pulse}Select the appropriate dial type for out-dialing.
signal {wink-start | immediate | delay-dial}Select the appropriate signal type for this
interface.
cptone {australia | brazil | china | finland |
france | germany | japan | northamerica |
Select the appropriate voice call progress tone for
this interface.
unitedkingdom}
operation {2-wire | 4-wire}Select the appropriate cabling scheme for this
voice port.
type {1 | 2 | 3 | 5}Select the appropriate E&M interface type.
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Configuring Voice-over-IP
Both PBXs in this example use E&M interface Type 2, with four-wire operation and immediate-start
signaling. The values for your configuration depend on your PBX and are available from your
telecommunications department or the PBX manufacturer. For more information about E&M interface
configuration commands, refer to the “VoIP Commands” chapter of the Cisco 1751 Router Software Configuration Guide.
In this example, Wes t users can dial 5 and a four-digit extension to reach telephones in the East Office.
East users can dial 5 and a four-digit extension to reach telephones in the West office.
The Wes t router connects to the PBX through an E&M VIC port 0/0. This port is associated with a POTS
dial peer for incoming calls. But you no longer need to associate every telephone number with its own
port. Instead, you can configure a local dial peer as if all the West telephones (represented by a wildcard
destination pattern) are connected directly to this port, as shown in the following commands:
Configure VoIP dial peers for outgoing calls and associate destination phone numbers on the East router
with that router IP address, as shown in Figure 20, and in the following commands:
Now configure number expansion so that numbers beginning with 5 (belonging to the East office) and
sent by the West PBX to the Wes t router are expanded into the full destination pattern:
West(config)# num-exp 5.... 1919555....
NoteYou do not need to configure number expansion for calls from one We st telephone to another We st
telephone because the PBX switches those calls.
Finally, enter the following global configuration voice-port command to configure the E&M port:
West(config)# voice-port 0/0
West(config-voice-port)# signal immediate
West(config-voice-port)# operation 4-wire
West(config-voice-port)# type 2
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Configuring Voice-over-IP
NoteFor these commands to take effect, you have to cycle the port by using the shutdown and no shutdown
commands.
NoteConfigure the PBX to pass all dual-tone multifrequency (DTMF) signals to the router.
Configure the East router similar to the Wes t router. The East router connects to the PBX through an
E&M VIC port 0/1. Enter the following commands to configure a POTS dial peer for all East telephones:
Enter the following commands to configure number expansion and to make it easy for East users to dial
numbers on the We st router:
West(config)# num-exp 5.... 1408555....
Finally, configure the E&M port:
East(config)# voice-port 0/1
East(config-voice-port)# signal immediate
East(config-voice-port)# operation 4-wire
East(config-voice-port)# type 2
Checking E&M Interface Configuration
If you configured the E&M interfaces correctly, you can place calls from a telephone served by one PBX
to a telephone served by the other PBX (using just the extension, if you configured number expansion).
If you have trouble placing calls, ping the remote router to make sure you have IP connectivity.
NoteRefer to the section “Tuning Parameters for Real-Time Voice Traffic” if you need to improve the quality
of voice connections.
Tuning Parameters for Real-Time Voice Traffic
Voice traffic is much more sensitive to timing variations than data traffic. For good voice performance,
you need to configure your data network so that voice packets are not lost or delayed. To achieve the
level of network performance needed for VoIP connections, use the following features to improve quality
of service (QoS):
• IP precedence
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• Resource Reservation Protocol (RSVP)
Cisco 1751 Voice-over-IP Quick Start Guide
25
Configuring Voice-over-IP
• Multilink Point-to Point Protocol (PPP) interleaving
• Real-Time Transport Protocol (RTP) header compression
Cisco IOS software provides many other tools for ensuring QoS, such as custom queuing, priority
queuing, and weighted fair queuing. For further information and more detailed examples of QoS
configuration, refer to the Cisco 1751 Router Software Configuration Guide.
NoteQoS measures the level of network performance. It does not directly measure the quality of the voice
signal.
Configuring VoIP on a Frame Relay link involves special considerations. These are discussed in the
section “Configuring Frame Relay for VoIP.”
