Planning Guide for Cisco Unified Customer Voice Portal Cisco Unified Customer Voice Portal 4.1(1)

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1 Planning Guide for Cisco Unified Customer Voice Portal Cisco Unified Customer Voice Portal 4.1(1) November 2007 Americas Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA USA Tel: NETS (6387) Fax:

2 THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS. THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET FORTH IN THE INFORMATION PACKET THAT SHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSE OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY. The Cisco implementation of TCP header compression is an adaptation of a program developed by the University of California, Berkeley (UCB) as part of UCBs public domain version of the UNI operating system. All rights reserved. Copyright  1981, Regents of the University of California. NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE PROVIDED "AS IS" WITH ALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EPRESSED OR IMPLIED, INCLUDING, WITHOUT LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OF DEALING, USAGE, OR TRADE PRACTICE. IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. CCVP, the Cisco logo, and Welcome to the Human Network are trademarks of Cisco Systems, Inc.; Changing the Way We Work, Live, Play, and Learn is a service mark of Cisco Systems, Inc.; and Access Registrar, Aironet, BP, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, Cisco, the Cisco Certified Internetwork Expert logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, Enterprise/Solver, EtherChannel, EtherFast, EtherSwitch, Fast Step, Follow Me Browsing, FormShare, GigaDrive, HomeLink, Internet Quotient, IOS, iphone, IP/TV, iq Expertise, the iq logo, iq Net Readiness Scorecard, iquick Study, LightStream, Linksys, MeetingPlace, MG, Networkers, Networking Academy, Network Registrar, PI, ProConnect, ScriptShare, SMARTnet, StackWise, The Fastest Way to Increase Your Internet Quotient, and TransPath are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries. All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (0710R) Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental. Copyright  2007 Cisco Systems, Inc. All rights reserved.

3 Table of Contents Preface...1 Purpose...1 Audience...1 Organization...1 Related Documentation...3 Conventions...4 Obtaining Documentation, Obtaining Support, and Security Guidelines Product Overview...7 Functional Overview...8 Unified CVP Solution Components...9 SIP and H Typical Call Flow...10 Unified CVP Product Components...12 Unified CVP Solution Components...16 Prerequisite Tasks...19 Internal Interfaces Table Choosing a Call Control Protocol...23 SIP, H.323 Comparison...24 H.323 and Unified CVP...24 SIP Restrictions...24 Queue and Transfer Model Choosing a Deployment Model...27 Call Flow Models...27 Call Director Scenario...28 VRU-Only Scenario...32 Comprehensive Scenario...35 VML Server (Standalone) Scenario...38 Geographic Models...42 Centralized Single-Site Geography...43 Centralized Multi-Site Geography...43 Centralized Branch Geography...43 Standalone Branch Geography...44 Physical Models Performance, Sizing, and Choosing Hardware Planning for Reporting...49 Sizing...49 Backup and Restore...49 Synchronizing Timestamps Creating a Failover Strategy (Designing for High Availability)...51 High Availability SIP-Based Call Flow...52 High Availability H.323-Based Call Flow...58 Survivability of Existing Calls...63 Non-Call Handling Components Licensing...67 i

4 8. Planning Network Topology...69 Selecting Codecs...69 WAN vs. LAN...69 High Availability VLANs...70 ASR and TTS...70 Quality of Service (QoS)...70 Security Best Practices Creating Dial Plans Scripting Alternatives: VoiceML vs. Unified ICME Scripting Developer Services...75 Glossary...77 Index...97 ii

5 List of Figures Figure 1: SIP-Based Unified CVP Solution...10 Figure 2: H.323-Based Unified CVP Solution...12 Figure 3: SIP-Based Call Director Call Flow...30 Figure 4: VRU-Only Call Flow...34 Figure 5: SIP-Based Comprehensive Call Flow...36 Figure 6: Standalone VML Call Flow...40 Figure 7: SIP-Based High Availability Component Layout...52 Figure 8: H.323-Based Unified CVP Solution...58 iii

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7 Preface Purpose This manual provides a product overview and describes how to plan for a Cisco Unified Customer Voice Portal (Unified CVP) deployment. Audience This document is intended for Call Center Managers, Unified CVP System Managers, Cisco Unified Contact Center Enterprise (Unified CCE)/Cisco Unified Intelligent Contact Management Enterprise (Unified ICME) and/or Cisco Unified Contact Center Hosted (Unified CCH)/Cisco Unified Intelligent Contact Management Hosted (Unified ICMH) System Managers, VoIP Technical Experts, and IVR application developers. There will also be a significant contingent of people familiar with TDM IVR products, unrelated to contact centers (these people will be buying the Cisco Unified Customer Voice Portal standalone Deployment). Readers should already have a general understanding of Unified CCE and/or Unified ICME and should be familiar with Unified CCE and/or Unified ICME installation and setup procedures. Readers should also be familiar with Cisco Unified Communications Manager. Organization This manual is divided into the following chapters: 1

8 Organization Preface Chapter Chapter 1, "Product Overview" (page 7) Chapter 2, "Choosing a Call Control Protocol" (page 23) Chapter 3, "Choosing a Deployment Model" (page 27) Description An overview of Cisco Unified Customer Voice Portal (CVP) features, benefits, components, and preparation. Discusses using SIP, H.323, or both as your call control protocol with Unified Customer Voice Portal. Discusses the various deployment models available with Unified Customer Voice Portal. Chapter 4, "Performance, Sizing, and Choosing Hardware" Mentions some key factors that determine choices in the (page 47) areas of performance, sizing, and hardware. Chapter 5, "Planning for Reporting" (page 49) Chapter 6, "Creating a Failover Strategy (Designing for High Availability)" (page 51) Chapter 7, "Licensing" (page 67) Chapter 8. "Planning Network Topology" (page 69) Chapter 9, "Creating Dial Plans" (page 71) Chapter 10, "Scripting Alternatives: VoiceML vs. Unified ICME Scripting" (page 73) Chapter 11, "Developer Services" (page 75) Mentions a few critical topics that must be addressed when using Unified CVP reporting. Discusses failover and high availability design considerations for the entire system. Discusses the licensing associated with Unified CVP. Comments on some key factors that affect network topology. Points out that there are differences between creating dial plans in a SIP environment versus an H.323 environment. Discusses the two script editors and their uses. Discusses the distinction between Cisco Technical Support and Cisco Developer Services. 2

9 Preface Related Documentation Related Documentation Unified CVP provides the following documentation: Cisco Security Agent Installation/Deployment for Cisco Unified Customer Voice Portal provides installation instructions and information about Cisco Security Agent for the Unified CVP deployment. We strongly urge you to read this document in its entirety. Configuration and Administration Guide for Cisco Unified Customer Voice Portal describes how to set up, run, and administer the Cisco Unified CVP product, including associated configuration. Element Specifications for Cisco Unified CVP VML Server and Cisco Unified Call Studio describes the settings, element data, exit states, and configuration options for Elements. Installation and Upgrade Guide for Cisco Unified Customer Voice Portal describes how to install Unified CVP software, perform initial configuration, and upgrade. Operations Console Online Help for Cisco Unified Customer Voice Portal describes how to use the Operations Console to configure Unified CVP solution components. Planning Guide for Cisco Unified Customer Voice Portal provides a product overview and describes how to plan for a Unified CVP deployment. Port Utilization Guide for Cisco Unified Customer Voice Portal describes the ports used in a Unified CVP deployment. Programming Guide for Cisco Unified CVP VML Server and Cisco Unified Call Studio describes how to build components that run on the Cisco Unified CVP VML Server. Reporting Guide for Cisco Unified Customer Voice Portal describes the Reporting Server, including how to configure and manage it, and discusses the hosted database. Say It Smart Specifications for Cisco Unified CVP VML Server and Cisco Unified Call Studio describes in detail the functionality and configuration options for all Say It Smart plugins included with the software. Troubleshooting Guide for Cisco Unified Customer Voice Portal describes how to isolate and solve problems in the Unified CVP solution. User Guide for Cisco Unified CVP VML Server and Cisco Unified Call Studio describes the functionality of Cisco Unified Call Studio including creating projects, using the Cisco Unified Call Studio environment, and deploying applications to the Cisco Unified CVP VML Server. For additional information about Unified ICME, see the Cisco web site en/us/products/sw/custcosw/ps1001/tsd_products_support_series_home.html listing Unified ICME documentation. 3

10 Conventions Preface Conventions This manual uses the following conventions: Convention boldface font Description Boldface font is used to indicate commands, such as user entries, keys, buttons, and folder and submenu names. For example: Choose Edit > Find. Click Finish. italic font Italic font is used to indicate the following: To introduce a new term. Example: A skill group is a collection of agents who share similar skills. For emphasis. Example: Do not use the numerical naming convention. A syntax value that the user must replace. Example: IF (condition, true-value, false-value) A book title. Example: See the Cisco CRS Installation Guide. window font Window font, such as Courier, is used for the following: Text as it appears in code or that the window displays. Example: <html><title>cisco Systems,Inc. </title></html> < > Angle brackets are used to indicate the following: For arguments where the context does not allow italic, such as ASCII output. A character string that the user enters but that does not appear on the window such as a password. 4

11 Preface Obtaining Documentation, Obtaining Support, and Security Guidelines Obtaining Documentation, Obtaining Support, and Security Guidelines For information on obtaining documentation, obtaining support, providing documentation feedback, security guidelines, and also recommended aliases and general Cisco documents, see the monthly What's New in Cisco Product Documentation, which also lists all new and revised Cisco technical documentation, at: 5

12 Obtaining Documentation, Obtaining Support, and Security Guidelines Preface 6

13 Chapter 1 Product Overview This chapter contains an overview of Cisco Unified Customer Voice Portal (CVP) features, benefits, components, and preparation. Note: What follows is a statement on product name usage in this manual: As is discussed below, Unified CVP has an intimate association with four related Cisco products that provide call routing: Cisco Unified Contact Center Enterprise (Unified CCE) formerly called Cisco IP Contact Center (IPCC) Enterprise Edition Cisco Unified Contact Center Hosted (Unified CCH) formerly called Cisco IP Contact Center (IPCC) Hosted Edition Cisco Unified Intelligent Contact Management Enterprise (Unified ICME) formerly called Cisco Intelligent Contact Management (ICM) Enterprise Edition Cisco Unified Intelligent Contact Management Hosted (Unified ICMH) formerly called Cisco Intelligent Contact Management (ICM) Hosted Edition Rather than constantly referring to all four of the above products, the following shorthand is used. Unless otherwise noted, Unified ICME stands for all four of the above products, while Unified ICMH stands for both Hosted products, and Unified CCE stands for both IP products. This section contains the following topics: Functional Overview, page 8 Unified CVP Solution Components, page 9 Prerequisite Tasks, page 19 Internal Interfaces Table, page 21 7

14 Functional Overview Chapter 1: - Product Overview Functional Overview Cisco Unified Customer Voice Portal (CVP) is a product with two distinct aspects. First, it provides the Voice over IP (VoIP) routing capability for Unified ICME. Unified ICME provides the services necessary to determine where calls should be routed, whether to Automatic Call Distributors (ACDs), specific agents, or Voice Response Units (VRUs), but the routing services themselves must be provided by an external routing client. Traditionally, Unified ICME s routing clients were various Public Switch Telephone Network (PSTN) network switches, or, in some cases, customer-provided ACDs. Unified CVP makes it possible for Unified ICME to use VoIP gateways as routing clients as well. This carries a number of advantages, not the least of which is that call traffic can be handled over the IP network rather than by the PSTN carrier. The second aspect is as a VRU platform. Long ago, intelligent PSTN carrier networks provided the capability to prompt for and collect basic data from the caller before delivering the call. Unified CVP now provides that capability, and greatly expands it with the use of its own VoiceML Interactive Voice Response (IVR) application platform. Also, when calls were delivered to an ACD, the ACD could provide some initial prompt and collect capability, and it could also provide some messaging or hold music to callers who are waiting in queue. When ICM took over the ACD queuing function, it needed a place to park calls while they were waiting. It introduced the notion of a Service Control VRU for that purpose, and third-party VRU vendors were invited to fill that role. Today, Unified CVP fills that role as well in VoIP environments. Unified Customer Voice Portal is both a product and a solution. As a product, it consists of a number of CVP-specific components. However, these components must be set into a solution which includes other products. The Unified CVP solution components are discussed in the next section. 8

