Cisco Unified Customer Voice Portal (CVP) Solution Reference Network Design (SRND)
Cisco Unified Customer Voice Portal (CVP) Solution Reference Network Design (SRND)
Cisco Unified Customer Voice Portal (CVP) Solution Reference Network Design (SRND)
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Cisco Unified Customer Voice Portal (CVP) 8.x Solution Reference Network Design (SRND)
© 2010 Cisco Systems, Inc. All rights reserved.
CONTENTS
Preface xiii
Audience xiii
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Comprehensive 2-6
SIP Protocol-Level Call Flow 2-8
H.323 Protocol-Level Call Flow 2-9
Transfers and Subsequent Call Control 2-10
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Overview 4-2
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Gateways 6-8
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Co-Resident Unified CVP Call Server, Media Server, and Unified CVP VXML Server 12-2
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INDEX
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Preface
This document provides design considerations and guidelines for deploying enterprise network solutions
that include the Cisco Unified Customer Voice Portal (CVP).
This document builds upon ideas and concepts presented in the latest version of the Cisco Unified
Contact Center Enterprise (Unified CCE) Solution Reference Network Design (SRND), which is
available online at
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_implementation_design_gui
des_list.html
Note Unless stated otherwise, the information in this document applies to Cisco Unified Customer Voice
Portal (CVP) 8.x (8.0 and all subsequent 8.x releases). Any differences between the various releases of
Cisco Unified CVP are specifically noted in the text.
Audience
This design guide is intended for the system architects, designers, engineers, and Cisco channel partners
who want to apply best design practices for the Cisco Unified Customer Voice Portal (CVP).
This document assumes that you are already familiar with basic contact center terms and concepts and
with the information presented in the Cisco Unified CCE SRND. To review those terms and concepts,
refer to the documentation at the preceding URL.
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Preface
Revision History
This document may be updated at any time without notice. You can obtain the latest version of this
document online at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_implementation_design_gui
des_list.html
Visit this Cisco.com website periodically and check for documentation updates by comparing the
revision date (on the front title page) of your copy with the revision date of the online document.
The following table lists the revision history for this document.
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CH A P T E R 1
Unified CVP Architecture Overview
Over the past two decades, many customers have invested in TDM-based interactive voice response
(IVR) applications to automate simple customer transactions such as checking account or 401K account
inquires. In addition, many TDM-based IVR platforms were based on proprietary development
environments and hardware platforms, which typically meant restricting the customer's integration
options with automatic speech recognition (ASR) and text-to-speech (TTS) solutions. Over the past few
years there has been a dramatic shift to using VoiceXML (VXML) standards-based technology to
support the next generation of IVR applications.
Because the implementation of Unified CVP is based on VXML, the discussion of Unified CVP begins
with the following overview of VXML as it relates to Unified CVP.
The chapter covers the following major topics:
• What is VoiceXML?, page 1-2
• What is the Cisco Unified Customer Voice Portal?, page 1-3
• Unified CVP Product and Solution Components, page 1-4
• Call Flows, page 1-17
• Design Process, page 1-19
• Quality of Service (QoS), page 1-24
• Licensing Information, page 1-24
Table 1-1 New or Changed Information Since the Previous Release of This Document
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What is VoiceXML?
Table 1-1 New or Changed Information Since the Previous Release of This Document
What is VoiceXML?
Voice eXtensible Markup Language, or VoiceXML VXML, is a markup language similar to HTML, that
is used for developing IVR services and leverages the power of web development and content delivery.
VoiceXML was designed for creating audio dialogs that feature synthesized speech, digitized audio,
recognition of speech or dual-tone multifrequency (DTMF) key input, and recording of spoken input. It
is a common language for content providers, tool providers, and platform providers, and it promotes
service portability across implementation platforms.
VoiceXML separates service logic from user interaction and presentation logic in VoiceXML voice web
pages. It also shields application authors from low-level, platform-specific IVR and call control details.
VoiceXML is easy to use for simple interactions, yet it provides language features to support complex
IVR dialogs.
VoiceXML programs are rendered (or executed) by a VoiceXML browser, much like an HTML program
is rendered via an internet browser (such as Internet Explorer). A Cisco Voice Gateway (or router) can
provide the VoiceXML browser function. For small deployments, the Ingress Voice Gateway and
VoiceXML Gateway are typically deployed in the same router. The Cisco IOS VoiceXML Gateway
provides both gateway and VoiceXML browser functions.
In the most simple call processing scenario, a new call arrives and the voice gateway dial peer matches
the call to an available VoiceXML gateway port. The VoiceXML gateway port represents a Voice over
IP (VoIP) endpoint and can be logically thought of as a voice response unit (VRU) port. Upon arrival of
the new call, the VoiceXML gateway (that is, the VRU) sends an HTTP request to a Cisco Unified CVP
VXML Server for instruction. The URL contained in the HTTP request correlates to a specific
VoiceXML doc.
In response to the HTTP request, the Unified CVP VXML Server sends the requested, dynamically
generated VoiceXML doc to the VoiceXML gateway (that is, the voice browser) to be rendered. A typical
VoiceXML doc is short and prompt the caller for some input, then includes the results in a new HTTP
request that redirects the caller to another URL and VoiceXML doc. Because a typical call requires
numerous prompts and caller inputs, there are numerous VoiceXML documents that need to be rendered
and a large number of possible paths through these VoiceXML documents.
To logically link the many different VoiceXML documents that may need to be rendered and to greatly
simplify the task of creating VoiceXML documents, a graphical scripting tool is often used to allow the
IVR service developer to easily develop complete IVR services with conditional logic and customer
relationship management (CRM) database integration. Cisco Unified Call Studio is one such scripting
tool. The Cisco Unified CVP VXML Server is capable of executing scripts developed with Cisco Unified
Call Studio, and both were designed to work with Cisco Unified CVP Server, Cisco Voice Gateways,
Cisco VoiceXML Gateways, Cisco Unified Communications Manager, Cisco Unified Contact Center,
and Cisco's VoIP-enabled LAN/WAN.
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What is the Cisco Unified Customer Voice Portal?
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Unified CVP Product and Solution Components
• VRU reporting
Access historical data using its included centralized reporting database. Design and run custom
reports using its well-documented schema.
• Compatibility and Integration
– Use with other Cisco Call Routing and VoIP Products, including, Cisco Unified Intelligent
Contact Management Hosted or Cisco Unified Intelligent Contact Management Enterprise,
Cisco Gatekeepers, Cisco Gateways, and Unified Contact Center Enterprise (UCCE).
– Use with Cisco Unified Communications Manager (Unified CM). Unified CM manages and
switches VoIP calls among IP phones. When combined with Unified ICME, Unified CM
becomes the UCCE product.
– Use with the Public Switch Telephone Network (PSTN). Calls can be moved onto an IP-based
network for Unified CVP treatment and then moved back out to a PSTN for further call routing
to a call center.
– Integration with Cisco Unified Contact Center (details)
Unified CVP integrates with Cisco Unified Contact Center via a VRU Peripheral Gateway (PG).
This integration enables Cisco Unified Contact Center Enterprise (Unified CCE) to control
Unified CVP VoIP switching and IVR services. It also enables Unified CCE to control the agent
selection application and to initiate the Real-Time Transport Protocol (RTP) stream transfer
from the VoiceXML gateway to the selected agent. Unified CVP integration with Unified CCE
requires that the traditional Cisco Unified Communications Manager PG be used for Unified
CCE integration with Cisco Unified Communications Manager.
Unified CCE can be integrated with Unified CVP via the Cisco Unified Intelligent Contact
Manager (ICM) System PG and the parent-child deployment model. This integration method
provides callers with some simple up-front menus and prompts by the parent Unified ICM and
Unified CVP, and it intelligently routes the calls via skill groups to the best Cisco Unified
Contact Center Express or Enterprise child. Queuing control and agent selection are handled by
the child contact center solution. In this model, it is also easy for a TDM automatic call
distributor (ACD) to play the role of a child. All call transfers between Unified CVP and
children will retain call data, and the ICM will provide enterprise-wide browser-based
consolidated reporting.
Unified CVP integration is not directly supported with the Unified CCE System PG (which is
also used by System Unified CCE). The Unified CCE System PG supports only the Cisco
Unified IP IVR. Unified CVP works only with System PG children via the parent-child
deployment model. Unified CVP can also provide IVR services for Unified CCE outbound IVR
campaigns and post-call customer surveys.
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Note A Unified CVP server can, optionally, be part of the enterprise domain.
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Integrated into this B2BUA is the ability to interact with the Cisco Unified ICM via the ICM Service.
This integration provides the ability for the SIP Service to query the Unified ICM for routing
instruction and service control. This integration also allows Unified ICM to initiate subsequent call
control to do things such as requesting that a caller be transferred from queue to an agent or
transferred from one agent to another agent.
• ICM Service
This service is responsible for all communication between Unified CVP components and Unified
ICM. It sends and receives messages on behalf of the SIP Service, the IVR Service, and the H.323
Service.
• IVR Service
This service creates the VoiceXML pages that implement the Unified CVP Microapplications based
on Run VRU Script instructions received from Unified ICM. The IVR Service functions as the VRU
leg (in Unified ICM Enterprise parlance), and calls must be transferred to it from the SIP Service in
order to execute microapplications. The VoiceXML pages created by this module are sent to the
VoiceXML gateway to be executed.
• H.323 Service (Formerly known as the Unified CVP Voice Browser)
This service interacts with the IVR Service to relay call arrival, release, and transfer call control
between it and the other H.323 components. This service is needed only for deployments using
H.323.
A Unified CVP Call Server can be deployed co-resident with the Unified CVP VXML Server or a media
server. Optionally, a Unified CVP Call Server can be deployed as part of the Enterprise Windows
Domain.
For hardware details, refer to the latest version of the Hardware and System Software Specification for
Cisco Unified CVP (formerly called the Bill of Materials), available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/prod_technical_reference_list.html
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The Operations Console also offers a direct link to Support Tools, which can collect trace logs and
perform other diagnostic and instrumentation functions on many solution components. The Operations
Console is, in effect, the dashboard from which an entire Unified CVP deployment can be managed.
The Operations Console must itself be configured with a map of the deployed solution network. It can
then collect and maintain configuration information from each deployed component. Both the network
map and the configuration information are stored locally on the server, where it can be backed up by
off-the-shelf backup tools. A web browser-based user interface, the Operations Console, provides the
ability to both display and modify the network map and the stored configuration data and to distribute
such modifications to the affected solution components.
The Operations Console can display two views of configuration parameters for managed components.
The runtime view shows the status of all configuration parameters as those components are currently
using them. The configured or offline view shows the status of all configuration parameters that are
stored in the Operations Server database and will be deployed to the device the next time a Save and
Deploy option is executed.
The Operations Console allows configuration parameters to be updated or preconfigured even when the
target component is not online or running. If the target server (without its services) comes online, the
user can apply the configured settings to that server. These settings will become active when that server's
services also come online. Only then will they be reflected in the runtime view.
The Operations Console Server is not a redundant component. As such, the Operations Console Server
cannot be duplicated within a deployment. Backups of the configuration database should be take
regularly or whenever changes are made.
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The Ingress Gateway can be deployed separately from the VoiceXML Gateway, but in most
implementations they are one and the same: one gateway performs both functions. Gateways are often
deployed in farms, for Centralized deployment models. In Branch deployment models, one combined
gateway is usually located at each branch office.
Video Endpoints
When using the Unified CVP Basic Video Service, the following video endpoints are supported:
• Cisco Unified IP Phone 7985G
• Cisco Unified Video Advantage
• Cisco TelePresence
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Cisco Gatekeeper
The gatekeeper is a network element used by H.323 gateways for call routing. The gatekeeper is required
for all H.323 installations for dial plan configuration and bandwidth management.
Scenarios for gatekeeper usage include:
• To map specific dialed numbers to specific Unified CVP Servers or VoiceXML gateways.
• To load-balance new calls to a set of Unified CVP Servers or VoiceXML gateways.
• To route the transfer of callers from a VoiceXML gateway port to a Cisco IP Phone.
• To provide failover capabilities for H.323 endpoints.
The Gatekeeper is the focus for high availability design in the H.323 protocol arena, and it is only used
in Unified CVP implementations that use H.323 for call control. Like the SIP Proxy Server, it mixes
directory lookup services with load balancing and failover capabilities, producing fault tolerance among
H.323 endpoints. Unlike the SIP Proxy Server, control messages do not pass through it to target
endpoints; instead, it uses a request/response server paradigm.
Two gatekeeper failover mechanisms are supported:
• HSRP. For redundancy, Gatekeepers can be deployed in pairs using the HSRP (Hot Standby Routing
Protocol), one redundant pair per site.
• Alternate Gatekeepers. The VBAdmin SetGatekeeper command allows multiple IP addresses to be
configured. The H.323 Service keeps track of a currently active Gatekeeper from that IP list. It
begins by sending all requests to the first gatekeeper on the list.
If the currently active Gatekeeper fails, it moves to the next one in the list, and that one becomes the
current gatekeeper. The H.323 Service continues to use that gatekeeper until it too fails, at which
time it begins using the subsequent Gatekeeper in the list. When the list is exhausted, the next
failover is back to the top of the list.
For sizing purposes, each Gatekeeper should be sized to handle the entire load.
For more information on H.323 gatekeepers, refer to the Overview of the Cisco IOS Gatekeeper available
online at: http://www.cisco.com/en/US/docs/ios/12_2/gktmp/gktmpv4_2/iosgk.html.
Also consult the Cisco Gatekeeper External Interface Reference, available at
http://www.cisco.com/en/US/docs/ios/12_3/gktmpv4_3/guide/gktmp4_3.html
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Unified CVP can also be deployed without a SIP Proxy Server depending on the design and complexity
of the solution. In such cases, some of the same functions can be provided by the Unified CVP Server
SIP Service. If a SIP Proxy Server is not used, then Ingress Gateways and Unified CMs must point
directly to Unified CVP. In such a deployment, load balancing is done via DNS SRV lookups from the
gateway to the DNS Server. Load balancing of calls outbound from Unified CVP (outbound call leg) can
be done in a similar fashion.
The benefits of using a SIP Proxy Server include:
• Priority and weight routing can be used with the routes for load balancing and failover.
• If a SIP Proxy Server is already used in your SIP network, Unified CVP can be an additional SIP
endpoint—it fits incrementally into the existing SIP network.
• If the Cisco Unified Presence Server is being used as the SIP Proxy Server, dial plan management
is available in the web administration of the static routes.
• If the Cisco Unified Presence Server is being used as the SIP Proxy Server, you are better positioned
to also take advantage of Presence and Cisco Unified Client, as a compliment to Unified CVP.
If a SIP Proxy Server is not used, then Ingress Gateways and Unified CMs need to point directly to
Unified CVP. In such a deployment:
• Load balancing is done via DNS SRV lookups from Gateway to DNS Server—SIP calls can be
balanced using this mechanism.
• Load balancing of calls outbound from Unified CVP (outbound call leg) can be done in similar
fashion.
• Failover of SIP rejections (code 503 only) can also be performed if SRV records are configured with
ordered priorities.
The following guidelines apply to the use of a Cisco Unified Presence server as a SIP Proxy:
• A Unified CM publisher is required in order to install Cisco Unified Presence. Therefore, you need
at least one Unified CM publisher if you plan on using the Cisco Unified Presence server as a SIP
Proxy (even for a TDM-only deployment with no Unified CM or Unified CCE agents). Unified CM
does not need any Device License Units to perform this function.
• A Cisco Unified CM 7.x publisher can support six Cisco Unified Presence nodes per cluster,
contained in three dual-node sub-clusters.
• The Cisco Unified Presence servers can be clustered over the WAN, provided the required
conditions are met. Refer to Cisco Unified Presence Server documentation for clustering over WAN
conditions. Document is available at:
http://www.cisco.com/en/US/products/ps6837/tsd_products_support_series_home.html
In situations where redundancy across two or more sites is required but clustering over the WAN
with Cisco Unified Presence servers is not needed or cannot be supported, you will need at least one
Unified CM publisher and one Cisco Unified Presence server at each site. Cisco Unified Presence
configuration data is not shared between clusters, so you must configure each Cisco Unified
Presence server with dial plan information.
• If you have multiple Cisco Unified Presence servers, in order for them to provide redundancy to
Unified CVP, you must configure a DNS SRV record that provides load balancing and/or failover
pointing at both servers. You then configure Unified CVP to use this single DNS SRV record as the
SIP Proxy Server.
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• If you have multiple Cisco Unified Communications Manager clusters, you do not need a Cisco
Unified Presence server attached to each cluster for SIP Proxy functionality. It is possible for a
single Cisco Unified Presence server to provide SIP Proxy services for multiple clusters. However,
depending on where the clusters are located, you might want to have multiple Cisco Unified
Presence servers for redundancy (such as with clustering over the WAN).
DNS Server
This optional component may be installed anywhere in the network. Its purpose in general is to resolve
hostnames to IP addresses. Unified CVP, can make both Type A record lookups and SRV Type record
lookups. If the DNS server is slow to respond, is unavailable, is across the WAN, or so forth, this will
affect performance.
The DNS Server comes into play during SIP interactions in the following situations:
• When a call arrives at an Ingress Gateway, the dial peer can use DNS to alternate calls between the
two SIP Proxy Servers. The SIP Proxy Servers can also use DNS to distribute incoming calls among
multiple SIP Services. If SIP Proxy Servers are not being used, then the Ingress Gateway can use
DNS directly to distribute inbound calls among multiple SIP Services.
• When the SIP Service is instructed by Unified CCE to transfer the call to the VRU leg, it can use
DNS to alternate such requests between two SIP Proxy Servers. If SIP Proxy Servers are not being
used, the SIP Service can use DNS directly to distribute VRU legs among multiple VoiceXML
Gateways.
• When transferring a call to an agent using a SIP Proxy Server, the Cisco Unified Presence Server
SIP Proxy cannot use DNS SRV for outbound calls; it must be configured with multiple static routes
in order to do load balancing and failover. (The Cisco Unified Presence Server supports DNS SRV,
but it has not been tested in Unified CVP deployments.) The static routes can point to an IP address
or a regular DNS A host record. If SIP Proxy Servers are not being used, then the SIP Service can
use DNS to locate the target agent's IP address.
The use of the DNS Server for SIP routing is entirely optional in Unified CVP. It is not required to have
a dedicated DNS Server, but the existing DNS server needs to handle the additional load of Unified CVP.
For every call destined for Unified CVP that comes into the network, there will be approximately 3 to 4
DNS lookups. You can determine the number of DNS queries per second by determining the number of
calls per second for the solution, and multiplying that number by 4.
DNS lookups are needed for DNS SRV queries, not necessarily for A record queries, which could also
be configured locally in the system "etc host" file. Unified CVP Server Groups can alternately be used
to avoid DNS SRV lookups.
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There are two ways to use the CSA feature of Unified CVP:
• If you want the protection of CSA but do not intend to customize its protection policies, install the
unmanaged Cisco Security Agent provided by Cisco.
• If you want to modify the behavior of CSA, buy and install the Cisco Security Agent Management
Console for CSA and import and modify the policies supplied with the unmanaged agent.
The Cisco Security Agent is not automatically installed by the Unified CVP installer. After installing
Unified CVP, do one of the following:
• Obtain the unmanaged version of CSA for Unified CVP and install it on the server. The unmanaged
version will include the supplied Cisco security policies which cannot be modified.
• Or, obtain the Cisco Security Management Console and install it. Then obtain the .export file that
contains the Cisco security policy, modify the policy as desired, and deploy CSA using your
modified policy.
For detailed information about Cisco Security Agent, its unmanaged and managed versions, and about
obtaining and installing the software, refer to Cisco Security Agent Installation/Deployment Guide for
Cisco Unified Customer Voice Portal, Release 8.0(1). The document is available under Install and
Upgrade Guides at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/prod_installation_guides_list.html
You can also download the software from the Cisco website, refer to:
http://tools.cisco.com/support/downloads/pub/Redirect.x?mdfid=270563413
Note Cisco does not support a user-modified version of the CSA security policy. Also, note that CSA for
Unified CVP is not supported on the device running Unified CVP Call Studio, nor on any non-CVP
devices. There are specific versions of CSA for other Cisco devices
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Note There are two features for the VXML Server that assist with load balancing:
• Limiting Load Balancer Involvement
• Enhanced HTTP Probes for Load Balancers
Refer to the configuration options ip_redirect and license_depletion_probe_error in the User Guide for
Unified CVP VXML Server and Cisco Unified Call Studio, available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_user_guide_list.html
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The Media Server can be installed co-resident with the Unified CVP Call Server or Unified CVP
VXML Server.
For the most current information on media servers, refer to the latest version of the Hardware and System
Software Specification for Cisco Unified CVP (formerly called the Bill of Materials), available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/prod_technical_reference_list.html
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Call Flows
CVP. For additional details on supported ASR and TTS products, consult the latest version of the
Hardware and System Software Specification for Cisco Unified CVP (formerly called the Bill of
Materials), available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/prod_technical_reference_list.html
Network Monitor
An SNMP management station can be used to monitor the solution deployment's health.
Call Flows
This section describes the Unified CVP call flows for both SIP and H.323 calls.
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Chapter 1 Unified CVP Architecture Overview
Call Flows
3. The SIP Service consults Unified ICME via the Unified CVP Server ICM Service, which causes
Unified ICME to run a routing script.
4. The routing script typically initiates a transfer of the call to a VoiceXML Gateway port via the SIP
service.
5. The VoiceXML Gateway sends a message to the IVR service, which requests scripted instructions
from Unified ICME.
6. Unified ICME exchanges VRU instructions with the VoiceXML gateway via the IVR service. The
instructions can include requests to invoke more sophisticated applications on the Unified CVP
VXML server. Such requests result in multiple exchanges between the Unified CVP VXML Server
and the VoiceXML Gateway to provide self-service.
7. If the customer wants to transfer to a live agent, the Unified ICME routing script queues the caller
for an available agent. While waiting for an available agent, Unified ICME provides additional
instructions to the VoiceXML Gateway to provide queueing treatment to the caller.
8. When an agent becomes available, Unified ICME sends a message to the Unified CVP Server SIP
Service, which forwards a message via the SIP Proxy Server to the Ingress Gateway and to
Unified CM to transfer the call away from the VoiceXML Gateway port and deliver it to the
Unified CM agent IP phone.
During the VRU exchanges, the VoiceXML Gateway interacts with an ASR/TTS Server to play text as
speech or recognize speech as data. It also interacts and with a Media Server (not shown in the diagram,
but connected to the VoiceXML Gateway) to fetch audio files and prompts. These two devices, as well
as the Unified CVP VXML Server, can be located behind a Content Services Switch (CSS), which offers
sophisticated failover and redundancy capability. (CSSs are optional, though recommended, and are not
displayed in the diagram.)
During this entire process, the SIP Service, the IVR Service, and the VXML Server send a stream of
reporting events to the Reporting Server (also not shown in the diagram, but connected to the Unified
CVP Call Server), which processes and stores the information in a database for later reporting. All these
devices use SNMP (Simple Network Management Protocol) to support a monitoring console. Cisco
Unified Operations Manager can also be configured to process and forward SNMP events to higher-level
monitoring stations such as HP OpenView.
All components in the solution can be managed by the Unified CVP Operations Console Server
(Operations Console). The Operations Console is not shown in the diagram, but is connected to all the
components that it manages. The Operations Console uses a variety of means to pull together the
configuration, management, and monitoring of the entire solution into a single station, which can be
accessed via a standard web browser.
VXML Server applications are designed and built using Call Studio (essentially an offline tool and not
shown in the diagram).
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Chapter 1 Unified CVP Architecture Overview
Design Process
Design Process
When designing a Unified CVP deployment consider the following high-level steps:
1. It is critical to first choose a call flow model for your functional deployment.
2. After choosing a call flow model, Unified CVP solution designers must determine where the Unified
CVP components are going to be deployed (in the data center or at a branch).
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Design Process
3. Then Unified CVP solution designers much choose the amount of availability and resiliency that is
justifiable or required.
4. Finally, Unified CVP solution designers must size the deployment to provide the justifiable or
required grade of service for the initial deployment and near-term growth.
Note For information on converting from H.323 to SIP, refer to the Configuration and Administration Guide
for Cisco Unified Customer Voice Portal.
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Design Process
How Unified CVP Routes Outbound Calls (Unified CVP Algorithm for Routing)
When you are configuring a dial plan and call routing, you can combine Unified CVP features (such as
Location Based CAC, SigDigits, SendToOriginator, LocalSRV, and Use Outbound Proxy) to achieve
your desired effect.
The following priority and conditionals are used to formulate the destination SIP URI for the outbound
calls made by Unified CVP. This description covers CONNECT messages which include labels from the
ICM (VXML GW, CUCM, etc), as well as calls to the ringtone service, recording servers, and error
message playback service.
Note The following algorithm only covers calls using the SIP subsystem, which includes audio only and basic
video SIP calls.
The algorithm for creating the destination SIP URI host portion for outbound calls, given the ICM label
is as follows.
1. At the start of the algorithm, the ICM label is provided. It may already have the Location siteID
inserted by the ICM subsystem, or SigDigits may be prepended if used. For network VRU labels,
the ICM subsystem passes in the entire prefix + correlation ID as the label.
2. If Send to Originator is matched for the Unified CCE label, the IP or hostname of the caller (ingress
gateway) is used by the Unified CVP algorithm, which returns the SIP URI.
The setting for SendtoOriginator only applies to callers who are on IOS gateways (the SIP
UserAgent header is checked), because non-IOS gateways do not have the CVP "bootstrap" service
used by the Cisco VXML gateway.
3. If use outbound proxy is set, then use the host of the proxy - return SIP URI.
4. If local static route is found for the label - return the SIP URI.
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Note • To avoid complex Dialed Number strings, do not use the Sigdigits feature if Locations CAC siteIDs
are used.
• An Outbound Proxy FQDN can be specified as a Server Group FQDN (local SRV FQDN). A local
static route destination can also be configured as a Server Group FQDN.
• Ringtone DN (91919191), Recording Server (93939393), and Error message services (92929292)
follow the same algorithm outlined above.
• SendToOriginator can work in conjunction with a REFER label.
• A REFER label can work with the SigDigits setting.
Note ASR100X platform is not supported for CUBE with CVP Solution.
CUBE on ISR gateways is supported. Also, survivability service is supported on the CUBE gateway.
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Design Process
Design Considerations
Please observe the following restrictions when deploying CUBE with SIP Trunks:
• Specifically with REFER call flows initiated with Unified CVP Call Server, CUBE does not
currently support passing the Refer-To header URI destination as-is from CVP. It rewrites the
destination address based on the dial peer configuration. The impact of this issue is that dial plan
needs to be duplicated on CVP and CUBE. The following note is further explanation.
Note IOS voice architecture and call routing is based on the assumption that dial-peer match is
required for all outbound calls. All voice components (SIP, H.323, Application) in IOS presume
that there is a matched dial-peer for an outgoing call and tries to get the call properties from that
dial-peer.
Routing a Unified CVP-initiated REFER transparently through CUBE, without a dial peer
configuration, is not supported.
• CUBE must be configured in media pass through mode in the Unified CVP deployment. Media flow
around mode cannot be used because it is not supported or validated. Only media pass through
mode, the default mode on the dial peer, is supported for CUBE.
• CUBE does not currently support a Unified CVP generated REFER message with GTD or NSS
mime body pass through. It will send the REFER, but will drop the mime body portion.
• If you plan to use the Alternate Destination Routing (ADR) feature of the service provider, do not
use Unified CVP survivability.