Need for Quality of Service
On a relatively low-bandwidth connection, such as a PPP or High-Level Data Link Control (HDLC)
serial link, you should consider using methods to ensure QoS. If you have a high-bandwidth network,
such as Ethernet or Fast Ethernet, and voice and data traffic together occupy only a small fraction of the
bandwidth available, you might not need to provide QoS. (See Figure 21.)
IP Precedence
Figure 21Bandwidth v. Quality of Service
Low bandwidth
needs QoS
10689
High bandwidth
might not need QoS
Use the ip precedence command to give voice packets a higher priority over other IP data traffic. Every
IP packet is given a precedence level. In IP precedence, the numbers 1 through 5 identify classes for IP
flows; the numbers 6 through 7 are used for network and backbone routing and updates. You can
configure voice packets for higher priority by setting the IP precedence value to 5. Internal routers using
weighted fair queuing will give these packets priority. This command only applies to VoIP dial peers.
The following example sets the IP precedence to 5:
Router> enable
Password:
Router# configure terminal
Router(config)# dial-peer voice 221 voip
Router(config-dial-peer)# ip precedence 5
26
The ip precedence command should also be used if RSVP is not enabled and the user would like to give
voice packets a higher priority over other IP data traffic.
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RSVP
NoteIf you configure multilink PPP interleaving, you can use the ip rtp reserve command instead of
Configuring RSVP
Configuring Voice-over-IP
Resource Reservation Protocol (RSVP) enables routers to reserve enough bandwidth on an interface for
reliable and quality performance. RSVP works well on PPP, HDLC, and similar serial line interfaces. It
does not work well on multi-access LANs.
You should configure RSVP if you have a serial interface and any of the following:
• Links slower than 2 Mbps
• Links with high utilization
• Need for the best possible voice quality
configuring RSVP. Refer to the next section on “Multilink PPP Interleaving.”
By default, RSVP is disabled so that it is backwards compatible with routers that do not implement it.
To enable RSVP on an IP network, enter the ip rsvp bandwidth command from interface configuration
mode. The following example enables RSVP on serial interface 0/0 and sets the maximum bandwidth to
96 kbps and the maximum bandwidth per single request to 24 kbps:
Router(config)# interface serial 0/0
Router(config-if)# ip rsvp bandwidth 96 24
The default maximum bandwidth is 75 percent of the bandwidth available on each interface.
RSVP must be enabled at each LAN or WAN interface that voice packets will travel across. After
enabling RSVP you must also use the req-qos dial-peer configuration command to request an RSVP
session on each VoIP dial peer. Otherwise, no bandwidth is reserved for voice traffic.
Router(config-dial-peer)# req-qos controlled-load
For further information about this command, refer to the Cisco 1751 Router Software Configuration
Guide.
Multilink PPP Interleaving
Multilink PPP interleaving allows large packets to be multilink-encapsulated and fragmented into
smaller packets to make sure that voice packets are transmitted without delay; small real-time packets,
which are not multilink-encapsulated, are transmitted between fragments of the large packets. The
interleaving feature also provides a special transmit queue for the smaller, delay-sensitive packets,
enabling them to be transmitted earlier than other flows.
You should configure multilink PPP interleaving if you have a dialer, ISDN PRI or BRI interface, or a
virtual template, and either of the following:
• Point-to-point connections using PPP encapsulation
• Links slower than 2 Mbps
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NoteDo not use multilink PPP on links faster than 2 Mbps.
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27
Configuring Voice-over-IP
Configuring Multilink PPP Interleaving
To configure multilink PPP and interleaving on a dialer, ISDN PRI or ISDN BRI interface, or a virtual
template, you must first configure multilink PPP and interleaving on the interface or template by entering
the following commands in interface configuration mode:
You can also reserve a special queue for real-time packet flows to specified destination User Datagram
Protocol (UDP) ports, allowing real-time traffic to have higher priority than other flows. You need the
following command only if you have not configured RSVP:
Router(config-if)# ip rtp reserve lowest-UDP-port range-of-ports
For virtual templates only, apply the virtual template to the multilink bundle:
Router(config-if)# multilink virtual-template 1
To create a virtual template interface, enter the following interface virtual-template global
configuration command:
Router(config)# interfacevirtual-template 1
RTP Header Compression
RTP header compression on a PPP, HDLC, or similar serial interface compresses the packet header to
reduce network overhead.