15 Chapter 1: Product Overview Unified CVP Solution Components The Unified CVP solution feature set includes: IP-based switching. Unified Customer Voice Portal can transfer calls over an IP network. IP-based Takeback. Unified Customer Voice Portal can take back a transferred call for further IVR treatment or transfer. IP-based IVR services. The classic prompt-and-collect functions: Press 1 for Sales, 2 for service, and so forth. IP-based queuing. Calls can be parked on Unified Customer Voice Portal for prompting, music on hold, and so forth, while waiting for a call center agent to be available. Compatibility with the Public Switch Telephone Network (PSTN). Calls can be moved onto an IP-based network for Unified Customer Voice Portal treatment and then moved back out to a PSTN for further call routing to a call center. Carrier-class platform. Unified Customer Voice Portal s reliability, redundancy, and scalability allow it to work with Service Provider and large Enterprise networks. IP-based Voice Enabled IVR Services. Unified Customer Voice Portal provides for sophisticated self-service applications, such as banking, brokerage, or airline reservations. Session Initiation Protocol (SIP). The ability to switch calls using SIP rather than, or in addition to, H.323. Operations Console. The ability to monitor and configure the entire Unified CVP solution from a single operations console. Reporting. The ability to provide custom reports on the activities of Unified CVP components, Unified CVP IVR applications, and Unified CVP IVR callers. Unified CVP Solution Components In its most sophisticated deployments, Cisco Unified Customer Voice Portal can be complex. The solution is made up of many kinds of components, most of which can be multiplied for scalability and/or reliability purposes. As indicated above, some of these components are part of the Unified CVP product, others are not. The introduction of the Operations Console in Unified CVP 4.0 has simplified the means by which these many components are dealt with, but it has not reduced the number of components. To be sure, any given deployment of Unified CVP need not use all the components that are available. The various deployments and ways of looking at them (call flow models, geographic models, physical models, models that involve Unified ICME) are discussed in Chapter 3, "Choosing a Deployment Model" (page 27). However, let us begin by looking at all of the components. One way to survey the various Unified CVP solution components is to review a typical call flow. 9

16 Unified CVP Solution Components Chapter 1: - Product Overview Detailed component descriptions will be provided later in this document, as we group them from other perspectives. However, it is worth noting, even at this point, that chief among the non-cvp-specific products is the Cisco IOS Gateway. This Gateway provides all the TDM-to-VoIP conversion, VoIP and TDM call switching, MRCP (Media Resource Control Protocol) interaction with ASR (Automatic Speech Recognition) and TTS (Text-to-Speech) servers, and of particular importance, the IOS VML Voice Browser. A Voice Browser acts like a web browser, but for voice rather than visual applications. As such, it uses HTTP to communicate with a web server. It sends URLs just as web browsers do, but it receives and renders VML documents rather than HTML documents. Rendering of a VML document amounts to playing voice and text-to-speech (TTS) prompts, and accepting DTMF (Dual Tone Multi-Frequency) and spoken input. Certain Unified CVP product components play the role of the web server in this paradigm. SIP and H.323 Unified CVP 4.0 and later provides the ability to switch calls using Session Initiation Protocol (SIP) rather than, or in addition to, H.323. Only H.323 was provided in earlier versions of CVP. SIP is the preferred protocol for Unified CVP 4.0 and later. H.323 support is primarily to provide backward compatibility for users of previous versions of CVP. These are referred to as legacy deployments. This topic is treated in more detail in Chapter 2, "Choosing a Call Control Protocol" (page 23). Typical Call Flow What follows is a fairly typical call flow scenario which corresponds roughly to the Comprehensive call flow model. It consists of an incoming call requiring initial self service, followed by queue treatment, and finally delivery to a Unified ICME agent. Note that this is only an illustration, presented not as a specification, but as an introduction to the overall flow of information in a Unified CVP solution. Call flows are discussed further in Chapter 3, "Choosing a Deployment Model" (page 27). Components are discussed in more detail immediately following this call flow discussion. In reading the following narrative, please consider the two associated diagrams. These represent the call handling components which are involved in a SIP implementation and an H.323 implementation. Note that important non-call handling components, such as the Reporting Server and the Operations Console, are not shown. Figure 1: SIP-Based Unified CVP Solution 10

17 Chapter 1: Product Overview Unified CVP Solution Components SIP-Based Call Flow Calls arrive from a PSTN via a TDM connection into an Ingress Gateway. The Ingress Gateway sends a new call message to the SIP Proxy Server, which forwards it to the SIP Service. The SIP Proxy Server, in addition to forwarding SIP messages to appropriate devices, also provides failover and redundancy support for those devices. The SIP Service consults Unified ICME (actually, in this particular case, Unified CCE) via the ICM Service (this and all subsequent interactions between any Unified CVP component and Unified ICME pass through the ICM Service). This consultation causes Unified ICME to run a routing script. The routing script requires the call to be transferred to a VRU, so Unified ICME instructs the SIP Service to extend a second call leg (that is, deliver the call) to a VML Gateway (which is often identical with the Ingress Gateway). The VML Gateway sends a message to the IVR Service, which then requests scripted instructions from Unified ICME. Unified ICME exchanges VRU instructions with the VML Gateway via the IVR Service. Among these VRU instructions can be requests to the VML Gateway to invoke more sophisticated applications on the VML Server. Such requests will result in multiple exchanges between the VML Server and the VML Gateway, ultimately ending in a reply to Unified ICME via the IVR Service. VML Server applications are designed and built using Call Studio (not shown in the diagram, since it is essentially an offline tool). Once the self service portion of the call is completed, the Unified ICME routing script places the caller into queue. This call is already at a VRU, and it waits there for the agent to become available. Meanwhile, further VRU instructions are exchanged between Unified ICME and the VML Gateway via the IVR Service to play messages and recorded music. When the agent becomes available, Unified ICME sends a message to the SIP Service, which forwards a message via the SIP Proxy Server to the Ingress Gateway and to Unified Unified CM to transfer the call away from the VML Gateway and deliver it to the Unified CM phone. During the VRU exchanges, the VML Gateway interacts with an ASR/TTS Server to have text played as speech or speech recognized as data, and with a Media Server (not shown in the diagram, but connected to the VML Gateway) to fetch audio files and prompts. These two devices, as well as the VML Server, can be located behind a Content Services Switch (CSS), which offers them 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 VML Server send a stream of reporting events to the Reporting Server (not shown in the diagram, but connected to the Call Server), which processes and stores the information in a database for later reporting. All these devices also 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 Operations Console (not shown in the diagram, but 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. 11

18 Unified CVP Solution Components Chapter 1: - Product Overview Figure 2: H.323-Based Unified CVP Solution H.323-Based Call Flow If you are using H.323-based call control instead of the SIP-based call control described above, then the SIP Proxy Server and SIP Service are replaced by a Gatekeeper and H.323 Service. The message paths are a bit different as well; the Ingress Gateway consults the Gatekeeper but does not pass messages through it, and the H.323 Service communicates with the IVR Service rather than directly with Unified ICME. Unified CVP Product Components As mentioned above, Unified Customer Voice Portal is both a product and a solution. The following subsections discuss the components specific to the Unified CVP product. Call Server The Call Server provides call control capabilities by means of the various services that make up the Call Server: ICM Service, SIP Service, IVR Service, and H.323 Service. Call Server: ICM Service This software module is responsible for all communication between Unified CVP components and Unified ICME. It sends and receives messages on behalf of the SIP Service, the IVR Service, and the VoiceML Service. The ICM Service is a software module that always resides in the Call Server. Call Server: SIP Service This software module is a SIP Back-to-Back User Agent (B2BUA). On the front end, it supports two SIP call legs, one incoming and one outgoing. This component does not terminate RTP (Real-time Transport Protocol) traffic; it only deals with SIP messages. On the back end, it works with the ICM Service to implement an interface to Unified ICME for call control functions only (no VRU scripts). The purpose of a B2BUA is to act as an active intermediary for a call, communicating with both the source and destination legs of the call, and providing third-party call control capabilities on behalf of a third-party device such as the Unified ICME CallRouter. Because it must track the call throughout its life, it becomes a key source of call-level reporting data and a natural place for moderating call switching resources. 12

19 Chapter 1: Product Overview Unified CVP Solution Components Generically speaking, a SIP B2BUA is two User Agents (UAs). The device receives incoming calls in one UA, and immediately creates an outgoing call via its other UA. Thereafter, all SIP messages from either one are forwarded to the other. What makes Unified CVP s B2BUA particularly useful is the fact that it puts Unified ICME in between the two legs. On incoming calls it asks Unified ICME where the outgoing call should be addressed. Once the call has been delivered, Unified ICME has the opportunity to end the outgoing leg and reconnect it to a subsequent destination. In practice, there will typically be a delivery to the VRU leg, a take-back and delivery to an agent, and subsequent take-backs and deliveries to subsequent agents or back to the VRU leg. The SIP Service is a software module that always resides in the Call Server. Call Server: IVR Service This software module creates the VoiceML pages which implement the Unified CVP micro-applications based on Run Script instructions received from Unified ICME. The IVR Service functions as the VRU leg (in Unified ICME parlance), and calls must be transferred to it from the SIP Service in order to execute micro-applications. The VML pages created by this module are sent to the VML Gateway to be executed. There is a special micro-application that causes the IVR Service to generate a "wrapper" VML page. This VML page instructs the VML Gateway to invoke an application on the VML Server. In deployments which use H.323 instead of SIP, the IVR Service can act as the switch leg as well. The deployment includes an H.323 Service, which interacts with the IVR Service for call control activities. Calls still need to be transferred to a VRU leg before micro-applications can be executed; however, in legacy deployments, that transferred leg may end up using the same IVR Service as the switch leg. This scenario is exactly the same as it was for CVP 3.0/3.1. The IVR Service plays a significant role in implementing Unified CVP s native failover mechanism those capabilities which can be achieved without a Content Services Switch for Media Servers, ASR/TTS Servers, and VML Servers. Up to two of each such server are supported, and the IVR Service orchestrates retries and failover between them. The IVR Service is a software module that always resides in the Call Server. Call Server: H.323 Service (Formerly known as the "CVP Voice Browser".) This software module interacts with the IVR Service to relay call arrival, departure, and transfer instructions between it and the other H.323 components (only used in H.323-based call flows). The H.323 Service is a software module that always resides in the Call Server. 13

20 Unified CVP Solution Components Chapter 1: - Product Overview VML Server This component executes complex IVR applications by exchanging VoiceML pages with the VML Gateway s built-in voice browser. VML Server runs within a J2EE (Java 2 platform Enterprise Edition) application server environment such as Tomcat or WebSphere. VML Server applications are written using Call Studio, and deployed to the VML Server for execution. The applications are invoked on an as-needed basis by a special micro-application which must be executed from within the Unified ICME routing script. VML Server can also be deployed in a standalone configuration one which includes no Unified ICME components. In this model, applications are invoked as a direct result of calls arriving in the VML Gateway. You can add the VoiceML Service software. This affords standalone applications the additional ability to forward reporting events to a Reporting Server, and to make ancillary routing requests and exchange call context information with a Unified ICME if one is present. However, the integration with Unified ICME is nominal. Call Studio This component is the service creation environment (script editor) for VML Server applications. It is based on the open source Eclipse framework, and provides advanced drag-and-drop graphical editing, as well as the ability to insert vendor-supplied and custom-developed plug-ins which allow applications to interact with other services in the network. Call Studio is essentially an offline tool, whose only interaction with the VML Server is to deliver compiled applications and plugged-in components for execution. Reporting Server The 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 VML Server. As stated, it is deployed together with an Informix database management system, and it transforms and writes this reporting data into this 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. There only needs to be one Reporting Server in a deployment. This does not represent a single point of failure, however, because data safety and security are provided by the database management system, and temporary outages are tolerated due to persistent buffering of 14