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Chapter 1 Unified CVP Architecture Overview
Quality of Service (QoS)
Scalability Options
After choosing the functional model and the distributed and high-availability deployment options,
Unified CVP solution designers must then size their solution and select appropriate hardware. To make
Unified CVP deployments larger, Unified CVP supports multiple gateways, Unified CVP Servers, and
Unified CVP VXML Servers.
To load-balance HTTP requests efficiently to multiple Unified CVP Servers, Unified CVP
VXML Servers, and media stores, you can use the Content Services Switch (CSS) or the Application
Content Engine (ACE) Refer to Content Services Switch, page 1-14 and Application Content Engine
(ACE), page 1-15.
For more details on choosing appropriate hardware for your deployment, see the chapter on Sizing,
page 14-1.
Virtualization
Unified CVP may be installed and run on Virtual Machines (VMs) provided by VMware software.
Running in a virtual environment has the potential for reducing the number of hardware boxes needed
to run a Unified CVP deployment, to facilitate the deployment's administration, and to leverage your
ESX (tm) infrastructure.
The following Unified CVP deployments are supported using VMware VMs:
• All SIP call flows, deployments, and features that could be installed on a physical server
• Mixed environments of physical and virtual servers
Note Deployments assume that you do not oversubscribe or overcommit the CPU and memory resources
beyond what is available physically on the host.
For specific information about virtualization with Unified CVP, refer to:
http://www.cisco.com/go/uc-virtualized.
Note For instructions on configuring QoS for Unified CVP, refer to the Operations Console online help.
For QoS design information, refer to the Enterprise QoS solution Reference Network Design Guide.
Licensing Information
For Unified CVP licensing information refer to the Cisco Customer Contact Solutions Ordering Guide.
This guide is a frequently updated source for all Unified CVP licensing information.
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Chapter 1 Unified CVP Architecture Overview
Licensing Information
Cisco employees and partners with a valid login account can access the ordering guide at:
http://www.cisco.com/web/partners/downloads/partner/WWChannels/technology/ipc/downloads/C
CBU_ordering_guide.pdf
If you need licensing information for Unified CVP but you cannot access the Ordering Guide, contact
your local Cisco Systems Engineer (SE) or Partner.
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Licensing Information
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CH A P T E R 2
Functional Deployment Models
This chapter covers the following functional deployment models for Unified CVP:
• Unified CVP VXML Server (Standalone), page 2-2
• Call Director, page 2-4
• Comprehensive, page 2-6
• VRU Only, page 2-11
For each model, this chapter provides a short discussion of the typical customer requirements for that
deployment model, a list of the required and optional components, and a step-by-step call flow.
The functional deployment models presented in this chapter assume all components are located in a
single site, and no discussion of failover is covered. Distributed deployment scenarios where
components are separated across a WAN link are discussed in the chapter on Distributed Deployments,
page 3-1. High-availability deployment options are covered in the chapter on Designing Unified CVP
for High Availability, page 4-1.
Table 2-1 New or Changed Information Since the Previous Release of This Document
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Chapter 2 Functional Deployment Models
Unified CVP VXML Server (Standalone)
Figure 2-1 Functional Deployment Model for a Unified CVP VXML Server (Standalone)
ASR/TTS
MRCP/RTSP
Unified CVP
VXML server Customer Data
(standalone) servers
VXML Backend
PSTN
270619
V Interface
IOS
Gateway
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Chapter 2 Functional Deployment Models
Unified CVP VXML Server (Standalone)
4. The Unified CVP VXML Server runs the application specified in the HTTP URL and returns a
dynamically generated VoiceXML document to the VoiceXML gateway. The Unified CVP
VXML Server application may access back-end systems to incorporate personalized data into the
VoiceXML document that is sent to the VoiceXML gateway.
5. The VoiceXML gateway parses and renders the VoiceXML document. For spoken output, the
VoiceXML gateway either retrieves and plays back prerecorded audio files referenced in the
VoiceXML document, or it streams media from a text-to-speech (TTS) server. Caller input can be
captured either by DTMF detection on the Ingress Gateway or via DTMF/speech recognition on an
ASR server.
6. As defined in the VoiceXML document, the VoiceXML gateway submits an HTTP request
containing the results of the caller input to the Unified CVP VXML Server. The Unified CVP
VXML Server again runs the application specified in the HTTP URL and returns a dynamically
generated VoiceXML document to the VoiceXML gateway. The dialog continues by repeating steps
5 and 6.
7. The IVR dialogue ends when either the caller hangs up, the application releases, or the application
initiates a transfer.
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Chapter 2 Functional Deployment Models
Call Director
Call Director
This functional deployment model provides organizations with a mechanism to route and transfer calls
across a VoIP network. The most common usage scenario for this model is for organizations with
multiple TDM ACD and TDM IVR locations that are integrated with Unified ICM via an ACD or IVR
PG, and they wish to use Unified ICM to route and transfer calls intelligently across these locations
without having to utilize PSTN pre-routing or release trunk transfer services. In this functional
deployment model, Unified CVP and Unified ICM can also pass call data between these ACD and IVR
locations. In this deployment model, Unified ICM can also provide cradle-to-grave reporting for all
calls. Although customers can have a Unified CVP Reporting Server in this deployment model, it is
optional because there is very little call information stored in the Unified CVP reporting database.
This functional deployment model is often the initial step in the migration from a TDM-based contact
center to a VoIP-based contact center. When the organization is ready to implement CVP-based IVR
services and Cisco Unified Contact Center Enterprise, the organization can migrate their Unified CVP
deployment to the comprehensive functional deployment model.
Callers can access Unified CVP via either local, long distance, or toll-free numbers terminating at
Unified CVP ingress voice gateways. Callers can also access Unified CVP from VoIP endpoints.
Call Director deployments can utilize either H.323, SIP, or a combination.
This model requires the following components:
• Ingress voice gateway(s)
• Egress voice gateway(s)
• Unified CVP Server
• Unified CVP Operations Console Server
• Cisco Unified ICM Enterprise
• H.323 gatekeeper (H.323 deployments)
• SIP Proxy Server (for SIP deployments)
Optional components for this model include:
• Unified CVP Reporting Server
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Chapter 2 Functional Deployment Models
Call Director
4. When the call arrives at the selected termination, the termination equipment sends a request to its
PG for routing instructions. This step resolves the translation route and allows any call data from the
previously run Unified ICM script to be passed to the selected termination. If the selected
termination is a TDM IVR platform, then self-service will be provided and the caller can either
release or request to be transferred to a live agent. If the selected termination is a TDM ACD
platform, then the caller will be queued until an available agent is selected by the TDM ACD. Call
data can then be popped onto the agent screen. After receiving live assistance, the caller can either
release or request to be transferred to another agent.
VoIP-based Transfer
1. Regardless of whether the call was initially routed to a TDM IVR or ACD location, the caller can
request to be transferred to another location. When this occurs, the TDM IVR or ACD sends a
post-route request with call data (via its PG) to Cisco Unified ICM.
2. When Unified ICM receives this post-route request, it runs a routing script based upon the transfer
dialed number and other criteria. The Unified ICM routing script selects a new target for the call and
then signals to the Unified CVP Server SIP Service to release the call leg to the originally selected
termination and to extend the call to a new termination.
3. When the call arrives at the new termination, the termination equipment sends a request to its PG
for routing instructions. This step resolves a translation route that was allocated for this call to this
new termination location, and it allows any call data from the previous location (IVR port or agent)
to be passed to the new termination. Calls can continue to be transferred between locations using
this same VoIP-based transfer call flow.
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Chapter 2 Functional Deployment Models
Comprehensive
platform, then the caller will be queued until an available agent is selected by the TDM ACD. Call
data can then be popped onto the agent screen. After receiving live assistance, the caller can either
release or request to be transferred to another agent.
VoIP-based Transfer
1. Regardless of whether the call was initially routed to a TDM IVR or ACD location, the caller can
request to be transferred to another location. When this occurs, the TDM IVR or ACD will send a
post-route request with call data (via its PG) to Cisco Unified ICM.
2. When Unified ICM receives this post-route request, it runs a routing script based upon the transfer
dialed number and other criteria. The Unified ICM routing script selects a new target for the call and
then signals to the Unified CVP Server H.323 Service to release the call leg to the originally selected
termination and to extend the call to a new termination. The H.323 Service queries the H.323
gatekeeper in order to get an IP address for the new termination.
3. When the call arrives at the new termination, the termination equipment sends a request to its PG
for routing instructions. This step resolves a translation route that was allocated for this call to this
new termination location, and it allows any call data from the previous location (IVR port or agent)
to be passed to the new termination. Calls can continue to be transferred between locations using
this same VoIP-based transfer call flow.
Comprehensive
This functional deployment model provides organizations with a mechanism to route and transfer calls
across a VoIP network, to offer IVR services, and to queue calls before being routed to a selected agent.
The most common usage scenario for this functional deployment model is for organizations wanting a
pure IP-based contact center. Callers are provided IVR services initially and then, upon request, are
provided queue treatment and are transferred to a selected Unified CCE agent. Upon request, callers can
also be transferred between Unified CCE agents. In this functional deployment model, Unified CVP and
Unified ICM can also pass call data between these endpoints and provide cradle-to-grave reporting for
all calls. Figure 2-2 illustrates this model.
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Chapter 2 Functional Deployment Models
Comprehensive
Customer Data
Unified CVP
ASR/TTS servers
VXML server
Backend
Interface
MRCP/RTSP
CVP
Call server
VXML GED-125
PSTN
V
SIP or H.323
IOS
Gateway
SIP or H.323
270617
Cisco Unified Agent
Communications Endpoint
Manager
This functional deployment model provides all the capabilities of the Standalone Unified CVP
VXML Server and Call Director functional deployment models, plus the ability to route and queue calls
to Unified CCE agents.
Callers can access Unified CVP via either local, long distance, or toll-free numbers terminating at the
Unified CVP ingress voice gateways. Callers can also access Unified CVP from VoIP endpoints.
Comprehensive deployments can utilize either H.323, SIP, or a combination.
This model requires the following components:
• Ingress voice gateway(s)
• VoiceXML gateway(s) (Can be co-resident with the ingress gateway)
• Unified CVP Server
• Unified CVP Operations Console Server
• Cisco Unified ICM Enterprise
• H.323 gatekeeper (for H.323 deployments)
• SIP Proxy Server (for SIP deployments)
Optional components for this model include:
• Unified CVP VXML Server
• Egress voice gateway(s)
• ASR / TTS server(s)
• Third-party media server(s)
• Content Services Switch(es)
• Application Content Engine (ACE)
• Unified CVP Reporting Server
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Chapter 2 Functional Deployment Models
Comprehensive
3. When a second Unified CCE agent becomes available, Unified ICM requests the Unified CVP
Server IVR Service to transfer the call the selected agent.
4. The IVR Service then requests the SIP Service to transfer the caller to the dialed number of the
selected agent. The SIP Service then sends a SIP invite message to the SIP Proxy Server, which finds
the Unified CM SIP Trunk IP address associated with the second agent DN, and then forwards the
SIP Invite message to Unified CM.
5. Unified CM accepts the incoming SIP trunk call and routes it to the second agent.
Note Due to a limitation in earlier versions of Cisco IOS, Cisco recommended configuring an MTP because
it was required for call flows in which the first agent consulted and was queued and then completed the
transfer before connecting to a second agent. This limitation no longer applies, and MTP configuration
is not required on SIP trunks if you are running the latest versions of Cisco IOS. Consult the Cisco
Unified CVP 7.0(2) Release Notes for details about this limitation. Also note that there are certain
situations where MTP usage can still be allocated dynamically (for example, when there is a SIP DTMF
capability mismatch).
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Chapter 2 Functional Deployment Models
VRU Only
failure indication to Unified ICM. This scenario causes a Router Requery operation. The Unified ICM
routing script then recovers control and has the opportunity to select a different target or take other
remedial action.
VRU Only
This functional deployment model provides self-service applications and queueing treatment for
organizations that are utilizing advanced PSTN switching services that are controlled via a Cisco Unified
ICM PSTN Network Interface Controller (NIC). Two Unified ICM PSTN NICs are available that allow
subsequent call control of calls in the PSTN. They are the SS7 NIC and the Carrier Routing Service
Protocol (CRSP) NIC. These NICs go beyond allowing Unified ICM to pre-route calls intelligently to
Unified ICM peripherals (such as ACDs and IVRs); they also allow Unified ICM to invoke mid-call
transfers in the PSTN. Figure 2-3 illustrates this model.
Customer Data
Unified CVP
ASR/TTS servers
VXML server
Backend
Interface
MRCP/RTSP
CVP
Call server
VXML GED-125
PSTN
V
IOS
Gateway ICM
PSTN NIC
or PGW
A typical call in this model would be pre-routed by Unified ICM to a Unified CVP Ingress Voice
Gateway for call treatment and queueing. When an agent becomes available, Unified ICM instructs the
PSTN to transfer the call to that agent. The agents can be Cisco Unified Contact Center Enterprise
agents, Cisco Unified Contact Center Express agents, or traditional ACD agents. If necessary, Unified
ICM can request the PSTN (via the NIC) to transfer the call again and again, just as Unified ICM can
request Unified CVP to transfer the call again and again. In this functional deployment model, the
Unified CVP Ingress Voice Gateway is just a Unified ICM-managed PSTN termination point that is
capable of providing VRU services via a VoiceXML gateway, the Unified CVP Server IVR Service, the
Unified CVP Server ICM Service, and Unified ICM. In this functional deployment model, neither the
Unified CVP Server H.323 Service nor the Unified CVP Server SIP Service is used for call control. All
call control and switching is controlled by Unified ICM and the PSTN. In this functional deployment
model, Unified ICM can pass call data between these termination points (for a screen pop or other
intelligent treatment) and provide cradle-to-grave reporting for all calls.
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VRU Only
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Chapter 2 Functional Deployment Models
Basic Video Service
CVP microapplications to generate VoiceXML documents for the VoiceXML gateway, and a
Unified CVP VXML Server is not required. If desired, the Unified ICM routing script can terminate
the call, and a disconnect message will be sent by the Unified ICM to the PSTN via the PSTN NIC.
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Chapter 2 Functional Deployment Models
Basic Video Service
• A TelePresence caller dials into Unified CVP, receives audio-only IVR and/or queuing treatment,
and then is transferred to an Agent on an audio-only IP Phone. MTP must be enabled on the SIP
trunk or else one-way audio is encountered.
Because the Basic Video Service is simply an extension of the SIP-based Comprehensive deployment
model, the required components and SIP protocol-level call flow details are identical.
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CH A P T E R 3
Distributed Deployments
In a distributed deployment, the ingress gateways are geographically separated from the Unified CVP
Call Server. This chapter discusses how these types of deployments should be designed as well as how
to handle call survivability and call admission control.
The chapter covers the following major topics:
• Distributed Gateways, page 3-1
• Call Survivability in Distributed Deployments, page 3-5
• Call Admission Control Considerations, page 3-6
Table 3-1 New or Changed Information Since the Previous Release of This Document
Distributed Gateways
Unified CVP can use several different types of gateways depending on the deployment model. This
section discusses each type of gateway and how a distributed deployment can affect them.
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Chapter 3 Distributed Deployments
Distributed Gateways
gateways are located at branches either for localized PSTN breakout or for integration of decentralized
TDM platforms into the Unified CVP switching solution. Apart from the gateways, all other Unified
CVP components are centrally located, and WAN links provide data connectivity from each branch
location to the central data center.
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Chapter 3 Distributed Deployments
Distributed Gateways
The term SigDigits is used to describe this feature because the command in Unified CVP to turn on
the feature and specify how many significant digits should be stripped is setsigdigits X for H.323,
and for SIP it is called Prepend Digits in the Operations Console.
This method is preferred because it involves the least amount of Unified ICM configuration
overhead; a single NetworkVRU and single set of VRU scripts and Unified ICM routing scripts is
all that is needed. This allows all of the Unified CVP servers and VoiceXML gateways to function
as a single network-wide virtual IVR from the perspective of Unified ICM.
The SigDigits feature can also be used to solve multi-cluster call admission control problems. (See
Call Admission Control Considerations, page 3-6, for more information.)
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Chapter 3 Distributed Deployments
Distributed Gateways
the ingress Unified CVP gateway correctly is complicated because the Unified CVP Call Server is the
component that is actually making the H.323 call to Unified CM. Therefore, Unified CM sees the call as
coming from the centralized Unified CVP Call Server rather than from the remote ingress gateway.
The Unified CVP Call Server is able to solve this problem by setting the sourcesignaladdress field
inside the H.323 setup to the IP address of the ingress gateway. Upon receiving the setup from Unified
CVP, Unified CM sees the source signaling address and knows that the address is the one that should be
used when determining from which location the call is coming. Because Unified CM has this ingress
gateway IP configured, Unified CM will use its Locations call admission control configuration to deduct
the bandwidth between the ingress gateway and the destination IP phone locations. The Unified CVP
Call Server should not be configured as a gateway in Unified CM; instead, the Unified CVP Call Server
should send calls to Unified CM via a gatekeeper-controlled H.323 trunk. (See Call Admission Control
Considerations, page 3-6, for more information on call admission control mechanisms.)
Design Considerations
The following considerations apply when using Multicast MOH:
• Do not set modem passthrough nse codec g711ulaw globally, or on a dial peer on the ingress/egress
gateway. This setting may cause Unified CM to stop the MOH after a timeout period of 10-12
seconds.
• Do not set media inactivity on the ingress gateway, because multicast MOH does not send RTP or
RTCP, so the call may get disconnected due to media inactivity configuration. The setting
media-inactivity-criteria all doesn't support multicast traffic.
• SIP-based Multicast MOH is not supported on the 5400 platform since ccm-manager-based MOH
subsystems are not supported on 5400 platform. This limitation also includes the ability of a TDM
caller to hear multicast packets broadcasted from the Unified CM MOH server.
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Call Survivability in Distributed Deployments
Note To take advantage of alternate routing upon signaling failures, you must use the survivability service on
all gateways pointing to Unified CVP. Always use this service, unless you have a specific
implementation that prevents using it.
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Note Router requery is not supported when using SIP Refer with Unified CVP Comprehensive Call Flow, and
the survivability service is handling the REFER message from Unified CVP. Router requery with Refer
can be supported in other call flows when IOS is handling the REFER without the survivability service,
or else Unified CM is handling the REFER. For third party SIP trunks, the support of router requery with
REFER is dependent on their implementation and support for SIP REFER itself.
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when using this model in a situation where more than one Unified CM cluster are being used to control
the remote sites. For further discussion and information on this topic, see H.323 Gatekeeper Call
Routing, page 3-10.
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deliver calls to Unified CM from Unified CVP without specifying Unified CVP as an H.323 gateway,
you must configure an H.323 Gatekeeper Trunk in Unified CM so that Unified CVP will sends calls to
Unified CM via the gatekeeper over the trunk.
The Unified CM service parameter Device Name of gatekeeper trunk that will use port 1720 is used
to force a gatekeeper-controlled trunk to register to a gatekeeper on port 1720. This feature causes any
inbound H323 call that is signaling on port 1720 to be treated immediately as a gatekeeper-controlled
trunk call. The H.225 signaling address is not examined in this case.
This behavior is not how Unified CM traditionally treats a gatekeeper-controlled H.323 call. Typically,
all gatekeeper-controlled calls come from the hub location (location None or Hub_None). These changes
ensure that the call is not treated as a gatekeeper-controlled call and that locations-based call admission
control is applied. Note that, in this model, if Unified CM does not match the gateway signaling address
in its list of configured gateways, it rejects the call. Figure 3-1 illustrates the decision tree for H.323 call
processing.
Check
Inbound H.323 call H.225 TCP connection Process call,
Match sourcing it as if from
to Cisco Unified peer IP address
CallManager against configured configured gateway.
H.323 gateways.
No Match
Equals No
Examine Port 1720 Is it configured as Accept inbound
signaling a gatekeeper H.225 TCP
port number. controlled trunk? connection.
Match
Check
H.225 signaling Process call,
Match
addressagainst sourcing it as if from
Process call as an configured H.323 configured gateway.
incoming gatekeeper gateways
controlled trunk call.
No Match
H225ReleaseComplete
Send ARQ to gatekeeper, with cause IE - Call
191324
To configure Unified CVP to work correctly with Unified CM call admission control, use VBAdmin to
set the Unified CVP Server parameter setlocationsbasedcac on. This setting tells Unified CVP to
populate the sourceCallSignalAddress field and to use TCP port 1720 when sending calls to Unified CM.
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When using this feature in conjunction with calls originated by Unified CM, the
sourceCallSignalAddress populated by Unified CVP will be the IP address of Unified CM. If the call is
transferred back to Unified CM, it will inspect this field and try to find a configured gateway with that
IP address, but the call will fail because normally Unified CM will never be configured as a gateway. As
a workaround to this problem, configure each Unified CM in the cluster as an H.323 gateway, but be sure
to never configure the Unified CM dial plan to send calls using those gateways.
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RSVP
Cisco Unified CM 5.0 introduced support for Resource Reservation Protocol (RSVP) between endpoints
within a cluster, and 8.0 UCM introduces RSVP over the SIP Trunk. RSVP is a protocol used for call
admission control, and it is used by the routers in the network to reserve bandwidth for calls. RSVP is
not qualified for call control signaling via the Unified CVP Call Server in SIP or H323 in the 8.0 release.
The recommended solution for CAC is to employ Locations configuration on Unified CVP and in
Unified CM.
For more information on RSVP, refer to the latest version of the Cisco Unified Communications SRND
Based on Cisco Unified Communications Manager, available at
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guides
_list.html
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Madison
.3 .1 M
WAN
M V
.2 M
M
Publisher
191325
group for the Madison gateway contains the session target ipv4:10.10.10.2
.1 and .2 servers.
While the example in Figure 3-2 is simple to understand, the configuration can become challenging to
maintain for several hundred remote sites over an extended period of time. If the Cisco Unified CM
gatekeeper redundancy group is changed, all the remote H.323 gateway dial-peer targets must be
changed to match the new IP address of the server added to the redundancy group. A gatekeeper can help
reduce this challenge.
When using the gatekeeper for configuration, the H.323 gateway makes a Registration Admission Status
(RAS) request to the gatekeeper for an IP address to which to send the call. The gatekeeper automatically
responds with one of the Unified CM server addresses defined in the redundancy group for the
gatekeeper trunk. If the redundancy group is changed, Unified CM must re-register to the gatekeeper.
However, no further configuration is necessary on the remote gateway.
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CH A P T E R 4
Designing Unified CVP for High Availability
This chapter describes guidelines and best practices for designing a high-availability Unified CVP
system.
This chapter covers the following topics:
• Overview, page 4-2
• Layer 2 Switch, page 4-3
• Originating Gateway, page 4-4
• SIP Proxy, page 4-5
• Unified CVP SIP Service, page 4-11
• Server Groups, page 4-14
• Gatekeeper, page 4-16
• Unified CVP H.323 Service, page 4-19
• Unified CVP IVR Service, page 4-21
• VoiceXML Gateway, page 4-23
• Content Services Switch (CSS), page 4-28
• Media Server, page 4-30
• Unified CVP VXML Server, page 4-31
• Automatic Speech Recognition (ASR) and Text-to-Speech (TTS) Server, page 4-32
• Cisco Unified Communications Manager, page 4-33
• Intelligent Contact Management (ICM), page 4-34
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Chapter 4 Designing Unified CVP for High Availability
What's New in This Chapter
Table 4-1 New or Changed Information Since the Previous Release of This Document
Overview
A high-availability design provides the highest level of failure protection. Your solution may vary
depending upon business needs such as:
• Tolerance for call failure
• Budget
• Topological considerations
Unified CVP can be deployed in many configurations that use numerous hardware and software
components. Each solution must be designed in such a way that a failure impacts the fewest resources
in the call center. The type and number of resources impacted depends on how stringent the business
requirements are and which design characteristics you choose for the various Unified CVP components,
including the network infrastructure. A good Unified CVP design is tolerant of most failures (defined
later in this chapter), but sometimes not all failures can be made transparent to the caller.
Unified CVP is a sophisticated solution designed for mission-critical call centers. The success of any
Unified CVP deployment requires a team with experience in data and voice internet working, system
administration, and Unified CVP application configuration.
Before implementing Unified CVP, use careful preparation and design planning to avoid costly upgrades
or maintenance later in the deployment cycle. Always design for the worst possible failure scenario, with
future scalability in mind for all Unified CVP sites.
In summary, plan ahead and follow all the design guidelines and recommendations presented in this
guide and in the latest version of the Cisco Unified Communications Solution Reference Network Design
(SRND) Based on Cisco Unified Communications Manager, available at:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guides
_list.html
For assistance in planning and designing your Unified CVP solution, consult your Cisco or certified
Partner Systems Engineer (SE).
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Layer 2 Switch
Layer 2 Switch
Figure 4-1 shows a high-level layout for a fault-tolerant Unified CVP system. Each component in the
Unified CVP site is duplicated for redundancy. The quantity of each of these components varies based
on the expected busy hour call attempts (BHCA) for a particular deployment. The following sections
describe the failover strategy for each of these components.
PSTN
SIP Proxy / SIP Proxy /
Gatekeeper Gatekeeper
Call V V Call
server server
VXML VXML
server server
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Originating Gateway
In Figure 4-1, two switches provide the first level of network redundancy for the Unified CVP Servers:
• If one switch fails, only a subset of the components becomes inaccessible. The components
connected to the remaining switch can still be accessed for call processing.
• If a Content Services Switch (CSS) is used, its redundant partner must reside on the same VLAN in
order to send keep-alive messages to each other via Virtual Router Redundancy Protocol (VRRP), a
protocol similar to Hot Standby Router Protocol (HSRP). If one of the switches fails, the other CSS
is still functional.
• Although Figure 4-1 shows a CSS being used for redundancy, you may also use the Application
Content Engine (ACE). Refer to Application Content Engine (ACE), page 1-15.
For more information on data center network design, refer to the Data Center documentation available at
http://www.cisco.com/go/designzone
Note NIC teaming is not currently supported in the Unified CVP solution.
Note Cisco recommends that the NIC card and ethernet switch be set to 100 MB full duplex for 10/100 links,
or set to auto-negotiate for gigabit links.
Originating Gateway
The function of the originating gateway in a Unified CVP solution is to accept calls from the PSTN and
direct them to Unified CVP for call routing and IVR treatment.
This section covers the following topics:
• Configuration, page 4-4
• Call Disposition, page 4-19
Configuration
For the most current information on how to provide redundancy and reliability for originating gateways
and T1/E1 lines, refer to the latest version of the Cisco Unified Contact Center Enterprise Solution
Reference Network Design (SRND), available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_implementation_design_gui
des_list.html
In addition, consider the following issues when designing gateways for high availability in a Unified
CVP solution:
• When used in ICM-integrated models, the originating gateway communicates with Unified CVP
using H.323 or SIP. Unlike MGCP, SIP and H.323 do not have redundancy features built into the
protocol. Instead, SIP and H.323 rely on the gateways and call processing components for
redundancy.
• When configuring the gateway, it is best to bind the H.323 or SIP signaling to the virtual loopback
interface, as illustrated in the following configuration examples:
H.323:
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SIP Proxy
interface Loopback0
ip address 10.0.0.10 255.255.255.255
h323-gateway voip interface
h323-gateway voip id sj-gk ipaddr 10.0.1.100 1719 <<- GK IP
h323-gateway voip h323-id sj-gw1
h323-gateway voip bind srcaddr 10.0.0.10
SIP:
voice service voip
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
This configuration allows call signaling to operate independent of the physical interfaces. In this
way, if one interface fails, the other interface can handle the traffic. Each gateway interface should
be connected to a different physical switch to provide redundancy in the event that one switch or
interface fails. Each interface on the gateway is configured with an IP address on a different subnet.
