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C H A P T E R

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Cisco ICS 7750 Installation and Configuration Guide

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6

Configuring the Cisco ICS 7750

Many tasks are required for fully configuring the

Cisco Integrated Communications System 7750 (Cisco ICS 7750) for data and

voice routing. This chapter lists common tasks required to configure the

Cisco ICS 7750, gives pointers to Cisco IOS and Cisco CallManagerdocumentation that tells how to perform these tasks, and describes any differences

between configuring Cisco IOS or Cisco CallManager software on the

Cisco ICS 7750 and configuring Cisco IOS or Cisco CallManager on other

platforms.

This chapter contains these sections:

Best Practices for Using the IOS CLI, page 6-2

Setting the System Date and Time, page 6-4

Configuring the SSP, page 6-8

Configuring MRPs and ASIs, page 6-11

Network Security Considerations, page 6-115

Configuring Cisco CallManager, page 6-122

Running Network Time Protocol, page 6-129

Installing and Configuring Cisco Unity Voice Messaging, page 6-130

Configuring the System for Voice Mail, page 6-131

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Chapter 6 Configuring the Cisco ICS 7750

Best Practices for Using the IOS CLI

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Best Practices for Using the IOS CLIICS System Manager is designed to communicate with and to monitor the status

of all the components in the chassis. To enable ICS System Manager to perform

these functions, a configuration program (ICSConfig) guides you through the

configuration process. ICSConfig enables you to change key system parameters,

such as the IP addresses of system cards, passwords, destination for syslog

messages, and Simple Network Management Protocol (SNMP) community

strings.

To enable ICS System Manager to properly function as a system management

tool, it is important that you use ICSConfig or ICS System Manager, as

appropriate, rather than the IOS command-line interface (CLI), when you enter

key system parameters.

Except for the procedures listed in ICSConfig Tasks, you can enter all IOS CLI

commands that are available for use in any IOS software release that is intendedfor use on the Cisco ICS 7750.

ICSConfig Tasks

You should always use ICSConfig for the following tasks:

Passwords

Changing the login password, which gives ICS System Manager

continued Telnet access to system cards

Changing the Windows 2000 administrator password, which grants those

with administrator privileges continued access to SPE310s

Changing the enable or secret password, which makes it possible for

administrators to enter certain IOS commands Card configurations

Assigning or changing the IP addresses or subnet mask of system cards

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Best Practices for Using the IOS CLI

SNMP settings

Changing read-only and read/write SNMP community strings of the

SNMP server

Changing the server destination of SNMP traps

Managing the SNMP server

Logging

Changing the syslog logging host

Note SNMP community strings and system passwords are case sensitive.

The following list includes tasks that should never be configured on the

Cisco ICS 7750 by using the IOS CLI under any circumstances:

Shutting down an Ethernet interface Changing an Ethernet or VLAN interface

Disabling Cisco Discovery Protocol (CDP) on an Ethernet or VLAN interface

Configuring Domain Name System (DNS) on SPEs

Disabling Network Time Protocol (NTP)

Invoking the Cisco Network Registrar (CNR) dhcp.exe from c:\program

files\network registrar\bin

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Setting the System Date and Time

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Saving Configuration Changes

To prevent loss of the ASI or MRP configuration, save the running-config file to

the startup-config file by following these steps:

Setting the System Date and TimeThis section explains how to set the date and time on Cisco ICS 7750 cards. It

contains the following tasks:

Setting the Date and Time on SPE310 Cards, page 6-5

Setting the Date and Time on SSP, MRP, and ASI Cards, page 6-5

Note For information about using NTP on the Cisco ICS 7750, see the Running

Network Time Protocol section on page 6-129.

Command Purpose

Step 1 MRP> enable

Password: password

MRP#

Enters enable mode. You have entered enable

mode when the prompt changes to MRP#.

Step 2 MRP# copy running-configstartup-config

Saves the configuration changes to the

startup-config file so that they are not lost during

resets, power cycles, or power outages.

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Setting the Date and Time on SPE310 Cards

Complete the following steps to set the date and time on Cisco

System Processing Engine 310 (SPE310) cards:

Step 1 On your PC, choose Start > Programs > Terminal Services Client > Client

Connection Manager.

Step 2 Use the Client Connection Manager to open a Terminal Services Clientconnection with the SPE310:

If you already have a Terminal Services Client connection defined for the

SPE310, select it, and choose File > Connect.

If you do not have a Terminal Services Client connection defined for the

SPE310, choose File > New Connection. Follow the instructions in the

wizard, and then choose File > Connect.

Step 3 Log in as an administrator (user ID administrator), and enter your password (the

default is changeme).

Step 4 On the SPE310, choose Start > Settings > Control Panel > Date/Time.

The Date/Time Properties dialog opens.

Step 5 Fill in the necessary fields. ClickOK to close the Date/Time Properties dialog

box.

Step 6 Repeat Step 2 through Step 6 for any additional SPE310s, if present.

Setting the Date and Time on SSP, MRP, and ASI Cards

The system switch processor (SSP) card, multiservice route processor (MRP)

cards, and analog station interface (ASI) cards each have a system clock that

begins to run from the point at which the card starts up. The system clock keeps

track of the date and time. The SSP stores its configuration data in

Flash-simulated NVRAM. The MRP300, MRP3-8FXS, and MRP3-16FXS cards

store their configuration data in NVRAM. The MRP200, ASI81, and ASI160

cards obtain their configuration data from the SPE310 running System Manager

when they boot.

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When you set the date and time, the setting remains accurate until the next card

restart.

Note When changing the date and time settings on SSP, MRP, and ASI cards, open a

Telnet session from the PC, not from the SPE310. If you open a Telnet session

from the SPE, the changes you make to the card configuration are not saved.

Complete the following steps to set the date and time on the SSP card and MRP

cards:

Step 1 From the PC, choose Start > Run.

Step 2 Enter the following command to open a Telnet session:

telnet IP address

whereIP address is the IP address of the card with which you wish tocommunicate.

Step 3 Enter your login password.

Step 4 Enter privileged EXEC mode by entering the following command:

ICS7750> enable

Step 5 Enter your enable password.Step 6 To enter global configuration mode, enter the following command:

ICS7750# configure terminal

Step 7 To set the time zone, enter the following command in global configuration mode:

ICS7750(config)# clock timezone zone hours [minutes]

where:

zone is the name of the time zone to be displayed when standard time is in

effect (such as Pacific Standard Time, or PST)

hours is the number of hours offset from Coordinated Universal Time (UTC)

(optional) minutes is the number of minutes offset from UTC

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For example, to set the time to PST, eight hours offset from UTC, enter the

following command:ICS7750(config)# clock timezone PST -8

Table 6-1 lists the time zones in North America and their offsets from UTC.

Step 8 To set the clock for a card, enter one of the following IOS commands in

privileged EXEC mode:

ICS7750(config)# clock set hh:mm:ss day month year

or

ICS7750(config)# clock set hh:mm:ss month day year

where:

hh:mm:ss is the current time in hours, minutes, and seconds. Note that this is

a 24-hour clock, so 10:03:00 p.m. would be entered as 22:03:00.

day is the current day in the month, entered as a two-digit date.

