Cisco Voip Guide

Customer Order Number: DOC-VOICEOIP-QSG=
Text Part Number: 78-4936-01
Voice over IP Quick Start Guide

Product Numbers: NM-1V=, NM-2V=
VIC-2E/M=, VIC-2FXO=, VIC-2FXS=
The voice over IP feature enables Cisco 3600 series modular routers to carry voice traffic, such as
telephone calls and faxes, over an IP network, simultaneously with data traffic. (See Figure 1.)
Figure 1 Voice and Data Traffic on an IP Network
IP cloud
10690
This guide explains briefly how to install voice hardware and how to set up basic configurations of
Cisco IOS software for a voice over IP network. It contains the following sections:
• What You Should Know, page 3—Describes what you should know before starting, what you
should do before configuring your voice network, conventions used in this guide, and how to find
more information.
• Installing Voice Network Modules and Voice Interface Cards, page 4—Describes voice
hardware and explains how to install and connect it.
• Voice Port Numbering, page 8—Explains the interface numbering convention for voice ports.
• Entering Configuration Mode, page 10—Offers hints for using Cisco IOS software to
configure a voice over IP network.
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170 West Tasman Drive
San Jose, CA 95134-1706
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All rights reserved. 1
• Configuring the IP Network for Real-Time Voice Traffic, page 12—Describes how to
configure the IP network to accommodate voice traffic.
• Configuring FXS Interfaces, page 16—Describes how to configure FXS voice interface cards
for connecting a router to telephones, fax machines, and similar devices. This section also
introduces the concepts of dial peers, which link voice ports or IP addresses with telephone
numbers, and number expansion, a shortcut to entering repeated voice-configuration commands.
• Configuring FXO Interfaces, page 24—Describes how to configure FXO voice interface cards
for connecting a router to a telephone company central office.
• Configuring E&M Interfaces, page 26—Describes how to configure E&M voice interface
cards for connections between PBXs.
• Saving the Configuration, page 29—Explains how to save the configuration so it is available
the next time you boot the router.
• List of Terms, page 30—Provides a list of terms and abbreviations used in this guide.
• Cisco Connection Online, page 32—Explains how to get service and support.
2 Voice over IP Quick Start Guide

What You Should Know
What You Should Know

This guide is intended for data communications managers or telecommunications managers who are
installing, configuring, or maintaining voice network modules and interface cards. It assumes that
you already have a working IP LAN or WAN, or that you know how to set one up, and that you want
to add voice capability to it. It does not tell you how to design or install a LAN or WAN, assign IP
addresses, install routers and configure them for IP traffic, or perform other basic tasks. It also
assumes that you are familiar with Cisco IOS software, and that you know how to configure the
routers on your LAN or WAN for IP service.
For E&M installations, you should also be familiar with telephony concepts and PBX operation.
The “List of Terms” near the end of this guide provides a glossary of many terms and concepts used
by voice over IP. For further information on voice over IP features, refer to the Voice
Network Module and Voice Interface Card Configuration Note, the Voice over IP Software
Configuration Guide, your router installation and configuration guide, the Regulatory Compliance
and Safety Information document for your router, and the Cisco IOS configuration guides and
command references.
Document Conventions
In this guide:
• Commands in the text are shown in boldface.
• Variables for which you supply values are in italics.
• Italics are also used for the titles of publications and for new words or concepts.
• Information the router displays on the console screen is in screen font.
• Information that you enter at the console keyboard is in boldface screen font.
To Learn More
Cisco documentation and additional literature are available in a CD-ROM package, which ships with
your product. The Documentation CD-ROM, a member of the Cisco Connection Family, is updated
monthly. Therefore, it might be more up to date than printed documentation. To order additional
copies of the Documentation CD-ROM, contact your local sales representative or call customer
service. The CD-ROM package is available as a single package or as an annual subscription.
You can also read Cisco documentation on the World Wide Web at http://www.cisco.com,
http://www-china.cisco.com, or http://www-europe.cisco.com. From here, you can submit
comments electronically. Click Feedback on the title bar, and then select Documentation. After you
complete the form, click Submit to send it to Cisco. We appreciate your comments.
If you have questions or need help, refer to the section “Cisco Connection Online” at the end of this
guide for further information.
Voice over IP Quick Start Guide 3

Installing Voice Network Modules and Voice Interface Cards

The voice over IP feature for Cisco 3600 series routers uses two types of hardware:
• Voice network modules
• Voice interface cards (VICs)
Note To transmit voice calls over an IP LAN or WAN, you need (in addition to the voice hardware)
at least one other network module or WAN interface card in the router to provide the connection to
the LAN or WAN. In most cases, this network module, or another network module or WAN interface
card in the router, also serves to carry data traffic.
You should install and cable voice modules and VICs before performing the software configuration
tasks explained later. If you need more detailed installation instructions, refer to the Voice
Network Module and Voice Interface Card Configuration Note.
Warning Be sure to observe all warnings and safety precautions in the configuration note.
Caution Network modules and voice interface cards do not support online insertion and removal
(hot swap). To avoid equipment damage, before you insert a network module into a chassis slot, or
insert a voice interface card into a voice network module in the router chassis, you must turn OFF
electrical power and disconnect network cables.
Voice Network Modules