21 Chapter 1: Product Overview Unified CVP Solution Components information on the source components. However, if more than one Reporting Server is used, be aware that: Each Call Server can be associated with only one Reporting Server Reports cannot span multiple Informix databases Operations Console This component is required in every Unified CVP deployment. It provides the administration and configuration interface for all Unified CVP product components using the JM (Java Management Extensions) protocol, and it offers shortcuts into the administration and configuration interfaces of all the remaining components. 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 by 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 or online 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 as those components will use them upon the next component restart. The Operations Console allows configuration parameters to be updated (or, indeed, pre-configured) 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. Unified CVP product components do not provide any means to modify their own configuration without the use of the Operations Console. A Unified CVP deployment can only have one Operations Console. The ability to access and update configurations at this level is not usually real-time critical. Health monitoring by network monitoring tools is real-time critical, but high availability for these tools is not Unified CVP s responsibility. The need to provide administration access to multiple users is also taken care of by the fact that the Operations Console can be accessed through any supported web browser. 15

22 Unified CVP Solution Components Chapter 1: - Product Overview Unified CVP Solution Components The following subsections discuss components that are part of an overall Unified CVP solution, but are not contained in the Unified CVP product itself. SIP Proxy Server A SIP Proxy Server is a device that routes individual SIP transport messages among SIP endpoints. It plays a key role in Unified CVP s high availability architecture for call switching. It is designed to support multiple SIP endpoints of various types, and implements load balancing and failover among these endpoints. SIP Proxy Servers themselves can be deployed alone or as a pair. Unified CVP can also be deployed without a SIP Proxy Server. In such cases, some of the Proxy Server functions can be provided by the SIP Service, because it provides the ability to configure a static table to look up destinations. 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 can also be performed if SRV records are configured with ordered priorities Ingress Gateway The Ingress Gateway is the point at which an incoming call enters the Unified CVP solution. It terminates TDM phone lines on one side and implements VoIP on the other side. It also 16

23 Chapter 1: Product Overview Unified CVP Solution Components provides for sophisticated call switching capabilities at the command of other Unified CVP solution components. It works with either SIP or H.323 protocols, and also supports Media Gateway Control Protocol (MGCP) for use with Unified CM. The Ingress Gateway can be deployed separately from the VML 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. VML Gateway The VML Gateway hosts the IOS Voice Browser, the component which interprets VML pages from either the IVR Service or the VML Server, plays.wav files and TTS, inputs voice and DTMF, and sends results back to the VML requestor. It also mediates between Media Servers, VML Servers, ASR and TTS Servers, and IVR Services. Unless it is combined with the Ingress Gateway (see above), the VML Gateway does not require any TDM hardware. All its interfaces are VoIP on one side and HTTP (carrying VML or.wav files) and MRCP (carrying ASR and TTS traffic) on the other side. As with Ingress Gateways, VML Gateways are often deployed in farms for Centralized deployment models, or one per office in Branch deployments. MRCP ASR/TTS Server This device provides speech recognition services and text-to-speech services for VML Gateway. For capacity and redundancy reasons, a CSS is usually used to mediate between a farm of such servers; if no CSS is used, then Unified CVP can support a maximum of two. Cisco does not sell, OEM, or support any ASR/TTS Servers. Cisco does, however, test Unified CVP with Nuance and IBM offerings. A certification process is available to allow additional vendors to qualify the interoperability of their products with Unified CVP. Such certification is part of the Cisco Technology Developer Program for more information see Media Server The Media Server is a simple web server, such as Microsoft IIS or Apache. Its only purpose within Unified CVP is to store and serve up.wav files to the VML Gateway, as required in order to render VML pages. As with ASR/TTS Servers, Media Servers can be deployed singly, as a redundant pair, or with CSS in a farm. Note that the VML Gateway caches.wav files it retrieves from the Media Server. In most deployments, the Media Server encounters extremely low traffic from Unified CVP. Unified CM This is Cisco s IP-based PB. It is used to manage and switch VoIP calls among IP phones. When combined with Unified ICME it becomes Unified CCE. 17

24 Unified CVP Solution Components Chapter 1: - Product Overview Unified CVP interacts with Unified CM primarily as a means for sending PSTN-originated calls to Unified CCE agents. However, several applications require that calls be originated by Unified CCE agents instead. Specifically, Cisco Unified Outbound Option when used with Unified CCE, and calls that are being warm-consultative-transferred from one agent to another, are originated in this way. "Help desk" calls, in which an agent or other IP phone user calls Unified CVP (or calls a skill group and gets queued on Unified CVP), also fall into this category. A single Unified CM can originate and receive calls from both SIP and H.323 devices. Gatekeeper 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; the paradigm is instead that of a request/response server. For redundancy, Gatekeepers can be deployed in pairs using the HSRP (Hot Standby Routing Protocol), one redundant pair per site. Additionally, the VBAdmin SetGatekeeper command allows multiple IP addresses to be configured. The H.323 Service then keeps track of a currently active Gatekeeper from that list, beginning with the first, and sends all requests to that Gatekeeper. If the currently active Gatekeeper fails, it moves to the next one in the list, and that one becomes current. The H.323 Service continues to use it 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. Note that for sizing purposes, each individual Gatekeeper should be sized to handle the entire load. DNS Server This optional component can be installed anywhere in the network. Its purpose, in general, is to resolve hostnames to IP addresses. However, it can also be configured with multiple IP addresses for the same hostname. Successive requests for a given hostname can be configured to return IP addresses in a round-robin fashion. Unified CVP s SIP design can make use of this capability in order to implement a sort of load balancing among multiple like components. The DNS (Domain Name System) 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 ICME to transfer the call to the VRU leg, it can use DNS to alternate such requests between two SIP Proxy Servers, and the SIP Proxy Servers can use DNS to distribute VRU legs among multiple VML Gateways. If SIP Proxy 18

25 Chapter 1: Product Overview Prerequisite Tasks Servers are not being used, the SIP Service can use DNS directly to distribute VRU legs among multiple VML Gateways. When transferring a call to an agent using a SIP Proxy Server, the SIP Proxy Server can use DNS to identify the IP address target for that agent. Also, the SIP Service can use DNS to alternate such requests between the two SIP Proxy Servers. If SIP Proxy Servers are not being used, then the SIP Service can use DNS to locate the target agent s IP address. Note that use of the DNS Server for SIP routing is entirely optional in Unified CVP. Content Services Switch The Content Services Switch (CSS) is the focus for high availability design in the TCP (Transmission Control Protocol) arena. The CSS can be placed between one (or more) VML Gateways and all its IVR Services, VML Servers, Media Servers, and ASR/TTS Servers. Various mechanisms allow it to implement transparent load balancing and failover across these types of devices. CSS is an optional device, but it comes highly recommended. Without it, the IVR Service implements a "poor man s failover" mechanism for up to two of each of the above components, but they are not load balanced, and various retries and delays are part of the algorithm all of which is avoided if CSS is used. The CSS is normally deployed as a Virtual Router Redundancy Protocol (VRRP) pair. It is useful in all deployment models except for Call Director call flows. Network Monitor An SNMP management station that can be used to monitor the solution deployment's health. Prerequisite Tasks At this point we have presented some of Unified Customer Voice Portal's features, mentioned that there are various deployment models, presented a typical call flow, and discussed the major components of a Unified CVP solution. This section indicates tasks that must be performed, or decisions that must be made, before purchasing and deploying a Unified CVP solution. These should be kept in mind when reading the rest of this manual. Choose a Deployment model. Choose a Call Flow model. Choose a Geographic model. 19

26 Prerequisite Tasks Chapter 1: - Product Overview Choose a Physical model. Work out the network topology for your system, including addresses and names of the solution components. Determine a failover strategy for all the components of your Unified CVP solution. In non-sip environments, determine your strategy for inbound call routing (that is, dial peers versus Gatekeeper). Decide on a naming resolution system for Gateways (DNS versus configured on the Gateway). If using a VRU other than Unified CVP, determine the VRU trunk group number and number of trunks. If ASR and/or TTS are part of your solution, determine the locale values to be used. Decide whether the same set of VRUs are to be used for all cases, or whether that will be determined separately for each customer (dialed number). Note: If all dialed numbers will use the same VRUs, it is easiest to use a default Network VRU, rather than to configure multiple Network VRUs. For more information, see the Common Configuration for Differentiating VRUs (Customer Voice Portals) Based on Dialed Number section in the Configuration and Administration Guide for Cisco Unified Customer Voice Portal. Plan how you will be routing calls through the network to the VRU. For Comprehensive deployment models and for deployment models with a NIC (Network Interface Controller): Determine the Network Routing Number. This number is the base for routing calls through the network to the VRU; a correlation ID is appended to this number to transfer calls to a Network VRU through the network. The Network Routing Number should be at least as long as the longest dialed number on which Unified CVP will receive incoming calls. For deployments with a Customer VRU and in Unified ICMH/CICM environments and for NIC Type 2 or 8 deployments: Determine the translation route pools to use for each VRU. Determine the labels to be sent to the network to connect the call to the VRU and the corresponding DNIS (Dialed Number Identification Service) that will be seen by the VRU. For example, the label for the network might be and the DNIS received by the VRU and sent back to the ICM to identify the call might be The pool must contain as many independent DNISs as the number of new call arrivals that might occur during the longest time it takes to get any to the VRU, that is, the time to execute a Translation Route to VRU node. This time needs to include the network delays and the possibility that alternate endpoints are used in the VoIP network. In practice, make sure the number is significantly larger than this to ensure there are always enough. For example, if the time to get to the VRU is 3 20

27 Chapter 1: Product Overview Internal Interfaces Table seconds and the maximum call arrival rate is 5 cps (instantaneous), 15 would be needed, so provision at least 30. If used, define naming schemes for Unified ICME peripheral gateways (PGs), peripherals, and routing clients. Submit design for Unified Customer Voice Portal solution to the Bid Assurance/Assessment to Quality (A2Q) process for design review and deployment assessment. (Contact your Cisco representative for additional information.) Internal Interfaces Table Table 1: Major Internal Interfaces The following table summarizes the major interfaces between the Unified CVP product components and other Cisco products. Interface Between This Component......and This Component Interface Characteristics Interface Type Selection Determined By Gateway (Ingress or Egress) Unified CM/IP Telephone Call Server SIP SIP Proxy Server or dial-peer configuration Gateway Unified CM/IP Telephone H.323 Service H.323 Gateway dial peers or Gatekeeper, based on H.323 Service availability and proximity Gateway Gatekeeper H.323 Gateway configuration H.323 Service IVR Service HTTP (URL with VML response). H.323 Service client, IVR Service server. H.323 Service configuration Media Server HTTP URL is given in VML generated by the IVR Service from information specified in Unified ICME. Normal web access (DNS, distributors, and so forth) is used to resolve it. Note: The H.323 to Media Server interface exists ONLY in legacy H.323 with Media Termination call flow models. IVR Service Unified ICME ICM VRU Messaging CVP configuration 21

28 Internal Interfaces Table Chapter 1: - Product Overview Interface Gateway VML Server Interface Characteristics HTTP (URL with VML response) Unified ICME script configuration (though this does not apply for VML Server (standalone)) Unified ICMH Unified ICME Proprietary Customer identified by calling information; Unified ICMH configuration determines Unified ICME Unified ICME Customer database SQL Gateway Unified ICME Configuration IVR Service Unified ICME Configuration Reporting Server Informix DB JDBC Operations Console VML Server Call Server Proprietary Operations Console, VoiceML script, and Unified ICME script configuration 22