Table 6-1 North American Time Zones

Time Zone Abbreviation UTC Offset

Atlantic Standard Time AST -4 hours

Atlantic Daylight Saving Time ADT -3 hours

Eastern Standard Time EST -5 hours

Eastern Daylight Saving Time EDT -4 hours

Central Standard Time CST -6 hours

Central Daylight Saving Time CDT -5 hours

Mountain Standard Time MST -7 hours

Mountain Daylight Saving Time MDT -6 hours

Pacific Standard Time PST -8 hours

Pacific Daylight Saving Time PDT -7 hours

Hawaiian Standard Time HST -10 hoursAlaska Standard Time AKST -9 hours

Alaska Standard Daylight Saving Time AKDT -8 hours

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Configuring the SSP

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month is the current month, entered as a three-letter abbreviation.November,

for example, would be entered as nov. year is the current year entered as a four-digit year, such as 2001.

Note This step must be performed after every reboot of the Cisco ICS 7750.

Step 9 To exit global configuration mode, enter the following command:

ICS7750(config)# exit

Step 10 To save your configuration, enter the following command:

copy running-config startup-config

Step 11 To verify your settings, enter the following command:

show clock

Step 12 Repeat Step 3 through Step 12 for additional cards, as necessary.

Step 13 Close the Telnet session by typing exit at the prompt.

Configuring the SSPThe SSP is an eight-port switching module in the Cisco ICS 7750. It has two

external ports for connecting to external network devices and has six internal

ports for connecting to the other cards in the Cisco ICS 7750.

The SSP serves the following purposes:

With the chassis backplane, it acts as a communications path for intrachassiscommunications among the installed cards in the Cisco ICS 7750.

When connected to an external Ethernet switch, it forwards voice and data

traffic between the IP network and the external network (such as the WAN and

the Public Switched Telephone Network [PSTN]) through the

Cisco ICS 7750.

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Configuring the SSP

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Network Time Protocol (NTP) to provide an external source for time-of-day

information Hot-swap support for removing and installing the SSP without having to

power down the system

Note Hot-swapping the SSP will result in system downtime since all the cards

in the ICS 7750 chassis will lose connectivity during the swap.

SSP Configuration Tasks

The SSP is, in the default configuration, network-ready. In most network

configurations, the SSP will not require any additional configuration. However,

many settings on the SSP are configurable. Table 6-2 lists tasks that you may need

or want to perform in order to configure the SSP. In addition, Table 6-2 givespointers to the locations in the Catalyst 2900 XL and Catalyst 3500 Software

Configuration Guide that provide instructions for performing those tasks. The

Catalyst 2900 XL and Catalyst 3500 Software Configuration Guide is available at

the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/lan/c2900xl/29_35wc/sc/inde

x.htm

Table 6-2 SSP Configuration Tasks

Tasks Documentation Locations

Configuring System Settings Managing the ARP Table

Configuring Device Settings Controlling IP Multicast Packets Through

CGMP

Configuring STP

Configuring UniDirectional Link Detection

Configuring Protected Ports

Configuring Port Settings Creating EtherChannel Port Groups

Enabling SPAN

Configuring Flooding Controls
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Configuring MRPs and ASIs

Configuring M RPs and ASIsThis section explains how to configure MRP and ASI cards and contains the

following sections:

System Card Overview, page 6-11

Codec/DSP Overview, page 6-14 Configuring Fast Ethernet Ports, page 6-26

Configuring WAN Interfaces, page 6-27

Voice over IP, page 6-36

H.323 Overview, page 6-85

MGCP Overview, page 6-87

Configuring Voice Ports

Configuring VLAN Settings Assigning VLAN Port Membership Modes

Overlapping VLANs and Multi-VLAN Ports

Using VTP

VTP Version 2VTP Pruning

VLANs in the VTP Database

How VLAN Trunks Work

Configuring 802.1p Class of Service

Load Sharing Using STP

How the VMPS Works

Configuring Security Settings Managing the MAC Address Tables

Enabling Port Security

Configuring TACACS+

Table 6-2 SSP Configuration Tasks (continued)

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Configuring M RPs and ASIs

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System Card Overview

This section lists the key features of the MRP200, MRP300, ASI81, ASI160,

MRP3-8FXS, and MRP3-16FXS system cards.

Key Features of M RP200 and M RP300 Cards

MRP200 and MRP300 cards have the following features:

Voice- and data-capable routers that support both digital and analog voice

trunks and WAN routing interfaces and that can link remote Ethernet LANs

to the PSTN and existing private branch exchanges (PBXs), as well as most

common analog devices such as fax machines and teleconferencing stations.

Slots for two WAN interface cards (WICs), voice interface cards (VICs), or

voice WAN interface cards (VWICs).

Support for the following T1/E1 configurations:

Two T1/E1 ports with voice payload (no more than 24 simultaneous calls

per MRP [T1] or no more than 30 simultaneous calls per MRP [E1]).

No more than one T1/E1 data port.

No more than one channelized T1/E1 group for data.

Up to two external clock sources.

H.323 and Quality of Service (QoS) for voice support.

G.711, G.723.1, G.726, and G.729 codec support.

Hot-swap support (MRP cards are hot-swappable, but any WICs, VICs, and

VWICs installed within the MRPs are not hot-swappable).

Configuration files for the MRP200 are stored on the SPE310 running System

Manager. Configuration files for the MRP300 are stored in NVRAM.

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Key Features of ASI81, ASI160, M RP3-8FXS, and M RP3-16FXS Cards

The key features of the ASI81, ASI160, MRP3-8FXS, and MRP3-16FXS cards

are delineated below:

ASI81 and MRP3-8FXSVoice-and-data-capable routers that can carry

voice traffic over an IP network, that can link small-to-medium-size remote

Ethernet LANs to central offices over WAN links (depending on the type of

card installed in its WIC/VIC/VWIC slot), and that can support eight

connections to analog telephones, fax machines, and polycoms ASI160 and MRP3-16FXSAnalog gateways that support 16 connections to

telephones, fax machines, and teleconferencing stations.

H.323 and QoS for voice support.

G.711, G.723.1, G.726, and G.729 codec support.

Hot-swap support (system cards are hot-swappable, but any WIC, VIC, or

VWIC installed within an ASI81 or MRP3-8FXS card is not hot-swappable).

ASI81 and ASI160Configuration files are stored on the SPE310 running

System Manager.

MRP3-8FXS and MRP3-16FXSConfiguration files are stored in NVRAM.

Note The MRP300, MRP3-8FXS and MRP3-16FXS cards have additional

functionality provided by 16 MB of onboard Flash memory, with 64 MB of

add-on Flash memory available as an option.

Supported WICs, VICs, and VWICs

For a list of the WICs, VICs, and VWICs that are supported in MRP200, MRP300,

MRP3-8FXS, and ASI81 cards, refer to the Cisco ICS 7750 System Description.For information about valid combinations of WICs, VICs, and VWICs on MRP

and ASI cards, see Appendix C, PVDM Requirements.

Ch 6 C fi i h Ci ICS 7750
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Codec/DSP Overview

VICs and VWICs installed in MRP cards or ASI cards might require additional

digital signal processors (DSPs) for processing heavier voice traffic. Each DSP

can perform a maximum of 100 million instructions per second (MIPS).

You can install up to two packet voice/data modules (PVDMs) on each MRP or

ASI card. PVDMs contain DSP chips that give MRP and ASI cards more

processing power.

Voice Compression Algorithms (Codecs)

The Cisco ICS 7750 supports several options for voice-compression algorithms.

These algorithms are commonly called codecs. The word codec is a combination

of the words coderand decoder. Coding is the process of encoding a digitized

signal into a more efficient form for transmission or storage. Decoding is the

process of restoring the coded signal to the original form.