Voice network modules install in a slot in a Cisco 3600 series router, and convert telephone voice
signals into a form that can be transmitted over an IP network. The one-slot voice network module
provides one slot for a voice interface card. The two-slot voice network module provides two slots
for voice interface cards. Figure 2 shows a two-slot voice network module.
Figure 2 Two-Slot Voice Network Module
Slots for voice interface cards
VOICE
2V
V1 V0
EN
10692

Voice Interface Cards (VICs)
To install a voice network module, slide it into a slot in the router chassis (Figure 3). Use a number 1
Phillips or small flat-blade screwdriver to fasten the module’s mounting screws.
Figure 3 Installing a Voice Network Module in a Router
3
2E
NT1
W1
ACT
2W
WO
B1
B2
BRI
NT1 SERIAL
0
SEE MANUAL BEFORE INSTALLATION
H10841
VOICE
4V
LNK
ACT
LNK
ACT
1 ETHERNET 1
ETHERNET 0 V1
V0
INPUT 100-240VAC 50/60HZ 3.0-1.5 AMPS
EN
Router
Module
Voice Interface Cards (VICs)

Voice interface cards install in slots in the voice network module, and provide connections to the
telephone equipment or network. Figure 4 shows a typical VIC.
Figure 4 Voice Interface Card
VIC ports
IN USE
IN USE
VIC
E&M
10695
1 0
There are three types of VIC interface:

• An 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.
• An FXO (foreign exchange office) interface connects local calls to a public switched telephone
network 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 (ear and mouth) 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.
Each VIC provides two ports. You need one VIC port for each voice connection.
You can install one VIC (two voice ports) in a one-slot voice network module, and two VICs (four
voice ports) in a two-slot voice network module.

To install a VIC, slide it into a slot in the voice network module, as shown in Figure 5. Use a
number 1 Phillips or small flat-blade screwdriver to fasten the VIC’s mounting screws.
Figure 5 Installing a Voice Interface Card in a Voice Network Module
3
2E
NT1
W1
ACT
2W VOICE
WO
B1 4V
0
IN USE
BRI
IN USE
B2 NT1 SERIAL
VIC
FXO
H10842
1
V1 SEE MANUAL BEFORE INSTALLATION 0
IN USE
IN USE
V0
LNK
ACT
LNK
ACT
VIC
FXO
1 ETHERNET 1
ETHERNET 0
1
SEE MANUAL BEFORE INSTALLATION 0 EN
INPUT 100-240VAC 50/60HZ 3.0-1.5 AMPS
Module
Router

Connecting VICs to the Network
Connecting VICs to the Network

Figure 6 shows how to connect VICs to the network:
• Use a standard RJ-11 modular telephone cable to connect FXS VIC ports (color-coded gray) to
a telephone or fax machine.
• Use a standard RJ-11 modular telephone cable to connect FXO VIC ports (color-coded pink) to
the public switched telephone network, or to a PBX that does not support E&M signaling,
through a telephone wall outlet.
• The E&M VIC uses an RJ-48S connector and cable. The cable wiring depends on the type of
connection. See the Voice Network Module and Voice Interface Card Configuration Note for
details.
Figure 6 Connecting VICs to the Network
FXS VIC
IN USE
IN USE
1 0
FXO VIC
IN USE
IN USE
1 0
E&M VIC
10691
PBX
IN USE
IN USE
1 0
When you are finished, reinstall any network interface cables you removed and turn ON power to
the router.
Checking the 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 dial tone when you lift the handset. 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. If
it is, try connecting a different handset to the VIC.

Voice Port Numbering
Voice Port Numbering

Cisco IOS configuration commands identify voice ports in the form
router-slot/voice-slot/VIC-port
12
9 3
Timesaver You can use the Cisco IOS show voice port command to identify the port numbers of
6
voice interfaces installed in your router.
Figure 7 shows Cisco 3620 router slot numbering.
Figure 7 Cisco 3620 Slot Numbers
H7238
2E
W1
NT1
2W
B1
ACT
BRI WO 2E W1
B2 NT1 2W DO NOT INSTALL WAN INTERFACE
SEE MANUAL BEFORE SERIAL CARDS WITH POWER APPLIED
ACT
INSTALLATION
ACT
LNK
SERIAL
ACT
LNK
AUI
1
ACT
LNK
ETHERNET 1 EN
ACT
LNK
ETHERNET 0 AUI
ETHERNET
ETH 1 1
EN 0
ETHERNET 0
Slot 1
Slot 0
Figure 8 shows Cisco 3640 router slot numbering.
Figure 8 Cisco 3640 Slot Numbers
Slot 3 Slot 2
3
2E
NT1
W1
ACT
2W
B1 WO 2E W1 DO NOT INSTALL WAN INTERFACE
2W
ACT
BRI CARDS WITH POWER APPLIED

B2 NT1 SERIAL
SERIAL
H6551
LNK
ACT
STP
LNK
ACT
AUI
LNK
ACT
ACT
LNK
AUI
EN
1 ETHERNET 1 ETHERNET 0
ETHERNET 1 ETHERNET 0
EN
INPUT 100-240VAC 50/60HZ 3.0-1.5 AMPS
Slot 1 Slot 0 Power supply

Example
Figure 9 shows voice slot numbering.
Figure 9 Voice Slot Numbering
Voice slot 1 Voice slot 0
VOICE
2V
V1 V0
EN
10693
Figure 10 shows VIC port numbering.
Figure 10 VIC Port Numbering
VIC port 1 VIC port 0

IN USE
IN USE
VIC
E&M
10694
1 0
Example
Suppose you install a two-slot voice network module in the upper right slot of a Cisco 3640 router,
and install two VICs in the module. Each VIC has two ports. From right to left, these ports would
be numbered 2/0/0, 2/0/1, 2/1/0, and 2/1/1.