29 Choosing a Call Control Protocol This chapter discusses using SIP, H.323, or both as your call control protocol with Unified Customer Voice Portal. Unified CVP 4.0 and later provides the ability to switch calls using Session Initiation Protocol (SIP) rather than, or in addition to, H.323. Only H.323 was provided in earlier versions of CVP. SIP is a network communication protocol for initiating, modifying, and terminating interactive sessions between clients that involves multimedia elements such as voice, video, instant messaging, and so forth. SIP is intended to provide a superset of the call processing features available in the PSTN. SIP is the preferred protocol for Unified CVP 4.0 and later. SIP provides improved scalability and performance to Unified CVP. With SIP, you are able to interoperate with both Cisco and non-cisco SIP edge devices. As a generic session protocol, SIP is better able to handle non-voice media. SIP is becoming a de facto industry standard. Chapter 2 H.323 support is primarily to provide backward compatibility for users of previous versions of CVP. These are referred to as legacy deployments. This section contains the following topics: SIP, H.323 Comparison, page 24 H.323 and Unified CVP, page 24 23

30 SIP, H.323 Comparison Chapter 2: - Choosing a Call Control Protocol SIP, H.323 Comparison Table 2: Comparison of SIP and H.323 SIP Clients (User Agents) are intelligent The following table provides a partial comparison of SIP and H.323. Servers (Proxy, Redirect, Registrar) provide network intelligence and services Follows the Internet/WWW model Uses UDP or TCP for signaling Uses RTP for media Is a full-featured multimedia protocol Is text-based (ASCII) H.323 Clients are intelligent Gatekeepers provide network intelligence and services Follows the Telephony/Q.SIG model Uses TCP for signaling (UDP optional in v3) Uses RTP for media Is a full-featured multimedia protocol Is binary-based (ASN.1 encoding) Utilizes IETF/IP protocols - SDP, HTTP/1.1, SMTP, MIME, Utilizes ITU / ISDN protocols - H.225, H.245, H.450. All and so forth portions fall under H.323 Only defines signaling, not all elements Interoperability is done with inow or IMTC H.323 and Unified CVP For users of previous versions of CVP, where SIP was not an alternative, upgrading while keeping their call flows unchanged (at least initially) is a useful option. Indeed, a Unified CVP solution is capable of running as a hybrid, that is, some call flows using H.323 and some using SIP. The software should all be upgraded first; then flows should be cut over in groups, perhaps by DNIS or by application (see the discussion in Installation and Upgrade Guide for Cisco Unified Customer Voice Portal). A single call can be controlled by either SIP or H.323, but not both. However, a single Unified CVP component can carry some calls in each category. Note that certain call control capabilities which are available in H.323 are not yet available in SIP. Customers who require these particular capabilities will wish to remain with H.323. In addition, most customers who are staying with H.323 are advised to move to Comprehensive call flow if they currently use Queue and Transfer, but there are some cases where these customers will not be able to do so. These situations are discussed below. SIP Restrictions SIP cannot perform all operations that H.323 can perform, because the endpoints, the Unified CVP implementation, or the protocol itself does not yet support those operations. Customers who require such call control capabilities must use H

31 Chapter 2: Choosing a Call Control Protocol H.323 and Unified CVP The following are reasons why new customers may need to use H.323 for call control: Cisco PGW media gateway controller (MGC) in call control mode is used to initiate calls to Unified CVP GKTMP NIC is used for pre-routing Hook-flash is used to transfer calls Queue and Transfer Model This model, which was supported in previous versions of CVP, is not supported for new customers in Unified CVP 4.0 and later. New customers are advised to use the Comprehensive model instead. However, existing customer who are upgrading to Unified CVP 4.0 while continuing to use ICM 6.0 or earlier, do not always have this option. Such customers who: Do have IP-originated calls, must stay with H.323 and continue terminating media on the H.323 Service (formerly called the CVP Voice Browser) Do not have IP-originated calls, are encouraged to move to SIP; they may choose to stay with H.323, but in that case they must remove media termination from the H.323 Service since, within the Unified CVP solution, Gateways and Unified CM devices are the only components that terminate voice 25

32 H.323 and Unified CVP Chapter 2: - Choosing a Call Control Protocol 26

33 Choosing a Deployment Model Deployments vary in terms of the call flow model, geographic model, and physical model used. This chapter discusses the various models available with Unified Customer Voice Portal. Note: High availability considerations are discussed in Chapter 6 (page 51). This section contains the following topics: Call Flow Models, page 27 Geographic Models, page 42 Physical Models, page 45 Chapter 3 Call Flow Models This section presents the options available in terms of the type of call flow required. Note that for each call flow scenario presented below, you can elect to use either SIP or H.323 for call control (where such call control is relevant). When using SIP, the SIP Service and an optional SIP Proxy Server are used; with H.323, the H.323 Service and an H.323 Gatekeeper are used. The Reporting Server and Operations Console are not discussed in connection with the call flows. A Reporting Server can be added to any scenario, and an Operations Console is required for all scenarios. The call flow scenarios discussed are: Call Director. For customers who: Want to use Unified CVP only to provide Unified ICME with VoIP call switching capabilities Do not need to use Unified CVP to control queued calls 27

34 Call Flow Models Chapter 3: - Choosing a Deployment Model Want to prompt/collect using third-party VRUs or ACDs Do not want to use VML Server VRU-Only. For customers who: Want to use Unified CVP to provide Unified ICME with VRU services including integrated self-service applications and/or initial prompt and collect Do not want to use Unified CVP for switching calls May want to use optional VML Server May want to prompt/collect with optional ASR/TTS Comprehensive. For customers who: Want to use Unified CVP to provide Unified ICME with VoIP call switching capabilities Want to use Unified CVP to provide Unified ICME with VRU services including integrated self-service applications, queuing, and/or initial prompt and collect May want to use optional VML Server May want to prompt/collect with optional ASR/TTS VML Server (Standalone). For customers who: Want to deploy self-service VoiceML applications For more details, and additional information, see the following discussions of the call flow scenarios. Note: The Queue and Transfer call flow model, which was supported in previous versions of CVP, is not supported for new customers in Unified CVP 4.0 and later. New customers are advised to use the Comprehensive model instead. For more information, see the discussion at the end of Chapter 2 (page 23). Call Director Scenario In this scenario, Unified CVP provides Unified ICME with VoIP call switching capabilities only. Customers provide their own Service Control VRU if they are using Unified ICME to queue calls, or they can queue calls directly on their ACD. Calls can be transferred multiple times, from ingress, to customer-provided VRU, to either Unified CCE or customer-provided ACD or agent, and back again. When calls are connected to customer-provided equipment, their voice paths must go to an egress gateway which is connected by TDM to that equipment. If the signaling is SIP, then Cisco will work with customer-provided SIP endpoints which have been tested and certified to interoperate with Unified CVP. Neither the VML Server nor any VML Gateways are used in this scenario. 28

35 Chapter 3: Choosing a Deployment Model Call Flow Models SIP-Based Call Director Scenario Table 3: Components Used for SIP-Based Call Director Scenario Component SIP Service (part of Call Server) Required Optional Installed (but Inactive) Not Used IVR Service (part of Call Server) ICM Service (part of Call Server) H.323 Service (part of Call Server) VML Server Call Studio Ingress Gateway VML Gateway SIP Proxy Server Gatekeeper Operations Console Reporting Server ASR/TTS Media Server DNS Server Content Services Switch Unified ICME A sample SIP-based Call Director call flow progresses as follows (the sample call flow assumes a third-party TDM VRU using Service Control Interface on a Type 8 Network VRU): 29

36 Call Flow Models Chapter 3: - Choosing a Deployment Model Figure 3: SIP-Based Call Director Call Flow 1. TDM call goes from PSTN to Ingress Gateway. 2. New call invitation goes from Ingress Gateway to SIP Proxy Server. 3. Invitation is forwarded from SIP Proxy Server to SIP Service. 4. SIP Service, wishing to consult Unified ICME, passes new call message to ICM Service. 5. New call message is passed from ICM Service to Unified ICME. 6. Unified ICME runs a routing script, which requires call to be transferred to VRU, and passes a Connect to VRU request to ICM Service. 7. Connect to VRU request passes from ICM Service to SIP Service. 8. Invitation to Connect to VRU goes from SIP Service to SIP Proxy Server. 9. Invitation goes from SIP Proxy Server to Egress Gateway, and call is then passed to Egress Gateway. 10. TDM call is passed from Egress Gateway to third-party VRU. 11. The VRU sends Request Instruction to Unified ICME. 12. Unified ICME sends Run Script request to VRU. 13. VRU executes the script and sends Run Script result to Unified ICME. 14. Unified ICME sends another Run Script request to VRU, which executes the script, and the call is dequeued. 15. Unified ICME passes a connect to agent request to ICM Service. 30

37 Chapter 3: Choosing a Deployment Model Call Flow Models 16. ICM Service passes connect to agent request to SIP Service. 17. A disconnect VRU request passes from SIP Service to SIP Proxy Server. 18. A disconnect request is passed from SIP Proxy Server to Egress Gateway. 19. Egress Gateway passes TDM hangup request to VRU. 20. SIP Service passes connect to agent request to SIP Proxy Server. 21. SIP Proxy Server passes connect to agent request to Unified CM. 22. Unified CM informs Unified ICME that call was received by agent. SIP Coresident with VRU PG In relatively low volume Call Director call flow scenarios, the SIP Service can be installed together with the ICM Service directly on a VRU PG. (That is, the VRU PG becomes the Call Server. Installation of the IVR Service and H.323 Service is also allowed, though these parts of the Call Server are inactive in the present case.) This provides a streamlined deployment option for customers who need little more than to SIP-enable their Unified ICME call control. H.323-Based Call Director Scenario Table 4: Components Used for H.323-Based Call Director Scenario Component SIP Service (part of Call Server) Required Optional Installed (but Inactive) Not Used IVR Service (part of Call Server) ICM Service (part of Call Server) H.323 Service (part of Call Server) VML Server Call Studio Ingress Gateway VML Gateway SIP Proxy Server Gatekeeper Operations Console Reporting Server ASR/TTS Media Server DNS Server 31

38 Call Flow Models Chapter 3: - Choosing a Deployment Model Component Content Services Switch Required Optional Installed (but Inactive) Not Used Unified ICME The H.323-based call flow is much like the SIP-based call flow. However, the SIP Service and the (optional) SIP Proxy Server are not used. Rather, the H.323 Service and (optionally) the Gatekeeper are used. The H.323 Service communicates with the IVR Service rather than directly with Unified ICME. If the Gatekeeper is used, the Ingress Gateway consults the Gatekeeper to find the destination for messages, but does not pass messages through it. This is in contrast to the SIP Proxy Server, where messages actually pass through the Proxy Server to the appropriate destination. VRU-Only Scenario This scenario (called "Advanced Speech" in earlier versions of CVP) has Unified CVP providing Unified ICME with VRU services for calls which are switched in some other manner, such as by a carrier switched network via a Unified ICME Network Interface Controller (NIC) interface. VRU services could be for initial prompt and collect, for integrated self-service applications, for queuing, or for any combination thereof. This scenario does not use SIP or H.323, and requires no Ingress Gateway. It does use VML Gateways, but VML Server is optional, as are ASR and TTS Servers, and Media Servers. Depending on which kind of routing client is in charge of call switching, Unified ICME may transfer the call to the VRU-Only Call Server either by a Translation Route to VRU node, or by a Send To VRU node. In the first case, the ICM Service portion of the Call Server will determine that the arriving call is a VRU leg call by matching the arriving DNIS with its configured list of arriving DNISs. In the second case, it will determine that it is a VRU leg call by the fact that the DNIS length is greater than its configured maximum DNIS length. Digits beyond the maximum DNIS length are taken to be the Correlation ID. Table 5: Components Used for VRU-Only Scenario Component SIP Service (part of Call Server) Required Optional Installed (but Inactive) Not Used IVR Service (part of Call Server) ICM Service (part of Call Server) H.323 Service (part of Call Server) VML Server Call Studio Ingress Gateway VML Gateway SIP Proxy Server 32