Codecs differ in terms of voice quality, compression rate and bandwidth, ability

to carry dual-tone multifrequency (DTMF) and modem traffic, and number of

channels (calls) that a single DSP can support. The more DSP channels, the

greater the number of calls that an MRP or ASI card can support. The number of

channels supported also depends on whether the DSP is running a digital image

or it is running an analog image. (Digital T1 and E1 VWICs process digital

signals, and analog VICs process analog signals.)

As Table 6-3 shows, some codec compression techniques require more processing

power than others. Multiple DSP firmware images are available for use on MRP

and ASI cards. High-complexity images support fewer calls than

medium-complexity images.

Table 6-3 MRP and ASI Card Codec Options

Channels per DSPDigital Image1

Channels per DSPAnalog Image

Codec Bandw idthMedium

Complexity

High

Complexity

Medium

Complexity2High

Complexity3

G.711 64 kbps 8 6 4 2

G.723.1 5.3 or 6.3 kbps none 2 none 2G.726 32, 24, or 16 kbps4 none 3 4 2

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G.711

G.711 performs pulse code modulation (PCM) and is the standard digital channel

used in the public telephone network. PCM provides no compression and

therefore no opportunity for bandwidth savings. Any services that operate over

the public network should operate with similar performance over a

Cisco ICS 7750 PCM channel (although the Cisco ICS 7750 connection might

have more delay).

G.723.1

G.723.1 is a compression technique that uses multi-pulse, multi-level

quantization (MP-MLQ) or code excited linear prediction (CELP) coding to

compress speech or audio signal components at 5.3 or 6.3 kilobits per second

(kbps), respectively. G.723.1, which is part of the H.324 family of standards, canbe used for compressing speech or audio signal components at very low bit rates.

G.723.1 Annex-A provides built-in voice activity detection (VAD) and Comfort

Noise Generation (CNG).

G.729a 8 kbps 4 3 4 2

Fax Relay Variable none 3 4 2

1. VWICs (VWIC-1MFT-T1, VWIC-1MFT-E1, VWIC-2MFT-T1, and VWIC-2MFT-E1) require digital DSP software.

2. Medium-complexity analog DSP software supports 8- and 16-port FXS modules (in the ASI 81 and the ASI 160,

respectively).

3. High-complexity analog DSP software supports all 2-port analog VICs (VIC-2DID, VIC-2BRI-NT/TE, VIC-2FXS,

VIC-2FXO-M1, VIC-2FXO-M2, VIC-2FXO-M3, VIC-2FXO, and VIC-2E/M) and the 8- and 16-port FXS modules (in the

ASI 81 and the ASI 160, respectively).

4. 32 kbps = 2:1 compression, 24 kbps = 3:1 compression, and 16 kbps = 4:1 compression.

Table 6-3 MRP and ASI Card Codec Options (continued)

Channels per DSPDigital Image1

Channels per DSPAnalog Image

Codec Bandw idthMedium

Complexity

High

Complexity

Medium

Complexity2High

Complexity3


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G.726

G.726 performs adaptive differential pulse code modulation (ADPCM) coding.

G.726 reduces network bandwidth requirements for transmitting voice by

encoding 64 kbps voice channels as 32, 24, or 16 kbps ADPCM (that is, 32 kbps

provides 2:1 compression, 24 kbps provides 3:1 compression, and 16 kbps

provides 4:1 compression). Generally, there is a trade-off between the amount of

compression and voice quality. ADPCM-encoded voice can be interchanged

between packet voice, PSTN, and private branch exchange (PBX) networks if the

PBX networks are configured to support ADPCM.

G.729

G.729 performs CELP coding, where voice is coded into 8-kbps streams. There

are two variations of this standard (G.729 and G.729 Annex A [G.729a]) that

differ mainly in terms of their computational complexity; both provide speech

quality similar to 32-kbps ADPCM. G.729 is a high complexity algorithm, andG.729a is a medium complexity variant of G.729 with slightly lower voice quality.

G.729a performs conjugate structure algebraic code excited linear predictive

(CS-ACELP) coding, providing speech quality similar to 32-kbps ADPCM.

G.729a offers the best compression rate (8:1), but it does not typically carry

modem traffic, and it degrades DTMF and music signals somewhat. Depending on

the type of traffic, using G.729a can produce cost savings of 40 percent, relative

to using G.711. Other algorithms in the G.729 family include G.729 Annex-B, a

high complexity algorithm, and G.729a Annex-B, a medium-complexity variantof G.729 Annex-B with slightly lower voice quality. The difference between the

G.729 and G.729 Annex-B codecs is that G.729 Annex-B provides built-in VAD

and CNG.

Codec Interoperability

Codec interoperability is the ability of one codec to decode another codec. If a

DSP is configured with a certain codec, the DSP should be able to decode the

voice codec using any codec with which the DSP is interoperable.

The following G.729 codec combinations interoperate:

G.729 and G.729a

G.729 and G.729

G.729a and G.729a


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G.729 Annex-B and G.729a Annex-B

G.729 Annex-B and G.729 Annex-B

G.729a Annex-B and G.729a Annex-B

The following G.723.1 codec combinations interoperate:

G.723.1 (5.3 kbps) and G.723.1 (6.3 kbps)

G.723.1 (5.3 kbps) and G.723.1 (5.3 kbps)

G.723.1 (6.3 kbps) and G.723.1 (6.3 kbps) G.723.1 Annex-A (5.3 kbps) and G.723.1 Annex-A (6.3 kbps)

G.723.1 Annex-A (5.3 kbps) and G.723.1 Annex-A (5.3 kbps)

G.723.1 Annex-A (6.3 kbps) and G.723.1 Annex-A (6.3 kbps)

Delay

Delay is the time it takes for packets to travel between two endpoints. In

traditional data networking, delay can be tolerated with little or no impact on

network users; however, in networks carrying voice traffic, delay is potentially

quite significant because it can affect the ability of users to carry on a telephone

conversation. For example, delay can introduce pauses or gaps in the

conversation, increasing the likelihood that one person will start talking before the

other person has finished.Because of the speed of network links and the limited processing power of many

devices, some delay is expected. Telephone users normally accept up to about 150

milliseconds (ms) of delay without noticing problems. You can measure delay by

using ping tests at various times of the day with different network traffic loads. If

network delay is excessive, reduce it before deploying a network that carries

Voice over IP (VoIP) traffic.

The two types of delay most commonly found in todays telephony networks arepropagation delay and handling delay. Propagation delay is caused by the

characteristics of the speed of light traveling via a fiber-optic-based or

copper-based medium. Handling delay (sometimes called serialization delay) is

caused by the devices that handle voice information. Handling delays have a

significant impact on voice quality in a packetized network. Codec-induced

delays are considered a handling delay.

Table 6-4 shows the delay that is introduced by different codecs.


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DSP Groups

ASIs and MRPs handle calls based on the grouping of the DSPs. The DSPs are

located on PVDMs. There can be up to five DSPs on a single PVDM. Each PVDM

corresponds to one DSP group. MRP200 and MRP300 cards each have two

PVDM slots and, therefore, can have a maximum of two DSP groups. Each DSP

group serves either an analog port or a T1 port on the VIC. Therefore, one analog

VIC and one T1 VWIC make up two groups, and two T1s with two different clock

sources (regardless of whether they are on the same VWIC) also make up two

groups.

DSP Group Serving a T1 Port

Each DSP group that serves a T1 port can support as many DSPs as there are in

the PVDM.