The IP Router(s) for the network is then configured with redundant routes to the Loopback address
through the use of static routes or a routing protocol. If a routing protocol is used, pay careful
attention to the number of routes being exchanged with the gateway, and consider using filters to
limit the routing updates so that the gateway is only advertising the loopback address and not
receiving routes.
Call Disposition
If the originating gateway fails, the following conditions apply to call disposition:
• Calls in progress are dropped. There is nothing that can be done to preserve these calls because the
PSTN switch has lost the D-channel to all T1/E1 trunks on this gateway.
• New calls are directed by the PSTN carrier to a T1/E1 at an alternate gateway, provided the PSTN
switch has its trunks and dial plan configured to do so.
SIP Proxy
A SIP Proxy server plays a similar role to the gatekeeper in a Unified CVP solution. The SIP Proxy
server provides dial plan resolution on behalf of SIP endpoints, allowing dial plan information to be
configured in a central place instead of statically on each SIP device. A SIP Proxy server is not required
in a Unified CVP solution, but it is used in most solutions because of the benefits of centralized
configuration and maintenance. Multiple SIP Proxy servers can be deployed in the network to provide
load balancing, redundancy, and regional SIP call routing services. In a Unified CVP solution, the
choices for SIP call routing are:
• SIP Proxy Server
– Advantages:
Weighted load balancing and redundancy.
Centralized dial-plan configuration.
SIP Proxy may already exist or be used for other applications for dial-plan resolution or
intercluster call routing.
– Disadvantages:
Additional server and/or hardware required for SIP Proxy if not already deployed.
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SIP Proxy
• Static routes using Server Groups (DNS SRV records) on a DNS Server
– Advantages:
Weighted load balancing and redundancy.
– Disadvantages:
Might not be able to use an existing DNS server, depending on the location of the DNS server.
The ability to share or delegate DNS server administration rights might not be possible in some
organizations.
Dial-plan configuration needs to be configured on each device individually (Cisco Unified
Communications Manager, Unified CVP, and gateways).
DNS SRV lookup is performed for each and every call by Unified CVP. If the DNS server is
slow to respond, is unavailable, is across the WAN, or so forth, this will affect performance.
• Static routes using local DNS SRV records
– Advantages:
Weighted load balancing and redundancy.
Does not depend on an external DNS Server, thus eliminating a point of failure, latency, and
DNS Server performance concerns.
– Disadvantages:
Dial plan must be configured on each device individually (Cisco Unified Communications
Manager, Unified CVP, and gateways).
Note The options for static routes using SRV with a DNS Server, or using Server Groups, can introduce some
unexpected, long delays during failover and load balancing with UDP transport on the Unified CVP Call
Server when the primary destination is shut down or is off the network. With UDP, the per-hop delay is
3.5 seconds (if retry count is 2) or 7.5 seconds (if retry count is 3). This delay is on every call or every
other call (on average) during failure, depending on load balancing, and is accordance with section
17.1.1.1 of RFC 3261 regrading the T1 timer.
Note Due to long delays when DNS is used with the Cisco Unified Presence proxy server, Cisco recommends
that you disable the DNS server on the Cisco Unified Presence server (CUP Server). Refer to CUP Server
release notes for more information.
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SIP Proxy
This method:
• Consists of 2 gateways for redundancy, geographically separated, 1 proxy module each, using SRV
priority for redundancy of proxies, no HSRP.
• The ISR is dedicated to the proxy blade function and is not co-located as a VXML gateway, or as a
TDM gateway, due to platform validation restrictions on CUSP.
• TDM gateways are configured with SRV or with Dial Peer Preferences to use the primary and
secondary CUSP proxies.
• CUSP is set with Server Groups to find primary and back up Unified CVP, Unified CM and VXML
gateways.
• Unified CVP is set up with Server Group to use the primary and secondary CUSP proxies.
• Cisco Unified CM is set up with a Route Group with multiple SIP Trunks, to use the primary and
secondary CUSP proxies.
Example of Option A
In this example, ISR1 is on the east coast and ISR2 is on the west coast. The TDM gateways will use the
closest ISR, and only cross the WAN when needing to failover to the secondary priority blades.
The SRV records look like this:
east-coast.proxy.atmycompany.com
blade 10.10.10.10 priority 1 weight 10 (this blade is in ISR1 on east coast)
blade 10.10.10.20 priority 2 weight 10 (this blade is in ISR2 on west coast)
west-coast.proxy.atmycompany.com
blade 10.10.10.20 priority 1 weight 10 (this blade is in ISR2 on west coast)
blade 10.10.10.10 priority 2 weight 10 (this blade is in ISR1 on east coast)
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SIP Proxy
This method:
• Consists of 2 gateways for redundancy, 2 proxy modules in each chassis. All 4 proxy servers are in
active mode with calls being balanced between them.
• Uses SRV to load balance across proxies with priority.
• The ISR is dedicated to the proxy blade function and is not co-located as a VXML gateway, nor as
a TDM gateway, due to platform validation restrictions on CUSP.
• TDM gateways are set with SRV or with Dial Peer Preferences to use the primary and secondary
CUSP proxies.
• CUSP is set with Server Groups to find primary and back up Unified CVP, Unified CM and VXML
gateways.
• Unified CVP is set up with Server Group to use the primary and secondary CUSP proxies.
• Cisco Unified CM is set up with Route Group with multiple SIP Trunks, to use the primary and
secondary CUSP proxies.
Example of Option B
With this example ISR1 is on the east coast and ISR2 is on the west coast. The TDM gateways will use
the closest ISR, and only cross the WAN when needing to failover to the secondary priority blades. The
SRV records look like this:
east-coast.proxy.atmycompany.com
blade 10.10.10.10 priority 1 weight 10 (this blade is in ISR1 on east coast)
blade 10.10.10.20 priority 1 weight 10 (this blade is in ISR1 on east coast)
blade 10.10.10.30 priority 2 weight 10 (this blade is in ISR2 on west coast)
blade 10.10.10.40 priority 2 weight 10 (this blade is in ISR2 on west coast)
west-coast.proxy.atmycompany.com
blade 10.10.10.30 priority 1 weight 10 (this blade is in ISR2 on west coast)
blade 10.10.10.40 priority 1 weight 10 (this blade is in ISR2 on west coast)
blade 10.10.10.10 priority 2 weight 10 (this blade is in ISR1 on east coast)
blade 10.10.10.20 priority 2 weight 10 (this blade is in ISR1 on east coast)
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SIP Proxy
The CUP proxy version 7.0(5) supports upstream route destination status using OPTIONS ping and
INVITE requests, but with a variation on how it was implemented in CUSP. CUP will only start pinging
a route only after it has failed a call attempt or a OPTIONS ping with any 5XX response. It will mark
the destination as out-of-service with any 5XX response to an INVITE or an OPTIONS message.
Configuration
The following sections discuss configuration of the SIP Proxy Server and Cisco IOS Gateways using SIP.
It is not meant to be an exhaustive list of configuration options but only highlights certain configuration
concepts.
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SIP Proxy
The sip-server command on the dial-peer tells the Cisco IOS gateway to use the globally defined
sip-server that is configured under the sip-ua settings. In order to configure multiple SIP Proxies for
redundancy, you can change the IP address to a DNS SRV record, as shown in the following example.
The DNS SRV record allows a single DNS name to be mapped to multiple servers.
sip-ua
sip-server dns:cvp.cisco.com
Alternatively, you can configure multiple dial-peers to point directly at multiple SIP Proxy servers, as
shown in the following example. This configuration allows you to specify IP addresses instead of relying
on DNS.
dial-peer voice 1000 voip
session target ipv4:10.4.1.100
preference 1
...
dial-peer voice 1000 voip
session target ipv4:10.4.1.101
preference 1
...
In the preceding examples, the calls are sent to the SIP Proxy server for dial plan resolution and call
routing. If there are multiple Unified CVP Call Servers, the SIP Proxy server would be configured with
multiple routes for load balancing and redundancy. It is possible for Cisco IOS gateways to provide load
balancing and redundancy without a SIP Proxy Server. The following example shows a Cisco IOS
gateway configuration with multiple dial-peers so that the calls are load-balanced across three Unified
CVP Call Servers.
dial-peer voice 1001 voip
session target ipv4:10.4.33.131
preference 1
...
dial-peer voice 1002 voip
session target ipv4:10.4.33.132
preference 1
...
dial-peer voice 1003 voip
session target ipv4:10.4.33.133
preference 1
...
DNS SRV records allow an administrator to configure redundancy and load balancing with finer
granularity than with DNS round-robin redundancy and load balancing. A DNS SRV record allows you
to define which hosts should be used for a particular service (the service in this case is SIP), and it allows
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Unified CVP SIP Service
you to define the load-balancing characteristics among those hosts. In the following example, the
redundancy provided by the three dial-peers configured above is replaced with a single dial-peer using
a DNS SRV record. Note that a DNS server is required in order to do the DNS lookups.
ip name-server 10.4.33.200
With Cisco IOS gateways, it is possible to define DNS SRV records statically, similar to static host
records. This capability allows you to simplify dial-peer configuration while also providing DNS SRV
load balancing and redundancy. The downside of this method is that, if the SRV record needs to be
changed, it must be changed on each gateway instead of on a centralized DNS server. The following
example shows the configuration of static SRV records for SIP services handled by cvp.cisco.com, and
the SIP SRV records for cvp.cisco.com are configured to load-balance across three servers.
ip host cvp4cc2.cisco.com 10.4.33.132
ip host cvp4cc3.cisco.com 10.4.33.133
ip host cvp4cc1.cisco.com 10.4.33.131
Call Disposition
Calls are handled as indicated for the following failure scenarios:
• Primary SIP Proxy Server fails
Active calls are preserved. Subsequent transfers of calls are successful, provided the backup SIP
Proxy is available and the RecordRoute header is not being populated by the SIP Proxy. If the
RecordRoute header is populated, signaling to the gateway will not be possible and subsequent
transfer attempts will fail.
• All SIP Proxy Servers fail or are unreachable
New calls arriving at the gateway are default-routed if survivability is configured on the gateway.
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Unified CVP SIP Service
Configuration
If only a single SIP Proxy server is needed for outbound call routing from the Call Server, choose the
SIP Proxy configuration when configuring the SIP Service. In the Unified CVP Operations Console
Server (Operations Console), configure the following:
• Add a SIP Proxy Server and specify the IP address of the server.
Under the Call Server SIP Service settings, configure the following:
• Enable Outbound Proxy = True
• Use DNS SRV type query = False
• Outbound Proxy Host = SIP Proxy Server configured above
When using multiple SIP Proxy servers for outbound redundancy from the Call Server, configure the SIP
Proxy with a DNS name and configure DNS SRV records in order to reach the SIP Proxy Servers. The
DNS SRV records can exist on an external DNS Server, or they can be configured in a local DNS SRV
record on each CVP server. In the OAMP Console, configure the following:
• Add a SIP Proxy Server and specify DNS name of the server.
Under the SIP Service configuration, configure the following:
• Enable Outbound Proxy = True
• Use DNS SRV type query = True
• The DNS SRV record should then be configured with the list of SIP Proxy Servers.
To configure the Local DNS SRV record on each server, under the SIP Service configuration, check
Resolve SRV records locally.
To use a Server Group for redundant proxy servers:
1. Select resolve SRV records locally and type in the name of the server group for the outbound proxy
domain name.
2. Under System > Server Groups, create a new server group with two proxy servers that have priority
1 and 2.
3. Deploy the server group configuration to the Call Server.
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The configuration discussed in this section protects against all of these situations. However, the
following two situations cannot be protected against:
• Someone stops the process with calls in progress. This situation occurs when a system administrator
forgets to put the Call Server out-of-service first to allow calls in progress to finish before stopping
the process.
• The Call Server exceeds the recommended call rate. Although there is a throttle for the absolute
number of calls allowed in the Call Server, there is no throttle for call rate. In general, exceeding the
recommended calls per second (cps) for an extended period of time can cause erratic and
unpredictable call behavior on certain components of the CVP solution if one of the components is
not sized correctly or if the call load is not balanced according to the weight and sizing of each call
processing component. (Refer to Chapter 14, “Sizing” for call server call rate details.)
For call survivability, configure the originating gateways as described in the latest version of the
Configuration and Administration Guide for Cisco Unified Customer Voice Portal (CVP), available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configurati
on_guides_list.html
The survivability.tcl script itself also contains some directions and useful information.
In the event of most downstream failures (including a Call Server failure), the call is default-routed by
the originating gateway. Note that survivability is not applicable in the Unified CVP Standalone and
NIC-routing models because there is no Unified CVP H.323 or SIP Service involved anywhere in those
models.
There is also a mechanism for detection of calls that have been cleared without Unified CVP’s
knowledge:
• Unified CVP checks every 2 minutes for inbound calls that have a duration older than a configured
time (the default is 120 minutes).
• For those calls, Unified CVP sends an UPDATE message. If the message receives a rejection or is
undeliverable, then the call is cleared and the license released.
The CVP SIP Service can also add the Session Expires header on calls so that endpoints such as the
originating gateway may perform session refreshing on their own. RFC 4028 (Session Timers in the
Session Initiation Protocol) has more details on the usage of Session Expires with SIP calls.
Call Disposition
Calls are handled as indicated for the following scenarios:
• Calls in progress
If the Unified CVP SIP Service fails after the caller has been transferred (transfers include transfer
to an IP phone, VoiceXML gateway, or other egress gateway), then the call continues normally until
a subsequent transfer activity (if applicable) is required from the Unified CVP SIP Service. If the
caller has not hung up and is awaiting further activity, there is a period of 9 to 18 seconds of silence
before the caller is default-routed by survivability to an alternate location.
If the call has not yet been transferred, the caller hears 9 to 18 seconds of silence before being
default-routed by survivability to an alternate location. (Survivability does not apply in NIC-routing
models.)
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Chapter 4 Designing Unified CVP for High Availability
Server Groups
• New calls
New calls are directed by the SIP Proxy to an alternate Unified CVP Call Server. If no Call Servers
are available, the call is default-routed to an alternate location by survivability. (Survivability does
not apply in NIC-routing models.)
Server Groups
A Server Group is a dynamic routing feature that enables the originating endpoint to have knowledge of
the status of the destination address before attempting to send the SIP INVITE. Whether the destination
is unreachable over the network, or is out of service at the application layer, the originating SIP user
agent can have fore-knowledge of the status through a heartbeat mechanism.
Although, there was already an H.323 endpoint registration mechanism, the Server Groups features adds
a heartbeat mechanism with endpoints for SIP.
This feature allows faster failover on call control by eliminating delays due to failed endpoints.
Note • Server Groups are not automatically created. Server Groups are not created by the upgrade to
8.0(1). You must explicitly configure Server Groups for their deployment, and turn the feature on
after upgrading, in order to take advantage of the feature.
• Upgrade for customers who already use Local SRV. Release 7.0(2) customers who already have
an srv.xml file configured with local SRV must run the import command mentioned below in order
to put their configuration into the Unified CVP Operations Console Server database. Do this before
saving and deploying any new server groups to avoid overwriting your previous configuration.
The Unified CVP SIP Subsystem builds on the local SRV configuration XML available with Release
7.0(1).
A Server Group consists of one or more destination addresses (endpoints), and is identified by a Server
Group domain name. This domain name is also known as the SRV cluster domain name, or FQDN. The
SRV mechanism is used, but the DNS server resolution of the record is not performed. Server Groups
remains the same as the Release 7.0(1) local SRV implementation (srv.xml), but the Server Groups
feature adds the extra heartbeat mechanism on top of it, as an option.
Note • Server Groups in Unified CVP and Server Groups in CUSP proxy servers work the same way.
• Only endpoints defined in a Server Group may have heartbeats sent to them.
The srv.xml configuration file was used in the 7.0(1) release to configure SRV records locally, to avoid
the overhead of DNS SRV querying. However, the method of configuration was manual, and could not
be pushed from the Unified CVP Operations Console Server (Operations Console). Also, there was no
validation on the min and max values for fields.
Release 8.0(1) adds this configuration into the Operations Console SIP subsystem using the Server
Groups concept. The Server Group term just refers to the local SRV configuration. When you turn on
Server Groups with Heartbeating, you get the dynamic routing capability for Unified CVP to
preemptively monitor the status of endpoints. This feature only covers outbound calls from Unified CVP.
To cover the inbound calls to Unified CVP, the CUSP proxy server can send similar heartbeats to Unified
CVP, which can respond with status responses.
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Server Groups
Note • Heartbeat Behavior Settings for Server Groups. To turn off pinging when the element is up, set
Up Endpoint Heartbeat Interval to zero (reactive pinging). To turn off pinging when the element is
down, set the Down Endpoint Heartbeat Interval to zero (proactive pinging). To ping when the
element is either up or down, set the heartbeat intervals to greater than zero (adaptive pinging).
• Heartbeat Response Handling. Any endpoint that CVP may route calls to should respond to
OPTIONS with some response, either a 200 OK or some other response. Any response to a heartbeat
will indicate the other side is alive and reachable. A 200 OK is usually returned, but proxy servers
like the Cisco Unified Presence Server (CUP Server) or Cisco Unified SIP Proxy Server (CUSP
Server) may return a 483 Too Many Hops response, since the max-forwards header is set to zero in
an OPTIONS message. Sometimes the endpoints may not allow OPTIONS or PING, and may return
405 Method Not Allowed, which is fine as well.
By default, Server Group heartbeats are performed using a UDP socket connection. The transport type
can be changed to TCP from the Operations Console Server Groups window.
Whenever an element has an unreachable or overloaded status, that element is marked as being down
completely, that is for both UDP and TCP transports. When the element is up again, it is allowed to be
routable for both UDP and TCP.
Duplicate Server Group Elements are excluded for heartbeating since the heartbeating is already
established for that element.
Note Refer to the Configuration and Administration Guide for Cisco Unified Customer Voice Portal for
typical configurations for the Server Groups feature. Document available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configuration_g
uides_list.html.
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Gatekeeper
Design Considerations
Observe the following design considerations when implementing Server Groups:
• When you are using the Local SRV configuration, that configuration does not work with the DNS
SRV configuration. However, elements may be declared as A record hostnames instead of IP
addresses, and resolved through a DNS server lookup or in the OS etc host file.
• If you are using the CUP proxy, typically the SRV cluster name (such as proxy-servers.cisco.com)
will need to be defined in the service parameters section of the proxy configuration. Otherwise a 404
not found rejection may result. The CUSP proxy has a similar configuration in the CLI.
Diagnostics
The CVP log file has traces which show endpoint status events. Refer to the Unified CVP System CLI
instructions in the Configuration and Administration Guide for Cisco Unified Customer Voice Portal,
available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configuration_g
uides_list.html
Gatekeeper
An H.323 gatekeeper is used when using H.323 in any of the ICM-integrated deployment models except
Model #4 (VRU Only with NIC Controlled Routing), which does not use Unified CVP for call control
at all. Additionally, if SIP is used as the call control protocol, the gatekeeper is not required. An
originating gateway can perform all of its H.323 call routing by using VoIP dial-peers that contain static
IP addresses, whereas the Unified CVP H.323 Service must always perform a gatekeeper Remote Access
Service (RAS) lookup to route calls.
Note In one particular situation, when using the VBAdmin SetTransferLabel option, the H.323 Service
ignores the IP address returned from the gatekeeper and instead routes the IVR call leg back to the
originating gateway from which the call arrived. This feature ensures that no WAN bandwidth is used
during IVR treatment or queuing. A gatekeeper is still required in this situation because the H.323
Service has to perform the gatekeeper lookup function to obtain possible alternate endpoints in the event
that the attempt to transfer the call to the originating gateway fails.
Unified CVP can use one of the following types of gatekeeper high-availability mechanisms:
• Gatekeeper Redundancy Using HSRP, page 4-16
• Gatekeeper Redundancy Using Alternate Gatekeeper, page 4-17
Only HSRP and alternate gatekeeper are supported by Unified CVP. Alternate gatekeeper support was
introduced in Unified CVP 3.1 SR1.
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Gatekeeper
The gatekeepers share the same IP and MAC addresses. Therefore, if one of the gatekeepers fails, the
hosts on the LAN are able to continue forwarding packets to a consistent IP and MAC address. The
process of transferring the routing responsibilities from one device to another is transparent to the user.
The H.323 endpoints (such as the Unified CVP H.323 Service, Cisco Unified Communications Manager,
and gateways) register to a virtual IP address that represents the HSRP gatekeeper pair.
If one gatekeeper fails, its partner assumes primary control. The major disadvantage of HSRP is that both
gatekeepers in the HSRP failover pair must reside on the same IP subnet or VLAN, therefore they
generally cannot be separated geographically. Gatekeepers using HSRP for redundancy also do not share
any state information. Therefore, when a failover occurs, all of the devices must re-register with the
gatekeeper from scratch.
As of Unified CVP 3.1 SR1, HSRP is no longer recommended. Instead gatekeeper clustering and
alternate gatekeeper configuration on Unified CVP is the preferred method of gatekeeper redundancy.
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Chapter 4 Designing Unified CVP for High Availability
Gatekeeper
Configuration
This section covers the following topics:
• HSRP Configuration, page 4-18
• Alternate Gatekeeper, page 4-18
HSRP Configuration
On the primary gatekeeper, enter these commands:
interface ethernet 0
ip address 10.0.1.98 255.255.255.0
! Unique IP address for this GK
standby 1 ip 10.0.1.100
! Member of standby group 1, sharing virtual address 10.0.1.100
standby 1 preempt
! Claim active role when it has higher priority.
standby 1 priority 110
! Priority is 110.
For more information, refer to the latest version of the Configuration and Administration Guide for
Cisco Unified Customer Voice Portal (CVP), available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configurati
on_guides_list.html
Alternate Gatekeeper
Configure alternate gatekeepers using Unified CVP VBAdmin, as shown in the following examples:
set GK "10.0.1.100, 10.0.2.100, 10.0.3.100"
This example sets up three gatekeepers to which the H.323 Service could possible register. In
each case, the H.323 Service registers to the first local zone that is configured in that gatekeeper.
It also uses the default RAS port 1719.
This example causes the H.323 Service to attempt to register first to gatekeeper 10.0.1.100 on
port 1718 with local zone zone1. If that gatekeeper fails, the H.323 Service subsequently
attempts to register to 10.0.2.100 on port 1719, with the first local zone defined on that
gatekeeper.
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Unified CVP H.323 Service
Call Disposition
The call dispositions presented in this section apply to both HSRP and alternate gatekeeper.
A gatekeeper can fail in any of the following ways:
• The primary gatekeeper fails
– Some calls in progress may not be transferred during the period that the endpoints are
re-registering to the backup gatekeeper. After the failed transfer, an error is returned to the ICM.
If the ICM script is coded to return an error (an END node does this) and survivability is
configured on the gateway, the call is default-routed.
– New calls arriving at the incoming gateway and Unified CVP are serviced correctly, although it
is possible that some of the calls might invoke survivability during the period that the endpoints
are re-registering to the backup gatekeeper.
• All gatekeepers fail
– The Unified CVP H.323 Service goes out of service.
– Calls in progress are not transferred. After the failed transfer, an error is returned to the ICM.
If the ICM script is coded to return an error (an END node does this) and survivability is
configured on the gateway, the call is default-routed.
– New calls arriving at the gateway are default-routed if survivability is configured on the
gateway.
• The primary gatekeeper degrades but does not fail
– There are two conditions that usually cause this behavior: low memory due to memory leaks or
excessive debug levels causing CPU overload.
– In this situation, call processing behavior is unpredictable due to the fact that there might be no
clean failover to the backup gatekeeper. If survivability is configured on the gateway, calls are
default-routed.
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Unified CVP H.323 Service
Configuration
Unified CVP H.323 configuration for high availability is performed primarily on the ingress gateways,
but it is also necessary to configure the H.323 Service to register to the gatekeeper.
The command session target ras instructs the gateway to send the call to its gatekeeper. The command
tech-prefix 2# instructs the gateway to prepend 2# to the DNIS number when sending the call to the
gatekeeper.
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Unified CVP IVR Service
The survivability.tcl script itself also contains some directions and useful information.
In the event of most downstream failures (including a Call Server failure), the call is default-routed by
the originating gateway. Note that survivability is not applicable in the Unified CVP Standalone and
NIC-routing models because there is no Unified CVP Call Server involved anywhere in those models.
This action allows the gateway to lose connectivity with the Call Server or Cisco Unified
Communications Manager but still retain active calls. If you do no t use this command, calls that are still
active that are otherwise unaffected by the failure (that is, the RTP stream is still streaming between the
endpoints) will be disconnected when the TCP session times out.
The following commands specify the RTP media timeout:
ip rtcp report interval 2000
gateway
timer receive-rtcp 4
When the gateway detects that RTCP messages have not been received in the specified interval, the call
is disconnected.
Call Disposition
If the Unified CVP H.323 Service fails, the following conditions apply:
• Calls in progress
If the Unified CVP H.323 Service fails after the caller has been transferred (transfers include
transfer to an IP phone, VoiceXML gateway, or other egress gateway), then the call continues
normally until a subsequent transfer activity (if applicable) is required from the Unified CVP H.323
Service. If the caller has not hung up and is awaiting further activity, there is a period of 9 to
18 seconds of silence before the caller is default-routed by survivability to an alternate location.
If the call has not yet been transferred, the caller hears 9 to 18 seconds of silence before being
default-routed by survivability to an alternate location. (Survivability does not apply in NIC-routing
models.)
• New calls
New calls are directed by the gatekeeper to an alternate Unified CVP Call Server. If no Call Servers
are available, the call is default-routed to an alternate location by survivability. (Survivability does
not apply in Unified CVP Standalone and NIC-routing models.)
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Chapter 4 Designing Unified CVP for High Availability
Unified CVP IVR Service
gateways with a list of application server IP addresses and/or using the Content Services Switch (CSS).
With Unified CVP 4.0 and later releases, the IVR Service is tightly coupled with the SIP Service or
H.323 Service. If the IVR Service goes out of service, the H.323 or SIP Service will be taken out of
service as well so that no further calls are accepted by the Unified CVP Call Server.
Configuration
No additional configuration is needed in order to tell the H.323 or SIP Service which IVR Service to use.
By default, the H.323 and SIP Service use the IVR Service that resides on the same server. It is also no
longer necessary to configure the VoiceXML gateway with the IP address of the Call Server’s IVR
Service. When SIP is used, the SIP Service inserts the URL of the Call Server's IVR Service into a header
in the SIP INVITE message when the call is sent to the VoiceXML gateway. The VoiceXML gateway
extracts this information from the SIP INVITE and uses it when determining which Call Server to use.
When H.323 is used, the VoiceXML gateway examines the source IP address of the incoming call from
the Call Server. This IP address is then used as the address for the Call Server’s IVR Service.
The following example illustrates the VoiceXML bootstrap service that is invoked when a call is
received:
service bootstrap flash:bootstrap.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
Unlike Unified CVP 3.1 and earlier releases, with Unified CVP 4.0 and later releases you do not have to
configure the IP address of the Call Server. The bootstrap.tcl learns the IP address of the source Call
Server and uses it as its call server. There is no need for a CSS or backup Call Server configuration
because receiving a call from the Call Server means that the server is up and operational.
The following files in flash memory on the gateway are also involved with high availability: handoff.tcl,
survivability.tcl, recovery.vxml, and several .wav files. Use Trivial File Transfer Protocol (TFTP) to load
the proper files into flash. Configuration information for each file can be found within the file itself. For
more information, refer to the latest version of the Configuration and Administration Guide for Cisco
Unified Customer Voice Portal (CVP), available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configurati
on_guides_list.html
Call Disposition
If the Unified CVP IVR Service fails, the following conditions apply to the call disposition:
• Calls in progress are default-routed to an alternate location by survivability on the originating
gateway. (Survivability does not apply in NIC-routing models.)