39 Chapter 3: Choosing a Deployment Model Call Flow Models Component Gatekeeper Required Optional Installed (but Inactive) Not Used Operations Console Reporting Server ASR/TTS Media Server DNS Server Content Services Switch Unified ICME A sample VRU-Only call flow progresses as follows: 33

40 Call Flow Models Chapter 3: - Choosing a Deployment Model Figure 4: VRU-Only Call Flow 1. New call is passed from PSTN to Unified ICME. 2. Unified ICME queues the call by sending a temporary Connect to VRU request to PSTN. 3. PSTN establishes audio connection with VML Gateway. 4. VML Gateway passes new call message to IVR Service. 5. IVR Service passes new call message to ICM Service. 6. ICM Service sends Requests Instruction to Unified ICME. 7. Unified ICME sends Run Script request to ICM Service. 8. ICM Service sends Run Script request to IVR Service. 9. IVR Service creates VoiceML pages based on Run Script instructions received from Unified ICME and sends the VML pages to VML Gateway where the appropriate micro-application is executed. 10. The execution of the micro-application/script invokes the VML Gateway's prompt/collect capabilities (possibly using ASR/TTS or VML Server) and sends result to IVR Service. 11. IVR Service sends result to ICM Service. 12. ICM Service sends result to Unified ICME. 13. Unified ICME sends Run Script request to ICM Service. 14. ICM Service sends Run Script request to IVR Service. 15. IVR Service creates VoiceML pages based on Run Script instructions received from Unified ICME and sends the VML pages to VML Gateway where the appropriate 34

41 Chapter 3: Choosing a Deployment Model Call Flow Models micro-application/script is executed. 16. Agent becomes available; Unified ICME dequeues the call and asks to be disconnected from VML Gateway; Unified ICME passes connect to agent request to PSTN. 17. PSTN establishes audio connection with ACD. 18. ACD informs Unified ICME that call was received by agent. Comprehensive Scenario This scenario combines the Call Director and the VRU-Only scenarios. It provides initial prompt and collect, self service IVR, queuing, and VoIP switching among all manner of Unified CCE and TDM agents. SIP-Based Comprehensive Scenario Table 6: Components Used for SIP-Based Comprehensive Scenario Component SIP Service (part of Call Server) Required Optional Installed (but Inactive) Not Used IVR Service (part of Call Server) ICM Service (part of Call Server) H.323 Service (part of Call Server) VML Server Call Studio Ingress Gateway VML Gateway SIP Proxy Server Gatekeeper Operations Console Reporting Server ASR/TTS Media Server DNS Server Content Services Switch Unified ICME A sample SIP-based Comprehensive call flow progresses as follows: 35

42 Call Flow Models Chapter 3: - Choosing a Deployment Model Figure 5: SIP-Based Comprehensive Call Flow 1. Call goes from PSTN to Ingress Gateway. 2. New call invitation goes from Ingress Gateway to SIP Proxy Server. 3. Invitation is forwarded from SIP Proxy Server to SIP Service. 4. SIP Service, wishing to consult Unified ICME, passes new call message to ICM Service. 5. New call message is passed from ICM Service to Unified ICME. 6. Unified ICME runs a routing script, which requires call to be transferred to VRU, and passes a Connect to VRU request to ICM Service. 7. Connect to VRU request passes from ICM Service to SIP Service. 8. Invitation to Connect to VRU goes from SIP Service to SIP Proxy Server. 9. Invitation (including information about Ingress Gateway) goes from SIP Proxy Server to VML Gateway, which then connects audio path back to Ingress Gateway. 10. Ingress Gateway establishes audio connection with VML Gateway. 11. New call message is passed from VML Gateway to IVR Service. 12. New call message is passed from IVR Service to ICM Service. 13. ICM Service sends Requests Instruction to Unified ICME. 14. Unified ICME sends Run Script request to ICM Service. 15. ICM Service sends Run Script request to IVR Service. 36

43 Chapter 3: Choosing a Deployment Model Call Flow Models 16. IVR Service creates VoiceML pages based on Run Script instructions received from Unified ICME and sends the VML pages to VML Gateway where the appropriate micro-application is executed. 17. The execution of the micro-application/script invokes the VML Gateway's prompt/collect capabilities (possibly using ASR/TTS or VML Server) and sends result to IVR Service. 18. IVR Service sends result to ICM Service. 19. ICM Service sends result to Unified ICME. 20. Unified ICME sends Run Script request to ICM Service. 21. ICM Service sends Run Script request to IVR Service. 22. IVR Service creates VoiceML pages based on Run Script instructions received from Unified ICME and sends the VML pages to VML Gateway where the appropriate micro-application/script is executed. 23. Agent becomes available; Unified ICME dequeues the call and asks to be disconnected from VML Gateway; Unified ICME passes connect to agent request to ICM Service. 24. ICM Service passes connect to agent request to SIP Service. 25. SIP Service passes disconnect VRU request to SIP Proxy Server. 26. SIP Proxy Server passes disconnect to VML Gateway. 27. SIP Service passes connect to agent request to SIP Proxy Server. 28. SIP Proxy Server passes connect to agent request (including information about Ingress Gateway) to Unified CM, which will then connect audio path back to Ingress Gateway. 29. Ingress Gateway establishes audio connection with agent's phone. 30. Unified CM informs Unified ICME that call was received by agent. H.323-Based Comprehensive Scenario Table 7: Components Used for H.323-Based Comprehensive Scenario Component SIP Service (part of Call Server) Required Optional Installed (but Inactive) Not Used IVR Service (part of Call Server) ICM Service (part of Call Server) H.323 Service (part of Call Server) VML Server 37

44 Call Flow Models Chapter 3: - Choosing a Deployment Model Component Call Studio Required Optional Installed (but Inactive) Not Used Ingress Gateway VML Gateway SIP Proxy Server Gatekeeper Operations Console Reporting Server ASR/TTS Media Server DNS Server Content Services Switch Unified ICME The H.323-based call flow is much like the SIP-based call flow. However, the SIP Service and the (optional) SIP Proxy Server are not used. Rather, the H.323 Service and the Gatekeeper are used. The H.323 Service communicates with the IVR Service rather than directly with Unified ICME. The Ingress Gateway consults the Gatekeeper to find the destination for messages, but does not pass messages through it. This is in contrast to the SIP Proxy Server, where messages actually pass through the Proxy Server to the appropriate destination. VML Server (Standalone) Scenario This scenario involves calls arriving through a gateway, and interacting directly with a VML Server to execute VoiceML applications. The gateway performs both ingress and VML functions. This is designed for customers who require a sophisticated VoiceML-based VRU, for applications which in many cases will not need to interact with Unified ICME at all. However, some standalone customers do require the ability to make back-end requests to Unified ICME, without relinquishing control of the call. To fill that need, Unified CVP has a Request ICM Label capability. The application generally acts on its own, but includes a special step to send a query to Unified ICME and receive a response. The query can contain full call context information, as can the response. 38

45 Chapter 3: Choosing a Deployment Model Call Flow Models The Request ICM Label facility gives rise to a number of possibilities: A Self Service application can ask Unified ICME to select an available Unified CCE or ACD agent to which the call should be transferred. Full call context is preserved during the transfer, but queuing would not be possible. A Self Service application can transfer its call to a separate full-blown Unified CVP system for agent selection and queuing. Full call context is preserved throughout. A Self Service application can ask Unified ICME to perform some calculation or ICM Application Gateway transaction that it already knows how to perform, and return the result to the application. A Self Service application can report intermediate or final call data to Unified ICME to be stored in its database. Table 8: Components Used for VML Server (Standalone) Scenario Component SIP Service (part of Call Server) Required Optional Installed (but Inactive) Not Used IVR Service (part of Call Server) ICM Service (part of Call Server) H.323 Service (part of Call Server) VML Server Call Studio Ingress Gateway VML Gateway SIP Proxy Server Gatekeeper Operations Console Reporting Server ASR/TTS Media Server DNS Server Content Services Switch Unified ICME A sample VML Server (Standalone) call flow progresses as follows: 39

46 Call Flow Models Chapter 3: - Choosing a Deployment Model Figure 6: Standalone VML Call Flow 1. New call is passed from PSTN to VML Gateway. 2. VML Gateway passes HTTP request to VML Server. 3. VML Server passes VML document to VML Gateway. 4. VML Gateway executes appropriate script; passes HTTP request to VML Server. 5. VML Server passes VML document to VML Gateway. 6. VML Gateway executes appropriate script; passes HTTP request to VML Server. 7. VML Server passes Request ICM Label for new call to ICM Service using internal message bus (see the discussion in "Special Considerations for standalone Models with ICM Lookup" just below). 8. ICM Service passes Request ICM Label for new call to Unified ICME; an ICM routing script is invoked. 9. Unified ICME passes connect label to ICM Service. 10. ICM Service passes label to VML Server. 11. VML Server passes transfer request based on label to VML Gateway. 12. VML Gateway establishes audio connection with Unified CM. 13. Unified CM informs Unified ICME that call was received by agent. Special Considerations for Standalone Models with ICM Lookup 40

47 Chapter 3: Choosing a Deployment Model Call Flow Models In standalone call flow models, with and without ICM Lookup capability, various combinations of VML Server and Reporting Server are possible. ICM Lookup capability requires a connection hub that both the VML Server and the Unified ICME VRU PIM (Peripheral Interface Manager) can connect to. Internally to Unified CVP, the connection hub is known as a message bus, and there is one always contained in every Call Server, Reporting Server, and Operations Console. Neither the VML Server itself, nor the VRU PIM, contains its own message bus. If a Reporting Server is in the system, one might be tempted to use its message bus for both the VML Server and the VRU PIM to connect to. However, the Reporting Server uses a different high availability paradigm than is used for components which are active in the processing of live calls, and the two paradigms are not compatible. (The same holds true for the Operations Console.) On the other hand, a VML Server can only connect to one message bus. If both Reporting and ICM Lookup are used in the system, then the VML Server must use a message bus which satisfies the high availability requirements of both. There is a way to explicitly deploy a message bus by itself: one can install a Call Server, without enabling any of its services. In some situations this will be the proper approach. The next question is to determine on what physical machine to deploy it so as not to increase the number of physical servers unnecessarily. Given all these considerations, the following table shows a set of guidelines for determining, in standalone call flow models, when and where separate Call Servers need to be deployed, and which components would connect to each other under each set of criteria. Table 9: Physical Placement for Call Server in Standalone Deployments Situation Standalone with no ICM Lookup and no Reporting Placement No Call Server needed Connectivity No special considerations 41

48 Geographic Models Chapter 3: - Choosing a Deployment Model Situation Standalone with ICM Lookup and no Reporting Placement Call Server on VRU PG Connectivity VRU PIM connects to Localhost VML Server connects to VRU PG Standalone with ICM Lookup and Reporting reporting data flow is not large Call Server on VRU PG VRU PIM connects to Localhost VML Server connects to VRU PG Reporting Server connects to VRU PG Standalone with ICM Lookup and Reporting reporting data flow is large Call Server on dedicated machine VRU PIM connects to dedicated Call Server VML Server connects to dedicated Call Server Reporting Server connects to dedicated Call Server Standalone with Reporting and no ICM Call Server on Reporting Server Lookup machine VML Server connects to Reporting Server Note: As the table above indicates in the first row, if standalone is not using the Request ICM Label capability, a Call Server is not required from the standalone perspective. However, since the Reporting Server is connected to the Call Server, you must install a Call Server even if its Services are inactive in order to be able to use reporting. Geographic Models This section presents the options available in terms of the geographic distribution required. That is, we present choices according to where components are physically positioned, and what kind of network connection links them together. In discussing geographic distribution, we consider the location of call centers (where agents are positioned), of data centers (where Unified CMs and Unified ICME and Unified CVP solution components are located), and of ingress points (where Ingress Gateways are located). However, generally speaking, the locations of the call centers are not relevant to Unified CVP, except in terms of network provisioning, which is beyond the scope of this discussion. 42