A DSP has a maximum capacity of 100 MIPS to handle a particular number of

simultaneous calls. One G.729a call requires 25 MIPS, and one G.711 call

requires 12.5 MIPS. The number of calls on a DSP is determined by the totalMIPS used reaching 100 on that DSP. The DSP resource manager rejects a call if

it cannot find a DSP with required unused MIPS for the selected codec.

Table 6-5 provides some examples of the number of calls that can be supported on a

single DSP, depending on the codec used. Table 6-6 lists some of the combinations

of calls that can be handled on a single DSP.

Table 6-4 Delay Introduced by Codecs

Compression M ethod Bit Rate (kbps) Compression Delay (ms)

G.711 PCM 64.0 0.75

G.726 ADPCM 32.0 1

G.729 CS-ACELP 8.0 10

G.729a CS-ACELP 8.0 10

G.723.1 MP-MLQ 6.3 30

G.723.1 ACELP 5.3 30


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Note The examples provided in Table 6-5and Table 6-6are based on the assumptionthat you are using a medium-complexity digital image.

DSP Group Serving Analog Ports

Each DSP group that serves analog ports requires the following:

MRP200s and MRP300sOne DSP for every two ports (using the

high-complexity image). For example, an MRP300 with two 2-port analog

VICs requires two DSPs. ASI81s and MRP3-8FXSsOne DSP for every two ports (using the

high-complexity image) or one DSP for every four ports (using the

medium-complexity image). For example, an MRP3-8FXSs with a 2-port

VIC installed (for a total of 10 analog ports) requires five DSPs (assuming

that the high-complexity image is used).

Table 6-5 Codec/ DSP Call-Processing Examples

ScenariosCalls perDSP Codecs

M IPS perSession M IPS Required Call Status

1 4 G.729a 25 25 x 4 = 100 4 calls accepted

2 8 G.711 12.5 12.5 x 8 = 100 8 calls accepted

3 4

1

G.729a

G.711

25

12.5

25 x 4 = 100

12.5 x 1 = 12.5 1 call rejected

Table 6-6 Sample Combinations of Calls on a Single DSP

G.711 Calls G.729a Calls

2 3

4 2

6 1


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Choosing Codecs

This section provides information that can help you choose the DSP image that is

best suited for a particular type of traffic. The following are some common

scenarios:

Intra-LAN or PSTN-to-LAN callsG.711 is recommended in situations such

as the following:

Traffic between analog telephones and Cisco IP Phones (normally on the

same LAN).

Traffic between an analog or digital trunk and a Cisco IP Phone.

Traffic between an analog telephone, an analog trunk, or a digital trunk

and a software application (such as Cisco Unity).

Calls across a WAN link with limited bandwidthIf one Cisco IP Phone calls

another Cisco IP Phone over a WAN link, a codec with voice

compression/decompression may be desirable to save bandwidth. Ciscorecommends G.711 encoding for LAN environments and G.729A across the

WAN. The use of the G.729 family, with a compressed bit rate of 8 kbps, can

result in bandwidth savings. Note that the actual bandwidth saving is also

affected by the packetizing overhead inherent in Real-Time Transport

Protocol (RTP), User Datagram Protocol (UDP), and IP headers.

Calls between an analog telephone and a Cisco IP Phone across a WAN

linkG.729a is recommended for traffic between an analog telephone and aremote Cisco IP Phone (using VoIP over the WAN). G.723.1 cannot be used

because Cisco IP Phones do not support G.723.1.

Choosing DSP Firmw are

When you choose DSP firmware, it is important to consider the following factors:

The codecs that must be supported

The number of voice channels required per DSP

Technical issues such as echo cancellation coverage

DSP firmware is included with each IOS release for the Cisco ICS 7750. Five DSP

firmware images are available for use on ASI and MRP cards. Two of the DSP

firmware images are intended for MRP200 and MRP300 cards (which contain


C fi i MRP d ASI

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analog VICs) and for MRP3-8FXS and ASI81 cards (which contain FXS ports);

two images are intended for digital trunks (such as T1 CAS and T1/E1 PRI); andone image is intended for transcoding.

Each DSP firmware image supports a particular set of codecs. High-complexity

DSP firmware supports more codecs than medium-complexity firmware supports.

However, in order to support more codecs, the number of voice channels

supported by the firmware has to be reduced.

Table 6-7 lists the number of channels supported by the DSP firmware images.

Note The abbreviations FIXHC, FIXMC, FLEX6, FLEX8, and XCODE represent the

names of the firmware images. These abbreviations appear in the output from the

show voice dsp command.

Table 6-7 Number of Channels Supported by DSP Firmware Images

DSP Firmware Image CodecNumber ofChannels per DSP Cards Supported

High-Complexity

Analog (FIXHC)

G.711, G.726, G.729

Annex-B, G.723.1, fax

relay

2 All 2-port analog

VICs1

8-port and 16-port FXS

modules (ASIs) VIC-2BRI-NT/TE

Medium-Complexity

Analog (FIXMC)

G.711, G.726, G.729a

Annex-B, fax relay

4 8-port and 16-port FXS

modules (ASIs)

High-Complexity

Digital (FLEX6)

G.711, G.726, G.729a

Annex-B, G.723.1, fax

relay

6 (G.711)

3 (G.729a

Annex-B) 3 (G.726)

3 (fax relay)

2 (G.723.1)

All digital VWICs2



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Determining How M any DSPs Are Needed

The number of DSPs needed for each voice interface depends on the following

two factors:

Codec complexity

Type of codec selected

Table 6-8 shows how to calculate the number of DSPs needed for each channel.For example, with a medium-complexity analog image and a G.726 codec, 1 DSP

is needed for 4 voice interfaces.

Note For additional information on PVDM selection, refer to the PVDM

Requirements appendix in the Cisco ICS 7750 Hardware Installation Guide.

Medium- Complexity

Digital (FLEX8)

G.711, G.726, G.729a

Annex-B

8 (G.711)

4 (G.729a

Annex-B)

4 (G.726)

All digital VWICs

Transcoding

(XCODE)

G.711, G.726,

G.729 Annex-B,

G.723.1

2 Not applicable

1. VIC-2DID, VIC-2E/M, VIC-2FXS, VIC-2FXO, VIC-2FXO-M1, VIC-2FXO-M2, VIC-2FXO-M3.

2. VWIC-1MFT-T1, VWIC-2MFT-T1, VWIC-1MFT-E1, VWIC-2MFT-E1.

Table 6-7 Number of Channels Supported by DSP Firmware Images (continued)

DSP Firmware Image CodecNumber ofChannels per DSP Cards Supported

Table 6-8 DSP Configuration Rules

Type of CardSuggestedNumber of DSPs Suggested DSP Firmw are

Total VoiceChannels

CodecsSupported

All 2-port analog VICs 1 1 (PVDM-4) High-complexity analog 2 All2

VIC-2BRI-NT/TE 2 (PVDM-8) High-complexity digital 4 All


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Note Table 6-8 does not address configuration rules for transcoding. See the

Determining How Many DSPs Are Needed for Transcoding section onpage 6-26.

Note On MFT-T1 and MFT-E1 cards, medium-complexity firmware is not

recommended, because this firmware restricts echo cancellation coverage to 16

ms.

Note The codec complexity for ASI cards defaults to medium-complexity but can be

changed to high-complexity with sufficient DSPs (see Table 6-8).