• New calls are directed to an in-service Unified CVP IVR Service.
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VoiceXML Gateway
VoiceXML Gateway
The VoiceXML gateway parses and renders VoiceXML documents obtained from one or several sources:
the Unified CVP Call Server (from its IVR Service), the Unified CVP VXML Servers, or some other
external VoiceXML source. Rendering a VoiceXML document consists of retrieving and playing
prerecorded audio files, collecting and processing user input, and/or connecting to an ASR/TTS server
for voice recognition and dynamic text-to-speech conversion.
Release 8.0(1) of CVP supports a flattened deployment of the g.711 codec on all call legs, or else a flat
g.729 deployment on all call legs. A mixture of the 2 codecs on different legs of the call, at any point
in the duration of the call, is not supported.
For a discussion of the upside and downside of each codec, refer to Voice Traffic, page 9-2.
Configuration
High availability configuration for VoiceXML gateways is controlled by the gatekeeper for H.323, the
SIP proxy for SIP, and/or the Unified CVP Call Server (Call Server). Whether the VoiceXML gateways
are distributed or centralized also influences how high availability is achieved.
In the event that a Call Server is unable to connect to a VoiceXML gateway, an error is returned to the
ICM script. In the ICM script, separate the Send to VRU node from the first Run External script node in
order to catch the VoiceXML gateway connection error. If an END script node is used off the X-path of
the Send to VRU node, the call is default-routed by survivability on the originating gateway.
(Survivability does not apply in VRU-only models.) A Queue to Skill group node could also be used, but
that method is effective only if there is an agent available. Otherwise, ICM tries to queue the caller, and
that attempt fails because the Call Server is once again unable to connect to a VoiceXML gateway. An
END node could then also be used off the X-path of the Queue to Skill Group node to default-route the
call.
Note There are two features for the VXML Server that assist with load balancing:
• Limiting Load Balancer Involvement
• Enhanced HTTP Probes for Load Balancers
Refer to the configuration options ip_redirect and license_depletion_probe_error in the User Guide for
Unified CVP VXML Server and Cisco Unified Call Studio, available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_user_guide_list.html
On the gatekeeper, configure a zone prefix list that contains the H.323 IDs of all VoiceXML gateways at
the data center. For example, assume that there are three VoiceXML gateways in the data center with
H.323 IDs of VoiceXMLgw1, VoiceXMLgw2, and VoiceXMLgw3, and that the ICM label for the
Network VRU is 5551000. In this example, the gatekeeper distributes calls in essentially a round-robin
scheme among all three VoiceXML gateways, provided they are all in service:
zone prefix gkzone-name 5551000* gw-priority 10 VoiceXMLgw1 VoiceXMLgw2 VoiceXMLgw3
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Chapter 4 Designing Unified CVP for High Availability
VoiceXML Gateway
If you are using Cisco Unified Presence Server: On the SIP proxy server, configure a static route for the
Network VRU label for each gateway. If the VRU label is 5551000, the static route pattern would be
5551000* in order to allow for the correlation-id to be appended and routed to the VoiceXML gateway.
If you are using SIP static routes on the Unified CVP Call Server: Under the SIP Service configuration
for the Call Server, configure a static route for each Network VRU label and gateway. If the VRU label
is 5551000, the static route pattern would be 5551000>. The > is a wildcard representing one or more
digits, and it is needed so that the correlation-id appended to the DNIS number can be passed to the
VoiceXML gateway correctly.
Note Other wildcard characters can be used. Refer to the topic Valid Formats for Dialed Numbers in the
Operations Console online help for complete wildcard format and precedence information.
In the case of both SIP proxy or Unified CVP static routes, the next-hop address of the route can be either
the IP address of the gateway or a DNS SRV record. If you are using an IP address, you must create
multiple static routes, one for each VoiceXML gateway. In the case of DNS SRV, only one route per
Network VRU label is needed, and the SRV record will provide for load-balancing and redundancy.
Use the SetTransferLabel in VBAdmin (not gatekeeper zone prefixes) to control the selection of the
VoiceXML gateway. The SetTransferLabel command is specified per Network VRU label. When the
Unified CVP Call Server receives a label from ICM that matches what is configured in the
SetTransferLabel, the Call Server performs a gatekeeper lookup but ignores the destination gateway
returned by the gatekeeper and sends the call back to the gateway that originated the call. The H.323
Service determines the originating gateway by looking at the source IP address of the H.323 signaling.
With SIP, the equivalent of the SetTransferLabel command is the Send to Originator configuration under
the SIP Service. If the Network VRU label is 5551000, the Send to Originator pattern would be
5551000>. The > is a wildcard pattern representing one or more digits. The SIP Service determines the
originating gateway by looking at the Remote-Party-ID header in the SIP INVITE message.
Note Other wildcard characters can be used. Refer to the topic Valid Formats for Dialed Numbers in the
Operations Console online help for complete wildcard format and precedence information.
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VoiceXML Gateway
When H.323 is used, the significant digit is prepended with a # sign so that the gatekeeper treats it as a
technology prefix. The VoiceXML gateway at the remote site should register to the gatekeeper with the
same technology prefix as the leading significant digit(s) that were stripped from the DNIS number. The
gatekeeper then routes the IVR leg of the call to the correct VoiceXML gateway. If you are using Cisco
Unified Communications Manager (Unified CM), remember that Unified CVP indiscriminately
prepends the sigdigits value to all transfers, including those to Unified CM. Therefore, when using
Unified CM in this scenario, it is necessary to define a gatekeeper-controlled trunk for each of the
VoiceXML gateway tech-prefixes and to add zone prefix configuration to the gatekeeper for the
Unified CM agents, as illustrated in the following example.
Assuming the DNIS number is 8002324444, the final DNIS string routed to Unified CVP is
2#38002324444.
Configuration in VB Admin:
setTechPrefix 2#
setSigDigits 1
Strip one digit from the DNIS number after stripping the 2# technology prefix.
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VoiceXML Gateway
Gatekeeper configuration:
Define zone prefixes to route calls appropriately to Unified CM agents (only if using Cisco Unified CM).
When SIP is used, the significant digits are prepended to the DNIS number, and a SIP Proxy can be
configured to route based on those prepended digits. The static routes in the SIP Proxy for the VoiceXML
gateway should have the digits prepended. Because these prepended digits were originally populated by
the ingress gateway, the SIP Proxy can use them to determine which VoiceXML gateway to use based
on the incoming gateway. In this way, calls arriving at a particular site can always be sent back to that
site for VoiceXML treatment, with the result that no WAN bandwidth is used to carry the voice RTP
stream. Keep in mind that Unified CVP indiscriminately prepends the sigdigits value to all transfers,
including those to Unified CM. Therefore, when using Unified CM in this scenario, it is necessary to
strip the prepended digits when the call arrives so that the real DNIS number of the phone can be used
by Unified CM to route the call, as illustrated in the following example.
Assuming the DNIS number is 8002324444, the final DNIS string routed to Unified CVP is
33338002324444.
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VoiceXML Gateway
Configure the Unified CVP bootstrap.tcl application with the sigdigits parameter, telling it how many
digits to strip off of the incoming DNIS string:
application
service bootstrap flash:bootstrap.tcl
param sigdigits 4
...
Call Disposition
If the VoiceXML gateway fails, the following conditions apply to the call disposition:
• Calls in progress are default-routed to an alternate location by survivability on the ingress gateway.
(Survivability does not apply in NIC-routing models.)
• New calls find an alternate VoiceXML gateway.
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Chapter 4 Designing Unified CVP for High Availability
Content Services Switch (CSS)
Note Cisco recommends using the CSS to find an available VXML Server via heartbeat and to perform load
balancing. Subsequent requests and responses between the VoiceXML gateway and the VXML Server
should bypass the CSS.
Note There are two features for the VXML Server that assist with load balancing:
• Limiting Load Balancer Involvement
• Enhanced HTTP Probes for Load Balancers
Refer to the configuration options ip_redirect and license_depletion_probe_error in the User Guide for
Unified CVP VXML Server and Cisco Unified Call Studio, available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_user_guide_list.html
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Content Services Switch (CSS)
Configuration
You can configure high availability for the CSS by using the Virtual IP (VIP) Redundancy method, as
described in the latest version of the Configuration and Administration Guide for Cisco Unified
Customer Voice Portal (CVP), available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configurati
on_guides_list.html
Also refer to the latest version of the CSS Redundancy Configuration Guide, available at
http://www.cisco.com/en/US/products/hw/contnetw/ps792/products_installation_and_configuratio
n_guides_list.html
Essentially, a master/backup pair of CSSs functions very much like an HSRP gatekeeper pair. They must
reside on the same VLAN and exchange heartbeats using Virtual Router Redundancy Protocol (VRRP),
a protocol very similar to HSRP. If the primary CSS fails, the backup CSS recognizes the failure within
three seconds and begins processing client requests to its configured virtual IP addresses. The
configuration of the master and backup CSSs must always be kept in synchronization.
Call Disposition
If the master CSS fails, then the following conditions apply to the call disposition:
• Calls in progress encounter various behaviors, depending on the type of service the VoiceXML
gateway client requested:
– Media server requests are unaffected.
The VoiceXML gateway has a very short-lived interaction with the CSS for audio files. Upon
receiving a media server request from the gateway, the CSS simply provides an HTTP redirect
IP address for the VoiceXML gateway. At that point, the gateway fetches the audio file directly
from the media server, bypassing any further interaction with the CSS. Additionally, media file
requests to the CSS are very infrequent because the VoiceXML gateway caches previously
retrieved media files.
– Unified CVP IVR Service requests are unaffected.
Only the initial VoiceXML document request to a Unified CVP IVR Service uses the CSS. The
CSS first picks a Unified CVP IVR Service to service the request. The first document passes
through the CSS on its return to the VoiceXML gateway. However, subsequent VoiceXML
requests are made directly from the VoiceXML gateway client to the Unified CVP IVR Service.
If the CSS fails during the very brief period that the first VoiceXML document is being returned,
the VoiceXML gateway simply retries the request. If the backup (now primary) CSS selects the
same Unified CVP IVR Service as the previous one, there is an error due to a duplicate call
instance. In that case, the caller is default-routed by survivability on the originating gateway.
– ASR/TTS requests typically fail but might be recoverable.
When the VoiceXML gateway first makes an ASR/TTS request to the CSS, a TCP connection
is opened from the VoiceXML gateway to the Media Resource Control Protocol (MRCP) server.
That TCP connection goes through the CSS and persists until the caller disconnects or is
transferred to an agent. If the primary CSS fails, that TCP connection is terminated. The
VoiceXML gateway returns an error code, which you could write a script to work around. The
worst-case scenario is that the caller is default-routed to an alternate location by survivability
on the originating gateway.
– Unified CVP VXML Server requests may fail.
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Chapter 4 Designing Unified CVP for High Availability
Media Server
The VoiceXML gateway is "sticky" to a particular Unified CVP VXML Server for the duration
of the VoiceXML session. It uses CSS cookies to provide that stickiness.
Configuring Adaptive Session Redundancy (ASR) on CSS peers in an active-backup VIP
redundancy and virtual interface redundancy environment will provide a stateful failover of
most existing calls. ASR ensures that, if the master CSS fails, the backup CSS has the necessary
flow-state information to continue most active calls without interruption when the backup CSS
assumes mastership.
For the few cases where the existing call cannot continue, the VoiceXML gateway returns an
error code, which you could write a script to work around. The worst-case scenario is that the
caller is default-routed to an alternate location by survivability on the originating gateway.
The Adaptive Session Redundancy (ASR) feature of CSS ensures that port licenses are not
temporarily and needlessly unavailable on the VXML Server. The VXML Server is stateful, and
the ASR feature minimizes VXML Server license port usage during a CSS failover.
New calls are directed transparently to the VIPs on the backup CSS,and service is unaffected.
Media Server
Audio files can be stored locally in flash memory on the VoiceXML gateway or on an HTTP/TFTP file
server. By definition, audio files stored locally are highly available. However, HTTP/TFTP file servers
provide the advantage of centralized administration of audio files.
The backup server is invoked only if the primary server is not accessible, and this is not a load-balancing
mechanism. Each new call attempts to connect to the primary server. If failover occurs, the backup server
is used for the duration of the call; the next new call will attempt to connect to the primary server.
Note that mediaserver is not a fixed name, and it needs to match whatever name was assigned to the
media_server ECC variable in the ICM script.
The VoiceXML gateway also uses the following VoiceXML gateway configuration parameters to locate
a server when using a CSS:
ip host mediaserver <ip-address-of-CSS-VIP-for-media-server>
ip host mediaserver-backup <ip-address-of-CSS-VIP-for-media-server>
Because the CSS almost always locates a media server on the first request, a backup server is rarely
invoked. However, it is helpful to configure the backup server when using a CSS for deployments where
there are multiple data centers with a CSS in each.
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Unified CVP VXML Server
Configuration
The Unified CVP VXML Server high-availability configuration and behavior differ between Standalone
deployments and deployments that are integrated with ICM, as described in the following sections.
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Chapter 4 Designing Unified CVP for High Availability
Automatic Speech Recognition (ASR) and Text-to-Speech (TTS) Server
Call Disposition
If the Unified CVP VXML Server fails, the following conditions apply to the call disposition:
• Calls in progress in a Standalone deployment are disconnected. Calls in progress in an
ICM-integrated deployment can be recovered using scripting techniques to work around the error as
shown in the script above (for example, retry the request, transfer to an agent or label, or force an
error with an END script node to invoke survivability on the originating gateway).
• New calls are directed transparently to an alternate Unified CVP VXML Server.
Note Without a CSS or ACE device, callers might experience a delay at the beginning of the call and
have to wait for the system to timeout while trying to connect to the primary Unified CVP
VXML Server.
Configuration
The ASR/TTS high-availability configuration and behavior differ between Standalone and
ICM-integrated deployments, as described in the following sections.
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Cisco Unified Communications Manager
The hostnames (such as asr-en-us) are fixed and cannot be modified. The only portion that may be
modified is the locale. In the following example, there is a set of primary and backup English ASR/TTS
servers and a set of Spanish servers. Configure the CSS, if used, according to the instructions in the latest
version of the Configuration and Administration Guide for Cisco Unified Customer Voice Portal (CVP),
available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configurati
on_guides_list.html
When a CSS is used, the IP addresses mentioned the following example would be the virtual IP address
for the ASR/TTS service on the CSS.
ip host asr-en-us <ip-address-of-primary-English-ASR-server>
ip host asr-en-us-backup <ip-address-of-secondary-English-ASR-server>
ip host tts-en-us <ip-address-of-primary-English-TTS-server>
ip host tts-en-us-backup <ip-address-of-secondary-English-TTS-server>
ip host asr-es-es <ip-address-of-primary-Spanish-ASR-server>
ip host asr-es-es-backup <ip-address-of-secondary-Spanish-ASR-server>
ip host tts-es-es <ip-address-of-primary-Spanish-TTS-server>
ip host tts-es-es-backup <ip-address-of-secondary-Spanish-TTS-server>
Release 8.0(1) of CVP supports a flattened deployment of the g.711 codec on all call legs, or else a flat
g.729 deployment on all call legs. A mixture of the 2 codecs on different legs of the call, at any point
in the duration of the call, is not supported.
For a discussion of the upside and downside of each codec, refer to Voice Traffic, page 9-2.
Note The ASR speech license is not released until the caller is transferred to the agent.
Call Disposition
If the ASR/TTS MRCP server fails, the following conditions apply to the call disposition:
• Calls in progress in Standalone deployments are disconnected. Calls in progress in ICM-integrated
deployments can be recovered using scripting techniques to work around the error (for example,
retry the request, transfer to an agent or label, switch to prerecorded prompts and DTMF-only input
for the remainder of the call, or label or force an error with an END script node to invoke
survivability on the originating gateway).
• New calls in Standalone or ICM-integrated deployments are directed transparently to an alternate
ASR/TTS server if a CSS is being used. New calls in ICM-integrated deployments are directed
transparently to an alternate ASR/TTS server if "-backup" gateway hostnames have been used.
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Chapter 4 Designing Unified CVP for High Availability
Intelligent Contact Management (ICM)
Unified CVP version 8.0(1) also supports the Analysis Manager. Refer to The Analysis Manager,
page 13-8.
Configuration
For the most current information on providing Unified CM for high availability, refer to the latest version
of the Cisco Unified Contact Center Enterprise Solution Reference Network Design (SRND) available
at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_implementation_design_guides_li
st.html.
Call Disposition
If the Unified CM process fails on the server that is either hosting the call or hosting the phone, the
following conditions apply to the call disposition:
• Calls in progress are preserved. Skinny Client Control Protocol (SCCP) phones have the ability to
preserve calls even when they detect the loss of their Unified CM. The caller-and-agent conversation
continues until either the caller or agent goes on-hook. The Unified CVP Call Server recognizes that
Unified CM has failed, assumes the call should be preserved, and maintains the signaling channel
to the originating gateway. In this way, the originating gateway has no knowledge that Unified CM
has failed. Note that additional activities in the call (such as hold, transfer, or conference) are not
possible. Once the parties go on-hook, the phone then re-homes to another Unified CM server. When
the agent goes on-hook, Real-Time Control Protocol (RTCP) packets cease transmitting to the
originating gateway, which causes the gateway to disconnect the caller 9 to 18 seconds after the
agent goes on-hook. If survivability has been configured on the gateway and the caller is waiting for
some additional activity (the agent might think the caller is being blind-transferred to another
destination), the caller is default-routed to an alternate location.
• New calls are directed to an alternate Unified CM server in the cluster.
Configuration
For the most current information on configuring ICM for high availability, refer to the latest version of
the Cisco Unified Contact Center Enterprise Solution Reference Network Design (SRND), available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_implementation_design_gui
des_list.html
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Intelligent Contact Management (ICM)
Call Disposition
There are many components in Cisco ICM, and call disposition varies depending on the component that
fails. Although there are a few exceptions, the following conditions apply to the call disposition:
• If the Voice Response Unit (VRU) Peripheral Gateway (PG) or any component on the VRU PG fails,
calls in progress are default-routed by survivability on the originating gateway.
• If the Logger fails, calls in progress are unaffected.
• If the primary router fails, calls in progress are unaffected. If both the Side A and Side B routers fail,
calls in progress are default-routed by survivability on the originating gateway.
• New calls are directed to the backup ICM component.
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Intelligent Contact Management (ICM)
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CH A P T E R 5
Interactions with Cisco Unified ICM
This chapter discusses Cisco Unified Intelligent Contact Management (ICM) from the perspective of its
relationship with Unified CVP. In some cases, the choice of deployment model has implications for
Unified ICM; and in other cases, certain choices about the Unified ICM configuration carry implications
for the Unified CVP deployment.
This chapter covers the following major topics:
• Network VRU Types, page 5-2
• Network VRU Types and Unified CVP Deployment Models, page 5-6
• Hosted Implementations, page 5-10
• Deployment Models and Sizing Implications for Calls Originated by Cisco Unified
Communications Manager and ACDs, page 5-13
• Using Third-Party VRUs, page 5-15
• DS0 Trunk Information, page 5-15
• Trunk Utilization Routing and Reporting, page 5-16
• Enhanced User-to-User Information, page 5-17
• Custom SIP Headers, page 5-19
• Courtesy Callback, page 5-21
• Post Call Survey, page 5-25
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Chapter 5 Interactions with Cisco Unified ICM
What's New in This Chapter
Table 5-1 New or Changed Information Since the Previous Release of This Document
Note The Generic PG is a consolidated PG that requires both separate peripherals for Unified CM and the
VRU. It is best not to use the Generic PG for Unified CVP. Use the VRU PG instead.
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Network VRU Types
Note The deployed VRU can be located in the network (Network VRU) or at the customer premises. In the
latter scenario, a Network Applications Manager (NAM) would be deployed in the network and a
Customer ICM (CICM) would be deployed at the customer premises. The corresponding Correlation ID
or Translation Route ID should be used accordingly, as described earlier, depending on the location of
the VRU.
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Chapter 5 Interactions with Cisco Unified ICM
Network VRU Types
ICM Customers are now supported, one cannot initiate a two-step transfer from the Unified CVP VRU
switch leg to a completely separate Unified CVP (for example, a two-steps CVP-to-CVP transfer using
SendToVRU). A translation route would have to be used in order for such a two-step transfers to work.
Type 10 Network VRU has the following behavior:
• There is a Handoff of routing client responsibilities to the Unified CVP switch leg.
• There is an automatic transfer to the Unified CVP VRU leg, resulting in a second transfer in the case
of calls originated by the VRU, ACD, or Cisco Unified Communications Manager (Unified CM).
• For calls originated by Unified CM, the Correlation ID transfer mechanism is used. The Correlation
ID is automatically added to the end of the transfer label defined in the Type 10 Network VRU
configuration.
• The final transfer to the Unified CVP VRU leg is similar to a Type 7 transfer, in that a RELEASE
message is sent to the VRU prior to any transfer.
In Unified CVP implementations without the ICM Customers feature (that is, in Unified CVP
implementations with a single Network VRU), a single Type 10 Network VRU should be defined, and
all Unified ICM VRU scripts should be associated with it. It requires one label for the Unified CVP
Switch leg routing client, which will transfer the call to the Unified CVP VRU leg. If calls will be
transferred to Unified CVP from Unified CM, it also needs another label for the Unified CM routing
client, and this label should be different from the label used for the CVP Routing Client. This label will
transfer the call to the Unified CVP Switch leg. The Unified ICM Router will send this label to
Unified CM with a Correlation ID concatenated to it. Unified CM must be configured to handle these
arbitrary extra digits.
The Unified CVP Switch leg peripheral should be configured to point to the same Type 10 Network
VRU. Also, all incoming dialed numbers for calls that are to be transferred to Unified CVP should be
associated with a Customer Instance that points to the same Type 10 Network VRU.
For calls that originate at a Call Routing Interface VRU or at a TDM ACD, a TranslationRouteToVRU
node should be used to transfer the call to Unified CVP’s Switch leg peripheral. For all other calls, use
either a SendToVRU node, a node that contains automatic SendToVRU behavior (such as the queuing
nodes), or a RunExternalScript.
Note Cisco Unified ICM 7.1 introduces the Type10 Network VRU. This VRU should be used for all new
implementations of Unified CVP using Unified ICM 7.1 or greater. The Type5 VRU can still be used for
existing customer deployments that have upgraded or for deployments that are not running Unified ICM
7.1 or later.
Type 5 and Type 6 are similar in the sense that the VRU entity functions both as a switch (call control)
and as the VRU (IVR). However, they differ on how to connect to the VRU.
In Type 6, the Switch and the VRU are the same device, therefore the call is already at the VRU. No
Connect and Request Instructions message sequence is needed from the point of view of Unified ICM.
On the other hand, in Type 5, the Switch and the VRU are different devices even though they are in the
same service node from the viewpoint of Unified ICM (that is, they both interact with Unified ICM
through the same PG interface). Therefore, Unified ICM uses a Connect and Request Instructions
sequence to complete the IVR call.
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Network VRU Types
Note In a Unified CVP application, there are two legs of the call as perceived by Unified ICM: the Switch leg
and the VRU leg. In the case where Unified CVP acts as the service node application (that is, when
Unified CVP receives the call from the network directly and not via pre-routing), Unified CVP will
appear to Unified ICM as Type 5 because the call control (Unified CVP) and the VRU device are
different. Hence, Unified CVP must be configured as VRU Type 5 in the Unified ICM and NAM
configuration for the Switch leg. The VRU leg requires a different setup, depending on the deployment
model (for example, the VRU leg could be Type 7 in the Comprehensive Unified ICM enterprise
deployment model). For examples of configuring Unified CVP as Type 5, refer to the latest version of
the Configuration and Administration Guide for Cisco Unified Customer Voice Portal (CVP), available
at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configuration_g
uides_list.html.
Neither Correlation ID nor Translation Route ID is needed when Unified CVP acts as a Type 5 VRU to
Unified ICM and the NAM. However, a dummy label is sometimes required.
Note Cisco Unified ICM 7.1 introduces the Type10 Network VRU. This VRU should be used for all new
implementations of Unified CVP using Unified ICM 7.1 or greater, except as VRU Only (Model #4a,
described below). The Type 3 or 7 VRU can still be used for existing customer deployments that have
upgraded or for deployments that are not running Unified ICM 7.1 or later.
When the VRU functions as an IVR with the Correlation ID mechanism, Unified ICM uses Type 3 and
Type 7 to designate sub-behaviors of the VRU via the PG in the Correlation ID scheme. Both Type 3 and
Type 7 VRUs can be reached via the Correlation ID mechanism, and a PG is needed to control the VRU.
However, the difference between these two types is in how they release the VRU leg and how they
connect the call to the final destination.
In Type 3, the switch that delivers the call to the VRU can take the call from the VRU and connect it to
a destination (or agent).
In Type 7, the switch cannot take the call away from the VRU. When the IVR treatment is complete,
Unified ICM must disconnect or release the VRU leg before the final connect message can be sent to the
Switch leg to instruct the switch to connect the call to the destination.
When used as an Intelligent Peripheral IVR, Unified CVP can function with either Type 3 or 7, but it is
somewhat more efficient under Type 7 because it gets a positive indication from Unified ICM when its
VRU leg is no longer needed (as opposed to waiting for the VoiceXML gateway to inform it that the call
has been pulled away).
As stated previously, there are two legs of the call: the Switch leg and the VRU leg. Different Unified
CVP hardware can be used for each leg, but from the perspective of Unified ICM functionality, there will
be a Unified CVP via PG acting as VRU Type 5 (that is, a service node) along with potentially a different
Unified CVP via another PG acting as VRU Type 7 to complete the IVR application (self service,
queuing, and so forth).
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Chapter 5 Interactions with Cisco Unified ICM
Network VRU Types and Unified CVP Deployment Models
For configuration examples of the Unified CVP application with VRU Type 3 or Type 7, refer to the
latest version of the Configuration and Administration Guide for Cisco Unified Customer Voice Portal
(CVP), available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configurati
on_guides_list.html
Note Cisco Unified ICM 7.1 introduces the Type10 Network VRU. This VRU should be used for all new
implementations of Unified CVP using Unified ICM 7.1 or greater, except as VRU Only (Model #4a,
described below). The Type 8 or 2 VRU can still be used for existing customer deployments that have
upgraded or for deployments that are not running Unified ICM 7.1 or later.
When the VRU functions as an IVR with the Translation Route ID mechanism, Unified ICM uses Type 8
or Type 2 to designate sub-behaviors of the VRU via the PG in the translation route scheme. Both Type 2
and Type 8 VRUs can be reached via the Translation Route mechanism, and PG is needed to control the
VRU. However, they differ in how they connect the call to the final destination.
In Type 8, the switch that delivers the call to the VRU can take the call from the VRU and connect it to
a destination/agent.
Type 2 is used when the switch does not have the ability to take the call away from the VRU to deliver
it to an agent. In that case, when the IVR treatment is complete, Unified ICM sends the final connect
message to the VRU (rather than to the original switch) to connect the call to the destination. The VRU
effectively assumes control of the switching responsibilities when it receives the call. This process is
known as a handoff.
Similarly to the Correlation ID case, there are two legs of the call: the Switch leg and the VRU leg.
Unified CVP can be used for either the Switch leg or the VRU leg. For example, when a Network
Interface Controller (NIC), NAM, or CICM is involved, Unified CVP should be configured as Type 2 or
Type 8 in the VRU leg.