You should configure RTP header compression on a serial interface if you have either of the following:
• Links slower than 2 Mbps
• Need to save bandwidth
NoteDo not use RTP header compression on links faster than 2 Mbps.
Configuring RTP Header Compression
Enable RTP header compression at both ends of the serial link by entering the following ip rtp
header-compression
Router(config-if)# ip rtp header-compression
NoteWhen you enter the show running-config command, the format of the ip rtp header-compression
command will change to ip rtp header-compression iphc-format.
interface configuration command:
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Configuring Frame Relay for VoIP
Configuring VoIP on a Frame Relay link involves certain special considerations to ensure acceptable
voice quality. For Frame Relay links with slow output rates (64 kbps or less) and with data and voice
being transmitted over the same permanent virtual circuit (PVC), you should configure the following
parameters:
• Lower maximum transmission unit (MTU) size—Voice packets are generally small. If you decrease
the MTU size to 300 bytes, large data packets can be broken up into smaller data packets that are
more easily interleaved with voice packets.
The following example configures an MTU size of 300 bytes over serial interface 0/0:
Router# interface serial 0/0
Router(config-if)# mtu 300
NoteLowering the MTU size affects data throughput speed.
• RSVP—Configure RSVP on the subinterfaces (which correspond to PVCs) to reserve bandwidth for
voice channels. Refer to the section on “RSVP” earlier in this guide.
The following example configures RSVP over serial subinterface 0/0.1:
Router(config-if)# interface serial 0/0.1 point-to-point
Router(config-if)# ip address 192.168.19.0 255.0.0.0
Router(config-if)# ip rsvp bandwidth 48 48
Configuring Voice-over-IP
• RTP header compression—Configure RTP header compression on the subinterfaces to minimize the
size of the voice packet. Refer to the section on “RTP Header Compression” earlier in this guide.
The following command configures RTP header compression on the selected subinterface:
Router(config-if)# frame-relay ip rtp header-compression
• Traffic shaping—Use traffic shaping to control the outbound traffic rate; otherwise, voice packets
can be discarded. In Cisco IOS Release 12.0T, Frame Relay traffic shaping is not compatible with
RSVP. Use generic traffic shaping instead, and set the committed information rate (CIR) equal to
the port speed. This prevents the router from exceeding the CIR rate and discarding the frames.
The following command configures generic traffic shaping with a CIR of 32000 bps:
Router(config-if)# traffic-shape rate 32000
For further information and more detailed examples of Frame Relay configuration, refer to the
Cisco 1751 Router Voice-over-IP Configuration Guide.
NoteWhen you finish configuring the router, you must save the new configuration to nonvolatile RAM
(NVRAM) by copying
running-config to startup-config. You should also save periodically during
the configuration process. Refer to the “Saving the Configuration” section on page 13 for details.
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List of Terms
List of Terms
This section defines some of the VoIP terms and concepts used in this guide.
BRI—Basic Rate Interface, an ISDN interface.
CIR—Committed information rate. The average rate of information transfer a subscriber (for example,
the network administrator) has stipulated for a Frame Relay PVC.
Call leg—A segment of a call path, for instance, between a telephone and a router, a router and a
network, a router and a PBX, or a router and the PSTN. Each call leg corresponds to a dial peer.
Destination pattern—The pattern of numbers that identifies the destination of an incoming call; in other
words, a phone number.
Dial peer—A software object that ties together a voice port and a local telephone number (local dial
peer or POTS dial peer) or an IP address and a remote telephone number (remote dial peer or Vo I P d i al
peer). Each dial peer corresponds to a call leg.
DTMF—Dual-tone multifrequency. Use of two simultaneous voice-band tones for dial (such as touch
tone).
E.164—The international public telecommunications numbering plan. A standard set by ITU-T
that addresses telephone numbers.