ASI81 (8-port FXS

module in slot 0)

2 (PVDM-8) Medium-complexity analog 8 All except

G.723.1

4 (PVDM-16) High-complexity analog 8 All

ASI160 (16-port FXS

module)

4 (PVDM-16) Medium-complexity analog 16 All except

G.723.1

8 (2 PVDM-16) High-complexity analog 16 All

MFT-T1 4 (PVDM-16) High-complexity digital 24 G.711

8 (2 PVDM-16) High-complexity digital 24 G.729a

MFT-E1 5 (PVDM-20) High-complexity digital 30 G.711

10 (2 PVDM-20) High-complexity digital 30 G.729a1. VIC-2DID, VIC-2E/M, VIC-2FXS, VIC-2FXO, VIC-2FXO-M1, VIC-2FXO-M2, VIC-2FXO-M3.

2. The codecs supported on the Cisco ICS 7750 are G.711, G.723.1, G.726, and the G.729 family.

Table 6-8 DSP Configuration Rules (continued)

Type of CardSuggestedNumber of DSPs Suggested DSP Firmw are

Total VoiceChannels

CodecsSupported



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Transcoding

Because some hardware and software currently support only

G.711 (uncompressed) connections, transcoding is available on MRP and ASI

cards. MRP and ASI cards are considered packet-to-packet gateways because they

have DSPs that transcode between voice streams using different compression

algorithms. For example, when a user on a Cisco IP Phone at a remote location

calls a user at the central location, Cisco CallManager can be configured so that

it causes the remote IP phone to use compressed voice (G.729a) for the WAN call.

However, if the called party at the central site is unavailable, the call potentiallycould be routed to an application that supports only G.711. In this case, the MRP

or ASI card transcodes the G.729a voice stream to G.711 so that a voice message

is stored by the G.711-compliant voice-messaging server.

Transcoding is required when a compressed voice stream is used to save WAN

bandwidth and when the local device does not support the codec. The transcoding

service compresses and decompresses voice streams to match the capabilities of

the endpoint device.

A transcoderis a device that takes the output stream of one codec and transcodes

(converts) it from one compression type to another compression type. For

example, a transcoder could take an output stream from a G.711 codec and

transcode (convert) it in real time to a G.729 input stream accepted by a G.729

codec.

Transcoding is supported under the following conditions: Low-bit-rate to high-bit-rate (G.729a or G.723.1 to G.711 a-law or to

G.711 U-law), or vice versa, configurations.

High-bit-rate to high-bit-rate (G.711 a-law to G.711 U-law), or vice versa,

configurations.

Each instance of Cisco CallManager must have access to its own transcoding

resources.

Deciding When to Use Transcoding

Transcoding is needed when the calling and called parties cannot use the same

codec type. Codec incompatibility may result from of a lack of support for a

particular codec. For example, some unified messaging systems support only

G.711, while Cisco IP Phones support G.711 and G.729. (Note that Cisco Unity

supports both G.729a and G.711.) Codec incompatibility could also be caused by



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a failure when negotiating a common codec. For example, in a lab, two Cisco

voice gateways (such as an MRP or an ASI card) can be forced to use different

codecs so that transcoding is required for them to communicate.

Suppose that an application is communicating with a G.711-only voice-mail

system over a WAN link. To conserve bandwidth, the caller on one side of WAN

link uses G.729, while the called party voice-mail system recognizes only G.711.

This is a situation that would require transcoding.

Transcoding is not required if all the called parties (except those on a voice-mail

system) are on the same LAN. You can configure the calling and called parties sothat they must negotiate a common codec when possible.

Here are some additional transcoding guidelines:

Calls between Cisco IP Phones on the same LAN do not need transcoding,

even if the Cisco IP Phones are assigned to different Cisco CallManager

regions. For example, if a Cisco IP Phone in a G.729 region calls a Cisco IP

Phone in a G.711 region (the default), the two Cisco IP Phones automatically

negotiate a common codec.

Calls between Cisco IP Phones and an MRP or ASI in the same LAN do not

need transcoding, even if the Cisco IP Phone and the MRP or ASI are in

different regions. For example, if a G.729 gateway calls a Cisco IP Phone in

a G.711 region, the Cisco IP Phone can communicate with the gateway.

If a gateway is configured to use G.723.1, transcoding is needed because

Cisco IP Phones do not support G.723.1 and, therefore, cannot communicatewith a G.723.1 gateway.

Choosing a DSP Firmw are Image for Transcoding

When a DSP is reserved for transcoding, a special DSP firmware image is

downloaded to the DSP. At present, the DSP firmware supports transcoding

between G.723.1/G.729 and G.711 U/a-law, as well as between G.711 U-law and

G.711 a-law. Transcoding between low-bit-rate codecs, such as between G.723.1

and G.729, is not supported.



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Determining How M any DSPs Are Needed for Transcoding

Before an MRP or ASI can act as a transcoder, DSP resources must be reserved

for transcoding. Unlike other Cisco gateways, the MRP or ASI provides the

flexibility to choose the number of channels that should be reserved for

transcoding. One DSP is required for every two transcoding channels (full

duplex).

Understanding How DSPs Are Allocated for Transcoding

When the MRP or ASI boots, DSP resources are statically allocated first for

analog VICs and the VIC-2BRI-NT/TE. These DSP resource allocations cannot

be changed. In the show voice dsp command output, these DSPs are represented

with a value of FIXMC or FIXHC in the Image field, depending on whether high-

or medium-complexity DSP firmware is being used. The remaining DSP

resources can be allocated to T1 VWICs, to E1 VWICs, or to transcoding, as

needed.

For T1 VWICs or E1 VWICs, DSPs are reserved by defining a ds0-group or

pri-group under the individual T1 or E1 controller. A DSP is reserved if it hosts a

signaling channel for the T1/E1 VWIC. Such a reserved DSP has a non-zero value

in the D-sig Allocate field, which can be seen in the show voice dsp command

output.

Configuring Fast Ethernet Ports

ASI and MRP cards have Fast Ethernet interfaces that can be

configured.Depending on your own requirements and the protocols you plan to

route, you might need to enter additional configuration commands. For more

information about basic configuration, including enabling the interface and

specifying IP routing on Fast Ethernet interfaces, see the section Configuring

Ethernet, Fast Ethernet, or Gigabit Ethernet Interfaces in the Cisco IOS Interface

Configuration Guide, Release 12.2.

Note See the Configuring Dial Plans section on page 7-29 for a sample configuration

to configure a FastEthernet interface on an ASI or MRP card.


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Note Use ICSConfig to assign or modify the IP address of an ASI or MRP card, asnecessary. Do not use the CLI.

Configuring WAN Interfaces

You can configure an MRP or ASI card with a WIC or VWIC installed for access

to the WAN. For example, if you are using a serial interface, you can configureFrame Relay, Point-to-Point Protocol (PPP), and High-Level Data Link Control

(HDLC) over that serial interface.

Information about the various types of connections is provided in the sections that

follow:

Configuring Asynchronous/Synchronous Serial WICs, page 6-28

Configuring ISDN BRI WICs, page 6-30

Configuring T1 and Fractional T1 WICs, page 6-34

Configuring VWICs for Data-Only Transmission, page 6-35

Configuring the TDM Clock, page 6-35

Table 6-9 lists tasks you might need to perform in order to configure WAN

interfaces on MRP or ASI cards and gives pointers to the location in Cisco IOS

documentation set that provides additional instructions on performing those tasks.

The various Cisco IOS configuration guides for version 12.2 are available at the

following URL:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/inde

x.htm



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Configuring Asynchronous/Synchronous Serial WICs

You can configure the serial interfaces on your asynchronous/synchronous serial

WIC (WIC-1T, WIC-2T, or WIC-2A/S) by entering IOS commands at the ASI or

MRP command prompt, in configuration mode.