For configuration examples of the Unified CVP application with VRU Type 8 or Type 2, refer to the
latest version of the Configuration and Administration Guide for Cisco Unified Customer Voice Portal
(CVP), available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configurati
on_guides_list.html
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Network VRU Types and Unified CVP Deployment Models
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Network VRU Types and Unified CVP Deployment Models
Note Type10 is available only in Unified ICM 7.1 and later, and new implementations should use this
configuration. For Unified ICM 7.0, a Type 2 Network VRU should be used in this case.
Associate all incoming dialed numbers with a Customer Instance that is associated with a Type 10
Network VRU. All the VRU Scripts that will be executed by this call must be associated with the same
Type 10 Network VRU. Although it is not always necessary, the best practice is for the Unified ICM
routing script to execute a SendToVRU node prior to the first RunExternalScript node.
Note Type10 is available only in Unified ICM 7.1 and later, and new implementations should use this
configuration. For Unified ICM 7.0, a Type 7 should be used in this case.
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Chapter 5 Interactions with Cisco Unified ICM
Network VRU Types and Unified CVP Deployment Models
There are two variants of this submodel, depending on whether the Correlation ID or the Translation
Route mechanism is used to transfer calls to the VRU. Most NICs (actually, most PSTN networks) are
not capable of transferring a call to a particular destination directory number and carrying an arbitrary
Correlation ID along with it, which the destination device can pass back to Unified ICM in order to make
the Correlation ID transfer mechanism function properly. For most NICs, therefore, the Translation
Route mechanism must be used.
There are a few exceptions to this rule, however, in which case the Correlation ID mechanism can be
used. The NICs that are capable of transmitting a Correlation ID include Call Routing Service Protocol
(CRSP), SS7 Intelligent Network (SS7IN), and Telecom Italia Mobile (TIM). However, because this
capability also depends on the PSTN devices that connect behind the NIC, check with your PSTN carrier
to determine whether the Correlation ID can be passed through to the destination.
If the NIC is capable of transmitting the Correlation ID, the incoming dialed numbers must all be
associated with a Customer Instance that is associated with a Type 7 Network VRU. The Type 7 Network
VRU must contain labels that are associated to the NIC routing client, and all the VRU Scripts must also
be associated with that same Type 7 Network VRU. The peripherals need not be associated with any
Network VRU. Although it is not always necessary, the best practice is for the Unified ICM routing script
to execute a SendToVRU node prior to the first RunExternalScript node.
If the NIC is not capable of transmitting a Correlation ID (the usual and safe case), then the incoming
dialed numbers must all be associated with a Customer Instance that is not associated with any Network
VRU. The Unified CVP peripherals must, however, be associated with a Network VRU of Type 8, and
all the VRU Scripts must also be associated with that same Type 8 Network VRU. In this case it is always
necessary to insert a TranslationRouteToVRU node in the routing script prior to the first
RunExternalScript node. If the call is going to the VRU leg because it is being queued, generally the
TranslationRouteToVRU node should appear after the Queue node. In that way, an unnecessary delivery
and removal from Unified CVP can be avoided when the requested agent is already available.
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Chapter 5 Interactions with Cisco Unified ICM
Hosted Implementations
because it is being queued, generally these two nodes should appear after the Queue node. In that way,
an unnecessary delivery and removal from Unified CVP can be avoided if the requested agent is already
available.
Note Two different VRU transfer nodes are required. The first one transfers the call away from the NIC with
a handoff, and it establishes Unified CVP as a Switch leg device for this call. Physically the call is
delivered to an ingress gateway. The second transfer delivers the call to the VoiceXML gateway and
establishes Unified CVP as the call's VRU device as well.
Hosted Implementations
This section covers the following topics:
• Overview of Hosted Implementations, page 5-10
• Using Unified CVP in Hosted Environments, page 5-11
• Unified CVP Placement and Call Routing in a Hosted Environment, page 5-11
• Network VRU Type in a Hosted Environment, page 5-13
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Hosted Implementations
In practice, however, the NAM and CICM architecture is flexible enough to enable a number of other
possibilities. Many Hosted customers use this topology simply as a way to get more calls or more PGs
through their ICM setup. Others use CICMs, not for customer contact centers, but for outsourcers. In
such cases, the NAM handles perhaps the same number of calls as the CICM, and the CICM machines
themselves might be located quite far away from the NAM. Also, the NAM and CICM architecture was
designed at a time when all contact centers ran on TDM-based ACDs. The addition of VoIP routing and
ACDs based on Unified CM (that is, Unified CCE) with direct agent routing made matters considerably
more complicated.
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Chapter 5 Interactions with Cisco Unified ICM
Hosted Implementations
– If Translation Route transfers are used, then the TranslationRouteToVRU node, together with
all RunExternalScript nodes, must be in the NAM routing script. The implication here is that
the call is queued (or treated with prompt-and-collect) before the particular CICM is selected.
This configuration does not make much sense for queuing, but it could be useful for service
providers who want to offer initial prompt-and-collect before delegating control to the CICM.
• If Unified CVP is placed at the CICM and a NIC handles the Switch leg while Unified CVP handles
the VRU leg, only the Translation Route transfer method can be used. The TranslationRouteToVRU
node, together with all RunExternalScript nodes, must be in the CICM routing script.
Adding calls initiated by Unified CM or an ACD brings additional constraints. Both of these devices are
considered ACDs from the ICM perspective, and they most likely are connected at the CICM level.
Assuming these are new calls (as opposed to continuations of existing calls), the route request will come
from the ACD and the resulting label will be returned to the ACD. Neither Unified CM nor any ACD is
capable of transmitting a Correlation ID upon transfer; only the Translation Route transfer method can
be used. This limitation further implies that the transfer destination (for example, Unified CVP) must
also be connected at the CICM level, not the NAM level.
If the calls are not new but continuations of existing calls, then they are attempts to transfer an existing
inbound caller from one agent to another agent. Customers might want these transfers to be either blind
network transfers (that is, the first agent drops off and the caller is delivered to a second agent or queued
for a second agent) or warm consultative transfers (that is, the caller goes on hold while the first agent
speaks to a second agent or waits in queue for a second agent and eventually hangs up, leaving the caller
talking to the second agent). The following guidelines apply to such transfers:
• Blind network transfers
Whether or not the original call was introduced to the NAM via a NIC or Unified CVP Switch leg,
the transfer label will be passed from the CICM to the NAM to the original Switch leg device. There
are two sub-cases of blind network transfers:
– If the Switch leg device is Unified CVP or a NIC that can handle Correlation ID, the Correlation
ID transfer mechanism can be used. The SendToVRU node and all RunExternalScript nodes
must be incorporated in the CICM routing script. The Unified CVP VRU leg can be connected
to the NAM. This combination of blind transfer with Correlation ID transfer is ideal for Unified
CVP and should be employed if at all possible.
– If the Switch leg device is a NIC that cannot handle Correlation ID, then the Translation Route
transfer method must be used, which further implies that the Unified CVP VRU leg device must
be connected to the CICM.
Note In this case, the customer might have to deploy additional dedicated Unified CVP Call
Servers at the CICM level because the NAM-level Unified CVP Call Servers cannot be
used.
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Deployment Models and Sizing Implications for Calls Originated by Cisco Unified Communications Manager and
In the one supported case where Unified CVP owns the initial Switch leg and the transfer is among
Unified CCE agents, the Translation Route transfer method must be used because Unified CM
cannot handle Correlation ID transfers. Again, this means that the Unified CVP VRU leg device
must be connected to the CICM.
Note In this case, the customer might have to deploy additional dedicated Unified CVP Call
Servers at the CICM level because the NAM-level Unified CVP Call Servers cannot be used.
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Chapter 5 Interactions with Cisco Unified ICM
Deployment Models and Sizing Implications for Calls Originated by Cisco Unified Communications Manager and ACDs
Each of the above calls invokes an Unified ICM routing script. The script might or might not search for
an available destination agent or service. If it finds an appropriate destination, it sends the corresponding
label either back to the ACD or, if blind-transferring an existing call, to the original caller's Switch leg
device. If it needs to queue the call or if the ultimate destination is intended to be a self-service
application rather than an agent or service, the script sends a VRU translation route label either back to
the ACD or, if blind-transferring an existing call, to the original caller's Switch leg device.
If the above sequence results in transferring the call to Unified CVP's VRU leg device, there is a second
transfer to deliver it to a VoiceXML gateway. To ensure that these events take place, the following
Unified ICM configuration elements are required:
• For new calls from the ACD or warm transfers of existing calls:
– The Unified CVP peripheral must be configured to be associated with a Type 10 Network VRU
(Type 2 if Unified ICM 7.0 is used).
– The dialed number that the ACD dialed must be associated with a Customer Instance that is
associated with a Type 10 Network VRU (Type 7 if Unified ICM 7.0 is used).
– With Unified ICM 7.0, or with a Unified ICM 7.1 and an ACD that is not Unified CM, the
routing script that was invoked by the ACD dialed number must contain a
TranslationRouteToVRU node to get the call to Unified CVP's Switch leg, followed by a
SendToVRU node to get the call to the VoiceXML gateway and Unified CVP's VRU leg.
– With Unified ICM 7.1 and Unified CM, the routing script that was invoked by Unified CM
should use a SendToVRU node to send the call to Unified CVP using the Correlation ID. The
Type10 VRU will perform an automatic second transfer to the VoiceXML gateway VRU leg.
– All the VRU scripts that are executed by that routing script must be associated with the Type 10
Network VRU (Type 7 if Unified ICM 7.0 is used).
• For blind transfers of existing calls:
– It does not matter to which Network VRU the Unified CVP peripheral is associated.
– The dialed number that the ACD dialed must be associated with a Customer Instance that is
associated with a Type 10 Network VRU (Type 7 if Unified ICM 7.0 is used).
– The routing script that was invoked by the ACD dialed number must contain a SendToVRU node
to get the call to the VoiceXML gateway and Unified CVP's VRU leg.
– All the VRU scripts that are executed by that routing script must be associated with the Type 10
Network VRU (Type 7 if Unified ICM 7.0 is used).
When Unified ICM chooses an agent or ACD destination label for a call, it tries to find one that lists a
routing client that can accept that label. For calls originated by an ACD or Unified CM which are not
blind transfers of existing calls, the only possible routing client is the ACD or Unified CM. Once the call
has been transferred to Unified CVP, because of the handoff operation, the only possible routing client
is the Unified CVP Switch leg. But in the case of blind transfers of existing calls, two routing clients are
possible: (1) the Unified CVP Call Server switch leg that delivered the original call, or (2) the ACD or
Unified CM. For calls that originate through Unified CVP, you can prioritize Unified CVP labels above
ACD or Unified CM labels by checking the Network Transfer Preferred box in the Unified ICM Setup
screen for the Unified CVP peripheral.
When using Unified CVP to do network transfers, an agent blind-transfers the caller to a new destination
and the Network Transfer Preferred option is used. In this scenario, the agent should use the CTI Agent
Desktop (and not the phone itself) to invoke the transfers. In addition to the CTI Agent Desktop, the
Unified ICM Dialed Number Plan should be used. If configured with the same DN as the CTI Route
Point, the Unified ICM Dialed Number Plan causes Unified ICM to intercept the transfer and run the
Unified ICM routing script without sending the transfer commands to Unified CM through JTAPI. When
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Chapter 5 Interactions with Cisco Unified ICM
Using Third-Party VRUs
the Unified ICM script returns a label, that label will be returned to the Network routing client (Unified
CVP), and the caller is sent directly to the new destination. This configuration avoids a timing problem
that can occur if an agent uses Unified CM CTI Route Points to initiate a network transfer.
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Chapter 5 Interactions with Cisco Unified ICM
Trunk Utilization Routing and Reporting
Note DS0 is the data line that provides utilization information about the number of trunks free on a gateway.
Deployment Considerations
Configuration and Deployment Considerations
• For Proxy Server Deployment with CUSP:
Configure TDM originating gateways for rai-targets to provide status in OPTIONS message to
primary and secondary Unified CVP Call Servers, for reporting purposes only. The data is only used
for reporting, not routing, so the data only needs to be sent to Call Servers that have reporting turned
on.
Configure primary and secondary CUSP proxy servers with Server Groups pinging to Unified CVP,
VXML gateways, and CUCM elements.
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Enhanced User-to-User Information
Configure Unified CVP with Server Group pinging to both primary and secondary CUSP proxies
for outbound calls.
• For Non Proxy Deployment:
Configure TDM originating gateways for RAI-targets to provide status in OPTIONS message to
primary and secondary Call Servers. Unified CVP can handle the messages for both reporting and
routing purposes. If used for routing, then the gateway will have to be in a server group by itself on
Unified CVP.
Configure Unified CVP with Server Groups pinging to Unified CVP, VXML gateways and CUCM
elements for outbound calls.
Configure VXML gateways for rai-targets to provide status in the OPTIONS message to primary
and secondary Call Servers.
• Refer to IOS documentation for recommendations on the high and low watermark settings.
• Unified CVP Call Servers can be configured to send the same hostname in the contact header of
OPTIONS requests to the gateways. This enables a single rai target to be configured to all Call
Servers. This is important since the limit is only 5 targets. The parameter to set is called Options
Header Override.
Limitations:
• RAI not currently supported on Proxy Servers.
CUP and CUSP servers do not currently handle the RAI header of OPTIONS messages, so they will
not mark the status of elements with that information. If VXML gateways are down, Unified CVP
could still send the call via the proxy, because the proxy does not handle incoming RAI headers in
OPTIONS. It is possible to use a local static route scheme on Unified CVP to send all calls to the
proxy except the VXML gateways calls, in order to create a server group for VXML gateways and
take advantage of RAI updates for routing.
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Chapter 5 Interactions with Cisco Unified ICM
Enhanced User-to-User Information
Whereas RTP media port and codec information is defined as a SDP body type, ISDN data is
encapsulated in a Generic Type Descriptor body type by the IOS gateways. When both RTP and ISDN
data are sent to Unified CVP via the TDM gateway, both body types are sent in a multipart/mixed mime
type, that includes both SDP and GTD parts.
The following configuration in the gateway is required to turn on the Enhanced UUI feature:
voice service voip
signaling forward unconditional
The rest of the GTD parameter values are preserved, keeping the values as they arrived from the
caller GTD.
• When GTD is not present in the inbound call leg, Unified CVP prints an informational message on
the trace stating No GTD Body present in Caller Body and the call continues as a regular call.
Note The modified UUI from Unified ICM is passed using the user.microapp.uui ECC variable, or the
Call.UsertoUserInfo variable.
Modified GTD is set in the outbound INVITE mime body from CVP SIP B2BUA, which includes IP
originated callers as well as TDM callers. If a DTMF label for outpulse transfer is received on a
connected call, then the BYE will be sent with the GTD only if UUI is passed by Unified ICM. The BYE
is sent immediately after the SIP INFO with DTMF.
Using UUI
Extract the UUI in your Unified ICM Script by looking at the Call ECC variable user.microapp.uui and
the Call.UserToUserInfo variable, such as in the IF node. By using the SET node on either one of these
variables, the variable can be set on the outbound direction of the call.
Setting Call.UserToUserInfo takes precedence over using the ECC variable.
Note Unified CVP will not send a BYE message on the DTMF label if UUI is not received from Unified ICM.
If a BYE message is received, then the GTD from the received BYE is used to send it on the other leg.
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Chapter 5 Interactions with Cisco Unified ICM
Custom SIP Headers
The ingress gateway should be configured with signaling forward unconditional, as in the following
example, so that GTD with UUI/UUS can be forwarded on the VoIP side.
Example:
voice service voip
signaling forward unconditional
Design Considerations
The UUI data transfer feature cannot be used with Hookflash or TBCT transfers.
Note Not all headers have a compact format. For example, P-Headers or private headers (for example
P-Asserted-Identity) may not have a compact form and hence the full header name shall be passed to
ICM.
Please refer to the table in the RFC3261 that defines the compact header abbreviations.
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Chapter 5 Interactions with Cisco Unified ICM
Custom SIP Headers
Note The SIP header modification feature is a powerful tool which can tweak SIP headers as needed. You
should exercise caution when applying SIP Profiles and ensure that the profile does not create interop
issues, rather than solving them. Unified CVP provides the flexibility to add/modify/remove outgoing
SIP header in the INVITE message only. This enables you to deploy Unified CVP in many scenarios to
facilitate inter-operability with third-party devices.
Outgoing SIP header feature do not allow you to remove or add Mandatory SIP headers. Only the modify
option is available for basic mandatory headers. These include SIP headers such as To, From, Via, CSeq,
Call-Id and Max-Forwards. There is no checking for the modifications in the ICM script editor, it is
actually enforced by the java SIP stack layer by throwing a DsSipParserException.
Typically, with Unified ICM, if the field is greater than 255 chars then it is truncated. In the SIP
subsystem, if there is a problem updating or adding a header with the string given from the Unified ICM
script, then you will either see an WARN type message in the Unified CVP log, if there is an
DsSipParserException, or else it will send the INVITE as - is with unexpected results on the receiver end.
This feature is applicable only for outgoing SIP INVITES (only the initial INVITE, not reinvites).
Changes to the INVITE are applied just before it is sent out. There is no restriction on the changes that
can be applied.
The header length (including header name) after modification should not exceed 255.
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Chapter 5 Interactions with Cisco Unified ICM
Courtesy Callback
Courtesy Callback
The Courtesy Callback feature is available in Unified CVP starting in Release 8.0(1). Courtesy Callback
reduces the time callers have to physically wait on hold or in a queue. The feature enables your system
to offer callers, who meet your criteria, the option to be called back by the system instead of waiting on
the phone for an agent. The caller who has been queued by Unified CVP can hang up and subsequently
be called back when an agent is close to becoming available (preemptive callback). This feature is
provided as a courtesy to the caller so that the caller does not have to wait on the phone for an agent.
Preemptive callback does not change the time a customer must wait to be connected to an agent, but
rather enables the caller hang up and not be required to remain in queue listening to music. Callers who
have remained in queue or have undergone the callback treatment will appear the same to agents
answering the call.
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Courtesy Callback
Note Scheduling a callback to occur at a specified time is not part of this feature.
The following illustration shows the components needed for the Courtesy Callback feature.
cvp_ccd_vxml.tcl CallbackEntry
CallbackQueue
CallbackEngine
CallbackWait
Customization Customization
of these of these
applications applications
is allowed is not allowed
Ingress/PreemptiveCallback
Unified ICM
Gateway
CourtesyCallback OAMP
survivability.tcl Script
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Courtesy Callback
• BillingQueue
This script is responsible for playing queue music to callers that either choose to not have a callback
occur or must reenter the queue for a short period after receiving a callback.
You may customize this script to suit your business needs.
• CallbackEngine
This script keeps the VoIP leg of a callback alive between when a caller elects to have a callback and
when a caller receives the callback.
Do not customize this script.
• Callback Entry
This script handles the initial IVR when a caller enters the system and when the caller is provided
with the opportunity to receive a callback.
You may customize this script to suit your business needs.
• CallbackQueue
This script handles the keepalive mechanism of a call while a caller is in queue and listening to the
music played within the BillingQueue script.
Do not customize this script.
• CallbackWait
This script handles the IVR portion of a call when a customer is called back.
You may customize this script to suit your business needs.
Callback Criteria
Examples of callback criteria you can establish include:
• Number of minutes a customer is expected to be waiting in queue exceeds some maximum number
of minutes (based on your average call handling time per customer)
Note The included sample scripts use this method for determining callback eligibility.
• Assigned status of a customer (gold customers may be offered the opportunity to be called back
instead of remaining on the line)
• The service a customer has requested (sales calls, or system upgrades, etc. may be established as
callback criteria)
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Courtesy Callback
2. The Call Studio and Unified ICM Courtesy Callback scripts determine if the caller is eligible for a
callback based on the rules of your organization (such as in the prior list of conditions).
3. If a courtesy callback can be offered, the system tells the caller the approximate wait time and offers
to call the customer back when an agent is available.
4. If the caller chooses not to use the callback feature, queuing continues as normal.
Otherwise, the call continues as indicated in the remaining steps.
5. If the caller chooses to receive a callback, the system prompts the caller to record their name and to
key in their phone number.
6. The system writes a database record to log the callback information.
Note If the database is not accessible, then the caller is not offered a callback and they are placed in
queue.
7. The caller is disconnected from the TDM side of the call. However, the IP side of the call in Unified
CVP and Unified ICM is still active. This keeps the call in the same queue position. No queue music
is played, so VXML gateway resources used during this time are less than if the caller had actually
been in queue.
8. When an agent in the service/skill category the caller is waiting for is close to being available (as
determined by your callback scripts), then the system calls the person back. The recorded name is
announced when the callback is made to insure the correct person accepts the call.
9. The system asks the caller, through an IVR session, to confirm that they are the person who was
waiting for the call and that they are ready for the callback.
If the system cannot reach the callback number provided by the caller (for example, the line is busy,
RNA, network problems, etc.) or they do not confirm they are the caller, then the call is not sent to
an agent. In this way, the agent is always guaranteed that someone is there waiting when they take
the call. The system assumes that the caller is already on the line by the time the agent gets the call.
This is why this feature is called preemptive callback - the system assumes that the caller is already
on the line by the time the agent gets the call and that the caller has to wait minimal time in queue
before speaking to an agent.
10. The system presents the call context on the agent screen-pop, as normal.
In the event that the caller cannot be reached after a configurable max number and frequency of retries,
the callback is aborted and the database status is updated appropriately. You can run reports to determine
if any manual callbacks are necessary based on your business rules.
Additional Conceptional Information
The Configuration and Administration Guide for Cisco Unified Customer Voice Portal guide provides a
call flow description of the function of the scripts that provide the Courtesy Callback feature.
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Post Call Survey
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Post Call Survey
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CH A P T E R 6
Calls Originated by Cisco Unified
Communications Manager
Table 6-1 New or Changed Information Since the Previous Release of This Document
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Chapter 6 Calls Originated by Cisco Unified Communications Manager
Customer Call Flows
destination specified by the returned label. As with other ACD post-route requests, the exchange ends
there. Unified ICM has no ability to send a subsequent label to that Unified CM unless Unified CM
issues another post-route request.
This limitation is the first difference between calls originated by Unified CM and calls originated
through a Unified CVP ingress gateway. Unified CVP can continue to route and re-route the call as many
times as necessary. For this reason, when calls are originated from Unified CM, routing client
responsibilities should be handed off to Unified CVP as soon as possible.
The second difference has to do with how calls are transferred to a VRU. ACD routing clients such as
Unified CM may be transferred only by using a TranslationRouteToVRU label. When Unified CVP is
the routing client, it can handle Translation Route labels as well as the Correlation ID labels that are
generated by SendToVRU nodes.
Locations-based call admission control is not supported for IP originated calls, including calls where the
agent performs a warm transfer or conference
The next sections provide more details on these differences.
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Chapter 6 Calls Originated by Cisco Unified Communications Manager
Protocol Call Flows
Note Model #4, VRU Only with NIC Controlled Routing, is not discussed here because there is no NIC
involved with calls originated by Unified CM.
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Chapter 6 Calls Originated by Cisco Unified Communications Manager
Protocol Call Flows
Note The script must not execute a Transfer node, unless it is to a TDM destination. Transfers to an IP
destination will result in an IP-to-IP call, which is supported, but requires that you add ip-ip-gw
commands (CUBE commands) to the gateway configuration for the transfer operation to another VoIP
destination to succeed.
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Chapter 6 Calls Originated by Cisco Unified Communications Manager
Protocol Call Flows
Now consider what happens if the target device issues another route request to Unified ICM. This part
of the call flow would not be possible without the initial TranslationRouteToVRU mentioned step 3.
12. Unified ICM invokes a new routing script.
13. The routing script encounters a Select or Label node, and it selects a target label.
14. Unified ICM returns the target label to the Unified CVP Call Server (not to the device that issued
the route request).
15. The Unified CVP Call Server consults the gatekeeper, DNS SRV, or SIP Proxy to locate the
destination device.
16. The Unified CVP Call Server communicates via H.323 or SIP with the target device and instructs
Unified CM to establish a media stream to the device.
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Chapter 6 Calls Originated by Cisco Unified Communications Manager
Deployment Implications
12. While the Unified CVP Microapplications are in progress, a target agent becomes available to take
the call.
13. Unified ICM determines a label for the target agent.
14. Unified ICM returns the target label to the Unified CVP Call Server.
15. The Unified CVP Call Server consults the gatekeeper, DNS SRV, or SIP Proxy to locate the
destination device.
16. The Unified CVP Call Server communicates via H.323 or SIP with the target device and instructs
Unified CM to establish a media stream to it, removing the VoiceXML gateway's media stream.
If the target device later issues another route request to Unified ICM, the call flow is again exactly as
described above. The call must again be transferred with Correlation ID via SendToVRU to the Unified
CVP Call Server and VoiceXML gateway to create the VRU leg. Microapplications might be executed,
and eventually the new target label is delivered to the Unified CVP Switch leg, which transfers the call
to that target.
Deployment Implications
This section presents guidelines for the following aspects of incorporating calls originated by
Unified CM into the deployment:
• Unified ICM Configuration, page 6-6
• Hosted Implementations, page 6-7
• Cisco Unified Communications Manager Configuration, page 6-7
• Sizing, page 6-8
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Chapter 6 Calls Originated by Cisco Unified Communications Manager
Deployment Implications
• With Cisco Unified ICM 7.1, if you want Unified CVP to be able to perform subsequent call control,
a translation route is not necessary if you use a Type 10 NetworkVRU. The Type 10 VRU uses the
Correlation ID mechanism to perform a transfer from Unified CM to Unified CVP using a
SendToVRU node. When the SendToVRU node is used with a Type 10 VRU, an initial transfer to
Unified CVP hands off call control to Unified CVP, and then an automatic second transfer to the
VRU leg is performed to deliver the call to a VoiceXML gateway for IVR treatment.
Note This call flow and all others in this document assume Cisco Unified ICM 7.0(0) or later.
• For additional configuration requirements, see Protocol Call Flows, page 6-3.
Hosted Implementations
Translation routes sent by one ICM router must be received by a peripheral that is connected to the same
ICM router. Therefore, you can translation-route a call from a Unified CM at the CICM level into
Unified CVP only if Unified CVP is also located at the CICM level. In Hosted environments, this means
you must provision Unified CVP Call Servers (Call Servers) at the CICM level even if you already have
other Call Servers at the NAM level. VoiceXML gateways and gatekeepers can, of course, be shared.
For more details on this subject, see the chapter on Interactions with Cisco Unified ICM, page 5-1.
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Chapter 6 Calls Originated by Cisco Unified Communications Manager
KPML Support
Sizing
Most customer implementations are not primarily designed for calls originated by Unified CM. The
main driver is usually incoming customer calls, although calls originated by Unified CM are frequently
a component, particularly in the case of warm transfers. Remember to consider those calls when sizing
equipment.
Gateways
Calls originated by Unified CM do not use ingress gateways at all. Calls are delivered directly from
Unified CM to the Unified CVP Call Server. They do, however, use VoiceXML gateways whenever a
VRU leg is in use. Therefore, for the purposes of sizing VoiceXML gateways, consider each Unified CM
call that is either in queue or conducting self-service activities.
KPML Support
KPML is an out-of-band dtmf method that passes key press information through SIP signaling instead
of through the RTP stream.
Typical Unified CVP Comprehensive call flows use the inband RFC2833 DTMF configuration on
endpoints. However, there are some endpoints that do not support inband RFC2833, such as the 7985
video phone, and CTI Ports used in UCCE Mobile Agent deployments.
For these endpoints, when the destination behind the SIP Trunk is set with RFC2833, Cisco Unified CM
allocates an MTP resource because the line side and the trunk side require a translation of the inband
packets to the out-of-band signaling messages for DTMF.