49 Chapter 3: Choosing a Deployment Model Geographic Models The geographic scenarios discussed are: Centralized Single-Site Centralized Multi-Site Centralized Branch Standalone Branch Note: A couple of geographic considerations have already been discussed with regard to specific call flow models. These are "Special Considerations for standalone Models with ICM Lookup", which was discussed under the standalone VML call flow scenario; and "SIP Coresident with VRU PG", which was discussed under the Call Director call flow scenario. Centralized Single-Site Geography This geography is most suited to small to medium centralized enterprises. It places all Unified CVP solution components at one site, including the Ingress Gateways. Agents need not be physically located at the data center. Callers generally dial a small array of toll-free numbers, paid for by the enterprise, to reach Unified CVP. An entire site outage obviously brings down the whole operation, but failures of individual components cause minimal operational impact if the solution is properly designed for high availability. For this geography, all of the call flow models are supported. Centralized Multi-Site Geography This geography is identical to the Single-Site version, except that many or all components are duplicated at two or more data centers. The sites can be designed such that one remains completely dormant unless one of its sibling sites is disabled, in which case it takes over all the traffic from the disabled site. Or, they can be designed such that activity is distributed across all sites all the time, but that each site contains enough excess capacity so that traffic from one failed site can be absorbed by the remaining sites. Under this geography, callers generally dial the same small array of toll-free numbers that were used in the Single-Site version. The PSTN distributes these calls across ingress points at all the sites. Once calls enter a site and need to be handled by various components, Unified CVP can be configured to prefer that call handling remain within one site, with the ability to involve components at other sites if the capacity is needed or if individual components fail over. As with the Single-Site geography, all of the call flow models are supported. Centralized Branch Geography Centralized Branch geography is very much the same as the two previous Centralized geographies, except that a combined Ingress and VML Gateway is physically located in each branch office. A retailer with a thousand stores, for example, would place a gateway in each 43

50 Geographic Models Chapter 3: - Choosing a Deployment Model store, but the rest of the solution components would reside in one or more data centers. Agents could also be located either in the branch, in the next nearest branch, in centralized call centers, or any combination thereof. Under this geography, callers dial local branch phone numbers, rather than centralized phone numbers, and the caller pays the toll charges. Furthermore, if the caller needs to be queued, his voice stream remains in the branch gateway until an agent is available. This preserves network bandwidth. All of the call flow models are supported with one partial exception: ASR is not generally used with this geography. This is because ASR requires G.711 encoding, which is very bandwidth intensive, and bandwidth from branch offices to data centers is usually at a premium. Users who require this capability should contact their Cisco support representative. Note, however, that TTS is not affected by the restriction that applies to ASR. Standalone Branch Geography The VML Server (Standalone) call flow scenario describes a combination Ingress/VML Gateway together with a VML Server, not integrated with Unified ICME (except potentially as a back-end server). If standalone is deployed in a Centralized geography, then one might see a data center containing some number of gateways sharing some number of VML Servers, each with the ability to make requests to back-end databases and servers which are also located in the data center. One Branch version of this is to place Ingress/VML Gateways at branch offices, and have them share some number of VML Servers located in the data center, which have the ability to access back-end databases and servers, also located in the data center. This leads to a potentially large number of gateways, but an easily manageable number of VML Servers (Content Services Switches should also be used here, for load balancing and fault tolerance). However, a problem that some retail customers have run into has to do with the relative unreliability of the WAN link between branch and data center. If that link goes down, then even local self service activities cannot take place. A second Branch version is to move the VML Servers to the branch. They may need access to centralized back-end services, but they can still provide local-only VRU services if the WAN goes down. Several challenges come with this geography, however: A VML Server requires an extra box in each branch A VML Server is not an appliance like the IOS gateway it may need to be locally managed by branch office personnel The prospect of remotely managing perhaps thousands of VML Servers is potentially daunting though the Operations Console available with Unified CVP 4.0 and later provides convenient handling not available with earlier versions of CVP Given these just-mentioned considerations, it is usually preferable to use the first Standalone Branch version if your WAN provisioning is reliable enough. 44

51 Chapter 3: Choosing a Deployment Model Physical Models Note that ASR is not generally used with either Standalone Branch version. This is because ASR requires G.711 encoding, which is very bandwidth intensive, and bandwidth from branch offices to data centers is usually at a premium. Users who require this capability should contact their Cisco support representative. Note, however, that TTS is not affected by the restriction that applies to ASR. Physical Models Physical models are essentially hardware configurations. They describe which logical servers are installed on which physical servers. Physical models are to a large extent determined by what geographic model you are following, the size of your system, and your performance requirements. With regard to these considerations, see the discussion of geographic models above, the Hardware and Software System Specification for Cisco Unified Customer Voice Portal Software Release 4.1(1), and the Cisco Unified Customer Voice Portal (CVP) Release 4.0 Solution Reference Network Design (SRND) document. There are four basic physical models. 1. Typical for Unified ICME Integrated: - Call Server (with IVR/H.323/SIP/ICM Services or combinations thereof) - VML Server - Reporting Server - Operations Console 2. Typical for Standalone: - Call Server (with only ICM Service activated) on PG (if using ICM Lookup) - VML Server - Reporting Server - Operations Console 3. Streamlined for SIP Call Director: - Call Server (with only SIP and ICM Services activated) on PG - Reporting Server (possible, though unlikely) - Operations Console 4. Laboratory (All-in-a-Box): This model, in which all Unified CVP components can be installed on a single physical machine, is intended solely for testing purposes and is not supported as a deployed system. 45

52 Physical Models Chapter 3: - Choosing a Deployment Model An All-in-a-Box server, with the reporting option, is required to have at least 50 GB of free space. The server must have at least two mirrored drives, for protection against the failure of a single drive. It does not support more than 10 simultaneous calls. 46

53 Performance, Sizing, and Choosing Hardware The subjects of performance, sizing, and hardware are intimately related. Factors that determine your choices include: call flow used geographic model followed overall system size failover strategy reporting usage (if any) Chapter 4 Specifics that are of special importance include: number of calls being treated at times of highest usage, and the states of these calls number of legs being handled by Call Servers what gateways are being used, for what purpose, how many calls per second are being handled, what is the maximum number of concurrent calls, what is the maximum number of VoiceML sessions if a reporting database is being used, database size is affected by the number of days of data to be retained, which in turn depends on configuration and data persistence choices For detailed discussions on performance, sizing, and hardware, please refer to the Cisco Unified Customer Voice Portal (CVP) Release 4.0 Solution Reference Network Design (SRND) document, and the Hardware and Software System Specification for Cisco Unified Customer Voice Portal Software Release 4.1(1). 47

54 48 Chapter 4: - Performance, Sizing, and Choosing Hardware

55 Chapter 5 Planning for Reporting The current chapter mentions a few topics that you must be aware of if you plan to use Unified CVP reporting. However, for a detailed discussion of this subject, see the Reporting Guide for Cisco Unified Customer Voice Portal, the Cisco Unified Customer Voice Portal (CVP) Release 4.0 Solution Reference Network Design (SRND) document, and the Hardware and Software System Specification for Cisco Unified Customer Voice Portal Software Release 4.1(1). This section contains the following topics: Sizing, page 49 Backup and Restore, page 49 Synchronizing Timestamps, page 50 Sizing The Unified CVP 4.0 reporting solution deployment options, together with related sizing requirements, are discussed in the Cisco Unified Customer Voice Portal (CVP) Release 4.0 Solution Reference Network Design (SRND) document. Backup and Restore Unified CVP utilizes RAID as protection against failure of a single drive in a mirrored pair. However, RAID 10 will not protect against the loss of a site, loss of a machine, or a loss of both mirrored drives. Unified CVP allows customers, by means of the Operations Console, to schedule daily database backups or to run database backups on-demand. This allows the customer to manually restore the database if needed to the last backup time, so that the worst case scenario is losing about 24 hours worth of data. 49

56 Synchronizing Timestamps Chapter 5: - Planning for Reporting Database backups are written to the local database server. However, storing backups only on a local machine does not protect the customer against server failure or the loss of a site. Cisco recommends that Unified CVP customers copy the backup files to a different machine, preferably at a different location. Customers who choose to do this must assume all security and backup management responsibilities. Database backups are essentially the same size as the originating database. Due to disk size limitations, Unified CVP can store a maximum of two backups. Customers who wish to store more copies of database backups must copy the backups to another location. Database restore is not supported through the Operations Console. To restore the Unified CVP database, a customer must manually run the Informix command from a command prompt. Synchronizing Timestamps Call Servers, VML Servers, and Reporting Servers must have their clocks synchronized in order to assure accurate timestamps in both the database and log files. Since Unified CVP components do not themselves synchronize machine times, a cross-component time synchronization mechanism, such as NTP, must be used. 50

57 Creating a Failover Strategy (Designing for High Availability) Unified CVP is designed with high availability in mind. Most Unified CVP call flow components those components that carry call traffic are designed for n+1 redundancy. If any single component fails, there is no caller-observable impact to the handling of new incoming calls, except possibly a prorated reduction in total call capacity. However, calls that are in progress using the impacted component may be affected, though they will usually be caught by the Survivability script on the ingress gateway and given default treatment or transferred to a default target rather than dropped. The Unified CVP solution components that support n+1 redundancy are: Call Server, VML Server, Ingress Gateway, VML Gateway, Media Server, and ASR/TTS Servers. However, Media Server and ASR/TTS Servers can only be duplexed if a Content Services Switch is not used. A second set of Unified CVP solution components SIP Proxy Server, Gatekeeper, Content Services Switch allow for redundant pair (duplex) configuration. Connections to the Reporting Server are resilient: information is buffered at the source in case of a connection failure, and then transmitted to the Server when it later becomes available. The Operations Console does not support redundancy because the ability to access and update configurations is not usually real-time critical. Similarly, Call Studio is an offline tool for application script development, which does not require high availability. Note: The material included in this chapter is intended to compliment, not to replace, the high availability information provided in the Cisco Unified Customer Voice Portal (CVP) Release 4.0 Solution Reference Network Design (SRND) document. You are urged to consult that document as well. This section contains the following topics: High Availability SIP-Based Call Flow, page 52 High Availability H.323-Based Call Flow, page 58 Survivability of Existing Calls, page 63 Chapter 6 51

58 High Availability SIP-Based Call Flow Chapter 6: - Creating a Failover Strategy (Designing for High Availability) Non-Call Handling Components, page 65 High Availability SIP-Based Call Flow Note: It is highly recommended that you use UDP instead of TCP for SIP signaling. TCP stack timeout delays can cause significant delays to the caller during failures. This section describes the relationship among Unified CVP solution components from the perspective of a SIP-based call flow. We begin with a hypothetical comprehensive model incoming call, and trace its effect on each component with a particular view toward how that component finds the next component downstream. In the process, we will be describing the system s high availability architecture. Please refer to the following diagram. Figure 7: SIP-Based High Availability Component Layout Table 10: High Availability SIP-Based Call Flow Although not shown in the diagram, a Media Server can be connected to the VML Gateway. Contact Server Switches (CSSs) can be placed between the VML Gateway and the VML Server, between the VML Gateway and the ASR/TTS Server, between the VML Gateway and the Media Server. Path New call from PSTN to Ingress Gateway New Call from Ingress Gateway to SIP Proxy Server Scalability Method Multiple Ingress Gateways can be deployed, each with its own array of TDM connections. High Availability Method PSTN should be contracted to fail over to other TDM connections when the selected connection is down. Up to two SIP Proxy Servers are supported. If the selected SIP Proxy Server either does not Either an Ingress dial peer respond within a time-out can reference DNS (which period, or responds with can return either SIP a 500-level message to Proxy Server), or multiple reject the call, the Ingress dial peers can be Gateway will try the other configured with SIP Proxy Server. If no preference attributes to SIP Proxy Server accepts cause one SIP Proxy the call, the Ingress 52