Note See the Configuring Synchronous Serial WICs section on page 7-3 for a sample

configuration.

Table 6-10 lists the half-duplex timer commands.

Table 6-9 WAN Interface Configuration Tasks


Configuring Asynchronous/Synchronous

Serial WICs

See Configuring a Synchronous Serial Interface and

Configuring Low-Speed Serial Interfaces in the Cisco IOS

Interface Configuration Guide, Release 12.2.

Configuring ISDN BRI WICs See Configuring ISDN BRI in the Cisco IOS Dial

Technologies Configuration Guide, Release 12.2

Configuring T1 and Fractional T1 WICs See Configuring Serial Interfaces for CSU/DSU ServiceModules in the Cisco IOS Interface Configuration Guide,

Release 12.2.

Table 6-10 Half-Duplex Timer Commands

Timer SyntaxDefault Setting(ms)

CTS1 delay half-duplex timer cts-delay 100

CTS drop timeout half-duplex timer cts-drop-timeout 5000

DCD2 drop delay half-duplex timer dcd-drop-delay 100

DCD transmission start delay half-duplex timer dcd-txstart-delay 100

RTS3 drop delay half-duplex timer rts-drop-delay 100


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Table 6-11 through Table 6-13 list clock rate settings in bits per second (bps) for

specific interfaces.

RTS timeout half-duplex timer rts-timeout 2000

Transmit delay half-duplex timer transmit-delay 0

1. CTS = Clear To Send

2. DCD = data carrier detect

3. RTS = Request To Send

Table 6-10 Half-Duplex Timer Commands (continued)

Timer SyntaxDefault Setting(ms)

Table 6-11 Clock Rate Settings for 1-Port/ 2-Port Serial WICs in Synchronous

Mode

1200 bps 38400 bps 148000 bps

2400 bps 56000 bps 500000 bps

4800 bps 57600 bps 800000 bps

9600 bps 64000 bps 1000000 bps

14400 bps 72000 bps 1300000 bps

19200 bps 115200 bps 2000000 bps

28800 bps 125000 bps 4000000 bps

32000 bps 128000 bps 148000 bps

Table 6-12 Clock Rate Settings for 1-Port/ 2-Port Serial WICs in Asynchronous

Mode

1200 bps 28800 bps 72000 bps

2400 bps 32000 bps 115200 bps

4800 bps 38400 bps 125000 bps

9600 bps 56000 bps 128000 bps



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Configuring ISDN BRI WICs

You can use an Integrated Services Digital Network (ISDN) Basic Rate Interface

(BRI) WIC to connect MRPs or ASIs with other ISDN routers. ISDN BRI is adial-up connection. Adding an ISDN BRI connection to the MRP creates a logical

dialer interface.

ISDN connections use one or both data channels for the connection to the ISDN

service provider. Normally, the ISDN provider is your local telephone company.

This section tells how to configure ISDN BRI WICs.

Note For information on how to configure ISDN voice interfaces, see the Configuring

ISDN Interfaces for Voice section on page 6-64.

14400 bps 57600 bps

19200 bps 64000 bps

Table 6-13 Clock Rate Settings for 2-Port Asynchronous/ Synchronous SerialWICs

1200 bps 28800 bps 72000 bps

2400 bps 32000 bps 115200 bps

4800 bps 38400 bps 125000 bps

9600 bps 56000 bps 128000 bps

14400 bps 57600 bps

19200 bps 64000 bps

Table 6-12 Clock Rate Settings for 1-Port/ 2-Port Serial WICs in Asynchronous

Mode (continued)

1200 bps 28800 bps 72000 bps



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ISDN BRI W IC Prerequisite Tasks

Before using an MRP with an ISDN BRI WIC, you must order a correctlyconfigured ISDN BRI line from your local telecommunications service provider.

The ordering process varies from provider to provider and from country to

country; however, here are some general guidelines:

Ask for two channels to be called by one number.

Ask for delivery of calling-line identification, also known as caller ID or

automatic number identification (ANI).

If the MRP or ASI will be the only device attached to the ISDN BRI line, ask

for point-to-point service and a data-only line.

If you plan to connect another ISDN device (such as an ISDN telephone) to

the ISDN BRI line through the MRP, ask for point-to-multipoint service

(subaddressing is required) and a voice-and-data line.

Note See the Configuring ISDN BRI WICs section on page 7-5 for a sample

configuration.

Table 6-14 lists the ISDN switch types for North America.

ISDN BRI Provisioning by Sw itch Type

ISDN BRI provisioning refers to the types of services provided by the ISDN BRI

line. Although provisioning is performed by your ISDN BRI service provider, you

must tell the provider what you want. Table 6-15 lists the provisioning that you

should order for switches used in North America.

Table 6-14 ISDN Switch Types for North America

ISDN Switch Type Description

basic-5ess Lucent basic rate switches

basic-dms100 NT DMS-100 basic rate switches

basic-nil1 National ISDN-1 switches


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Table 6-15 North American ISDN BRI Switch Type Configuration Information

Sw itch Type Provisioning

DMS-100 BRI Custom Two B channels for voice and data.

Two directory numbers assigned by service provider.

Two SPIDs1 required; assigned by service provider.

Functional signaling.

Dynamic TEI2 assignment.

Maximum number of keys = 64.

Release key = no, or key number = no.

Ringing indicator = no.

EKTS = no.

PVC = 2.Request delivery of calling line ID on Centrex lines.

Set speed for ISDN calls to 56 kbps outside local exchange.

Directory number 1 can hunt to directory number 2.

5ESS Custom BRI For data only:

Two B channels for data.

Point to point.

Terminal type = E.

One directory number (DN) assigned by service provider.

MTERM = 1.

Request delivery of calling line ID on Centrex lines.




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Defining ISDN SPIDs

Some service providers use service profile identifiers (SPIDs) to define the

services subscribed to by the ISDN device that is accessing the ISDN service

provider. The service provider assigns the ISDN device one or more SPIDs when

you first subscribe to the service. If you are using a service provider that requires

5ESS National ISDN (NI-1) BRI Terminal type = A.

Two B channels.


Two SPIDs required, assigned by service provider.



DMS-100 BRI Two B channels.


Two SPIDs required, assigned by service provider.

Functional signaling.Dynamic TEI assignment.

Maximum number of keys = 64.

Release key = no, or key number = no.

Ringing indicator = no.

EKTS = no.

PVC = 2.

Request delivery of calling line ID on Centrex lines.



1. SPID = service profile identifier

2. TEI = terminal endpoint identifier

Table 6-15 North American ISDN BRI Switch Type Configuration Information (continued)

Sw itch Type Provisioning



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SPIDs, your ISDN device cannot place or receive calls until it sends a valid,

assigned SPID to the service provider when accessing the switch to initialize the

connection.

At present, only the DMS-100 and NI switch types require SPIDs. The AT&T

5ESS switch type may support a SPID, but we recommend that you set up that

ISDN service without SPIDs. In addition, SPIDs have significance only at the

local access ISDN interface. Remote routers never receive the SPID.

A SPID is usually a seven-digit telephone number with some optional numbers.

However, service providers may use different numbering schemes. For theDMS-100 switch type, two SPIDs are assigned, one for each B channel.

To define SPIDs and the local directory number (LDN) for both ISDN BRI B

channels, use the following isdn spid commands in interface configuration mode:

MRP (config-if)# isdn spid1 spid-number [ldn]

MRP (config-if)# isdn spid2 spid-number [ldn]

Note Although the LDN is an optional parameter, you might need to enter it so that the

MRP or ASI can answer calls made to the second directory number.