E&M—A signaling technique for two-wire and four-wire interfaces that provide connections to PBX
trunk lines (tie lines).
FXO—Foreign exchange office, a type of VIC interface. Using a standard RJ-11 modular telephone
cable, the FXO VIC connects local calls to a PSTN central office or to a PBX. This is the interface a
standard telephone provides.
FXS—Foreign exchange station, a type of VIC interface. The FXS VIC connects directly to a standard
telephone, fax machine, or similar device over a standard RJ-11 modular telephone cable, and supplies
ringing voltage, dial tone, and similar signals to it.
HDLC—High-Level Data Link Control. A data link layer protocol that specifies a data encapsulation
method on synchronous serial links using frame characters and checksums.
MTU—Maximum transmission unit. Maximum packet size that a particular interface can handle.
Multilink PPP—Multilink Point-to-Point Protocol, a method of splitting, recombining, and sequencing
datagrams across multiple logical data links.
NANP—North American Numbering Plan.
PBX—Private branch exchange, a private telephone switching system.
POTS—Plain old telephone service. Basic telephone service supplying standard single line telephones,
telephone lines, and access to the public switched telephone network.
POTS dial peer—A software object that ties together a voice port and the telephone number of a device
attached to the port (also called local dial peer).
PRI—Primary Rate Interface, an ISDN interface.
PSTN—Public switched telephone network.
PVC—Permanent virtual circuit.
QoS—Quality of service, a measure of the level of performance needed for a particular application, such
as a VoIP connection.
RSVP—Resource Reservation Protocol, a network protocol that enables routers to reserve the
bandwidth necessary for reliable performance.
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Obtaining Documentation
RTP—Real-Time Transport Protocol, a network protocol used to carry audio and video packet traffic
over an IP network.
session target—A remote IP or DNS address specified for a dial peer.
tag—A positive integer in the range 1 to 2
UDP—User Datagram Protocol, a simple protocol that exchanges datagrams without acknowledgments
or guaranteed delivery.
VIC—Voice interface card. VICs install in a slot in the router, and provide the connection to the
telephone equipment or network. There are three types of VICs: FXS, FXO, and E&M. Each VIC
provides two ports of the same type.
VoI P—Voice-over-IP, a feature that carries voice traffic, such as telephone calls and faxes, over an IP
network, simultaneously with data traffic.
VoIP dial peer—A software object that ties together an IP address and a telephone number at a remote
site reached over the IP network (also called remote dial peer).
Priority 4 (P4)—You require information or assistance with Cisco product capabilities, installation, or
configuration. There is little or no effect on your business operations.
31
-1 (2147483647) used to identify a dial peer.
Obtaining Documentation
Cisco documentation and additional literature are available on Cisco.com. Cisco also provides several
ways to obtain technical assistance and other technical resources. These sections explain how to obtain
technical information from Cisco Systems.
Cisco.com
You can access the most current Cisco documentation at this URL:
http://www.cisco.com/univercd/home/home.htm
You can access the Cisco website at this URL:
http://www.cisco.com
You can access international Cisco websites at this URL:
• Nonregistered Cisco.com users can order documentation through a local account representative by
calling Cisco Systems Corporate Headquarters (California, USA) at 408 526-7208 or, elsewhere in
North America, by calling 800 553-NETS (6387).
Cisco 1751 Voice-over-IP Quick Start Guide
31
Documentation Feedback
Documentation Feedback
You can send comments about technical documentation to bug-doc@cisco.com.
You can submit comments by using the response card (if present) behind the front cover of your
document or by writing to the following address:
Cisco Systems
Attn: Customer Document Ordering
170 West Tasman Drive
San Jose, CA 95134-9883
We appreciate your comments.
Obtaining Technical Assistance
For all customers, partners, resellers, and distributors who hold valid Cisco service contracts, Cisco
Technical Support provides 24-hour-a-day, award-winning technical assistance. The Cisco Technical
Support Website on Cisco.com features extensive online support resources. In addition, Cisco Technical
Assistance Center (TAC) engineers provide telephone support. If you do not hold a valid Cisco service
contract, contact your reseller.