To avoid MTP allocation, the destination of the SIP Trunk needs to be configured using the SIP KPML
DTMF method (that is, the No Preference setting). Also, the VXML bootstrap dial peer requires SIP and
KPML settings.
The Unified CVP SIP subsystem can pass through Subscribe and Notify events related to KPML DTMF
digits (out-of-band DTMF).
A typical use for the KPML feature would be when a you want to use the Mobile Agent solution with
Unified CVP, and do not want to use MTP resources on Unified CM.
Design Considerations
The following limitations apply when using the KPLM feature:
1. If you have configured KPML for SIP on a dial-peer, you cannot use the same dial-peer for H.323.
Another bootstrap DP can be configured with the same pattern and a priority may be put on it to
handle the H.323 calls.
2. ASR/TTS is not supported with KPML.
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KPML Support
The SIP Trunk should be configured for DTMF No Preference if KPML is set on the gateway. If the SIP
Trunk points to the Cisco Unified Presence Server (CUP Server) or Unified CVP directly, the DTMF
Preference should still be set to No Preference, because the Unified CVP B2BUA is in the middle of the
call, and the SDP attributes are passed through as though they came directly from the VXML gateway.
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KPML Support
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CH A P T E R 7
Gateway Options
Cisco offers a large range of voice gateway models to cover a large range of requirements. Many, but not
all, of these gateways have been qualified for use with Unified CVP. For the list of currently supported
gateway models, always check the latest version of the Hardware and System Software Specification for
Cisco Unified CVP (formerly called the Bill of Materials), available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/prod_technical_reference_list.html
Gateways are used in Unified CVP for conversion of TDM to IP and for executing VoiceXML
instructions. The following sections help you determine which gateways to incorporate into your design:
• PSTN Gateway, page 7-2
• VoiceXML Gateway with DTMF or ASR/TTS, page 7-2
• VoiceXML and PSTN Gateway with DTMF or ASR/TTS, page 7-3
• TDM Interfaces, page 7-3
• Gateway Choices, page 7-5
• Gateway Sizing, page 7-6
• Using MGCP Gateways, page 7-10
Table 7-1 New or Changed Information Since the Previous Release of This Document
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Chapter 7 Gateway Options
PSTN Gateway
PSTN Gateway
In this type of deployment, the voice gateway is used as the PSTN voice gateway. The voice gateway is
responsible for converting TDM speech to IP and for recognizing DTMF digits and converting them to
H.245 or RFC2833 events. Unified CVP does not currently support passing SIP-Notify or KPML DTMF
events.
In a centralized Unified CVP deployment, you can separate the VoiceXML functionality from the ingress
gateway to provide a separate PSTN ingress layer. The separate PSTN layer and VoiceXML farm enables
the deployment to support a large number of VoiceXML sessions and PSTN interfaces. For example, the
Cisco AS5400XM can accept a DS3 connection, providing support for up to 648 DSOs. However, a
gateway that is handling that many ingress calls cannot also support as many VoiceXML sessions. In
such cases, the VoiceXML sessions should be off-loaded to a separate farm of VoiceXML-only gateways.
Note Any TDM interface can be used as long as it is supported by the Cisco IOS gateway and by the Cisco IOS
version compatible with CVP.
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Chapter 7 Gateway Options
VoiceXML and PSTN Gateway with DTMF or ASR/TTS
TDM Interfaces
The Cisco AS5400XM Universal Gateway offers unparalleled capacity in only two rack units (2 RUs)
and provides best-of-class voice, fax, and remote-access services. High density (up to one Channelized
T3 (CT3) of voice over IP (VoIP) and two CT3s of time-division multiplexing (TDM) switching), low
power consumption (as low as 2.4 A at 48 VDC per G.711 CT3), high-density packet voice digital signal
processor (DSP) modules, universal port DSPs, and session border control (SBC) features make the
Cisco AS5400XM Universal Gateway ideal for many network deployment architectures, especially
co-location environments and mega points of presence (POPs).
The Cisco AS5350XM Universal Gateway is the one-rack-unit (1 RU) gateway that supports 2-, 4-,8-,
or 16-port T1/12-port E1 configurations and provides universal port data, voice, and fax services on any
port at any time. The Cisco AS5350XM Universal Gateway offers high performance and high reliability
in a compact, modular design. This cost-effective platform is ideally suited for internet service providers
(ISPs) and enterprise companies that require innovative universal services.
The Cisco 2800 Series and 3800 Series and the newer 2900 Series and 3900 Series Routers support the
widest range of packet telephony-based voice interfaces and signaling protocols within the industry,
providing connectivity support for more than 90 percent of the world's private branch exchanges (PBXs)
and public switched telephone network (PSTN) connection points. Signaling support includes T1/E1
Primary Rate Interface (PRI), T1 channel associated signaling (CAS), E1-R2, T1/E1 QSIG Protocol, T1
Feature Group D (FGD), Basic Rate Interface (BRI), foreign exchange office (FXO), E&M, and foreign
exchange station (FXS). The Cisco 2800 Series and 3800 Series Routers can be configured to support
from two to 540 voice channels. The Cisco 2900 Series and 3900 Series Routers can be configured to
support from two to 720 voice channels.
For the most current information about the various digital (T1/E1) and analog interfaces supported by
the various voice gateways, refer to the latest product documentation available at the following sites:
• Cisco 2800 Series
http://www.cisco.com/en/US/products/ps5854/tsd_products_support_series_home.html
• Cisco 3800 Series
http://www.cisco.com/en/US/products/ps5855/tsd_products_support_series_home.html
• Cisco AS5300
http://www.cisco.com/en/US/products/hw/univgate/ps501/tsd_products_support_series_home.html
• Cisco 2900 Series
http://www.cisco.com/en/US/products/ps10537/index.htm.
• Cisco 3900 Series
http://www.cisco.com/en/US/products/ps10536/index.htm
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Chapter 7 Gateway Options
Cisco Unified Border Element
Note Unlike flow-through mode, with flow-around mode, customers lose the ability to do DTMF
interworking, transcoding, and other key functions such as tcl and media capabilities that flow-through
will otherwise allow.
A Unified Border Element is typically needed when replacing a TDM voice circuit with an IP voice
trunk, such as a SIP trunk, from a telephone company. The Cisco Unified Border Element serves as a
feature-rich demarcation point for connecting enterprises to service providers over IP voice trunks.
The Cisco Unified Border Element has been tested with, and can be used in, any of the following
scenarios:
• SIP-to-SIP connectivity between a third-party SIP device and Cisco Unified CVP
• SIP-to-SIP connectivity between Cisco Unified Communications Manager and Cisco Unified CVP
• Co-residency of the VoiceXML Gateway and the Cisco Unified Border Element for any of the above
scenarios
For CUBE session numbers, refer to:
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/order_guide_c07_4
62222.html.
• Transcoding between G.711 and G.729
For more information about using the Cisco Unified Border Element with Unified CVP, including
topologies and configurations, refer to the document Cisco Unified Border Element for Contact Center
Solutions, available at
http://cisco.com/en/US/docs/voice_ip_comm/unified_communications/cubecc.html
Note Due to a limitation in Cisco IOS, the Cisco Unified Border Element does not support mid-call escalation
or de-escalation from audio to video, and vice versa.
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Chapter 7 Gateway Options
Gateway Choices
Gateway Choices
Unified CVP uses gateways for two purposes: TDM ingress and VoiceXML rendering. Any Cisco
gateway that is supported by Unified CVP can usually be used for either purpose or both. However,
depending on your deployment model, you might need only one of the functions:
• Model #1: Standalone Self-Service
All calls use both ingress and VoiceXML.
• Model #2: Call Director
All calls use ingress only.
• Model #3a: Comprehensive Using ICM Micro-Apps
All calls use ingress, and some calls use VoiceXML.
• Model #3b: Comprehensive Using Unified CVP VXML Server
All calls use ingress, and some calls use VoiceXML.
• Model #4: VRU Only with NIC Controlled Routing
All calls use both ingress and VoiceXML.
In cases where both ingress and VoiceXML are required, you can choose to run both functions on the
same gateways or you can choose to designate some gateways for ingress and others for VoiceXML. Use
the following guidelines to determine whether the functions should be combined or split:
• In classical branch office deployments, where the call needs to be queued at the branch where it
arrived, ingress and VoiceXML functions must always be combined.
• In cases where a large number of non-CVP PSTN connections will share the gateways, it is
recommended to dedicated Ingress for that purpose, and use separate VXML gateways.
• VoiceXML-only gateways are less costly because they do not require DSP farms or TDM cards. Use
a spreadsheet to determine which way you obtain the best price.
• With relatively low call volume, it is usually better to combine the functions for redundancy
purposes. Two combined gateways are better than one of each because the loss of one gateway still
allows calls to be processed, though at a lower capacity.
The next decision is whether to use Cisco Integrated Service Router (ISR) gateways (Cisco 2800, 2900
series routers), ISGR-G2 (3800, or 3900 Series routers), or the Cisco AS5x00 Series Gateways.
Guidelines state that ISR gateways should be used only in branch office sites, whereas AS5x00 Series
gateways should be used in centralized data center sites.
Sometimes it is difficult to determine what constitutes a branch office, and therefore which gateway
should be used. The following guidelines can help with that determination:
• The classical definition of branch offices, for which you should use ISR gateways, includes:
– Multiple sites where TDM calls will be arriving from the PSTN.
– Those sites are separated from the data centers where most of the Unified CVP equipment
resides.
– You will be placing only one gateway at each site.
• If you have sites where you will be stacking multiple gateways for any reason, then those sites are
data center sites and should use Cisco AS5x00 Series gateways.
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Gateway Sizing
For more information on the Cisco AS5x00 Series Gateways, refer to the technical specifications
available at
http://www.cisco.com/en/US/products/hw/iad/index.html
For more information on the Cisco Integrated Service Routers (ISRs), refer to the documentation
available at
http://www.cisco.com/en/US/products/hw/routers/index.html
Gateway Sizing
Individual Cisco gateways can handle various call capacities depending on whether they are doing
ingress only, VoiceXML only, or a combination of the two. Gateways doing VoiceXML activities also
have different call capacities depending on whether or not they are supporting ASR or TTS activities and
on the type of VoiceXML application being executed. For instance, an intensive JavaScript application
reduces call capacity. Gateways doing HTTPS experience lower call capacity as compared to HTTP.
In general, gateways performing ingress-only can be sized according to the number of TDM cables that
can be connected to them. For gateways that are combined or VoiceXML-only, it is important to ensure
that the overall CPU usage will be less than 75% on average. The numbers in Table 7-3 are based on
Unified CVP VoiceXML documents; other applications that generate more complex VoiceXML
documents will have a higher impact on performance. The following factors affect CPU usage:
• Calls per second (cps)
• Maximum concurrent calls
• Maximum concurrent VoiceXML sessions
Before sizing the voice gateways, use the UCCE Resource Calculator to determine the maximum number
of trunks (DS0s) and VoiceXML IVR ports needed to support the entire solution.
For almost all Unified CVP deployment models, sizing is based on the maximum number of concurrent
VoiceXML sessions and VoIP calls. The following tables list this information for different IOS release
versions as follows:
• Table 7-2 (IOS version 15.1.1T
• Table 7-3, Table 7-4, and Table 7-5(IOS version 12.4.15 and greater, but not including 15.1.1.T)
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Table 7-3 For IOS Version 12.4.(15)T5 and greater, but prior to version 15.0.1M or 15.1. 1T
Maximum Number of VoiceXML Sessions Supported by Cisco Voice Gateways
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Table 7-4 For IOS Version 12.4.(15)T5 and greater, but prior to version 15.0.1M or 15.1. 1T
Maximum Number of VoiceXML Sessions Supported by Cisco Voice Gateways
Executing Intensive JavaScript Applications
Table 7-5 For IOS Version 12.4.(15)T5 and greater, but prior to version 15.0.1M or 15.1. 1T
Maximum Number of VoiceXML Sessions Supported by Cisco Voice Gateways Using
HTTPS
Note The following note does not apply to IOS 15.0.1M and IOS 15.1.1T.
Performance numbers for the Cisco 3825 Series and 3845 Series Integrated Services Routers (ISRs) are
higher when the voice gateway and the VoiceXML gateway functions reside on the same router
(co-resident deployment). When the call is connected to the VoiceXML gateway from the ingress voice
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gateway, the media flows directly between the two. In a co-resident deployment, the gateway does not
have to spend CPU cycles to packetize and de-packetize the RTP packets. Hence, by saving these CPU
cycles, the gateway can support increased VoiceXML sessions.
The numbers in Table 7-3, Table 7-4, and Table 7-5 assume the only activities running on the gateway
are VXML with basic routing and IP connectivity. If you intend to run additional applications such as
fax, security, normal business calls, and so forth, then the capacity numbers presented here should be
prorated accordingly. The numbers mentioned in the "Voice Gateway and VoiceXML" column mean that
the indicated number of VoiceXML sessions and voice calls can be supported simultaneous on the same
gateway. For example, in Table 7-3 the AS5350XM can terminate a maximum of 240 PSTN calls, and
those 240 PSTN calls could have 240 corresponding VoiceXML sessions at the same time.
The numbers represent performance with scripts generated by Unified CVP Studio running on the
Unified CVP VXML Server. Other VoiceXML applications might perform differently. These figures
apply if the CPU utilization does not exceed more than 75%, Voice Activity Detection (VAD) is turned
off, and your system is running VoiceXML v2.0 and MRCP v2 with Cisco IOS Release 12.4(15)T5. The
Cisco 1861 Integrated Services Router requires Cisco IOS 12.4(20)T1 as the minimum release.
Note These performance numbers are accurate when used with either the Cisco Call Server or Cisco
Unified CVP VXML Server. Performance can, and often does, vary with different applications.
Performance from external VoiceXML applications (such as Nuance OSDMs) might not be
representative of the performance when interoperating with non-Cisco applications. You must ensure
that the CPU usage is less than 75% on average and that adequate memory is available on Cisco gateways
at full load when running external VoiceXML applications. Users should contact the application provider
of the desired VoiceXML application for performance and availability information. Be aware that
external VoiceXML applications are not provided by Cisco, and Cisco makes no claims or warranties
regarding the performance, stability, or feature capabilities of the application when interoperating in a
Cisco environment.
Note Cisco does not specifically test or qualify mixes of traffic because there are infinite combinations. All
numbers should be seen as guidelines only and will vary from one implementation to the next based on
configurations and traffic patterns. It is recommended that the systems be engineered for worst-case
traffic (all ASR) if it is not known or cannot be predicted what kinds of calls will be offered to the VXML
gateway.
Note If you run VoiceXML on one of the Cisco 1800, 2800, 3800 or 2900 and 3900 Series gateways, additional
licenses (FL-VXML-1 or FL-VXML-12) are required.
Also consult the following links to ensure that the concurrent call load and call arrival rates do not exceed
the listed capacities:
• Model Comparison:
http://www.cisco.com/en/US/products/ps10536/prod_series_comparison.html
• Gateway Sizing for Contact Center Traffic:
http://cisco.biz/en/US/docs/voice_ip_comm/cucm/srnd/8x/gateways.html#wp1043594
In addition to these capacities, also consider how much DRAM and flash memory to order. The capacity
that comes with the machine by default is usually sufficient for most purposes. However, if your
application requires large numbers of distinct .wav files (as with complex self-service applications) or
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Chapter 7 Gateway Options
Using MGCP Gateways
if your application has unusually large .wav files (as with extended voice messages or music files), you
might want to increase the amount of DRAM in order to accommodate more cache space. The .wav files
are recorded at 8 kbps. Additionally, if you plan to use the flash memory itself rather than a media server
to house media files, you might want to expand your flash memory order. The use of DRAM for prompt
caching is discussed in detail in the chapter on Media File Options, page 12-1.
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Design Implications for Unified CVP
VXML Server
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Chapter 8 Design Implications for Unified CVP VXML Server
Multi-Language Support
Multi-Language Support
The Cisco IOS Voice Browser or the Media Resource Control Protocol (MRCP) specification does not
impose restrictions on support for multiple languages. However, there might be restrictions on the
automatic speech recognition (ASR) or TTS server. Check with your preferred ASR/TTS vendor about
their support for your languages before preparing a multilingual application.
You can dynamically change the ASR server value by using the command cisco property
com.cisco.asr-server in the VXML script. This property overrides any previous value set by the VXML
script.
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Where to Install Cisco Unified Call Studio
Either a Tomcat or WebSphere Application server running Unified CVP VXML can support up to
1200 simultaneous calls per Cisco MCS-7845-I3-CCE2 server.
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Where to Install Cisco Unified Call Studio
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CH A P T E R 9
Network Infrastructure Considerations
This chapter presents deployment characteristics and provisioning requirements of the Unified CVP
network. Provisioning guidelines are presented for network traffic flows between remote components
over the WAN, including recommendations for applying proper Quality of Service (QoS) to WAN traffic
flows.
For the most current information on network considerations, refer to the sections on deployment models,
bandwidth, and QoS presented in the latest version of the Cisco Unified Contact Center Enterprise
Solution Reference Network Design (SRND), available at
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_implementation_design_gui
des_list.html
This chapter covers the following topics:
• What's New in This Chapter, page 9-1
• Bandwidth Sizing, page 9-5
• Call Admission Control, page 9-9
• QoS Marking, page 9-13
• Network Latency, page 9-13
• Blocking Initial G.711 Media Burst, page 9-14
• Network Security Using Firewalls, page 9-15
Table 9-1 New or Changed Information Since the Previous Release of This Document
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Bandwidth Provisioning and QoS Considerations
Note RSVP. Cisco Unified CM 5.0 introduced support for Resource Reservation Protocol (RSVP) between
endpoints within a cluster, and 8.0 Unified CM introduces RSVP over the SIP Trunk. RSVP is a protocol
used for call admission control, and it is used by the routers in the network to reserve bandwidth for calls.
RSVP is not qualified for call control signaling via the Unified CVP Call Server in SIP or H.323 in the
8.0(1) release. The recommended solution for Call Admission Control is to employ Locations
configuration on CVP and in UCM. Refer to Local Branch Call Admission Control
(LBCAC/Queue-at-the-Edge), page 9-9.
Voice Traffic
Voice calls consist of Real-Time Transport Protocol (RTP) packets that contain actual voice samples.
RTP packets are transmitted in the following cases:
• Between the ingress PSTN gateway or originating IP phone and one of the following:
– Another IP phone, such as an agent
The destination phone might or might not be co-located with the ingress gateway or calling IP
phone, and the connection can be over a WAN or LAN.
– An egress gateway front-ending a TDM ACD (for legacy ACDs or IVRs)
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The egress gateway might or might not be co-located with the ingress gateway, and the
connection can be over a WAN or LAN.
– A VoiceXML gateway performing prompt-and-collect treatment
The VoiceXML gateway will usually be the same gateway as the ingress gateway, but it can be
different. In either case, both the ingress and VoiceXML gateways are typically co-located
(located on the same LAN). The connection is typically over a LAN but can be over a WAN.
• Between the VoiceXML gateway and the ASR/TTS server. The RTP stream between the VoiceXML
gateway and ASR/TTS server must be G.711
G.729 Versus G.711 Codec Support
Release 8.0(1) of CVP supports a flattened deployment of the g.711 codec on all call legs, or else a flat
g.729 deployment on all call legs. A mixture of the 2 codecs on different legs of the call, at any point
in the duration of the call, is not supported.
Benefits and drawbacks for G.711 codec include:
• You do not have to convert prompts to G.729
• However, the solution requires significantly more bandwidth over the WAN link.
Benefits and drawbacks for G.729 codec include:
• No extra bandwidth is required.
• You must convert all prompts to G.729
• G.729 prompts have an inferior audio quality to G.711 prompts
• ASR/TTS cannot be used.
H.323 or SIP
Unified CVP is currently certified with three types of VoIP endpoints: Cisco IOS voice gateways, Cisco
Unified Communications Manager (Unified CM), and the PGW in either Call Control mode or Signaling
mode. Call Control traffic flows between the following endpoints:
• Ingress gateway to/from Unified CVP Call Server
The ingress gateway can be a PGW, Unified CM, or a Cisco IOS gateway, or other SIP device in the
case of SIP. The connection can be over a WAN or LAN.
• Unified CVP Call Server to/from egress gateway
The egress gateway can be Unified CM or a Cisco IOS gateway. The egress gateway is either a
VoiceXML gateway used to provide prompt-and-collect treatment to the caller, or it is the target of
a transfer to an agent (Unified CCE or TDM) or a legacy TDM IVR. The connection can be over a
WAN or LAN.
Note Currently approved deployment designs do not support SIP for interoperability between the
PGW and Unified CVP. If your design requires this functionality, contact the Cisco Assessment
to Quality (A2Q) team.
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GED-125
The Unified CVP Call Server and the Unified ICM VRU PG communicate using the GED-125 protocol.
The GED-125 protocol includes:
• Messages that control the caller experience, such as notification when a call arrives
• Instructions to transfer or disconnect the caller
• Instructions that control the IVR treatment the caller experiences
The VRU PG normally connects to Unified CVP over a LAN connection. However, in deployments that
use clustering over the WAN, it is possible for Unified CVP to connect to the redundant VRU PG across
the WAN.
At this time, no tool exists that specifically addresses communications between the VRU PG and
Unified CVP. However, bandwidth consumed between the Unified ICM Central Controller and VRU PG
is very similar to the bandwidth consumed between the VRU PG and Unified CVP.
The VRU Peripheral Gateway to ICM Central Controller Bandwidth Calculator tool is available (with
proper login authentication) through the Cisco Steps to Success Portal at
http://tools.cisco.com/s2s/HomePage.do?method=browseHomePage
You can also access the Bandwidth Calculator directly (with proper login authentication) at
http://tools.cisco.com/s2slv2/ViewDocument?docName=EXT-AS-100901
If the VRU PGs are split across the WAN, the total bandwidth required would be double what the
calculator tool reports: once for Unified ICM Central Controller to VRU PG and once for VRU PG to
Unified CVP.
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Bandwidth Sizing
Data Traffic
Data traffic includes VoiceXML documents and prerecorded media files returned as a result of HTTP
requests executed by the VoiceXML gateway. Specifically:
• The VoiceXML gateway requests media files in an HTTP request to a media file server. The media
server response returns the media file in the body of the HTTP message. The VoiceXML gateway
then converts the media files to RTP packets and plays them to the caller. The connection in this case
can be over a WAN or LAN.
• The VoiceXML gateway requests VoiceXML documents from either the Unified CVP
VXML Server or the Unified CVP IVR Service. The connection in this case can be over a WAN or
LAN.
This chapter focuses primarily on the types of data flows and bandwidth used between a remote ingress
gateway and the components with which it interfaces:
• Unified CVP VXML Server
• Unified CVP Call Server IVR Service
• Unified CVP Call Server SIP or H.323 Service
• IP phones
• Media servers
• Egress gateways
• ASR or TTS servers
Guidelines and examples are presented to help estimate required bandwidth and, where applicable,
provision QoS for these network segments.
Bandwidth Sizing
As discussed above, most of the bandwidth requirements in a Unified CVP solution occur in a
Distributed Unified CVP topology, due primarily to the fact that the ingress and/or VoiceXML gateway
is separated from the servers that provide it with media files, VoiceXML documents, and call control
signaling. For purposes of the following discussion, assume all calls to a branch begin with one minute
of IVR treatment followed by a single transfer to an agent that also lasts one minute. Each branch has
20 agents, and each agent handles 30 calls per hour for a total of 600 calls per hour per branch. The call
average rate is therefore 0.166 calls per second (cps) per branch.
Note that even a slight change in these variables might have a large impact on sizing. It is important to
remember that .166 calls per second is an average for the entire hour. Typically, calls do not come in
uniformly across an entire hour, and there are usually peaks and valleys within the busy hour. Try to find
the busiest traffic period, and calculate the call arrival rate based on the worst-case scenario.
VoiceXML Documents
VoiceXML (VXML) documents are generated based on voice application scripts written using either
Unified ICM scripts or Cisco Unified Call Studio, or both. A VoiceXML document is generated for every
prompt that is played to the caller. The VoiceXML documents vary in size, depending on the type of
prompt being used; menu prompts with many selections are much larger than a simple prompt that
simply plays an announcement.
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Bandwidth Sizing
On average, a VoiceXML document between the Unified CVP Call Server or Unified CVP
VXML Server and the gateway is about 7 kilobytes. You can calculate the bandwidth used by
approximating the number of prompts that will be used per call, per minute. The calculation, for this
example, is as follows:
7,000 bytes ∗ 8 bits = 56,000 bits per prompt
(.166 call/second) * (56,000 bit/prompt) * (# of prompts / call) = kbps per branch
However, if you are going to use a more complex application that uses many menu prompts (more than
the average estimated above) or if you want to calculate the bandwidth more exactly, you can use the
VoiceXML document sizes listed in Table 9-2 to calculate the amount of bandwidth needed. The
document sizes in Table 9-2 are measured from the Unified CVP VXML Server to the VXML Gateway.
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H.323 Signaling
Every call that is processed by the branch gateway requires 6000 bytes, plus 1000 bytes for each
transferred call to an agent, giving a total of 56,000 bits per call (7000 bytes ∗ 8 bits). Thus, the average
bandwidth required would be (0.166 ∗ 56 kbps) = 9.3 kbps for the WAN link to the remote branch.
SIP Signaling
SIP is a text-based protocol, therefore the packets used are larger than with H.323. The typical SIP call
flow uses about 17,000 bytes per call. Using the previous bandwidth formulas based on calls per second,
the average bandwidth usage would be:
(17,000 bytes/call) ∗ (8 bits/byte) = 136,000 bits per call
(0.166 calls/second) ∗ (136 kilobits/call) = 22.5 average kbps per branch
Classifying RTP Media Traffic Between VoiceXML Gateways and ASR/TTS Servers
The RTP port range used by the VoiceXML gateway is the normal Cisco IOS RTP UDP port range of
16384 to 32767; however, the RTP UDP port range used by the ASR/TTS server can vary by OS and
ASR/TTS vendor. It is possible to construct an ACL to match the traffic from the ASR/TTS server based
on the VoiceXML gateway UDP port range; but if possible, Cisco recommends finding the ports used by
the ASR/TTS server as well. The RTP traffic should be marked with DSCP EF so that it is placed in the
priority queue with other voice traffic.
The QoS priority queue must also be configured to support the maximum number of ASR/TTS sessions
anticipated. If a call admission control mechanism such as Cisco Unified CM locations or Resource
Reservation Protocol (RSVP) is used, this extra priority queue bandwidth should not be included when
configuring the locations or RSVP bandwidth. For example, if you want to support two ASR/TTS G.711
sessions (80 kbps each) as well as four IP telephony phone calls using G.729 (24 kbps each), the priority
queue total bandwidth would be 256 kbps. The locations call admission control or RSVP bandwidth
should be limited to only the IP telephony bandwidth (96 kbps in this example). Configuring the
locations or RSVP bandwidth with 256 kbps would allow IP telephony calls to use all of the bandwidth
and conflict with the ASR/TTS sessions.
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Note 80 kbps is the rate for G.711 full-duplex with no VAD, including IP/RTP headers and no compression.
24 kbps is the rate for G.729 full-duplex with no VAD, including IP/RTP headers and no compression.
For more information on VoIP bandwidth usage, refer to the Voice Codec Bandwidth Calculator (login
authentication required), available at http://tools.cisco.com/Support/VBC/do/CodecCalc1.do.
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Call Admission Control
Note RSVP. Cisco Unified CM 5.0 introduced support for Resource Reservation Protocol (RSVP) between
endpoints within a cluster and Unified CM Release 8.0 introduces RSVP over the SIP trunk. RSVP is a
protocol used for call admission control, and it is used by the routers in the network to reserve bandwidth
for calls. RSVP is not qualified for call control signaling via the Unified CVP Call Server in SIP or H.323
in the 8.0(1) release. The recommended solution for Call Admission Control is to employ Locations
configuration on Unified CVP and in Unified CM.