For further information on configuring ISDN, refer to the Configuring ISDN

BRI chapter in the

Cisco IOS Dial Technologies Configuration Guide.

Configuring T1 and Fractional T1 WICs

The 1-port T1 WIC (WIC-1T) and fractional T1 WIC (WIC-1DSU-T1) include an

integrated data service unit /channel service unit (DSU/CSU) and can be

configured either for full T1 service (1.544 Mbps) or for fractional T1 service

(less than 1.544 Mbps). You can configure the interfaces on your T1 WICs by

entering IOS commands at the ASI or MRP command prompt, in configuration

mode.

The IOS software provides a default configuration for CSU/DSU- and T1-specific

parameters. To view the current configuration, enter the show service-module

serial slot/portcommand.


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Note See the Configuring T1 and Fractional T1 WICs section on page 7-7 to see thedefault configuration and a sample configuration to configure a new T1 or

fractional T1 interface or to change the configuration of an existing interface.

For further information about these commands, refer to the Configuring Serial

Interfaces for CSU/DSU Service Modules section in the Configuring Serial

Interfaces chapter in the Cisco IOS Interface Configuration Guide.

Configuring VWICs for Data-Only Transmission

You can configure the multiflex trunk (MFT) interface card as a WIC (for

data-only transmission). In the WIC mode, an MRP treats the T1 or E1 as a single

serial interface for data. You can specify the number of channels (up to 24 [T1] or

up to 30 [E1]) for this connection. On a data T1 or E1, you can configure only one

channelized group. The rest of the channels are not used.In a data-only configuration, an MRP supports the following T1 or E1

configurations:

Maximum of one T1 or E1 data port

Only one channelized T1 or E1 group for data

Maximum of two external clock sources

This section describes basic configuration, including enabling the interface and

specifying IP routing. Depending on your own requirements and on the protocols

you plan to route, you might need to enter other configuration commands as well.

Note See the Configuring VWIC Interfaces for Data section on page 7-8 for a sample

configuration to configure a new T1 or E1 VWIC interface or to change the

configuration of an existing interface.

Configuring the TDM Clock

Digital T1 and E1 interfaces require not only that you set timing, but also that you

consider the source of the timers. You must configure the time-division

multiplexing (TDM) clock to specify the clock source. You can specify up to two


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external clock sources for each MRP. This means that only two of the T1 or E1

ports can use line as the clock source. The clock source is selected via the tdm

clock global configuration command.

Scenarios for TDM Clocking

For TDM clocking scenarios and topologies, see the TDM Clocking Scenarios

section on page 6-79, which describes the basic timing scenarios that can occur

when a digital T1 or E1 interface is connected to a PBX, to a central office (CO),

or to both.

Voice over IP

This section contains information on VoIP. VoIP is a Layer 3 network protocol that

uses various Layer 2 point-to-point or link-layer protocols such as PPP, Frame

Relay, or Asynchronous Transfer Mode (ATM) for its transport. VoIP enables

Cisco routers, access servers, and multiservice access concentrators to carry and

send voice and fax traffic over an IP network. DSPs segment the voice signal into

frames and store them in voice packets. These voice packets are transported via

IP in compliance with a voice communications protocol or standard such as

H.323, MGCP, or SIP.

This section contains the following subsections:

Voice Ports Overview, page 6-37

Configuring Dial Plans, page 6-40

Configuring Analog Voice Ports, page 6-44

Configuring Digital Voice Ports, page 6-60

Configuring ISDN Interfaces for Voice, page 6-64

Configuring VoIP for Frame Relay, page 6-69

Configuring Quality of Service, page 6-71

TDM Clocking Scenarios, page 6-79

Note See the Sample H.323 Call Routing Configurations section on page 7-29 for

sample configurations,



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Table 6-16 lists tasks that you might need to perform in order to configure VoIP

on your MRP or ASI cards and gives pointers to the locations in the Cisco IOS

Voice, Video, and Fax Configuration Guide, Release 12.2, that provide

instructions on performing those tasks. The Cisco IOS Voice, Video, and Fax

Configuration Guide, Release 12.2, is available at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122cgcr/fvvf

ax_c/index.htm

Voice Ports Overview

MRP and ASI cards can provide analog and digital voice ports for

implementations of VoIP. Voice ports emulate physical telephony switch

connections so that voice calls and their associated signaling can be transferred

intact between a packet network and a circuit-switched network or device.

Table 6-16 VoIP Configuration Task Checklist


Understanding Analog Voice Port

Configuration

See Voice Port Configuration Overview in the

Cisco IOS Voice, Video, and Fax Configuration Guide,

Release 12.2.

Configuring Analog Voice Ports See Configuring Basic Parameters on Analog FXO, FXS,

or E&M Voice Ports in the Cisco IOS Voice, Video, and FaxConfiguration Guide, Release 12.2.

Configuring Digital Voice Ports See Configuring Digital Voice Ports in the Cisco IOS

Voice, Video, and Fax Configuration Guide, Release 12.2.

Configuring ISDN Interfaces for Voice See Configuring ISDN Interfaces for Voice in the


Release 12.2.

Configuring Dial Plans See Dial Plan Overview in the Cisco IOS Voice, Video,

and Fax Configuration Guide, Release 12.2.

Configuring VoIP for Frame Relay See Configuring Voice over Frame Relay in the Cisco IOS

Voice, Video, and Fax Configuration Guide, Release 12.2.

Configuring Quality of Service See Configuring Quality of Service for Voice in the


Release 12.2.


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For a voice call to occur, certain information must be passed between the

telephony devices at either end of the call, such as the devices on-hook status, the

lines availability, and whether an incoming call is trying to reach a device. This

information is referred to as signaling, and to process it properly, the devices at

both ends of the call segment (that is, those directly connected to each other) must

use the same type of signaling.

The type of signaling associated with the analog voice ports on MRP or ASI cards

depends on the VIC installed in the device.

You can install either a VIC or a VWIC in an MRP or an ASI to makevoice-related calls through the network. A VIC connects the system directly to a

regular analog phone, a fax, or a PBX. A VWIC enables 24 channels on a T1 or

30 channels on an E1 (where each channel represents a simultaneous incoming or

outgoing call). A VWIC also provides the flexibility to combine channels to form

channel groups.

Each VIC is specific to a particular signaling type; therefore, VICs determine the

type of signaling for the voice ports. Voice-port commands define thecharacteristics associated with a particular voice-port signaling type.

The voice ports support the following voice signaling types:

FXSThe foreign exchange station (FXS) interface connects directly to a

standard telephone, fax machine, PBX, or similar device and supplies ring,

voltage, and dial tone.

FXOThe foreign exchange office (FXO) interface connects local calls to aPSTN CO or to a PBX that does not support E&M signaling. This interface

is used for off-premises extension applications.

E&MThe E&M interface connects remote calls from an IP network to PBX

trunk lines (tie lines) for local distribution. It is a signaling technique used for

two-wire and four-wire telephone and trunk interfaces.

ISDN BRIThe ISDN BRI interface connects directly to PSTNs, PBXs, and

private access branch exchanges (PABXs). ISDN BRI supports on-premisesand off-premises connections.



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MFT-T1 CASThe MFT-T1 CAS interface connects remote calls from an IP

network to the PBX and the CO.

T1/E1 PRIThe T1 PRI interface provides transmission of digital data over

23 B-channels and one D-channel (or 23B+D), for a total bandwidth of 1.544

Mbps. The E1 PRI provides transmission of digital data over 30 B-channels

(64 Kbps) and one D-channel (64 Kbps), plus one framing channel (64 Kbps),

for a total bandwidth of 2.048 Mbps.