Cisco Technical Support Website
The Cisco Technical Support Website provides online documents and tools for troubleshooting and
resolving technical issues with Cisco products and technologies. The website is available 24 hours a day,
365 days a year at this URL:
http://www.cisco.com/techsupport
Access to all tools on the Cisco Technical Support Website requires a Cisco.com user ID and password.
If you have a valid service contract but do not have a user ID or password, you can register at this URL:
http://tools.cisco.com/RPF/register/register.do
Submitting a Service Request
Using the online TAC Service Request Tool is the fastest way to open S3 and S4 service requests. (S3
and S4 service requests are those in which your network is minimally impaired or for which you require
product information.) After you describe your situation, the TAC Service Request Tool automatically
provides recommended solutions. If your issue is not resolved using the recommended resources, your
service request will be assigned to a Cisco TAC engineer. The TAC Service Request Tool is located at
this URL:
http://www.cisco.com/techsupport/servicerequest
For S1 or S2 service requests or if you do not have Internet access, contact the Cisco TAC by telephone.
(S1 or S2 service requests are those in which your production network is down or severely degraded.)
Cisco TAC engineers are assigned immediately to S1 and S2 service requests to help keep your business
operations running smoothly.
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To open a service request by telephone, use one of the following numbers:
For a complete list of Cisco TAC contacts, go to this URL:
http://www.cisco.com/techsupport/contacts
Definitions of Service Request Severity
To ensure that all service requests are reported in a standard format, Cisco has established severity
definitions.
Severity 1 (S1)—Your network is “down,” or there is a critical impact to your business operations. You
and Cisco will commit all necessary resources around the clock to resolve the situation.
Severity 2 (S2)—Operation of an existing network is severely degraded, or significant aspects of your
business operation are negatively affected by inadequate performance of Cisco products. You and Cisco
will commit full-time resources during normal business hours to resolve the situation.
Severity 3 (S3)—Operational performance of your network is impaired, but most business operations
remain functional. You and Cisco will commit resources during normal business hours to restore service
to satisfactory levels.
Severity 4 (S4)—You require information or assistance with Cisco product capabilities, installation, or
configuration. There is little or no effect on your business operations.
Obtaining Additional Publications and Information
Obtaining Additional Publications and Information
Information about Cisco products, technologies, and network solutions is available from various online
and printed sources.
• Cisco Marketplace provides a variety of Cisco books, reference guides, and logo merchandise. Visit
Cisco Marketplace, the company store, at this URL:
http://www.cisco.com/go/marketplace/
• The Cisco Product Catalog describes the networking products offered by Cisco Systems, as well as
ordering and customer support services. Access the Cisco Product Catalog at this URL:
http://cisco.com/univercd/cc/td/doc/pcat/
• Cisco Press publishes a wide range of general networking, training and certification titles. Both new
and experienced users will benefit from these publications. For current Cisco Press titles and other
information, go to Cisco Press at this URL:
http://www.ciscopress.com
• Pack et magazine is the Cisco Systems technical user magazine for maximizing Internet and
networking investments. Each quarter, Packet delivers coverage of the latest industry trends,
technology breakthroughs, and Cisco products and solutions, as well as network deployment and
troubleshooting tips, configuration examples, customer case studies, certification and training
information, and links to scores of in-depth online resources. You can access Packet magazine at this
URL:
http://www.cisco.com/packet
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33
Obtaining Additional Publications and Information
• iQ Magazine is the quarterly publication from Cisco Systems designed to help growing companies
learn how they can use technology to increase revenue, streamline their business, and expand
services. The publication identifies the challenges facing these companies and the technologies to
help solve them, using real-world case studies and business strategies to help readers make sound
technology investment decisions. You can access iQ Magazine at this URL:
http://www.cisco.com/go/iqmagazine
• Internet Protocol Journal is a quarterly journal published by Cisco Systems for engineering
professionals involved in designing, developing, and operating public and private internets and
intranets. You can access the Internet Protocol Journal at this URL:
http://www.cisco.com/ipj
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Cisco 1751 Voice-over-IP Quick Start Guide
78-11259-05
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