Entering Configuration Mode

You configure Cisco IOS software for voice over IP by typing commands on the command line. This
method of entering commands is called configuration mode.
Note Voice over IP commands require the IP Plus, Desktop Plus, or Enterprise Plus image,
Cisco IOS Release 11.3(1)T or later.
To enter configuration mode, follow this procedure:

Step 1 Connect a console to the router. Configure the console for 9600 baud, 8 data bits, 1 stop
bit, and no parity. Power up the router.
Step 2 If the current configuration is no longer valid (for instance, because you added an
interface), after about one minute you see the following prompt:
Would you like to enter the initial configuration dialog? [yes]:
Answer no. You now enter the normal operating mode of the router.
Note If the current configuration is valid, you enter the normal operating mode
automatically.
Step 3 After a few seconds, you see the user EXEC prompt (Router>). Type 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#):
Router#
Step 4 Enter the config terminal command to enter configuration mode:

Router# config terminal
Router(config)#
The router enters global configuration mode, indicated by the Router(config)# prompt.
Step 5 If you have not configured the router before, or want to change the configuration,
configure 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
For complete information about 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 Cisco IOS software for voice over IP traffic.
12
9 3
Timesaver Voice configuration uses a number of new Cisco IOS commands. For complete
6
information about these commands, see the Voice over IP Software Configuration Guide. You can
also enter a question mark after a command or partial command at the Router(config)# prompt
to get help with syntax and arguments.
Remember that when you finish configuring the software, you must save the new configuration to
NVRAM. This procedure is described in the section “Saving the Configuration” later in this guide.
You may also want to save periodically during the configuration process.
12
9 3
Timesaver At any point, you can see the current operating configuration, including changes you
6
just made, by entering the show running-config command.

Configuring the IP Network for Real-Time Voice Traffic

Voice traffic is much more sensitive to timing variations than data traffic. For good voice
performance, you might need to configure your data network so voice packets are not lost or delayed.
This section describes three important methods of improving quality of service, or QoS (the level of
network performance needed for voice over IP connections):
• RSVP
• Multilink PPP interleaving
• RTP header compression
Cisco IOS software provides many other tools for ensuring QoS, such as custom queuing, priority
queuing, weighted fair queuing, and IP precedence. For further information and more detailed
examples of QoS configuration, see the Voice over IP Software Configuration Guide.
Note QoS measures the level of network performance. It does not directly measure the quality of
the voice signal.
Configuring voice over IP on a Frame Relay link involves special considerations. These are
discussed in the section “Configuring Frame Relay for Voice over IP.”
Do You Need QoS?

On a relatively low-bandwidth connection, such as a PPP or 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 may not need to provide QoS. (See Figure 11.) If you are not sure, you may want to try skipping
the following sections and continuing with “Configuring FXS Interfaces” later in this guide. Return
to this section if you need to improve the reliability of voice connections.
Figure 11 Bandwidth v. Quality of Service
Low bandwidth
Need QoS
High bandwidth
10689
May not need QoS

Do You Need RSVP?
Do You Need RSVP?

Resource Reservation Protocol (RSVP) enables routers to reserve enough bandwidth for reliable
performance. RSVP works well on PPP, HDLC, and similar serial line interfaces. It does not work
well on multiaccess 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
Note If you configure multilink PPP interleaving, you can use the ip rtp reserve command instead
of configuring RSVP. See the next section on “Do You Need Multilink PPP Interleaving?”
Configuring RSVP
By default, RSVP is disabled for compatibility 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 shows how to configure RSVP on serial interface 0/0:
Router> enable
Password:
Router# configure terminal
Router(config)# interface serial 0/0
Router(config-if)# ip rsvp bandwidth
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. You
must also configure each remote dial peer to request an RSVP session, using the req-qos command.
See the section “Calling Between Routers” later in this guide.
Do You Need Multilink PPP Interleaving?

On a dialer, ISDN PRI or BRI interface, or a virtual template, you can configure multilink PPP
interleaving, which helps to make sure that voice packets are transmitted without delay.
You should configure multilink PPP interleaving if you have one of these interfaces, and either of
the following:
• Point-to-point connections using PPP encapsulation
Note Do not use multilink PPP on links faster than 2 Mbps.

