Chapter 1: Multisite Deployments

Cisco Press

This chapter reviews issues and solutions related to multisite deployments. It also covers the connections between these sites and dial plans for such configurations.

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	Excerpt from CCVP CIPT2 Quick Reference.
By Anthony Sequeira Published by Cisco Press ISBN-10: 1-58705-811-1 ISBN-13: 978-1-58705-811-0

This chapter reviews issues and solutions related to multisite deployments. It also covers the connections between these sites and dial plans for such configurations.

Issues

This section identifies the issues that can arise in a multisite Cisco Unified Communications Manager deployment.

Issues Overview

Here are the major issues that can result:

• Quality of Service—Jitter must be avoided.

• Bandwidth—Optimize the use of bandwidth to ensure correct amounts for voice and video.

• Availability—Fallback solutions need to be implemented.

• Dial plan—Overlapping directory numbers must be solved.

• NAT and Security—Translation results in visibility on the Internet; IPsec and VPN tunnels can be used between sites.

Solutions

This section reviews the solutions to issues that occur in Cisco Unified Communications Manager multisite deployments.

Quality of Service

To solve bandwidth and jitter (quality) issues, Quality of Service (QoS) is used. Traffic is identified and divided into classes, and then a QoS policy is applied per class. With QoS enabled, voice traffic is given absolute priority over all other traffic.

Bandwidth

Bandwidth on the IP WAN can be conserved using the following techniques:

• Using low-bandwidth codecs

• Using RTP-header compression

• Deploying local annunciators or disabling remote annunciators

• Deploying local conference bridges

• Deploying local Media Termination Points (MTP)

• Deploying transcoders or mixed conference bridges

• Deploying local Music on Hold (MOH) servers

• Using multicast MOH from branch router flash

• Limiting the number of voice calls using call admission control (CAC)

Availability

Solutions include the following:

• PSTN backup

• MGCP fallback

• Fallback for IP Phones

• Call Forward Unregistered (CFUR)

• Automated alternate routing (AAR) and Call Forward on No Bandwidth (CFNB)

• Mobility solutions

Dial Plan

Solutions include the following:

• Access codes and site codes for intersite dialing of Multisite Connections

• Variable-length numbering plans

• Direct inward dialing (DID) ranges and E.164 addressing

• Different number presentation in ISDN (Type of Number [TON])

• Toll bypass, tail-end hop-off (TEHO), and PSTN backup

NAT

To solve NAT security issues, use Cisco Unified Border Element as an application proxy.

Multisite Connections

This section reviews the configuration of gateways and trunks in multisite environments.

MGCP Gateway Implementation

First, add the MGCP gateway to Cisco Unified Communications Manager. Then, add MGCP endpoints to the gateway, and configure the endpoints. Next, configure the gateway. Cisco Unified Communications Manager stores an XML configuration file in its TFTP server. Use the commands ccm-manager config server and ccm-manager config.

H.323 Gateway Implementation

In Cisco Unified Communications Manager, complete the following:

Step 1. Create the gateway and specify its IP address.
Step 2. Create a route group and put the gateway into it.
Step 3. Create a route list and put the route group into it.
Step 4. Create one or more route patterns pointing to the route list.

The Cisco IOS gateway configuration includes these steps:

Step 1. Configure the H.323 gateway, specifying its H.323 ID and the IP address to use.
Step 2. Configure one or more VoIP dial peers pointing to Cisco Unified Communications Manager.
Step 3. Configure one or more POTS dial peers pointing to the PSTN.

Implementing SIP Trunks

To add a session initiation protocol (SIP) trunk in Cisco Unified Communications Manager, complete the following:

Step 1. Navigate to Device > Trunk and click Add New.
Step 2. In the Trunk Type drop-down list, choose SIP Trunk and click Next.
Step 3. In the Trunk Configuration window, enter a name and description for the SIP trunk and choose the device pool that should be used.
Step 4. In the SIP Information area of the Trunk Configuration window, enter the destination address.
Step 5. Choose a SIP Trunk Security Profile and a SIP Profile.

Dial Plans for Multisites

This section reviews the implementation of a dial plan to support inbound and outbound PSTN dialing, site-code dialing, and tail-end hop-off.

Overview

Dial plan solutions for a multisite environment with centralized or distributed call processing include the following:

• Access and site codes

• Implementing PSTN access

• Implementing PSTN backup

Implementing Site Codes for On-Net Calls

Site-code dialing is used when two sites have overlapping and nonconsecutive directory numbers. Users dial an access code followed by a three-digit site code. When distributed call processing is used, each Cisco Unified Communications Manager cluster is only aware of its own directory numbers in detail. For all directory numbers located at the other site, the call is routed to a Cisco Unified Communications Manager server at the other site based on the dialed site code. The access and site code have to be stripped off from the Dialed Number Identification Service (DNIS) on outgoing calls. For incoming calls, the access code and appropriate site code that are used to get to the caller’s site have to be added; this is done using translation patterns.

If overlapping directory numbers exist in a centralized call-processing deployment, access and site codes are implemented in a different way. Partitions and CSSs need to be deployed so that phones at the remote site do not see directory numbers of main-site phones and vice versa. A translation pattern then is added per site.

Implementing PSTN Access

When implementing PSTN access, both outgoing calls and incoming calls require Automatic Number Identification (ANI) transformation and DNIS transformation. For outgoing calls, if no direct inward dialing range (DID) is used at the PSTN, transform all directory numbers to the same, single PSTN number in the ANI. If DID is used, extend the directory numbers to a full PSTN number. For the DNIS transformation, strip off the access code. For incoming calls, transform ANI to the full number and add the access code. If DID is used, strip off the office code, area code, and country code to get to the directory number. If DID is not used, route the call to attendant or interactive voice response (IVR) application.

Implementing Selective PSTN Breakout

Multisite deployments often feature one PSTN gateway per site. Selective PSTN breakout ensures that local gateways are used to access the PSTN. This is ensured with route patterns. These route patterns are put into different partitions and point to different PSTN gateways. IP Phones need to be configured with a CSS that includes only the route patterns that refer to the local gateway.

Considerations When Using Backup PSTN Gateways

There are two options for configuring the ANI for the outgoing call. Use the primary gateway’s PSTN number on the secondary gateway or use the secondary gateway’s PSTN number.

Implementing PSTN Backup for On-Net Intersite Calls

Intersite calls should use the intercluster trunk (ICT) over the IP WAN. If the IP WAN is down, the PSTN should be used as a backup for intersite calls. PSTN backup for on-net calls can be easily provided by route lists and route groups, giving priority to the intercluster trunk versus the PSTN gateway.

Implementing Tail-End Hop-Off

When implementing TEHO, PSTN breakout occurs at the gateway that is closest to the dialed PSTN destination. Create a route pattern for each destination area that can be reached at different costs. These route patterns are required once per site and must be put into different partitions.

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