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Voice-over-IP (VoIP) enables a Cisco 1750 router (hereafter referred to as the router) to carry voice traffic (for example, telephone calls and faxes) over an IP network, simultaneously with data traffic. (See Figure 1.) This guide describes the configuration tools and the major configuration tasks to configure the router.
You can configure VoIP by using one of the following tools:
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:
Step 2 Close all programs active on your PC.
Step 3 Insert the CD into your CD-ROM drive.
Step 4 The Cisco ConfigMaker installer starts automatically.
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 5 Follow 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.
To configure your router from the command-line interface (CLI), refer to this guide, the Cisco 1700 Series Software Configuration Guide, or the Cisco 1750 Router Voice-over-IP Configuration Guide available on the Documentation CD-ROM.
Follow these major steps to configure VoIP on your router:
Step 2 Configure your router to support voice ports. Follow these steps for your specific requirements:
Step 3 Enter configuration mode to configure your router for VoIP traffic. Refer to the "Entering Configuration Mode" section.
Step 4 Save the configuration. Refer to the "Saving the Configuration" section.
Step 5 Establish 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.
Step 6 Fine-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.
International users must set the values of certain voice-port commands, like cptone, that are specific to their country. Refer to the Cisco 1750 Router Voice-over-IP Configuration Guide for details.
In the following sections, this guide explains how to install the voice interface cards (VICs) and how to set up basic configurations for a VoIP network:
For further information on VoIP features, refer to the Cisco 1750 Router Voice-over-IP Configuration Guide, the Cisco WAN Interface Cards Hardware Installation Guide, and the Cisco IOS configuration guides and command references.
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.
This document uses the following conventions:
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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.
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.
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Note To 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 three types of VIC interfaces. (See Figure 4.)
Figure 2 shows a typical VIC.
You should install and cable VICs before performing the software configuration tasks explained later in this guide.
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Warning Be sure to observe all warnings and safety precautions in the installation guide. |
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Caution VICs do not support online insertion and removal (hot swap). To avoid equipment damage, turn off electrical power and disconnect network cables before you insert a VIC in the router. |
Follow these steps, and refer to Figure 3 to insert a VIC in the router:
Step 2 Loosen the thumbscrews on the 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.
Step 3 Remove the metal plate that covers the card slot.
Step 4 Hold 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 3.
Step 5 Insert 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 6 Tighten the screws.
If you need more detailed installation instructions, refer to the Cisco WAN Interface Cards Hardware Installation Guide.
Use the following cables, and refer to Figure 4 to connect the VICs to the network:
When you are finished, reinstall any network interface cables you removed, and turn on power to the router.
Figure 5 shows the wiring patterns for FXS and FXO VICs connector, and Table 1 lists the pinouts.
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Note Pins that are not used should not be connected. |
| Pin | Signal |
|---|---|
1 | - |
2 | - |
3 | Ring |
4 | Tip |
5 | - |
6 | - |
Figure 6 shows the wiring pattern for an E&M connector, and Table 1 lists the pinout.
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Note The E&M VIC pinout depends on the PBX type and connection. Pins that are not used should not be connected. |
| Pin | Signal | Description |
|---|---|---|
1 | SB | -48V signaling battery |
2 | M-lead | Signaling input |
3 | R | Ring, audio input |
4 | R or R1 | Ring, audio input/output, or output |
5 | T or T1 | Tip, audio input/output, or output |
6 | T | Tip, audio input |
7 | E-lead | Signaling output |
8 | SG | Signaling ground return |
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.
Enter the show voice port command to identify the port numbers of voice interfaces installed in your router:
Router# show voice port slot-number/port
Figure 7 shows Cisco 1750 router slot numbering.
Figure 8 shows the VIC port numbering.
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
You can configure your router for VoIP by entering commands at the CLI. This method of entering commands is called configuration mode.
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Note VoIP 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:
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 2 Power up the router.
Step 3 If 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.
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Note If the current configuration is valid, you enter the normal operating mode automatically. |
Step 4 At 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#):
Router#
Step 5 Enter the following command to enter global configuration mode:
Router# config terminal Router(config)#
Step 6 If you have not configured the router before or want to change the configuration, you need to 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 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.
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Note Voice configuration uses a number of Cisco IOS commands. For complete information about these commands, refer to the Cisco 1750 Router Voice-Over-IP Configuration Guide. You can also enter a question mark after a command or partial command at the Router(config)# prompt to get syntax and argument help. |
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Note At 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. |
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 2 To see the startup configuration stored in NVRAM, enter the following command at the enable prompt:
Router# show startup-config
Step 3 To see the operating configuration, including any changes you have just made, enter the following command:
Router# show running-config
Step 4 To write your changes to NVRAM and make them permanent, enter the following command:
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, refer to the last section, "Cisco Connection Online."