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:
Router(config-if)# encapsulation ppp
Router(config-if)# ppp multilink
Router(config-if)# ppp multilink interleave
Optionally, configure a maximum fragment delay:

Router(config-if)# ppp multilink fragment-delay milliseconds
You can also reserve a special queue for real-time packet flows to specified destination UDP ports,
allowing real-time traffic to have higher priority than other flows. This command is needed 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
Do You Need 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:
• Need to save bandwidth
Note Do 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 command
in interface configuration mode:
Router(config-if)# ip rtp header-compression

Configuring Frame Relay for Voice over IP
Configuring Frame Relay for Voice over IP

Configuring voice over IP 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), where data
and voice are being transmitted over the same PVC, you should configure the following:
• Lower 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
• RSVP—Configure RSVP on the subinterfaces (which correspond to PVCs) to reserve bandwidth

for voice channels. See the section on “Do You Need 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
• RTP header compression—Configure RTP header compression on the subinterfaces to minimize

the size of the voice packet. See the previous section on “Do You Need RTP Header
Compression?”
The following command configures RTP header compression over the currently 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
might be discarded. In Cisco IOS Release 11.3, Frame Relay traffic shaping is not compatible
with RSVP. Use generic traffic shaping instead, setting the CIR equal to the port speed. Doing
this prevents the router from exceeding the CIR rate, which could lead to frames being discarded.
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, see the Voice
over IP Software Configuration Guide.

Configuring FXS Interfaces

This section explains how to configure FXS VICs. Ports on this VIC connect directly to a standard
telephone, fax machine, or similar device.
Figure 12 shows a basic voice network. A small business uses a Cisco 3600 series router (named
West) to provide telephone and fax connections among employees in its office. Figure 12 shows two
of these telephones, each connected to an FXS VIC port in the West router.
Figure 12 Basic Voice Network (West Router)
408 555-3737
FXS VIC IP cloud

2/0/0
West
FXS VIC
2/0/1
408 555-4141
10007
Note You can name your router by using the hostname command in global configuration mode.
Table 1 lists telephone numbers and voice ports for the West router. (For information about port
numbering, see the section on “Voice Port Numbering” earlier in this guide.)
Table 1 West Router Telephone Numbers and Voice Ports
Telephone Number Voice Port

408 555-3737 2/0/0
408 555-4141 2/0/1
Note Additional telephones and fax machines could be connected in this example, up to a total of
12 if the router is a Cisco 3640 configured with three two-slot voice network modules. The module
in the remaining slot would provide interfaces for IP connectivity to the LAN or WAN and for data
traffic. To accommodate more than 12 voice devices, you would need to add more routers, or to use
an E&M VIC and a local PBX, rather than connecting every telephone to its own FXS VIC.

Local Dial Peers
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 2/0/0 (as shown in Figure 12). In other words,
the router needs to know the information in Table 1.
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.
Dial peers are identified by numbers, but to avoid confusing these numbers with telephone numbers,
they are usually referred to as tags. Dial peer tags are integers that can range from 1 to 231 -1
(2147483647), which should be enough for most purposes. Dial peers on the same router must have
unique tags, but you can reuse the tags on other routers.
Table 2 assigns each phone number-voice port pair on the West router a dial peer tag. Within the
allowed range, you can choose any dial peer tag or any system that is convenient or makes sense to
you. 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. (Another type of dial peer is explained in the section “Calling Between Routers.”)
Table 2 West Router Local Dial Peers
Telephone Number Voice Port Dial Peer Tag

408 555-3737 2/0/0 401
408 555-4141 2/0/1 402
You should construct a table similar to Table 2 for your own routers, assigning your own telephone
numbers and dial peer tags.
Note The telephone numbers used in this guide are only examples, and are generally 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 information in the table, enter the following commands in global
configuration mode:
West(config)# dial-peer voice 401 pots
West(config-dial-peer)# dest-pat +14085553737
West(config-dial-peer)# port 2/0/0
West(config-dial-peer)# dest-pat +14085554141
West(config-dial-peer)# exit
West(config)#
These commands are summarized in Figure 13.

Figure 13 West Router Configured for Local Dial Peers
dial-peer voice 401 pots

dest-pat +14085553737
port 2/0/0
FXS VIC IP cloud

2/0/0
West
FXS VIC
2/0/1
10008
dest-pat +14085554141
port 2/0/1
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
(abbreviated dest-pat) command. A destination pattern always begins with a plus sign (+). It can
also include asterisks (*) and pound signs (#) from the telephone keypad, and commas (,) and
periods (.), which have special meaning. 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 append the prefix 1 (used in the United States to indicate
a long-distance number) and an area code to the destination pattern. You may need to include similar
codes for your country if the voice over IP equipment needs to establish a connection to the PSTN.
In other situations, you might be able to simplify configuration by omitting this information.
Note The 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 West router also has a branch office in the East. Figure 14 shows the East
network, and Table 3 lists phone numbers, voice ports, and dial peer tags for this office.
Figure 14 Basic Voice Network (East Router)
919 958-8282
IP cloud FXS VIC

3/1/0
East
FXS VIC
3/1/1
10009
919 958-9595

Local Dial Peers
Table 3 East Router Local Dial Peers
Telephone Number Destination Pattern Voice Port Dial Peer Tag

919 958-8282 +19199588282 3/1/0 901
919 958-9595 +19199589595 3/1/1 902
The following commands configure local ports on the East Router:

East(config)# dial-peer voice 901 pots
East(config-dial-peer)# dest-pat +19199588282
East(config-dial-peer)# port 3/1/0
East(config-dial-peer)# dest-pat +19199589595
East(config-dial-peer)# exit
East(config)#
Figure 15 East Router Configured for Local Dial Peers

dest-pat +19199588282
port 3/1/0
IP cloud FXS VIC
3/1/0
East
FXS VIC
3/1/1
10010
dest-pat +19199589595
port 3/1/1
Checking the Configuration

If you configured your router following these examples, you should now be able to 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.
12
9 3
Timesaver If the voice port is offline, use the no shutdown command from interface
6
configuration mode to enable it.
Note Although placing calls directly between ports on the same router helps to verify your
configuration, it is not recommended for general telecommunications use.

Wild Cards and Number Expansion

Office PBXs are usually configured so 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 voice over IP network by using the number-expansion
(num-exp) command. This command tells the router to expand a particular sequence of dialed
numbers into a complete telephone number (destination pattern). For instance, to expand 3737 into
408 555-3737, enter the following command:
West(config)# num-exp 3737 +14085553737
To expand 4141 into 1 408 555-4141, enter the following command:

West(config)# num-exp 4141 +14085554141
More generally, you can use the period (.) as a wild-card character representing a single digit. For
instance, the command
West(config)# num-exp .... +1408555....
expands any dialed sequence of four digits by prefixing +1408555 to it.
Note You must still configure each telephone number in full as a local dial peer so the router can
find the voice port it belongs to.
To use five-digit extensions beginning with the numeral 5 rather than four-digit extensions, you
would enter the following command:
West(config)# num-exp 5.... +1408555....
The corresponding commands for the East router would be (for four-digit extensions):
East(config)# num-exp .... +1919958....
or for five-digit extensions:

East(config)# num-exp 8.... +1919958....

If you followed the examples, you should now be able to place calls between telephones connected
to the same router using the extension instead of the full telephone number. You can use the show
num-exp command to verify that the data you configured is correct.

Calling Between Routers

Naturally, the West and East offices would like to send voice traffic to each other over the same IP
network they use for data traffic. A WAN port of some type on each router connects to the IP WAN,
as shown in Figure 16.
Figure 16 IP Connection Between Routers
919 958-8282
408 555-3737
FXS VIC
FXS VIC IP cloud 3/1/0
2/0/0
192.168.11.3
East
192.168.19.27
West FXS VIC
FXS VIC 3/1/1
2/0/1
408 555-4141
10015
919 958-9595
Look at the connection between the West 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 West router has to associate a telephone number in the East office with the IP address of the East
router.
This is done by associating both pieces of information with a remote dial peer or VoIP dial peer on
the West router, as shown in Table 4. (Remember, the dial peer tags are arbitrary.) The term “VoIP”
(voice over IP) means that the dial peer associates a telephone number with an IP address.
Table 4 West Router Remote Dial Peers
Remote Location Telephone Number Destination Pattern IP Address Dial Peer Tag
East 919 958-8282 +19199588282 192.168.11.3 501
East 919 958-9595 +19199589595 192.168.11.3 502
To do this, you could create a remote dial peer on the West router for every telephone on the East
router, all associated with the same IP address. But it is much easier to use periods as wild cards, as
shown in Table 5.
Table 5 West Router Remote Dial Peers with Wild Cards
Remote Location Telephone Number Destination Pattern IP Address Dial Peer Tag
East 919 958-xxxx +1919958.... 192.168.11.3 501
You should construct a table similar to Table 5 for your own routers, assigning your own telephone
numbers, IP addresses, and dial peer tags.

Note The IP addresses shown in this guide are reserved, and are meant only as examples. When
you configure your network, be sure to substitute your own IP addresses.
Now you need to enter only the following commands on the West router from global configuration
mode:
West(config)# dial-peer voice 501 voip
West(config-dial-peer)# dest-pat +1919958....
West(config-dial-peer)# session-target ipv4:192.168.11.3
Cisco IOS software calls the remote network 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 DNS name—for instance:
West(config-dial-peer)# session-target dns:voice.eastrouter.com
Figure 17 West Router Configured for Remote Dial Peers
IP cloud
192.168.11.3 919 958-xxxx
West East
dial-peer voice 501 voip

dest-pat +1919958….
session-target ipv4:192.168.11.3
num-exp 8…. +1919958….
10011
Source Destination
In general, you have to configure a dial peer on each router for every telephone number (or group of
wild cards) on every other router.
You can make things even easier by configuring number expansion for East router telephone
numbers on the West router:
Now users can dial a five-digit extension beginning with 8 from a telephone on the West router to
reach a telephone on the East router.
Now the West router is configured to send calls to the East router. Table 6 shows how to configure
the East router to send calls to the West router.
Table 6 East Router Remote Dial Peers with Wild Cards
Remote Location Telephone Number IP Address Dial Peer Tag