59 Chapter 6: Creating a Failover Strategy (Designing for High Availability) High Availability SIP-Based Call Flow Path New Call from SIP Proxy Server to Call Server (SIP Service) New Call from Ingress Gateway to Call Server (SIP Service) [for deployments which do not use a SIP Proxy Server] New Call from Call Server (SIP Service) to VRU PG Scalability Method Server to be preferred over the other. High Availability Method Gateway will not answer the call from the PSTN. The SIP Proxy Server is If the selected endpoint configured via a static either does not respond route table with all SIP within a time-out period, Service endpoints; each or responds with a endpoint also has a 500-level message to preference value. The reject the call, the SIP Proxy Server selects the Proxy Server will try the endpoint with the highest next most preferred preference value, or will endpoint in its static route round robin among table. If no endpoint endpoints with the same accepts the call, the SIP preference value. Proxy Server rejects the call by returning a 500-level message to the Ingress Gateway. The SIP Service may also reject the call if other required components in the same Call Server will not be able to perform their tasks. Multiple Call Servers can If the selected SIP Service be configured using either does not respond successive dial peers on within a time-out period, the gateway and varying or responds with a preference values. The 500-level message to Ingress Gateway selects reject the call, the Ingress the Call Server with the Gateway will try the next highest preference value, most preferred endpoint or will round robin among according to its dial-peer endpoints with the same configuration. If no Call preference value. Pre-established relationship. If this static connection is not operational, the SIP Server accepts the call, the Ingress Gateway will not answer the call from the PSTN. (Note: This is a trial-and-error type of failover.) The time-out period is configurable in the IOS configuration for SIP User Agent it is the "expires" timer value. The VRU PG can be deployed in pairs. If one side goes down, the other side immediately creates 53

60 High Availability SIP-Based Call Flow Chapter 6: - Creating a Failover Strategy (Designing for High Availability) Path New Call from VRU PG to Unified ICME VRU Transfer label from VRU PG to Call Server (SIP Service) Sticky Gateway Semantics (that is, SetTransfer-Label; inbound call always uses same VML Gateway) Non-Sticky Gateway Semantics (that is, inbound call can transfer to any VML Gateway) VRU Transfer Label from Call Server VRU Transfer label from Call Server (SIP Service) to SIP Proxy Server VRU Transfer label from SIP Proxy Server to VML Gateway Scalability Method Service will reject the incoming call from the SIP Proxy Server (or the Ingress Gateway if there is no SIP Proxy Server). High Availability Method a connection to the Call Server. Pre-established The Unified ICME relationship. If this static CallRouter may return a connection is not BUSY, RNA, or operational, the VRU PG NETWORK will not connect to the CONGESTION label to Call Server. the ICM Service, which will be returned to the SIP Service. These cause the SIP Service to return a 500-level message to the SIP Proxy Server (or to the Ingress Gateway if there is no SIP Proxy Server) and reject the call. Pre-established relationship. Message goes directly back to the Ingress Gateway. Message goes to one of the SIP Proxy Servers which are statically configured by IP address or DNS hostname in the SIP Service. The SIP Proxy Server is configured via a static route table with all VML Gateway endpoints; each endpoint N/A Ingress Gateway is assumed to be able to accept a SIP transfer request. If the selected SIP Proxy Server either does not respond within a time-out period, or responds with a 500-level message, the SIP Service will try the next most preferred SIP Proxy Server in its static table. If no endpoint accepts the call, the SIP Service rejects the inbound call leg by sending a 500-level message back to the inbound SIP Proxy Server. If the selected endpoint either does not respond within a time-out period, or responds with a 500-level message to 54

61 Chapter 6: Creating a Failover Strategy (Designing for High Availability) High Availability SIP-Based Call Flow Path VRU Transfer label from Call Server (SIP Service) to VML Gateway [for deployments which do not use a SIP Proxy Server] HTTP New Call from VML Gateway to Call Server (IVR Service) Request Instruction from IVR Service to VRU PG and Unified ICME Micro-application Request from Unified ICME and VRU PG to IVR Service VML Micro-application Request from IVR Service to VML Gateway Request from VML Gateway to MRCP Server Scalability Method High Availability Method also has a preference reject the call, the SIP value. The Proxy Server Proxy Server will try the selects the endpoint with next most preferred the highest preference endpoint in its static route value, or will round robin table. If no endpoint among endpoints with the accepts the call, the SIP same preference value. Proxy Server rejects the DNS can also be used to call by returning a provide multiple 500-level message to the endpoints for the same SIP Service, which sends hostname. a 500-level message back to the SIP Proxy Server to reject the inbound call leg. Message goes to one of the VML Gateways which are statically configured by IP address or DNS hostname in the SIP Service. If the selected VML Gateway either does not respond within a time-out period, or responds with a 500-level message, the SIP Service will try the next most preferred VML Gateway in its static table. If no endpoint accepts the call, the SIP Service rejects the inbound call leg. Since call was transferred The Call Server which from a Call Server, then sent the transfer is the New Call HTTP assumed to be able to request always goes back accept the transfer as well. to the same Call Server. This forces both legs of a call to use the came Call Server. Pre-established relationship. Pre-established relationship. Pre-established relationship. N/A N/A N/A CSS can be used to scale ASR/TTS requests to an - If the request is a micro-application request, array of servers. If no and CSS is not used, then CSS is used, then these the Unified ICME routing requests which originate script can identify a locale as micro-applications can name, which maps to an support one primary and ASR/TTS server with one 55

62 High Availability SIP-Based Call Flow Chapter 6: - Creating a Failover Strategy (Designing for High Availability) Path Scalability Method one backup ASR/TTS server for each locale. Without a CSS, requests from VML Server applications can only go to a single ASR/TTS server defined in the gateway. High Availability Method backup. The IVR Service always tries the primary ASR or TTS Server first, and fails over to the backup if necessary, remaining there for the life of the call. Subsequent calls try the primary first again. - If the request originates from a VML Server script and CSS is not used, then there is no automatic failover to a backup ASR/TTS server. Request from VML Gateway to Media Server Request from VML Gateway to VML Server HTTP Call Result from VML Gateway to IVR Service Micro-application Result from IVR Service to VRU PG and Unified ICME - In either case, if a CSS is used, it implements its own automatic failover to an array of ASR/TTS servers. Unified ICME routing IVR Service always tries script identifies a Media the primary Media Server Server and one backup, or first, and fails over to the if CSS is used, an array of backup if necessary, Media Servers can be remaining there for the used. life of the call. Subsequent calls try the primary first again. Note that most files fetched from the Media Server remain cached on the VML Gateway. (The above happens only if CSS is not used.) Unified ICME routing If request fails, routing script identifies a VML script may select a Server by name or IP different server and try address. If CSS is used, a again. There is no farm of VML Servers automatic failover to a can be used. backup VML Server without a CSS. Pre-established relationship. Pre-established relationship. N/A N/A 56

63 Chapter 6: Creating a Failover Strategy (Designing for High Availability) High Availability SIP-Based Call Flow Path Agent label from Unified ICME and VRU PG to SIP Service Agent label from SIP Service to SIP Proxy Server Transfer from SIP Proxy Server to target endpoint Agent label from SIP Service to target endpoint [for deployments which do not use a SIP Proxy Server] Scalability Method Pre-established relationship. Message goes to one of the SIP Proxy Servers which are statically configured by IP address or DNS hostname in the SIP Service. High Availability Method N/A If the selected SIP Proxy Server either does not respond within a time-out period, or responds with a 500-level message, the SIP Service will try the next most preferred SIP Proxy Server in its static table. If no SIP Proxy Server accepts the call, the SIP Service returns an error event to Unified ICME. If Unified ICME has requery enabled, this may cause the same sequence to repeat for each requery label, with the same results. SIP Proxy Server may use If no endpoint accepts the its static route table with transfer, or no appropriate preferences, optionally in endpoint is configured in combination with DNS, the Proxy Server, then a or consult its dynamic 500-level message is registration tables, to try returned to the SIP multiple endpoints. Service, which returns an error event to Unified ICME. Unified ICME executes a requery. Message goes to one of the endpoints which are statically configured by IP address or DNS hostname in the SIP Service. If the selected endpoint does not accept the transfer, the SIP Service will try the next most preferred endpoint in its static table. If no appropriate endpoint accepts the call, the SIP Service returns an error event to Unified ICME. If Unified ICME has requery enabled, this may cause the same sequence to repeat for each requery label, with the same results. 57

64 High Availability H.323-Based Call Flow Chapter 6: - Creating a Failover Strategy (Designing for High Availability) High Availability H.323-Based Call Flow This section describes the relationship among Unified CVP solution components from the perspective of an H.323-based call flow. We begin with a hypothetical comprehensive model incoming call, and trace its effect on each component with a particular view toward how that component finds the next component downstream. In the process, we will be describing the system s high availability architecture. Please refer to the following diagram. Figure 8: H.323-Based Unified CVP Solution Table 11: High Availability H.323-Based Call Flow Although not shown in the diagram, a Media Server can be connected to the VML Gateway. Contact Server Switches (CSSs) can be placed between the VML Gateway and the VML Server, between the VML Gateway and the ASR/TTS Server, between the VML Gateway and the Media Server. Path New call from PSTN to Ingress Gateway New Call from Ingress Gateway to Call Server (H.323 Service) Scalability Method Multiple Ingress Gateways can be deployed, each with its own array of TDM connections. High Availability Method PSTN should be contracted to fail over to other TDM connections when the selected connection is down. If dial peer session target is RAS (Registration, H.323 Services that are not ready to take calls are Admission, and Status), not registered with the then one of up to two gatekeeper and will not be gatekeepers are consulted given calls. If no H.323 for the IP address of an Services are available, the H.323 Service that is gatekeeper returns no known to be available and destination, and the ready to take calls. Ingress Gateway will Gatekeeper can distribute reject the call. calls to targets in round Alternatively, this robin or preference order, situation can be caught by and can also track the Survivability script. 58

65 Chapter 6: Creating a Failover Strategy (Designing for High Availability) High Availability H.323-Based Call Flow Path HTTP New Call from H.323 Service to IVR Service New Call from Call Server (IVR Service) to VRU PG New Call from VRU PG to Unified ICME VRU Transfer label from VRU PG to Call Server (IVR Service) VRU Transfer label from IVR Service to H.323 Service Scalability Method bandwidth usage to each target. H.323 Service can now (in Unified CVP 4.0 and later) only send calls to the IVR Service that is coresident with it on the same Call Server. Pre-established relationship. High Availability Method If the coresident IVR Service is not ready to take calls, then the H.323 Service takes itself "out of service" (note that an "out of service" component continues to handle active calls). An out of service H.323 Service deregisters itself from the Gatekeeper, so new calls are no longer sent to it. The VRU PG can be deployed in pairs. If one side goes down, the other side immediately creates a connection to the Call Server. If neither side is operational, the IVR Service will stop responding to long polls from the H.323 Service. Since this is the only IVR Service that this H.323 Service is configured for, the H.323 Service takes itself out of service, deregisters from the Gatekeeper, and does not receive new calls. Pre-established If the Unified ICME relationship. If this static CallRouter cannot accept connection is not the call, a failure is operational, the VRU PG returned to the VRU PG will not connect to the which fails the entire call. Call Server. There is no way to prevent the call from being answered by the Ingress Gateway. Pre-established relationship. Pre-established relationship. N/A N/A 59