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:
West(config)# dial-peer voice 401 pots West(config-dial-peer)# destination-pattern 14085553737 West(config-dial-peer)# port 0/0
West(config)# dial-peer voice 501 voip West(config-dial-peer)# destination-pattern 1919555.... West(config-dial-peer)# session target ipv4:192.168.11.3
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.
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....
This command tells the router to expand a particular sequence of dialed numbers into a complete telephone number (destination pattern) as shown in Table 3.
| Extension | Destination Pattern | Num-Exp Command Entry | Description |
|---|---|---|---|
3737 | 14085553737 | num-exp 3737 14085553737 | To expand 3737 to 14085553737 |
.... | 1408555.... | num-exp .... 1408555.... | To expand any dialed sequence of four digits (represented by four periods) by prefixing 1408555 to it |
5.... | 1408555.... | num-exp 5.... 1408555.... | To expand a five-digit extension beginning with the numeral 5 by prefixing 140855 to it |
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]
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Note You 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. |
This section explains how to configure ports on FXS VICs that connect directly to a standard telephone, fax machine, or similar device.
Figure 9 shows a basic voice network. A small business uses a Cisco 1750 router (named West) to provide telephone and fax connections among employees in its office. Two of these telephones are connected to an FXS VIC port in the West router.
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Note You can name your router by using the global configuration hostname command. |
Table 4 lists telephone numbers and voice ports for the West router. (For information about port numbering, refer to the section on "Voice Port Numbering" earlier in this guide.)
| Telephone Number | Voice Port |
|---|---|
408 555-3737 | 0/0 |
408 555-4141 | 0/1 |
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Note If 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. |
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 9.) In other words, the router needs to know the information in Table 4.
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 231-1(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 4 assigns a dial-peer tag to each telephone number and its associated voice port on the West 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. (VoIP dial peer is explained in the section "Calling Between Routers" later in this guide.)
| Telephone Number | Voice Port | Dial-Peer Tag |
|---|---|---|
408 555-3737 | 0/0 | 401 |
408 555-4141 | 0/1 | 402 |
You should construct a table similar to Table 4 for your own routers, assigning your own telephone numbers and dial-peer tags.
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Note The 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 Table 4, enter the following global configuration commands:
West> enable Password: West# configure terminal West(config)# dial-peer voice 401 pots West(config-dial-peer)# destination-pattern 14085553737 West(config-dial-peer)# port 0/0 West(config)# dial-peer voice 402 pots West(config-dial-peer)# destination-pattern 14085554141 West(config-dial-peer)# port 0/1 West(config-dial-peer)# exit West(config)#
These commands are summarized in Figure 10.
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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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 11 shows the East office network, and Table 6 lists the phone numbers, voice ports, and dial-peer tags for this office.
| Telephone Number | Destination Pattern | Voice Port | Dial-Peer Tag |
|---|---|---|---|
919 555-8282 | 19195558282 | 1/0 | 901 |
919 555-9595 | 19195559595 | 1/1 | 902 |
Enter the following commands to configure the local ports on the East Router with the dial-peer information in Table 6:
East(config)# dial-peer voice 901 pots East(config-dial-peer)# destination-pattern 19195558282 East(config-dial-peer)# port 1/0 East(config)# dial-peer voice 902 pots East(config-dial-peer)# destination-pattern 19195559595 East(config-dial-peer)# port 1/1 East(config-dial-peer)# exit East(config)#
These commands are summarized in Figure 12.
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.
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Note If the voice port is offline, use the interface configuration no shutdown command to enable it. |
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Note Although placing calls directly between ports on the same router helps to verify your configuration, it is not recommended for general telecommunications use. |
To enable the West 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 13.
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.
Table 7 assigns a dial-peer tag to each telephone number and its associated IP address on the West 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 VoIP means that the dial peer associates a telephone number with an IP address.
| Remote Location | Telephone Number | Destination Pattern | IP Address | Dial-Peer Tag |
|---|---|---|---|---|
East | 919 555-8282 | 19195558282 | 192.168.11.3 | 501 |
East | 919 555-9595 | 19195559595 | 192.168.11.3 | 502 |
Create a VoIP 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 wildcards, as shown in Table 8.
| Remote Location | Telephone Number | Destination Pattern | IP Address | Dial-Peer Tag |
|---|---|---|---|---|
East | 919 555-xxxx | 1919555.... | 192.168.11.3 | 501 |
Construct a table similar to Table 8 for your own routers, assigning your own telephone numbers, IP addresses, and dial-peer tags.
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Note The 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 West router to create the dial-peer configuration given in Table 8:
West(config)# dial-peer voice 501 voip West(config-dial-peer)# destination-pattern 1919555.... West(config-dial-peer)# session target ipv4:192.168.11.3
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:
West(config-dial-peer)# session target dns:voice.eastrouter.com
Configure a dial peer on each router for each telephone number on every other router connected to it.
Make things easier by configuring number expansion for East router telephone numbers on the West router:
West(config)# num-exp 5.... 1919555....
Now users can dial a five-digit extension beginning with 5 from a telephone on the West router to reach a telephone on the East router.