West 408 555-xxxx 192.168.19.27 801

To create this configuration, you would enter the following commands on the East router:
East(config)# num-exp 5.... +1408555....
East(config)# dial-peer voice 801 voip
East(config-dial-peer)# dest-pat +1408555....
East(config-dial-peer)# session-target ipv4:192.168.19.27
Figure 18 East Router Configured for Remote Dial Peers
IP cloud
408 555-xxxx 192.168.19.27
West East

dest-pat +1408555….
session-target ipv:192.168.19.27
10012
Destination Source
Requesting RSVP
If you configured RSVP on the WAN interface (see the section “Do You Need RSVP?” earlier in this
guide), then you must also configure each VoIP dial peer to request an RSVP session, using the
req-qos command:
West(config-dial-peer)# req-qos controlled-load
Otherwise no bandwidth is reserved for voice traffic. For further information about this command,
see the Voice over IP Software Configuration Guide.
Other Routers on the Network

If the path between endpoints of a voice call travels through intermediate routers, you should
configure those routers for VoIP traffic, as described in the section “Configuring the IP Network for
Real-Time Voice Traffic” earlier in this guide.
You need to configure local or remote dial peers on an intermediate router only if that router also has
voice devices attached.

After you configure remote dial peers (VoIP dial peers) on a router, you should be able to place calls
from that router to telephones on the remote routers (using just the extension if you configured
number expansion). If you have trouble, use the show dial-peer voice command to verify that the
data you configured is correct, and ping the remote router to make sure you have connectivity.

Configuring FXO Interfaces
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 users can reach telephones and fax machines outside the VoIP
network. Figure 19 shows a typical PSTN gateway attached to the West router.
Figure 19 FXO Gateway to PSTN
IP cloud
Source
West
FXO VIC
1/0/0
Destination
dest-pat +9………
port 1/0/0
PSTN
10013
cloud
To create a local (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. When you
create a local dial peer for an FXO interface, however, the destination pattern refers to outgoing calls,
and you can include wild cards in it, because the PSTN performs the switching.
The VoIP feature can also remove digits that you don’t want to send to the PSTN. For instance,
suppose you want to dial 9 to reach an outside line (that is, the analog PSTN). You would enter
commands similar to these:
West(config-dial-peer)# dest-pat +9...........
Now, when you dial 9 followed by an 11-digit telephone number from a telephone attached to the
West router, your call is connected through voice port 1/0/0 to the PSTN. The router software
automatically removes the fixed part of the destination pattern—in this case, the digit 9—and sends
the remaining 11 digits to the PSTN.
Note The fixed part of the destination pattern is removed only on calls sent to analog (FXO)
interfaces. The entire number is always sent to digital (FXS and E&M) interfaces.
To enable East router users to make calls over the West router’s local PSTN, you could enter
commands similar to these:
East(config-dial-peer)# dest-pat +7...........

West(config-dial-peer)# dest-pat +7...........

The East router now sends all calls whose numbers begin with the special prefix 7 over the IP
network to the West router. (The entire number is sent, including the 7.) The West router removes
the 7 and passes the calls through its analog FXO gateway to the local PSTN.
Note In this example, West voice port 1/0/0 has two separate POTS dials peers associated with it.
Dial peer 201 matches calls beginning with the digit 9, and handles PSTN calls originating from the
West router. Dial peer 601 matches calls beginning with the digit 7, and handles calls to the West
PSTN originating from the East router.

If you configured your FXO interface according to this example, you should now be able to place
outgoing calls over the PSTN. If you have trouble, 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.

Configuring E&M Interfaces

If you have more than a few voice users at each location, then 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 may be more efficient to use a PBX at each location to switch
local traffic and to direct incoming calls, and to connect the PBXs over an IP network using E&M
voice interface cards.
The following example again shows a company with two offices, West and East. Now each office
has a PBX to operate its internal telephone network, while the IP network carries voice traffic
between the offices. Figure 20 illustrates the topology of this connection. Each PBX connects to the
IP router over an E&M interface connection.
Figure 20 Linking PBXs over the IP Network (Local Dial Peers)
E&M VIC IP cloud

408 555-xxxx 3/0/0 192.168.11.3 919 958-xxxx
PBX PBX
West East

dest-pat +1408555….
port 3/0/0
10014
Destination Source
Both PBXs in this example use E&M interface Type 2, with four-wire operation and immediate-start
signaling. The values you should use for your configuration depend on your PBX, and should be
available from your telecommunications department or the PBX manufacturer. For more
information about E&M interface configuration commands, see the “Voice over IP Commands”
chapter of the Voice over IP Software Configuration Guide.
In the following configuration, West users can dial 8 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 West router is connected to the PBX over E&M port 3/0/0. That means that this port is
associated with local (POTS) dial peers 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 wild-card destination pattern) are connected directly to this port, as
shown in the following commands:
Remote (VoIP) dial peers for outgoing calls associate destination phone numbers on the East router
with that router’s IP address, just as before (see Figure 21):
West(config)# dial-peer voice 121 voip
West(config-dial-peer)# session-target ipv4:192.168.11.3
West(config-dial-peer)# exit
West(config)#