Connecting FXS, FXO, and E&M VICs to the Telephone N etw ork

VICs provide the connection to the telephone equipment or network, as follows:

FXS Interfaces, page 6-39

FXO Interfaces, page 6-39

E&M Interfaces, page 6-40

FXS Interfaces

Interfaces on FXS VICs are color-coded gray. Use a standard RJ11 modular

telephone cable to connect this interface to a telephone or fax machine.

Caution Do not connect an FXS interface directly to the PSTN.

FXO Interfaces

Interfaces on FXO VICs are color-coded pink. The following types of FXO

interfaces are available:

VIC-2FXOIntended for use in North America. In the United States,

Canada, and Mexico, use a standard RJ11 modular telephone cable to connect

the VIC-2FXO to the PSTN or PBX through a telephone wall outlet.

VIC-2FXO-M1Provides battery reversal detection and caller ID support

(for use in North America).


(for use in Europe).


(for use in Australia).



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In countries where the PSTN does not use RJ11 wall outlets, use a suitable adapter

to convert the plug on an RJ11 modular cable to the connector used by the local

wall outlet. These adapters are not sold by Cisco Systems, but they are availablefrom other vendors, such as TeleAdapt. You can obtain additional information

from TeleAdapt at http://www.teleadapt.com .

Caution Connect only an FXO interface approved for use in your country to the PSTN.

Otherwise, connect the FXO interface only to a PBX. Connections from the PBX

to the PSTN are permitted.

E&M Interfaces

Interfaces on E&M VICs are color-coded brown. The E&M voice interface card

uses an RJ48S connector. The pinout depends on the PBX type and connection.

Caution Do not connect an E&M interface directly to the PSTN.

Note If your MRP is configured with two VICs, a total of four telephones and fax

machines can be connected to it. As the MRP has only two 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 four voice devices, you

need to add an ASI, add MRPs, or use an E&M VIC and a local PBX, rather than

connecting every telephone to its own FXS VIC.

Configuring Dial Plans

Use a dial plan to map the destination telephone numbers with the voice ports onthe MRP. 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

countrys dialing plan follows the country code.


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In most corporate environments, the telephone network is configured so that users

can reach a destination by dialing only a portion (an extension number) of the full

E.164 telephone number. VoIP can be configured to recognize extension numbersand expand them into their full E.164 dialed numbers by using two commands in

tandem: destination-pattern and num-exp. Before you configure these two

commands, it is useful to 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:

A POTS dial peer associates a physical voice port with a local telephone

device. The key commands you need to configure are the port and

destination-pattern .

The port command associates the POTS dial peer with a specific logical

dial interface, which is typically the voice port connecting the MRP to the

local POTS network. The destination-pattern command defines the telephone number

associated with the POTS dial peer.

A VoIP dial peer associates a telephone number with an IP address. The key

commands you need to configure are the destination-pattern and session

target.

The destination-pattern command defines the telephone number

associated with the VoIP dial peer.

The session target command specifies a destination IP address for the

VoIP dial peer.

Note See the Configuring Dial Plans section on page 7-29 for sample configurations.

Use the dial-peer voice command to define dial peers and to change to dial-peer

configuration mode. See the Configuring Analog Voice Ports section on

page 6-44 for additional information.



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Create a Number Expansion Table

In Figure 6-1, a small company decides to use VoIP to integrate its telephonynetwork with its existing IP network. The destination pattern (or expanded

telephone number) associated with MRP 1 (at the left of the IP cloud) is

(408) 555-xxxx, where xxxx identifies the individual dial peers by extension. The

destination pattern (or expanded telephone number) associated with MRP 2 (at the

right of the IP cloud) is (729) 555-xxxx.

Figure 6-1 Sample VoIP Network

Table 6-17 shows the number expansion table for this scenario.

IP IP

IP cloud

408 555-1001

Voiceport 0/0

Voiceport 0/1

Voiceport 1/0

Voiceport 0/0

Cisco ICS 7750MRP 1

Cisco ICS 7750MRP 2

WAN

10.1.1.1

WAN

10.1.1.2Voiceport 0/1

Voiceport 1/0

408 555-1002

Analogphone

408 555-2001 408 555-2002 IP phone IP phone

729 555-4001 729 555-4002

729 555-3001

729 555-3002

3

3022



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Note You can use a period (.) to represent variables (such as extension numbers) in a

telephone number. A period is similar to a wildcard, which matches any entered

digit.

The configuration shown in Table 6-17 needs to be made to both MRP 1 and MRP

2. Based on this configuration, MRP 1 can call any number string that begins with

the digits 17295553 or 17295554 to connect to MRP 2. Similarly, MRP 2 can call

any number string that begins with the digits 14085551 and 14085552 to connect

to MRP 1.

Note See the Configuring Number Expansion section on page 7-30 for a sample

configuration that expands an extension number into a particular destination

pattern.

Table 6-17 Sample Number Expansion Table

ExtensionDestinationPattern

Num-ExpCommand Entry Description

1... 14085551... num-exp 1...

14085551...

Expands a 4-digit extension beginning

with the numeral 1 by prefixing 1408555

to it

2... 14085552... num-exp 2...

14085552...


with the numeral 2 by prefixing 1408555to it

3... 17295553... num-exp 3...

17295553...



to it

4... 17295554... num-exp 4...

17295554...



to it



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Configuring Analog Voice Ports

This section contains the following subsections:

Configuring FXS Interfaces, page 6-44

Local Dial Peers, page 6-45

Calling Between MRPs, page 6-49

Other MRPs on the Network, page 6-52

Configuring DID for ISDN, page 6-53

Configuring FXO Interfaces, page 6-55

Configuring E&M Interfaces, page 6-57

Configuring FXS Interfaces

This section explains how to configure ports on FXS VICs that connect directly

to a standard telephone, fax machine, or similar device.

Figure 6-2shows a basic voice network. A small business uses a MRP card

(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

MRP.

Figure 6-2 Basic Voice Network (West M RP)

408 555-3737

408 555-4141

FXS VIC0/0

FXS VIC0/1

West IP cloud

22634



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Note You can name your MRP by using the global configuration hostname command.

Table 6-18 lists telephone numbers and voice ports for the West MRP.

Note If your MRP is configured with two VICs, a total of four telephones and fax

machines can be connected to it. As the MRP has only two slots, you need to

replace one VIC with a WIC to provide an interface for IP connectivity to theWAN and for data traffic. To accommodate more than four voice devices, you

need to add an ASI, add MRPs, or use an E&M VIC and a local PBX, rather than

connecting every telephone to its own FXS VIC.

Local Dial Peers

To route a received voice call to the proper destination, the MRP needs to haveaccess to the telephone number that belongs to each voice port. For instance, if a

call comes in for 408 555-3737, the MRP needs to correlate that phone number

with a particular voice port (voice port 0/0, as shown in Figure 6-2.) In other

words, the MRP needs to have access to the information in Table 6-18.

To hold this information, 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 configuringstatic IP routesyou are telling the MRP 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 example, segments occur

between a telephone and an MRP, an MRP and a network, an MRP and a PBX, or

an MRP 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 thatcan range from 1 to 2311(2147483647). Within the allowed range, you can

Table 6-18 West MRP Telephone Numbers and Voice Ports

Te le phone Numbe r Voice Port

408 555-3737 0/0408 555-4141 0/1



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