These commands are summarized in Figure 14.
The West router is now configured to send calls to the East router.
Table 9 shows how to configure the East router to send calls to the West router.
| Remote Location | Telephone Number | IP Address | Dial-Peer Tag |
|---|---|---|---|
West | 408 555-xxxx | 192.168.19.27 | 801 |
Enter the following information on the East router to create the dial-peer configuration given in Table 9:
East(config)# num-exp 5.... 1408555.... East(config)# dial-peer voice 801 voip East(config-dial-peer)# destination-pattern 1408555.... East(config-dial-peer)# session target ipv4:192.168.19.27
These commands are summarized in Figure 15.
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.
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.
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Note Refer to the section "Tuning Parameters for Real-Time Voice Traffic" if you need to improve the quality of voice connections. |
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 16 shows a typical FXO gateway attached to the West router.
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:
West> enable Password: West# configure terminal West(config)# dial-peer voice 201 pots West(config-dial-peer)# destination-pattern 9 West(config-dial-peer)# port 0/0
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 enable East router users to make calls over the West router local PSTN, enter the following commands:
East(config)# dial-peer voice 701 voip East(config-dial-peer)# destination-pattern 7 East(config-dial-peer)# session target ipv4:192.168.19.27 West(config)# dial-peer voice 601 pots West(config-dial-peer)# destination-pattern 7 West(config-dial-peer)# port 0/0
When you dial 7 on the East router, the call is connected to the PSTN on the West router. The PSTN then provides a dial tone, and any digits you enter on the telephone thereafter are interpreted on the PSTN.
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Note In this example, West 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 West router. Dial peer 601 matches calls beginning with the digit 7 and handles calls to the PSTN originating from the East router. |
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.
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Note Refer to the section "Tuning Parameters for Real-Time Voice Traffic" if you need to improve the quality of voice connections. |
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 17 shows a company with two offices, West 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.
To configure E&M voice ports, you need to use the following commands beginning in privileged EXEC mode:
| Command | Task |
|---|---|
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 | unitedkingdom} | Select the appropriate voice call progress tone for this interface. |
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. |
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 1750 Router Voice-over-IP Configuration Guide.
In this example, West 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 West 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:
West> enable Password: West# configure terminal West(config)# dial-peer voice 111 pots West(config-dial-peer)# destination-pattern 1408555.... West(config-dial-peer)# port 0/0
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 18, and in the following commands:
West(config)# dial-peer voice 121 voip West(config-dial-peer)# destination-pattern 1919555.... West(config-dial-peer)# session target ipv4:192.168.11.3 West(config-dial-peer)# exit West(config)#
Now configure number expansion so that numbers beginning with 5 (belonging to the East office) and sent by the West PBX to the West router are expanded into the full destination pattern:
West(config)# num-exp 5.... 1919555....
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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, 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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Note For these commands to take effect, you have to cycle the port by using the shutdown and no shutdown commands. |
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Note Configure the PBX to pass all dual-tone multifrequency (DTMF) signals to the router. |
Configure the East router similar to the West 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:
East(config)# dial-peer voice 211 pots East(config-dial-peer)# destination-pattern 1919555.... East(config-dial-peer)# port 0/1
Enter the following commands to configure a VoIP dial peer for telephones on the West router:
East(config)# dial-peer voice 221 voip East(config-dial-peer)# destination-pattern 1408555.... East(config-dial-peer)# session target ipv4:192.168.19.27 East(config-dial-peer)# exit East(config)#
Enter the following commands to configure number expansion and to make it easy for East users to dial numbers on the West 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
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.
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Note Refer to the section "Tuning Parameters for Real-Time Voice Traffic" if you need to improve the quality of voice connections. |
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 QoS:
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 1750 Router Voice-over-IP Configuration Guide.
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Note QoS 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."
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 19.)
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 VoIPdial 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
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.
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:
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Note If you configure multilink PPP interleaving, you can use the ip rtp reserve command instead of 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 1750 Router Voice-over-IP Configuration Guide.
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:
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Note Do not use multilink PPP on links faster than 2 Mbps. |
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 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)# interface virtual-template 1
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:
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Note Do not use RTP header compression on links faster than 2 Mbps. |
Enable RTP header compression at both ends of the serial link by entering the following ip rtp header-compression interface configuration command:
Router(config-if)# ip rtp header-compression
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Note When 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. |
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:
Router# interface serial 0/0 Router(config-if)# mtu 300
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Note Lowering the MTU size affects data throughput speed. |
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
Router(config-if)# frame-relay ip rtp header-compression
Router(config-if)# traffic-shape rate 32000
For further information and more detailed examples of Frame Relay configuration, refer to the Cisco 1750 Router Voice-over-IP Configuration Guide.
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Note When 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 for details. |
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 VoIP dial 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.
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 231 -1 (2147483647) used to identify a dial peer.
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.
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).
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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. |
Posted: Fri Jul 21 10:03:03 PDT 2000
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