Figure 21 Linking PBXs over the IP Network (Remote Dial Peers)
E&M VIC IP cloud

408 555-xxxx 3/0/0 192.168.11.3 919 958-xxxx
PBX PBX
West East
dest-pat +1919958….
session-target ipv4:192.168.11.3
num-exp 8…. +1919958….
10696
Source Destination
Now configure number expansion, so that numbers beginning with 8 (belonging to the East office)
and sent by the West PBX to the West router are expanded into the full destination pattern:
Note You do not need to configure number expansion for calls from one West telephone to another
West telephone, because the PBX switches those calls.
Finally, configure the E&M port similarly to any other network interface, using the voice-port
command from global configuration mode:
West(config)# voice-port 3/0/0
West(config-voice-port)# signal immediate
West(config-voice-port)# operation 4-wire
West(config-voice-port)# type 2
Note Configure the PBX to pass all DTMF signals to the router.
Configuration for the East router is similar. The PBX for this router is connected to E&M port 2/0/1.
The following commands configure a local dial peer for all East telephones:
These commands configure a remote dial peer for telephones on the West router:
East(config-dial-peer)# exit
East(config)#
Configure number expansion, to make it easy for East users to dial numbers on the West router:

Finally, configure the E&M port:

East(config)# voice-port 2/0/1
East(config-voice-port)# signal immediate
East(config-voice-port)# operation 4-wire
East(config-voice-port)# type 2

If you configured the E&M interfaces correctly, you should now be able to 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, ping the remote router to make sure you
have connectivity.

Saving the Configuration
Saving the Configuration

When you finish router configuration, follow this procedure for each router to write the new
configuration to NVRAM:
Step 1 Exit configuration mode and return to the enable prompt by pressing Ctrl-Z. To see the
current operating configuration, including any changes you just made, enter the show
running-config command:
Router# show running-config
To see the configuration currently stored in NVRAM, enter the show startup-config
command at the enable prompt:
Router# show startup-config
Step 2 The results of the show running-config and show startup-config commands differ from
each other if you have made changes to the configuration, but have not yet written them
to NVRAM. To write your changes to NVRAM, making them permanent, enter the
copy running-config startup-config command at the enable prompt:
Router# copy running-config startup-config
Building configuration. . .
[OK]
Router#
The router is now configured to boot in the new configuration.

If you have questions or need help, see the last section, “Cisco Connection Online.”

List of Terms
List of Terms
This section defines some of the terms and concepts used by voice over IP.
BRI—Basic Rate Interface.
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.
CIR—committed information rate. The average rate of information transfer a subscriber (for
example, the network administrator) has stipulated for a Frame Relay PVC.
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 VoIP
dial peer). Each dial peer corresponds to a call leg.
E&M—ear and mouth (or recEive and transMit), a signaling technique for two-wire and four-wire
interfaces between PSTN central offices or PBXs.
FXO—foreign exchange office, a type of VIC interface. The FXO VIC connects local calls to a
PSTN central office or PBX over a standard RJ-11 modular telephone cable. 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.
IPX—Internetwork Packet Exchange.
ISDN—Integrated Services Digital Network.
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).
MTU—maximum transmission unit.
Multilink PPP—Multilink Point-to-Point Protocol, a method of splitting, recombining, and
sequencing datagrams across multiple logical data links.
PBX—private branch exchange, a private telephone switching system.
POTS—plain old telephone service.
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.
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 voice over IP connection.
remote 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 VoIP dial peer).
RSVP—resource reservation protocol, a network protocol that enables routers to reserve the
bandwidth necessary for reliable performance.
RTP—real-time transport protocol (RTP), a network protocol used to carry packetized audio and
video traffic over an IP network.
List of Terms
session target—a remote IP or DNS address specified as one end of a voice connection.
tag—a positive integer in the range 1 to 231 -1 (2147483647) used to identify a dial peer.
UDP—User Datagram Protocol.
VIC—voice interface card. VICs install in a slot in a voice network module, and provide the
connection to the telephone equipment or network. There are three types: FXS, FXO, and E&M
interface. Each VIC provides two ports of the same type.
voice network module—a network module that installs in a slot in a Cisco 3600 series router,
converts telephone voice signals into a form that can be transmitted over an IP network, and provides
one or two slots for voice interface cards.
VoIP—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).