For more information on RSVP, refer to the latest version of the Cisco Unified Communications SRND
Based on Cisco Unified Communications Manager, available at
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guides
_list.html
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Call Admission Control
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Chapter 9 Network Infrastructure Considerations
Call Admission Control
The goal of this model is to first route the calls locally to an agent available in the branch office, if
possible, and keep the media streams local. If the local agent is not available, only the call gets routed
to the agent on another branch office over the WAN link; the originating call and the initial VRU
treatment are done locally.
Another advantage of this deployment configuration is that in the event of WAN link failure, the call can
still be routed locally using the CVP survivability application running on the pots dial-peer for TDM
originated calls.
Design Considerations
The following considerations apply when using LBCAC:
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Chapter 9 Network Infrastructure Considerations
Call Admission Control
• A trunk configured with MTP required will not work with the LBCAC siteID feature. The reason is
when MTP is inserted, the media is terminated between the end point and MTP resource, not
between the two end points.
• If a MTP/Transcoder/TRP media resource is inserted by the UCM media layer, the incoming
location information is not used.
• If the inter cluster call is not hair-pin/loop back to the same cluster, the former behavior of Location
CAC logic will apply.
• Each site is uniquely identified by one siteID. Multiple gateways at the same site would need to align
to the same siteID, but if two clusters happen to use the same location name, then two siteIDs can
map to the same physical branch.
• A second Unified CM cluster may have the same location as the first cluster, but will still be required
to use a unique siteID on Unified CVP. You can define a route in the proxy server to send those
cluster calls to the common VXML gateway at the same location, but used by both the clusters.
• Each cluster would manage the bandwidth for devices in its cluster. If two clusters happen to use the
same physical location, then they would each separately manage the bandwidth for the phones that
they manage.
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Chapter 9 Network Infrastructure Considerations
QoS Marking
QoS Marking
The Unified CVP Call Server marks only the QoS DSCP for SIP messages. If QoS is needed for Unified
CVP H.323 signaling and data traffic across a WAN, configure network routers for QoS using the IP
address and ports to classify and mark the traffic as recommended in Table 9-3.
Maximum
Latency
Component Port Queue PHB DSCP (Round Trip)
1
Media Server TCP 80 CVP-Data AF11 101 1 sec
Unified CVP Call Server, H.323 TCP 1720 Call Signaling CS3 24 200 ms
Unified CVP Call Server, SIP TCP or UDP 5060 Call Signaling CS3 24 200 ms
1 1
Unified CVP IVR Service TCP 8000 CVP-Data AF11 10 1 sec
Unified CVP VXML Server TCP 7000 CVP-Data AF111 101 1 sec
Ingress Gateway, H.323 TCP 1720 Call Signaling CS3 24 200 ms
Ingress Gateway, SIP TCP or UDP 5060 Call Signaling CS3 24 200 ms
VoiceXML Gateway, H.323 TCP 1720 Call Signaling CS3 24 200 ms
VoiceXML Gateway, SIP TCP or UDP 5060 Call Signaling CS3 24 200 ms
H.323 Gatekeeper UDP 1719 Call Signaling CS3 24 200 ms
SIP Proxy Server TCP or UDP 5060 Call Signaling CS3 24 200 ms
MRCP TCP 554 Call Signaling CS3 24 200 ms
1. The DSCP (or PHB) value for CVP-Data traffic is only a recommendation. You can choose the actual DSCP value used to mark the traffic according to
your preference.
Neither the CVP-Data queue nor the Signaling queue is a priority queue as described in Cisco IOS router
terminology. The priority queue is used for voice or other real-time traffic, while call signaling and
Unified CVP traffic are reserved a certain amount of bandwidth based on the call volume.
Network Latency
Once proper application bandwidth and QOS policies are in place, another important consideration in a
distributed CVP deployment is that of network latency. With sufficient network bandwidth, the primary
contributor to latency is distance. In distributed CVP deployments, it is important to minimize this
latency and to also understand its effect on solution performance.
The primary effect of network latency between CVP components is on the end user’s calling experience.
Call signaling latency, either SIP or H.323, between the CVP Call Servers and voice gateways will affect
the call setup time and may add a period of silence during this setup. This includes the initial call setup
and subsequent transfers and/or conferences that are part of the final call flow. VXML application
document download time is also significantly affected by network latency and will have a pronounced
effect on the ultimate caller experience.
For the best caller experience, network latency of less than 200 ms round-trip is recommended. However,
the solution will tolerate round-trip times of up to 400 ms, but with an impact on caller experience. The
solution also makes heavy use of the HTTP protocol to transfer Voice XML documents and other media
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Chapter 9 Network Infrastructure Considerations
Blocking Initial G.711 Media Burst
files that are ultimately played to the caller. For the best end user calling experience, this HTTP traffic
should be treated with a priority higher than that of normal HTTP traffic in an enterprise network. The
recommendation is to treat this HTTP traffic the same as CVP call signaling traffic if possible.
Measures that may be used to work around latency issues include moving the VXML server to the same
local area as the VXML gateway, or using Cisco Wide Area Application Services (WAAS).
Otherwise, new configuration, listed in the following bullets, can help with WAN delays by providing
audio during times of dead air so that the caller does not disconnect.
• On the survivability service, the setting for "wan-delay-ringback" can be set to 1 to add a ringback
tone during longer than normal call setup times with IVR.
• IVR subsystem settings for IVR.FetchAudioDelay and IVR.FetchAudioMinimum are added. They
are WAN Delay settings for when root doc fetch is delayed over the WAN link.
– A default setting of 2 is needed to avoid a blip sound in a normal network scenario
– Setting WAN Delay to zero will have the effect of immediately playing holdmusic.wav and then
playing it for a minimum of 5 seconds.
– ECC variables such as user.microapp.fetchdelay, user.microapp.fetchminimum and
user.microapp.fetchaudio may be used to override these values in between invocations of
getSpeechExternal microapps.
• Specify the value for IVR.FetchAudio as follows: fetchaudio="flash:holdmusic.wav". Leave the
default empty so that nothing will be played in a normal scenario.
interface serial0/0
ip access-group 100 out
In the preceding example, 10.0.0.1 is the voice gateway’s H.323-bound IP address and 10.10.0.100 is the
Call Server. If there are multiple Call Servers, add one ACL entry for each. The interface serial0/0 is the
WAN interface connecting to the central site that is hosting the Call Server.
Note The preceding workaround will also avoid sending ICMP “unreachable” messages that occur due to the
failure to establish the RTP connection.
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Chapter 9 Network Infrastructure Considerations
Network Security Using Firewalls
Note Because the Unified CVP Operations Console Server uses dynamic ports for communication with other
components, it cannot be deployed outside of a firewall while the rest of the Unified CVP components
reside inside the firewall.
Table 9-4 TCP/UDP Ports Used by Unified CVP, Voice Gateways, and VoiceXML Gateways
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Network Security Using Firewalls
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CH A P T E R 10
Call Transfer Options
Designing for call transfers is one of the major steps required when designing a Unified CVP
deployment. There are numerous transfer options that can be used with Unified CVP. The goal of this
chapter is to explain each of the various options and to provide pros, cons, and considerations associated
with each.
This chapter covers the following topics:
• Release Trunk Transfers, page 10-2
• ICM Managed Transfers, page 10-5
• Network Transfer, page 10-6
• SIP Refer Transfer, page 10-7
• H.323 Refer Transfer, page 10-7
• Intelligent Network (IN) Release Trunk Transfers, page 10-7
• VoiceXML Transfers, page 10-8
Table 10-1 New or Changed Information Since the Previous Release of This Document
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Chapter 10 Call Transfer Options
Release Trunk Transfers
Takeback-and-Transfer (TNT)
TNT (also known as Transfer Connect) is a transfer mechanism offered by some U.S. PSTN service
providers (such as AT&T and Verizon). With this transfer method, inband DTMF tones are outpulsed to
the PSTN by Unified CVP. These inband tones act as a signaling mechanism to the PSTN to request a
transfer to be completed. A typical DTMF sequence is *8xxxx, where xxxx represents a new routing label
that the PSTN understands. Upon detection of a TNT DTMF sequence, the PSTN drops the call leg to
the ingress gateway port and then re-routes the caller to a new PSTN location (such as a TDM ACD
location).
This behavior might be necessary for a customer with existing ACD site(s) but no IVR, who wants to
use Unified CVP initially as just an IVR. Over time, the customer might want to transition agents from
the TDM ACD(s) to Cisco Unified CCE and use Unified CVP as an IVR, queueing point, and transfer
pivot point (thus eliminating the need for TNT services).
In Unified CVP deployments with the ICM, the DTMF routing label outpulsed could have been a Unified
ICM translation routing label to enable passing of call data to another Unified ICM peripheral (such as
a TDM ACD). In this scenario, Unified CVP views the call as completed, and Unified CVP call control
is ended. With TNT, if the transfer to the termination point fails, there is nothing Unified CVP can do to
re-route the call. While some TNT services do have the ability to re-route the call back to Unified CVP,
Unified CVP sees this call as a new call.
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Release Trunk Transfers
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Chapter 10 Call Transfer Options
Release Trunk Transfers
An alternate method that can be used on some PBXs is a "converse on step," whereby DTMF tones
indicating DNIS and ANI are sent to the IVR. This method requires a single main Unified ICM
routing script to input DNIS digits using a Get Data (GD) Microapplication and to invoke the correct
sub-script based on the collected DNIS digits. This method requires close coordination between
Cisco, the PBX vendor, and the customer, and it has not yet been tested.
Design Considerations
The following limitations apply to using the SIP Hookflash feature:
• This feature is only supported on 2X and 3X gateways. It is not supported on 5X gateways (e.g.
5400XM) due to the fact that the 5X gateway cannot generate a line-side hookflash.
• Hookflash only applies to TDM originated calls. Once hookflash is invoked by Unified CVP,
Unified CVP is no longer in control of the call.
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Chapter 10 Call Transfer Options
ICM Managed Transfers
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Chapter 10 Call Transfer Options
Network Transfer
Network Transfer
Unified CVP provides the capability to transfer calls to another destination after they have been
answered by an agent. This capability are referred to as Network Transfer.
When a call is transferred from Unified CVP to an agent, and that agent wants to transfer the call to
another agent, the agent can make that transfer using either the agent IP phone or agent desktop.
Transfers from the IP phone are made using CTI route points that point to a Unified ICME script.
Transfers from the agent desktop are made using the Dialed Number Plan.
There are two flags in Unified ICME to control the Network Transfer:
• NetworkTransferEnabled — This is a flag in the Unified ICME script. If enabled, it instructs the
Unified ICM to save the information about the initial routing client (the routing client that sent the
NewCall route request).
• NetworkTransferPreferred — This flag is checked on the Unified CVP PG configuration. If it is
checked, then any route request from this routing client (where Unified ICME knows about the
initial routing client) will send the route response to the initial routing client instead of the routing
client that sent the route request.
The following recommendations apply when using Network Transfer:
• Network Transfer using the two flags listed above can be used to perform a blind transfer only from
agent 1 to agent 2 via Unified CVP. In this case, Unified CVP will get instruction from
Unified ICME to pull the call back from agent 1 and route it either to a VoiceXML gateway (for IVR
treatment) or to another destination (to agent 2, for example).
• Network Transfer cannot be used to perform a warm transfer or conference with Unified CVP
because the call leg to agent 1 must be active while agent 1 performs a consultation or conference.
Unified CVP cannot pull the call back from agent 1 during the warm transfer and/or conference.
If a caller would like to dial the same number regardless of a blind transfer, warm transfer, or conference,
then the following recommendations and best practices can be used:
• Do not enable the NetworkTransferEnable flag in the Unified ICME script.
• Any transfer or conference request from an agent must dial the CTI Route Point of the same
Unified CCE PG to preserve the call context during the transfer. Dialing the Route Pattern or CTI
Route Point of another PG will not preserve the call context.
• Always use SendToVru as the first node in the Unified ICME routing script.
• In H.323-based deployments, there are two timers in Cisco Unified Communications Manager that
must be set to a value greater than the Unified CVP RONA timer. These timers are used to handle
the situation of consultation completion while agent 2’s phone is ringing. The timers are:
– Clusterwide Parameters (Service) ->Media Exchange Timer
– Clusterwide Parameters (Service) ->Media Resource Allocation Timer
• In H.323-based deployments, if you are using Cisco Unified CM 6.1.3 or earlier release, then you
have to uncheck the flag "Wait for Far End H.245 Terminal Capability Set" on the Unified CM
trunks configured with the UCCE PG Routing client label.
• Extra ports will be used during the consultation, blind transfer, and/or conference. They are released
only when the originating consultation is terminated.
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SIP Refer Transfer
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Chapter 10 Call Transfer Options
VoiceXML Transfers
VoiceXML Transfers
VoiceXML call control is supported only in standalone Unified CVP deployments (Deployment Model
#1) in which call control is provided by the Unified CVP VXML Server. Deployment Model #3b, which
also incorporates the Unified CVP VXML Server, does not support VoiceXML call control. In those and
all Unified ICM integrated deployments, Unified ICM must make all call control decisions.
The Unified CVP VXML Server can invoke three types of transfers: Release Trunk transfers, VoiceXML
blind transfers, and VoiceXML bridged transfers. Release Trunk transfers result in the incoming call
being released from the ingress voice gateway. VoiceXML blind transfers result in the call being bridged
to an egress voice gateway or a VoIP endpoint, but the Unified CVP VXML Server releases all
subsequent call control. VoiceXML bridged transfers result in the call being bridged to an egress voice
gateway or a VoIP endpoint, but the Unified CVP VXML Server retains call control so that it can return
a caller to an IVR application or transfer the caller to another termination point.
Release Trunk transfers from the Unified CVP VXML Server are invoked using the subdialog_return
element. The Unified CVP VXML Server can invoke a TNT transfer, TBCT transfer, and
HookFlash/Wink transfers as well as SIP Refer transfers. For TDM Release Trunk transfers (TNT, TBCT
and Hookflash/Wink), the VoiceXML gateway must be combined with the ingress gateway in order for
the Release Trunk transfer to work.
VoiceXML blind and bridged transfers are invoked using the Transfer element in Cisco Unified Call
Studio. VoiceXML Transfers will transfer the call to any dial-peer that is configured in the gateway.
VoiceXML Blind Transfers differ from VoiceXML Bridged Transfers in the following ways:
• VoiceXML blind transfers do not support call progress supervision, whereas Bridged transfers do.
This means that if a blind transfer fails, the Unified CVP VXML Server script does not recover
control and cannot attempt a different destination or take remedial action.
• VoiceXML blind transfers cause the Unified CVP VXML Server script to end. Always connect the
"done exit" branch from a Blind transfer node to a subdialog_return and a hang-up node.
Bridged transfers do not terminate the script. The Unified CVP VXML Server waits until either the
ingress or the destination call ends. The script ends only if the ingress call leg hangs up. If the destination
call leg hangs up first, the script recovers control and continues with additional self-service activity. Note
that the Unified CVP VXML Server port license remains in use for the duration of a bridged transfer,
even though the script is not actually performing any processing.
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CH A P T E R 11
Using the GKTMP NIC
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Chapter 11 Using the GKTMP NIC
Typical Applications of GKTMP with Unified CVP
To force the gatekeeper to reject the Admission Request (ARQ) and return an Admission Reject (ARJ)
to the requesting endpoint, the Unified ICM script can return a BUSY label, optionally with an additional
reason code (for example, DESTINATION_UNKNOWN).
For information on how to configure the GKTMP NIC, consult the ICM Setup and Installation Guide,
available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1001/tsd_products_support_series_home.ht
ml
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Typical Applications of GKTMP with Unified CVP
• Filtering calls that might be sent immediately to an available destination and bypass Unified CVP
While this approach might be seen as a way to avoid using Call Server resources, it limits the
functionality available. For example, there can be no intelligent re-query for alternative destinations
on ring-no-answer, nor will this approach allow the call to be taken back to Unified CVP for
subsequent call treatment and transfers.
• Pre-routing with context passing to Unified CVP
This method is used if call context collected during the pre-routing phase of the call needs to be
passed to Unified CVP rather than simply performing a standalone routing request via the GKTMP
NIC. In this example, the Call Server must be configured as a Type 2 VRU so that the call can be
translation-routed to it and still perform a subsequent transfer.
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Chapter 11 Using the GKTMP NIC
Typical Applications of GKTMP with Unified CVP
returning modified information and the selected target IP endpoints. In this case the gatekeeper
regards the request as completed, does no further processing of the request, and returns the ACF to
the endpoint that issued the ARQ. The Unified ICM routing script ends at this point.
6. The gatekeeper returns the selected IP address to the ingress gateway.
7. If the target is not a Unified CVP device, the ingress gateway sets up a VoIP call to that target.
8. If the target is a Unified CVP Call Server, the ingress gateway sets up a new call to it.
9. The Unified CVP Call Server sends a New Call message to Unified ICM.
10. Unified ICM starts an independent routing script to handle the incoming call.
11. Normal call flow continues. Transfer to VRU leg, Transfer to agent, as well as subsequent blind
Network VRU Transfers to secondary agents or return-to-queue are fully supported.
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Chapter 11 Using the GKTMP NIC
Typical Applications of GKTMP with Unified CVP
9. The Unified CVP Call Server performs and Empty Capability Set transfer, communicating with the
ingress gateway and the transfer destination endpoint to establish a VoIP call between them.
Deployment Implications
GKTMP is a simple request/response protocol. From the perspective of Unified ICM, this means that the
GKTMP NIC cannot perform any call control other than returning a single label and/or endpoint IP
address. Third-party call control through the GKTMP NIC is not possible once that single destination
has been returned. However, it is possible when call control responsibilities are handed off to Unified
CVP. The following discussion covers these call control implications for each of the three types of call
flows described in the previous section:
• Pre-routing of incoming calls, but call context passing is not required, page 11-5
• Pre-routing of incoming calls, and call context passing is required, page 11-5
• Routing of post-ICM calls, page 11-5
• Other implications, page 11-6
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Chapter 11 Using the GKTMP NIC
Typical Applications of GKTMP with Unified CVP
Other implications
Note that inserting the GKTMP NIC into the call flow does result in additional route requests being
processed by the Unified ICM Router for each call processed in this way, and additional call detail
records are written by the Unified ICM Logger. Where possible, configure the GKTMP server triggers
on the gatekeeper so that only those calls specifically requiring the additional routing functionality
afforded by the GKTMP NIC generate a Request ARQ to Unified ICM.
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CH A P T E R 12
Media File Options
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Chapter 12 Media File Options
Co-Resident Unified CVP Call Server, Media Server, and Unified CVP VXML Server
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Chapter 12 Media File Options
Bandwidth Calculation for Prompt Retrieval
should be used as indicated below whenever you want to use a Micro-App for prompting. If you
subsequently want to use the VXML Server, you will have to rewrite this variable by following the
instructions above.
1. When setting up the media_server ECC variable that specifies your Media server in the ICM script,
use the Formula Editor to set the media_server ECC variable to
concatenate("http://",Call.RoutingClient), where Call.RoutingClient is the built-in call variable
that ICM sets automatically for you. The routing client name in ICM is not necessarily the same as
the Unified CVP Server's hostname (and usually is not the same).
2. You can then use the name of the routing client as a hostname in the VXML gateway. However, do
not use non-compliant characters such as an underscore as part of the hostname because the router
cannot translate the hostname to an IP address if it contains any non-complaint characters. Cisco
also recommends using the ip hostname strict command in the router to prevent the use of invalid
characters in the hostname. This will ensure that the hostname is acceptable to Unified CVP.
3. Configure the routing client hostname for every Unified CVP Server Routing Client.
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Chapter 12 Media File Options
Configuring Caching and Streaming in Cisco IOS
interactions from fetching the media file in chunks, it deteriorates performance. Additionally, the ability
to cache the files in memory reduces the advantage of streaming large files directly from the HTTP
server.
The recommendation for a Unified CVP VoiceXML gateway is to use non-streaming mode for the
prompts in combination with caching. The Cisco IOS command to configure non-streaming mode is:
ivr prompt streamed none
Caching
There are two types of cache involved in storing media files: the IVR Media Player cache and the HTTP
Client cache. The HTTP Client cache is used for storing files that are downloaded from the HTTP server.
In non-streaming mode, the entire media file is stored inside the HTTP Client cache. In streaming mode,
the first chunk of the media file is stored in the HTTP Client cache and in the IVR cache, and all
subsequent chunks of the file are saved in the IVR cache only.
Because of the above recommendation to use only non-streaming mode, the IVR prompt cache is never
used and the HTTP Client cache is the primary cache. The HTTP Client cache also has the advantage of
being able to store 100 MB of prompts, whereas the IVR cache is limited to 16 MB.
To configure the HTTP Client cache, use the following IOS commands:
http client cache memory file <1-10000>
Where <1-10000> is the file size in kilobytes. The default maximum file size is 50 kB, but the
recommended file size is 600 kB. Any file that is larger than the configured HTTP Client
memory file size will not be cached.
http client cache memory pool <0-100000>
Where <0-100000> is the total memory size available for all prompts, expressed in kilobytes.
A value of zero disables HTTP caching. The default memory pool size for the HTTP Client
cache is 10 Mb. The recommended memory pool size is the total size of all prompts stored on
the media server, up to 100 MB.
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Chapter 12 Media File Options
Configuring Caching and Streaming in Cisco IOS
connections have the same host IP address and port number. This kind of connection is referred to as a
persistent connection. As the name implies, the connection can last over a long period of time without
being shut down.
To establish a persistent connection, both the client and the server must agree that the connection is going
to be a persistent one. To configure the Cisco IOS HTTP Client to request a persistent connection from
the server, configure the following command:
http client connection persistent
Cache Aging
The HTTP Client manages its cache by the "freshness" of each cached entry. Whether a cached entry is
fresh or stale depends on two numbers: Age and FreshTime. Age is the elapsed time since the file was
last downloaded from the server. FreshTime is the duration that the file is expected to stay fresh in the
HTTP Client cache since the file was last downloaded.
There are several variables that can affect the FreshTime of a file, such as HTTP message headers from
the server and the cache refresh value configured via the command line interface (CLI).
The FreshTime of a file is determined in the following sequence:
1. When a file is downloaded from the HTTP server, if one of the HTTP message headers contains the
following:
Cache-Control: max-age = <value in seconds>
Then the max-age is used as the FreshTime for this file.
2. If step 1 does not apply, but the following two headers are included in the HTTP message:
Expires: <expiration date time>
Date: <Current date time>
Then the difference (Expires – Date) is used as the FreshTime for this file.
3. The HTTP/1.1 spec, RFC 2616 (HyperText Transport Protocol), recommends that either one of the
HTTP message headers as described in step 1 or 2 above should be present. If the server fails to send
both 1 and 2 in its HTTP response, then take 10% of the difference between Date and Last-Modified
from the following message headers:
Last-Modified: <last-modified date time>
Date: <Current date time>
So the FreshTime for this file is calculated as:
FreshTime = 10% ∗ ((Date) – (Last-Modified))
4. The CLI allows the user to assign a FreshTime value to the files as a provisional value in case none
of the message headers in steps 1 to 3 are present:
http client cache refresh <1-864000>
The default refresh value is 86400 seconds (24 hours). The configured HTTP Client cache refresh
has no effect on files when any of the message headers in steps 1 to 3 are present. But if the resultant
FreshTime from the CLI command calculation turns out to be less than the system default (which is
86400 seconds), the FreshTime will be set to the default value (86400 seconds). This command is
also not retroactive. That is, the newly configured refresh value applies only to new incoming files,
and it has no effect on the entries already in the cache.
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Chapter 12 Media File Options
Branch Office Implications
Stale files are refreshed on an as-needed basis only. This means that a stale cached entry can stay in the
cache for a long time until it is removed to make room for either a fresh copy of the same file or another
file that needs its memory space in the cache.
A stale cached entry is removed on an as-needed basis when all of the following conditions are true:
• The cached entry becomes stale.
• Its refresh count is zero (0); that is, the cached entry is not being used.
• Its memory space is needed to make room for other entries.
When the Age exceeds the FreshTime and the file needs to be played, the HTTP Client will check with
the media server to determine whether or not the file has been updated. When the HTTP Client issues a
GET request to the server, it uses a conditional GET to minimize its impact on network traffic. The GET
request includes an If-Modified-Since in the headers sent to the server. With this header, the server will
either reply with a 304 response code (Not Modified) or return the entire file if the file was indeed
updated recently.
Note that this conditional GET applies only to non-streaming mode. Under streaming mode, the HTTP
Client always issues an unconditional GET; that is, no If-Modified-Since header is included in the GET
request, thus resulting in an unconditional reload for each GET in streaming mode.
You can reload individual files into cache by issuing the following command:
test http client get http://10.0.0.130/en-us/sys/1.wav reload
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CH A P T E R 13
Managing, Monitoring, and Reporting
This chapter discusses various types of managing, monitoring, and reporting functions that can be used
with Unified CVP. It covers the following areas:
• What's New in This Chapter, page 13-1
• End-to-End Tracking of Individual Calls: Log Files, page 13-3
• Formal Reporting, page 13-3
• Unified System CLI and Web Services Manager (WSM), page 13-6
Table 13-1 New or Changed Information Since the Previous Release of This Document
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Chapter 13 Managing, Monitoring, and Reporting
DS0 Trunk Information for Reporting
You can manage the following Unified CVP components directly from the Operations Console:
• Unified CVP Call Server
• Unified CVP VXML Server
• Unified CVP Reporting Server
The Operations Console provides web-based interfaces for mapping and summarizing the solution
network configuration, setting and displaying configuration information on a batch or per-node basis,
and storing local copies of these configurations. The Operations Console also provides the ability to
distribute Cisco Unified Call Studio applications to Unified CVP VXML Servers. Finally, the
Operations Console provides basic visual indications as to which managed components are functioning
properly and which are having problems.
The Operations Console provides access to the following operations:
• Troubleshooting
Use the Operations Console to access the Support Tools interface, which provides the ability to
retrieve and process trace logs from most components, plus the ability to set or reset trace levels on
these components.
• Health Monitoring
You can use any SNMP-standard monitoring tool to get a detailed visual and tabular representation
of the health of the solution network. All Unified CVP product components and most Unified CVP
solution components also issue SNMP traps and statistics that can be delivered to any standard
SNMP management station or monitoring tool.
• Statistical Monitoring
Unified CVP infrastructure statistics include real-time and interval data on the Java Virtual Machine
(JVM), threading, and licensing. You can access these statistics by selecting the Control Center from
the System menu and then selecting a device. SNMP statistics can also be used.
• Direct administration of individual Cisco IOS-based components
Administrators can select an individual gateway, gatekeeper, or Content Services Switch for direct
administration. Secure Shell (SSH) is used for the gateway and gatekeeper, while Telnet is used for
the Content Services Switch (CSS).
Note Internally, the Operations Console is occasionally referred to as the OAMP (Operate, Administer,
Maintain, Provision). The Operations Console manages individual components through the Unified CVP
Resource Manager, which is co-located with each managed Unified CVP component. The Resource
Manager is invisible to the end-user.
For more information on the Operations Console, see the Operations Console online help.
For information about the many new features for the Operations Console, refer to the new Operations
Console Guide for Cisco Unified Customer Voice Portal, available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_user_guide_list.html
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Chapter 13 Managing, Monitoring, and Reporting
End-to-End Tracking of Individual Calls: Log Files
Refer to the topic DS0 Trunk Information, page 5-15 and the topic Trunk Utilization Routing and
Reporting, page 5-16.