Cisco Connection Online
Cisco Connection Online

Cisco Connection Online (CCO) is Cisco Systems’ primary, real-time support channel. Maintenance
customers and partners can self-register on CCO to obtain additional information and services.
Available 24 hours a day, 7 days a week, CCO provides a wealth of standard and value-added
services to Cisco’s customers and business partners. CCO services include product information,
product documentation, software updates, release notes, technical tips, the Bug Navigator,
configuration notes, brochures, descriptions of service offerings, and download access to public and
authorized files.
CCO serves a wide variety of users through two interfaces that are updated and enhanced
simultaneously: a character-based version and a multimedia version that resides on the World Wide
Web (WWW). The character-based CCO supports Zmodem, Kermit, Xmodem, FTP, and Internet
e-mail, and it is excellent for quick access to information over lower bandwidths. The WWW version
of CCO provides richly formatted documents with photographs, figures, graphics, and video, as well
as hyperlinks to related information.
You can access CCO in the following ways:
• WWW: http://www.cisco.com
• WWW: http://www-europe.cisco.com
• WWW: http://www-china.cisco.com
• Telnet: cco.cisco.com
• Modem: From North America, 408 526-8070; from Europe, 33 1 64 46 40 82. Use the
following terminal settings: VT100 emulation; databits: 8; parity: none; stop bits: 1; and
connection rates up to 28.8 kbps.
For a copy of CCO’s Frequently Asked Questions (FAQ), contact cco-help@cisco.com. For
additional information, contact cco-team@cisco.com.
Note If you are a network administrator and need personal technical assistance with a Cisco
product that is under warranty or covered by a maintenance contract, contact Cisco’s Technical
Assistance Center (TAC) at 800 553-2447, 408 526-7209, or tac@cisco.com. To obtain general
information about Cisco Systems, Cisco products, or upgrades, contact 800 553-6387,
408 526-7208, or cs-rep@cisco.com.
Use this document in conjunction with your router installation and configuration guide, the Regulatory Compliance and Safety Information document for your router, the Voice
Network Module and Voice Interface Card Configuration Note, the Voice over IP Software Configuration Guide, and the Cisco IOS configuration guides and command references.
AccessPath, AtmDirector, the CCIE logo, CD-PAC, Centri, Centri Bronze, Centri Gold, Centri Security Manager, Centri Silver, the Cisco Capital logo, Cisco IOS, the Cisco IOS logo,
CiscoLink, the Cisco NetWorks logo, the Cisco Powered Network logo, the Cisco Press logo, ClickStart, ControlStream, Fast Step, FragmentFree, IGX, JumpStart, Kernel Proxy, LAN2LAN
Enterprise, LAN2LAN Remote Office, MGX, MICA, Natural Network Viewer, NetBeyond, NetRanger, NetSonar, Netsys Technologies, Packet, PIX, Point and Click Internetworking,
Policy Builder, RouteStream, Secure Script, SMARTnet, StrataSphere, StrataSphere BILLder, StrataSphere Connection Manager, StrataSphere Modeler, StrataSphere Optimizer, Stratm,
StreamView, SwitchProbe, The Cell, TrafficDirector, TransPath, VirtualStream, VlanDirector, Workgroup Director, Workgroup Stack, and XCI are trademarks; Empowering the Internet
Generation and The Network Works. No Excuses. are service marks; and BPX, Catalyst, Cisco, Cisco Systems, the Cisco Systems logo, EtherChannel, FastHub, FastPacket, ForeSight, IPX,
LightStream, OptiClass, Phase/IP, StrataCom, and StrataView Plus are registered trademarks of Cisco Systems, Inc. in the U.S. and certain other countries. All other trademarks mentioned
in this document are the property of their respective owners.
Copyright © 1998, Cisco Systems, Inc.

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Voice over IP Quick Start Guide

Figure 1 Voice and Data Traffic on an IP Network

2 Voice over IP Quick Start Guide

What You Should Know

Voice over IP Quick Start Guide 3

Installing Voice Network Modules and Voice Interface Cards

Voice Network Modules

Figure 2 Two-Slot Voice Network Module

Slots for voice interface cards

4 Voice over IP Quick Start Guide

Figure 3 Installing a Voice Network Module in a Router

Voice Interface Cards (VICs)

Figure 4 Voice Interface Card

There are three types of VIC interface:

Voice over IP Quick Start Guide 5

Figure 5 Installing a Voice Interface Card in a Voice Network Module

INPUT 100-240VAC 50/60HZ 3.0-1.5 AMPS

6 Voice over IP Quick Start Guide

Connecting VICs to the Network

Figure 6 Connecting VICs to the Network

Checking the Installation

Voice over IP Quick Start Guide 7

Voice Port Numbering

Figure 7 shows Cisco 3620 router slot numbering.

Figure 7 Cisco 3620 Slot Numbers

Figure 8 shows Cisco 3640 router slot numbering.

Figure 8 Cisco 3640 Slot Numbers

BRI CARDS WITH POWER APPLIED

INPUT 100-240VAC 50/60HZ 3.0-1.5 AMPS

Slot 1 Slot 0 Power supply

8 Voice over IP Quick Start Guide

Figure 9 shows voice slot numbering.

Figure 9 Voice Slot Numbering

Voice slot 1 Voice slot 0

Figure 10 VIC Port Numbering

VIC port 1 VIC port 0

Voice over IP Quick Start Guide 9

Entering Configuration Mode

To enter configuration mode, follow this procedure:

Step 4 Enter the config terminal command to enter configuration mode:

10 Voice over IP Quick Start Guide

Voice over IP Quick Start Guide 11

Configuring the IP Network for Real-Time Voice Traffic

Do You Need QoS?

Figure 11 Bandwidth v. Quality of Service

May not need QoS

12 Voice over IP Quick Start Guide

Do You Need RSVP?

Do You Need Multilink PPP Interleaving?

Note Do not use multilink PPP on links faster than 2 Mbps.

Voice over IP Quick Start Guide 13

Configuring Multilink PPP Interleaving

Optionally, configure a maximum fragment delay:

Do You Need RTP Header Compression?

Configuring RTP Header Compression

14 Voice over IP Quick Start Guide

Configuring Frame Relay for Voice over IP

• RSVP—Configure RSVP on the subinterfaces (which correspond to PVCs) to reserve bandwidth