Note Although Unified CVP components do not themselves synchronize machine times, customers must
provide a cross-component time synchronization mechanism, such as NTP, in order to assure accurate
time stamps for logging and reporting.
Formal Reporting
The Unified CVP Reporting Server houses the Reporting Service and hosts an IBM Informix Dynamic
Server (IDS) database management system.
The Reporting Service provides historical reporting to a distributed self-service deployment in a call
center environment. The system is used to assist call center managers with call activity summary
information to manage daily operations. It can also provide operational analysis of various IVR
applications.
The Reporting Service receives reporting data from the IVR Service, the SIP Service (if used), and the
Unified CVP VXML Server. (To capture the data from the Unified CVP VXML Server in the
Unified CVP Reporting Server's database, the Unified CVP VXML Server should be added by using the
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Chapter 13 Managing, Monitoring, and Reporting
Formal Reporting
CVP VXML Server device in the Unified CVP Operations Console Server (Operations Console).
Selecting the VXML Server Standalone device option will not capture the Unified CVP Reporting
data.) As stated, the Reporting Service is deployed together with an Informix database management
system, and it transforms and writes this reporting data into that database. The database schema is
prescribed by the Unified CVP product, but the schema is fully published so that customers can develop
custom reports based on it.
The Reporting Service does not itself perform database administrative and maintenance activities such
as backups or purges. However, Unified CVP provides access to such maintenance tasks through the
Operations Console.
A single Reporting Server may be used in a deployment. If a single Reporting Server is used, it does not
necessary represent a single point of failure, because data safety and security are provided by the
database management system, and temporary outages are tolerated due to persistent buffering of
information on the source components.
If more than one Reporting Server is used, be aware of the following restrictions:
• Each Unified CVP Call Server can be associated with only one Unified CVP Reporting Server.
• Reports cannot span multiple Informix databases.
Note Although Unified CVP components do not themselves synchronize machine times, customers must
provide a cross-component time synchronization mechanism, such as NTP, in order to assure accurate
time stamps for logging and reporting.
Note For reporting requirements related to the Courtesy Callback feature, refer to Courtesy Callback,
page 5-21.
The following is the list of features introduced in Unified CVP Release 8.0(1) for the Unified CVP
Reporting server (Reporting Server).
1. The Reporting server enables you to integrate with Cisco Unified Intelligence Center (Unified IC),
enabling you to run user friendly custom reports in the Unified IC environment. Unified IC
templates are shipped with all Unified CVP installations. These templates provide examples to
report against Call, Application, Callback, and Trunk Grout Utilization structures.
2. The Reporting Server provides increased data retention times by increasing the database space
requirements
3. All database backup files are compressed and stored on the Reporting Server. The backup file is
called cvp_backup_data.gz and is stored on the %INFORMIXBACKUP% drive in a folder named
cvp_db_backup.
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Chapter 13 Managing, Monitoring, and Reporting
Formal Reporting
4. Using the new System CLI you can make the request to list log files on the Reporting Server (show
log). This request includes the Informix Database Server Engine logs. The show tech-support
command also includes these files.
5. Debug can now be turned on (or off) from within the System CLI with the debug level 3 (or 0)
command. When on, this command generates trace files for all administrative procedures, Purge,
Statistics and Aggregator. Care should be taken when turning this on because the trace files place an
elevated burden on the database.
6. Log data for administrative procedures are now written on a nightly basis to the
%CVP_HOME%\logs folder.
7. All StartDateTime, EndDateTime and EventDateTime values are stored as UTC in the various
Reporting Server tables.
8. The Reporting Server supports the Analysis Manager tool by allowing Analysis Manager to query
the Reporting Server as long as the user is authenticated. This user would typically be the
cvp_dbuser login.
9. Transfer Type data and Transfer Labels for SIP and H.323 call events are now stored in the call event
table.
10. There is a data aggregator, which aggregates Unified CVP data in fifteen minute increments. Cisco
Unified Intelligence Center templates are created to capture this information. Call data is
summarized at 15 minute, daily, and weekly intervals. Dominant Path information is summarized at
the same intervals. These summaries are stored in the call_15, call_daily, call_weekly,
applicationsummary_15, applicationsummary_daily, and applicationsummary_weekly tables. Call
data is summarized into the Call_* structure, while an aggregate of each element invoked by each
application is stored in the ApplicationSummary_* structures.
11. To perform a post installation or upgrade of the Reporting Server, you only need to run a single file
(%CVP_HOME%\bin\CVP_Database_Config.bat). Log in as Informix user and run this file. This
file is a replacement for the two files used in the past. (ReportingRunAsInformix.bat and
ReportingRunAsCVP_DbAdmin.bat). This script is run after an installation or an upgrade and is
designed to upgrade from CVP 4.x or CVP 7.x.
12. Summary purge results are now logged in the log table.
13. Three new scheduled tasks have been added or the Reporting Server scheduler.
– CVPSummary, which builds summary tables
– CVPCallArchive, which archives Callback data to maintain callback database performance.
– CVPLogDump, which extracts the administrative logs on a nightly basis.
14. All metadata for administrative processes has been moved into a new Ciscoadmin database. This
removes the tables from normal view of reporting users.
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Chapter 13 Managing, Monitoring, and Reporting
Unified System CLI and Web Services Manager (WSM)
• Reporting Guide for Cisco Unified ICM Enterprise & Hosted available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps4145/products_user_guide_list.html
More Information
For more information on Unified CVP reporting, see the Reporting Guide for Cisco Unified Customer
Voice Portal, available at
www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configuration_gu
ides_list
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Unified System CLI and Web Services Manager (WSM)
– Licenses are only valid if the new license feature, CVP_SOFTWARE, is added. This new feature
will be used to ensure that customers have the right to run the current version of CVP
4. Serviceability Across Products.
– Enhanced Log & Trace messages
The CVP WebService Manager (WSM) is a new component that is installed automatically on all Unified
CVP Servers, including Remote Operations Manager (ROM) only installations. WSM interacts with
various subsystems and infrastructure handlers, consolidates the response, and publishes an xml
response. WSM supports secure authentication and data encryption on each of the interfaces.
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Unified System CLI and Web Services Manager (WSM)
Note For a discussion of the Analysis Manager, and a related discussion of the Analysis Call Path tool, refer
to: Cisco Unified Analysis Manager available at:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/service/8_0_1/rtmt/ch1_overview.html
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Unified System CLI and Web Services Manager (WSM)
In addition to the interactive user interface, the Unified System CLI can be used as a batch command.
This feature allows the System CLI to be used in scheduled jobs.
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Unified System CLI and Web Services Manager (WSM)
Note For detailed information on using the Unified System CLI, refer to: Unified System CLI in:
Configuration and Administration Guide for Cisco Unified Customer Voice Portal, available at:
http://www.cisco.com/en/US/products/sw/custcosw/ps1006/products_installation_and_configuration_g
uides_list.html.
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CH A P T E R 14
Sizing
This chapter discusses how to determine how many physical machines to order and, in the case of
gateways and gatekeepers, what kind to order.
This chapter covers the following topics:
• Sizing Overview, page 14-2
• Unified CVP Call Server, page 14-3
• Unified CVP VXML Server (VXML Server), page 14-4
• Unified CVP Co-Residency, page 14-5
• Unified Presence Server, page 14-7
• Unified CVP Video Service, page 14-7
• Unified CVP Reporting Server, page 14-8
Table 14-1 New or Changed Information Since the Previous Release of This Document
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Chapter 14 Sizing
Sizing Overview
Sizing Overview
When sizing a contact center, first determine the worst-case contact center profile in terms of the number
of calls that are in each state. In other words, if you were to observe the contact center at its busiest
instant in the busiest hour, how many calls would you find are in each of the following states:
• Self-service — Calls that are executing applications using the Unified CVP VXML Server
• Queue and collect — Calls that are in queue for an agent or that are executing prompt-and-collect
type self-service applications
• Talking — Calls that are connected to agents or to third-party TDM VRU applications
In counting the number of calls that are in the talking state, count only calls that are using Unified CVP
or gateway resources. To determine whether a talking call is using resources, you must consider how the
call gets transferred to that VRU or agent. If the call was transferred via VoIP, it continues to use an
ingress gateway port and it continues to use a Unified CVP resource because Unified CVP continues to
monitor the call and provides the ability to retrieve it and re-deliver it at a later time. The same is true of
calls that are tromboned to a TDM target, using both an incoming and an outgoing TDM port on the same
gateway or on a different gateway (that is, toll bypass). Calls that are transferred to VRUs or agents in
this manner should be counted as talking calls.
However, if the call was transferred via *8 TNT, hookflash, Two B Channel Transfer (TBCT), or an ICM
NIC, neither the gateway nor Unified CVP play any role in the call. Both components have reclaimed
their resources, therefore such calls should not be counted as talking calls.
Finally, include in the overall call counts those calls that have been transferred back into Unified CVP
for queuing or self-service, via either blind or warm methods. For instance, if a warm transfer is used
and the agent is queued at Unified CVP during the post-route phase, the call would use two ports due to
two separate call control sessions at Unified CVP. Because these calls usually do not amount to more
than 5% or 10% of the overall call volume, it is easy to overlook them.
The definitions of these call states differ somewhat from the definitions used for port licensing purposes
(see Licensing, page 15-1). The use of automatic speech recognition (ASR) or text-to-speech (TTS) has
nothing to do with delineating which calls are in which state, whereas it does for licensing purposes.
Similarly, the call state determination has nothing to do with whether the agents are Unified CCE agents
or ACD agents, nor does it matter whether the customer intends to use Unified CVP's ability to retrieve
and re-deliver the call to another agent or back into self-service.
Note The solution must be sized for the number of ports in use for calls in a talking state to agents. Even
though licenses for those ports do not have to be purchased when using Unified CCE agents, TDM agents
do require a Call Director license.
In addition to the overall snapshot profile of calls in the contact center, you must also consider the busiest
period call arrival rate in terms of calls per second. You will need this information for the contact center
as a whole. Because it is hard to identify a true maximum arrival rate, you can use statistical means to
arrive at this number. Except in fairly small implementations, this is seldom the critical factor in
determining sizing.
With the above data, you can begin sizing each component in the network. This section next considers
the Unified CVP products: Unified CVP Call Server and Unified CVP VXML Server followed by the
Unified Presence Server and Unified CVP Reporting Server. This section deals entirely with the number
and type of physical components required to support the Unified CVP system, but it does not include
any discussion of redundancy. For an understanding of how to extend these numbers to support higher
reliability, see Designing Unified CVP for High Availability, page 4-1.
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Chapter 14 Sizing
Unified CVP Call Server
Unified CVP Call Servers are sized according to the number of calls they can handle, in addition to their
maximum call arrival rate.
Table 14-2 Call Server Call Rate by Server Model Number
Note The following Example Call Server call rate calculations pertain to the MCS-7845-I3-CCE2 server.
Each Call Server can handle 1200 SIP calls or 500 H.323 calls. Each Call Server is further limited to a
sustained call arrival rate of 14 call per second (cps) for SIP and 7 cps for H.323. However, Model #4 is
exempt from this limitation because the Call Server in that model does not perform any H.323 or SIP
processing.
Specifically, the number of Call Servers required is the larger of:
((Self Service) + (Queue and Collect) + Talking) / 1200 [or 500 for H.323], rounded up
or
(Average call arrival rate) / 14, rounded up [7 for H.323; except in Model #4]
If you have a Call Server servicing both SIP and H.323 calls, you can size the server by prorating the
performance used for each call type. For example, if 60% of the calls will be H.323 and 40% SIP, the
maximum load in terms of active calls in this case will be:
60% ∗ (500 H.323 calls) + 40% ∗ (1200 SIP calls) = 780 active calls
In addition, calls delivered to the Cisco Unified Communications Manager cluster should be
load-balanced among the subscribers in the cluster and should not exceed 2 calls per second (cps) per
subscriber.
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Chapter 14 Sizing
Unified CVP VXML Server (VXML Server)
Note The following VXML Server example call rate calculations pertain to the MCS-7845-I3-CCE2 server.
Unified CVP VXML Server sizing with HTTP is simple: one Unified CVP VXML Server can handle up
to 1200 calls. If you are using Unified CVP VXML Servers, you should size those machines according
to the following formula:
Calls / 1200, rounded up,
where Calls refers to the number of calls that are actually in Unified CVP VXML Server self-service
applications at that busy moment snapshot in time.
Unified CVP can also be configured to use HTTPS on the Unified VXML Server and on the Unified CVP
IVR Service. (IVR Service can generate very basic VoiceXML documents and is part of the
Unified CVP Call Server.) Due to the large processing overhead of HTTPS, the Tomcat application
server can achieve a maximum of only 100 simultaneous connections, depending on the configuration.
Table 14-4 provides simultaneous call information for HTTPS calls using various applications and call
flow models.
Unified CVP Call Server Type, Application, and Call Flow Model: Model:
Model MCS-7845-I3-CCE2 USC-C210-M1 or
Later
Unified CVP VXML Server 250 ———
Max Simultaneous HTTPS Connections with
WebSphere (Standalone Call Flow Model)
Unified CVP VXML Server 100 ———
Max Simultaneous HTTPS Connections with Tomcat
(Standalone Call Flow Model)
Unified CVP Call Server and VXML Server 100 ———
Max Simultaneous HTTPS Connections with
WebSphere (Comprehensive Call Flow Model)
Unified CVP Call Server and VXML Server 100 ———
Max Simultaneous HTTPS Connections with
Tomcat (Comprehensive Call Flow Model
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Chapter 14 Sizing
Unified CVP Co-Residency
In all of the above scenarios, the Reporting and Datafeed options were disabled. Also note that:
• Cisco IOS Release 12.4(15)T5 or later release is required on the gateway to support the HTTPS
option. (Mainline Cisco IOS currently is not supported.)
• The HTTPS option has not been tested with co-resident Unified CVP Call Server and Unified CVP
VXML Server and hence is not a supported configuration.
Cisco recommends the following configuration on the Cisco IOS VoiceXML Gateway with HTTPS
option. Not having this configuration setting can severely impact the performance and sizing of the
VXML gateway and the overall solution in general with HTTPS.
http client connection persistence
http client cache memory pool 15000
http client cache memory file 1000
The following components can be installed on the same physical server (co-resident):
• Unified CVP Call Server (Call Server)
• Unified CVP VXML Server (VXML Server)
• Media Server
A SIP-based co-resident server can handle 1200 SIP calls as well as 1200 VXML Server sessions
simultaneously, and it can handle a sustained call arrival rate of 14 calls per second.
Note This means you can run 1200 ports of Call Server doing SIP call control, and 1200 ports of VXML Server
on one server with a 1200 port license.
An H.323 co-resident server can handle 500 H.323 calls as well as 500 VXML Server sessions
simultaneously, and it can handle a sustained call arrival rate of 6 calls per second.
Note This means you can run 500 ports of Call Server doing H.323 call control, and 500 ports of VXML
Server on one server with a 500 port license.
The number of Unified CVP Call Servers required is the larger of:
((Self Service) + (Queue and Collect) + Talking) / 1200 [or 500 for H.323], rounded up,
or
(Average call arrival rate) / 14 [or 6 for H.323], rounded up, except in the VRU-only model
The co-resident media server can be used for up to 1200 calls [or 500 for H.323], assuming that prompt
caching is enabled in the VoiceXML gateways. If multiple co-resident servers are to be used, you must
load-balance across the co-resident media servers in order to spread the load of the calls across all of the
servers. To reduce the administrative overhead of managing content on multiple media servers, separate
dedicated media servers can be used.
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Chapter 14 Sizing
Unified CVP Co-Residency
This means you can run 1200 ports of the Call Server with SIP call control, and 1200 ports of the
VXML Server, all on one server with 1200 port licenses. An H.323 co-resident server can handle 500
H.323 calls as well as 500 VXML Server sessions simultaneously, and it can handle a sustained call
arrival rate of 6 calls per second. This means you can run 500 ports of the Call Server with H.323 call
control, and 500 ports of the VXML Server, all on one server with 500 licensed ports.
Example
For example, assume that your deployment must be sized for 1200 self-service ports, 500 queue and
collect ports, and 3700 simultaneous calls to agents.
Note In the above example definition, self-service means that a call requires SIP or H.323 call control and
runs an application on the VXML Server. Queue and collect means that a call requires SIP or H.323 call
control and runs an application using Microapps only on the Call Server.
The following example applies for VXML and HTTP sessions only. The same values apply to both
co-resident and distributed deployments of Call Servers and VXML Servers.
The number of servers required using SIP call control would be as follows:
((Self Service) + (Queue and Collect) + Talking) / 1200 [or 500 for H.323], rounded up
((1200) + (500) + 3700) / 1200 = 5 servers
If you use the Cisco Unified Border Element as a Session Border Controller (SBC) for flow-through calls
to handle VXML requirements, then you must use the sizing information presented above. The Cisco
Unified Border Element is limited to the maximum number of simultaneous VXML sessions or calls as
outlined above for the particular situation and hardware platform.
If you use the Cisco Unified Border Element as an SBC to handle flow-through calls only (no VXML),
then take Voice Activity Detection (VAD) into consideration and refer to the sizing information in the
Cisco Unified Border Element Ordering Guide, available at
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/order_guide_c07_4
62222.html
Co-Resident Unified CVP Reporting Server and Unified CVP Call Server
The Unified CVP Reporting Server can also be co-resident with the Unified CVP Call Server, but only
for Standalone VoiceXML deployments. The Call Server is normally not needed in a Standalone
VoiceXML deployment; but if reporting is desired, a Call Server is required in order to send the reporting
data from the VXML Server to the Reporting Server. Thus, when the Reporting Server is co-resident
with a Call Server, the Call Server is not processing any SIP or H.323 calls but is simply relaying
reporting data from the VXML Server.
The co-resident Call Server does not have a significant impact on performance in this model, therefore
the sizing information in the section on the Unified CVP Reporting Server, page 14-8, does not change.
Note If CUBE is to be used as an SBC to handle flow-thru or flow-around calls only (no vxml), with VAD in
consideration, we can use the CUBE Ordering Guide for sizing.
If CUBE is to be used as an SBC for flow-thru or flow-around calls AND is to handle vxml requirements,
we must use the sizing information in the CVP SRND. CUBE will be limited to the maximum number
of simultaneous vxml sessions\calls as outlined there for the particular situation and hardware platform.
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Chapter 14 Sizing
Unified Presence Server
Table 14-5 Call Handling Capacities for Cisco Unified Presence Servers
The sizing numbers in Table 14-5 assume a dedicated SIP Proxy. If you are using the same Cisco Unified
Presence server for presence services, you should adjust the numbers accordingly, based on the capacity
used. For instance, if the Cisco Unified Presence server supports 2500 presence users but you will
actually have only 1250 (50%), then that leaves approximately 50% of the SIP Proxy capacity.
The capacities in Table 14-5 are measured in calls per second (cps). However, one call coming in from
the PSTN is not equivalent to one call through Cisco Unified Presence. Multiple calls are actually
generated per inbound customer call for queuing, ringback, and subsequent agent transfers. A typical
incoming call will be transferred by Unified CVP four times, so the inbound PSTN call rate should be
multiplied by 4.
For example:
If Unified CVP receives 20 PSTN calls per second, Cisco Unified Presence will see about 80 calls
per second.
Table #2 in the CUSP public data sheet “Performance Measured in the Number of New Call Attempts
per Second" shows additional performance data for the CUSP server.
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Unified CVP Reporting Server
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Chapter 14 Sizing
Unified CVP Reporting Server
Next, estimate the total number of reporting messages that your deployment will generate per second by
summing the values obtained from the previous calculation for each application:
A(total) = A1+ A2+…..+An
This is the total number of reporting messages generated per second by your VoiceXML applications.
The Cisco MCS-7845 Reporting Servers can handle 420 messages per second. If the total number of
reporting messages per second for your deployment is less than 420, you can use a single Reporting
Server. If it is greater, you need to use multiple Reporting Servers and partition the VoiceXML
applications to use specific Reporting Servers.
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Unified CVP Reporting Server
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Chapter 14 Sizing
Unified CVP Reporting Server
Example Applications
This section presents some examples of applications that can be used to estimate the number of reporting
messages that will be generated by your particular application.
Low Complexity
Total: 16 reporting messages per minute per call.
Approximate Number of
Element Type Reporting Messages
Start 2
Subdialog_start 2
Play element 2
Play element 2
Play element 2
Play element 2
Subdialog_end 2
End 2
Approximate Number of
Element Type Reporting Messages
Start 2
Subdialog_strart 2
Play element 2
Get input 5
Play element 2
Get input 5
Form 10
Input 5
Transfer with audio 2
Subdialog_end 2
End 2
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Chapter 14 Sizing
Unified CVP Reporting Server
Approximate Number of
Element Type Reporting Messages
Start 2
Subdialog_strart 2
Play element 2
Get input 9
Play element 2
Get input 9
Form 10
Input 9
Transfer with audio 2
Subdialog_end 2
End 2
Approximate Number of
Element Type Reporting Messages
Start 2
Subdialog_strart 2
Icmrequrestlabel 30
Form 10
ASR capture 9
Digit with confirm 20
Form 10
Digit with confirm 20
Subdialog_end 2
End 2
Cisco Unified Customer Voice Portal (CVP) 8.x Solution Reference Network Design (SRND)
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CH A P T E R 15
Licensing
The licensing information for Cisco Unified CVP has been consolidated and moved into the Cisco
Customer Contact Solutions Ordering Guide. The Ordering Guide provides a single, frequently updated
source for all the Unified CVP licensing information. Cisco employees and Partners with a valid login
account can access the Ordering Guide at:
http://www.cisco.com/web/partners/downloads/partner/WWChannels/technology/ipc/downloads/C
CBU_ordering_guide.pdf
If you need licensing information for Unified CVP but you cannot access the Ordering Guide, contact
your local Cisco Systems Engineer (SE) or Partner.
Cisco Unified Customer Voice Portal (CVP) 8.x Solution Reference Network Design (SRND)
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Chapter 15 Licensing
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INDEX
A C
ACD 5-13 cache aging 12-5
admission control for calls 3-6 caching
aging cache 12-5 prompts 12-3, 12-4
alternate query URLs 12-4
endpoints 4-27 call admission control 9-9
gatekeeper 4-17, 4-18 Call Director
Application Content Engine (ACE) call flow model described 1-20
discussed 1-15 deployment model 2-4, 5-8, 6-4
migrate from CSS to ACE 1-15 calls
minimum license information 1-15 admission control 3-6
application examples 14-11 control of 2-3, 2-6, 2-10
architecture 1-1 control traffic 9-3
ASR 1-16, 4-32, 7-2, 7-3, 9-7 disposition of 4-5, 4-11, 4-13, 4-19, 4-21, 4-22, 4-27, 4-29,
automatic call distributor (ACD) 5-13 4-30, 4-32, 4-33, 4-34, 4-35
Automatic Speech Recognition (ASR) 1-16, 4-32, 7-2, 7-3, failures 4-19
9-7 flows 1-17, 2-2, 2-4, 2-5, 2-8, 2-9, 2-12, 3-7, 3-10, 6-2, 6-3, 11-3
help desk 6-2
initial treatment 2-8, 2-9, 2-12
B
in progress 4-12, 4-20
backup and restore 13-6 log files 13-3
bandwidth maximum number 9-8
for retrieving prompts 12-3 originated by Cisco Unified CM 5-13, 6-1
provisioning 9-2, 9-5 outbound 6-2
Basic Video Service 2-13, 14-8 post-ICM 11-4, 11-5
blind transfer 6-3 pre-routing 11-3, 11-4, 11-5
blocking media streams 9-14 queue and collect 14-2
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Index
Cisco Unified Customer Voice Portal (CVP) 8.x Solution Reference Network Design (SRND)
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Index
reporting 13-1
E
traffic 9-5
deployment models Egress Gateway 1-9
Call Director 2-4 enterprise domain, CVP part of 1-5
Comprehensive models 2-6 example applications 14-11
distributed models 3-1
functional models 2-1
F
hosted implementations 5-10, 6-7
Model #1 - Standalone Self-Service 5-7, 6-3 firewalls 9-15
Model #2 - Call Director 5-8, 6-4 flow of calls 6-2, 6-3, 11-3
Model #3a - Comprehensive Using ICM formal reporting 13-3
Micro-Apps 5-8, 6-5
functional deployment models 1-20, 2-1
Model #3b - Comprehensive Using Unified CVP
VXML Server 5-8, 6-6
Model #4a - VRU Only with NIC Controlled G
Routing 5-8
Model #4b - VRU Only with NIC Controlled G.711 9-14
Pre-Routing 5-9 G.711 and G.729 support 9-9
Model #4 - VRU Only 5-8
gatekeeper
Network VRU types 5-6
alternate 4-17, 4-18
standalone self-service 4-31, 4-32
call admission control 3-6
types and their uses, summarized 1-20
call routing 3-10
Unified CVP VXML Server (Standalone) 2-2
configuration 4-18, 6-7
VRU only 2-11
described 1-11
design process H.323 3-10
overall steps 1-19
high availability 4-16
SIP protocol recommended 1-19
HSRP 4-16, 4-18
dial peers 3-10
redundancy 4-16
dial plan 6-7
required for all H.323 installations 1-11
disposition of calls 4-5, 4-11, 4-13, 4-19, 4-21, 4-22, 4-27, 4-29,
Gatekeeper Transaction Message Protocol
4-30, 4-32, 4-33, 4-34, 4-35
(GKTMP) 11-1
distributed gateways
deployments 3-1
at a branch office 3-1
gateways 3-1
centralized 4-23
network options 1-22
Cisco IOS 4-10, 4-21
VoiceXML gateways 4-24, 4-25
co-located with VXML Servers 3-3
DNS Server 1-13
distributed 3-1, 4-24, 4-25
domain, CVP part of 1-5
maximum VoiceXML sessions 7-7, 7-8
DTMF 7-2, 7-3
MGCP 7-10
originating calls 4-4
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Index
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Index
OAMP 13-2
R
OAMP Resource Manager (ORM) 13-2
Operate, Administer, Maintain, Provision (OAMP) 13-2 RAID 13-6
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Index
S
T
SBC 7-4
Takeback-and-Transfer (TNT) 10-2
scalability options 1-24
TBCT 10-4, 14-2
scripting 4-31
TCP socket persistence 12-4
security
TDM interface 7-3
on the network 9-15
Telecom Italia Mobile (TIM) 5-8
security agent for CVP 1-13
Text-to-Speech (TTS) 1-16, 4-32, 7-2, 7-3, 9-7
self-service
third-party
calls 2-8, 2-9, 2-12, 14-2
media server 1-15
deployment model 4-31, 4-32
VRUs 5-15
separate ingress gateway and VoiceXML 4-25
TIM 5-8
server group elements, proxy server 1-11
TNT 10-2
servers
traffic
Cisco Unified Presence 14-7
marking 9-13
co-resident 14-5
voice 9-2, 9-9
multiple 14-9
transfers
reporting 14-8
blind 6-3
sizing 6-8
call transfer options 10-1
VoiceXML 8-1
consultative 6-3
session border controller (SBC) 7-4
in Call Director deployments 2-6
SIP
in Comprehensive deployments 2-10
call flow 1-17, 2-4, 2-8, 3-10
in standalone VoiceXML deployments 2-3
call transfers 10-7
to live agent 2-8, 2-10, 2-13
dial plan 6-7
VoIP-based 2-5, 2-6
Proxy Server 1-11, 4-5, 4-12
warm 6-3
signaling 9-7
Translation Route ID 5-3, 5-6
SIP Service 4-11
troubleshooting 13-2
SIP protocol
TTS 1-16, 4-32, 7-2, 7-3, 9-7
recommended for deployments 1-19
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Index
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