66 High Availability H.323-Based Call Flow Chapter 6: - Creating a Failover Strategy (Designing for High Availability) Path Sticky Gateway Semantics (that is, SetTransfer-Label; inbound call always uses same VML Gateway) Non-Sticky Gateway Semantics (that is, inbound call can transfer to any VML Gateway) VRU Transfer Label from H.323 Service Directory query from H.323 Service to Gatekeeper Scalability Method Message goes directly back to the Ingress Gateway. High Availability Method Ingress Gateway should be able to accept a transfer request back to itself. Request goes to the If the Gatekeeper goes Gatekeeper, which returns down, its HSRP partner the IP addresses of one or begins handling requests more VML Gateways. at the same IP address. If Gatekeeper can distribute the entire pair is down, calls to targets in round the H.323 Service takes itself out of service and will not receive new calls. robin or preference order, and can also track bandwidth usage to each target. Alternatively, H.323 Service now supports Alternate Gatekeepers. You can configure VRU Transfer label from H.323 Service Message is sent to the to VML Gateway VML Gateway specified by the Gatekeeper. HTTP New Call from VML Gateway to Call Server (IVR Service) multiple gatekeepers; if the H.323 Service s primary one stops responding, the H.323 Service will make the next one primary and use that until it stops responding. If alternate endpoints are configured in the Gatekeeper, then the H.323 Service will attempt to transfer the call to each VML Gateway defined there. If no VML Gateway can accept the transfer, H.323 Service sends a connect failure message back to IVR Service, which sends a failure back to Unified ICME. Since call was transferred The Call Server that sent from a Call Server, the the transfer is assumed to New Call HTTP request be able to accept the always goes back to the transfer as well. same Call Server. This forces both legs of a call to use the came Call Server. 60

67 Chapter 6: Creating a Failover Strategy (Designing for High Availability) High Availability H.323-Based Call Flow Path Request Instruction from IVR Service to VRU PG and Unified ICME Micro-application Request from Unified ICME and VRU PG to IVR Service VML Micro-application Request from IVR Service to VML Gateway Request from VML Gateway to MRCP Server Scalability Method Pre-established relationship. Pre-established relationship. Pre-established relationship. High Availability Method N/A N/A N/A CSS can be used to scale ASR/TTS requests to an - If the request is a micro-application request, array of servers. If no and CSS is not used, then CSS is used, then those the Unified ICME routing requests which originate script can identify a locale as micro-applications can name, which maps to an support one primary and ASR/TTS server with one one backup ASR/TTS backup. IVR Service server for each locale. always tries the primary Without a CSS, requests ASR or TTS Server first, from VML Server and fails over to the applications can only go backup if necessary, to a single ASR/TTS remaining there for the server defined in the life of the call. gateway. Subsequent calls try the primary first again. - If the request originates from a VML Server script and CSS is not used, then there is no automatic failover to a backup ASR/TTS server. Request from VML Gateway to Media Server - In either case, if a CSS is used, it implements its own automatic failover to an array of ASR/TTS servers. Unified ICME routing IVR Service always tries script identifies a Media the primary Media Server Server and one backup, or first, and fails over to the if CSS is used, an array of backup if necessary, Media Servers can be remaining there for the used. life of the call. Subsequent calls try the primary first again. Note that most files fetched from the Media Server remain cached on the VML Gateway. This 61

68 High Availability H.323-Based Call Flow Chapter 6: - Creating a Failover Strategy (Designing for High Availability) Path Request from VML Gateway to VML Server HTTP Call Result from VML Gateway to IVR Service Micro-application Result from IVR Service to VRU PG and Unified ICME Agent label from Unified ICME and VRU PG to IVR Service Agent label from IVR Service to H.323 Service Directory query from H.323 Service to Gatekeeper Scalability Method High Availability Method algorithm is not in effect if CSS is used. Unified ICME routing If request fails, routing script identifies a VML script may select a Server by name or IP different server and try address. If CSS is used, a again. There is no farm of VML Servers automatic failover to a can be used. backup VML Server without a CSS. Pre-established relationship. Pre-established relationship. Pre-established relationship. Pre-established relationship. N/A N/A N/A N/A Request goes to the If the Gatekeeper goes Gatekeeper, which returns down, its HSRP partner the IP addresses of one or begins handling requests more endpoints at the same IP address. If represented by the agent the entire pair is down, label. the H.323 Service takes itself out of service and will not receive new calls. Agent transfer instructions from H.323 Service to agent destination Call is sent to Unified CM and IP phone specified by the Gatekeeper. Alternatively, H.323 Service now supports Alternate Gatekeepers. You can configure multiple gatekeepers; if the H.323 Service s primary one stops responding, the H.323 Service will make the next one primary and use that until it stops responding. If the Gatekeeper has alternate endpoints configured for the agent destination, then the H.323 Service will try each endpoint in succession. If no target can accept the transfer, 62

69 Chapter 6: Creating a Failover Strategy (Designing for High Availability) Survivability of Existing Calls Path Scalability Method High Availability Method H.323 Service sends a connect failure message back to IVR Service, which sends a failure back to Unified ICME. Survivability of Existing Calls Table 12: High Availability Call Survivability The sections above touch on high availability with respect to handling of new incoming calls in the face of failed components. In this section we discuss the handling of existing calls when components fail. If this component fails Ingress Gateway SIP Proxy Server Call Server (SIP Service) Existing calls experience this Call is lost. [SIP or H.323] If single Proxy Server, call will be lost. But if proxy pair is deployed, further SIP messaging will simply go through the other Proxy Server in the pair. Even if failure occurs during an Invite transaction, the Ingress Gateway will time-out waiting for a response and continue the transaction with the other Proxy Server in the pair. Media path remains connected to the VML Gateway or to the agent, but signaling through the SIP and IVR Services is lost. If connected to the VML Gateway, the IOS Voice Browser will time-out, but will not be able to inform the Ingress Gateway of the failure. The Ingress Gateway will time-out either its SIP session timer or its RTP media inactivity timer and invoke the Survivability script. If connected to an agent, the parties can continue talking for some time. The RTP media inactivity timer will not expire, but the SIP session timer will time-out and try to "refresh" the session. This involves a new Invite transaction, which, if the Call Server came back up in the meantime, will succeed and essentially restart the call. If refreshing fails, then the call is dropped. VRU PG [SIP] ICM Service sends a state change event of partial service. IVR Service sends the partial service state change to the IOS Voice Browser in a long-poll response. IVR Service sends a 5xx response to any pending or new request from the IOS Voice Browser. If that causes the call to be dropped, the SIP Service will be notified by the Ingress Gateway and 63

70 Survivability of Existing Calls Chapter 6: - Creating a Failover Strategy (Designing for High Availability) If this component fails Existing calls experience this drop the call. If SIP Service gets any new requests, it sends a 5xx message back to the requester. If the caller was talking to an agent, then the parties can continue to talk, but further call control through Unified ICME will not be possible. IVR Service returns an HTTP 5xx message to the VML Gateway in response to any active requests and any new requests it receives. The VML Gateway sends a SIP BYE message to the SIP Service, containing a reason code. SIP Service forwards that SIP BYE message with the reason code to the Ingress Gateway, which invokes Survivability if the reason code is non-normal. Thus, calls which are at the VRU will be handled by Survivability. If the caller was talking to an agent, then the parties can continue to talk, but further call control through Unified ICME will not be possible. [H.323] The IVR Service sends a state change to the H.323 Service, not the IOS Voice Browser mentioned above. The H.323 Service will then attempt to route calls to the next Call Server it has in its list. If there are no other options, the H.323 Service will reject new call requests. If a call is terminated for any abnormal reason after the call is connected, then the survivability script on the Ingress Gateway will play back a critical error message. Unified ICME VML Gateway A second Call Server is not supported in Unified CVP 4.0 and later. If the first Call Server fails, the H.323 Service will reject new call requests. Same as when VRU PG fails. [SIP] If IVR Service does not receive a new request from the VML Gateway within 7200 seconds, it will terminate the call. It will report the error, but will remain in full service to handle requests from other VML Gateways. The Ingress Gateway will time-out its SIP Session Timer (default: 1800 seconds), and issue a new Invite via the SIP Service to the VML Gateway. SIP Service will not be able to forward the Invite, and will return a 5xx message to the Ingress Gateway, containing a non-normal reason code. The call will then be handled by the Survivability script. [H.323] If IVR Service does not receive a new request from the VML Gateway within 7200 seconds, it will terminate the 64

71 Chapter 6: Creating a Failover Strategy (Designing for High Availability) Non-Call Handling Components If this component fails Existing calls experience this call. If the IOS Voice Browser does not get a response from the IVR Service within 7320 seconds, it will terminate the call. The call will then be handled by the Survivability script. Non-Call Handling Components The sections above deal with call-oriented availability requirements. The Unified CVP Solution also contains components that are not strictly call-handling components, but which have their own high availability requirements nonetheless. Table 13: High Availability Design (Non-Call Handling Components) Type of Traffic Reporting events from Call Server and VML Server to the Reporting Service Scalability Method High Availability Method Each Call Server and each VML - If any Call Server or VML Server Server can be associated with only one loses connectivity to its Reporting Reporting Service. Reports cannot span Service, it will send an SNMP event, multiple Informix databases. and start persisting reporting events on disk, until a preconfigured file size threshold is reached. At that point it will issue another SNMP event, and begin dropping reporting events. Configuration data between the Operations Console and its managed components. Only one Operations Console is supported. - If a Reporting Service loses the ability to write to a database management system, it will disconnect from its reporting event sources, causing those servers to begin storing their events. Since events travel asynchronously, it is possible for the Reporting Service to receive (and store) some number of additional events before they stop coming. When the links are restored, all reporting event sources begin draining their stored events in FIFO order, though the transmission rate is throttled to prevent any overloads. High availability is not generally required for this component because the ability to access and update configurations is not usually real-time critical. 65

72 Non-Call Handling Components Chapter 6: - Creating a Failover Strategy (Designing for High Availability) 66

73 Chapter 7 Licensing Call Servers are licensed on both a per-instance basis and port usage. For Unified CVP 4.0 and later, customers will receive a license to operate the server at maximum capacity with the server license and software (a separate part must be ordered for each server the software will reside on). For SIP Service and IVR Service, customers will be provided a 'paper' right-to-use license for the number of ports ordered. Customers who order the Call Director option will receive a complete Call Server license, but no VML Server license. While the licensing does not disable queuing or caller segmentation, customers who use this product are only entitled to call control. With Unified CVP server licensing, when the maximum number of licenses shipped with a server is exceeded, a Call Server places itself into Partial Service. Partial Service means that existing calls continue, but new calls are not accepted on this Call Server. Once the number of ports in use falls below the licensed limit, the Call Server places itself back into Full Service and begins accepting new calls again. Please note that this licensing scheme may change in future releases, and should customers order an insufficient number of licenses, they will be impacted in future releases when licensing tracks the number of ports actually ordered. For more comprehensive information on licensing, please see the Cisco Unified Customer Voice Portal (CVP) Release 4.0 Solution Reference Network Design (SRND) document. 67

74 68 Chapter 7: - Licensing

75 Planning Network Topology This chapter comments on some key factors that affect network topology. For detailed discussion of network considerations, please refer to the Cisco Unified Customer Voice Portal (CVP) Release 4.0 Solution Reference Network Design (SRND) document. This section contains the following topics: Selecting Codecs, page 69 WAN vs. LAN, page 69 High Availability VLANs, page 70 ASR and TTS, page 70 Quality of Service (QoS), page 70 Security Best Practices, page 70 Chapter 8 Selecting Codecs Generally, G.711 should be used. G.729 is typically used when RTP traffic must traverse a WAN link, in order to save bandwidth. ASR cannot use silence suppression and must use the G.711 codec. This applies to the entire RTP stream, from Ingress Gateway to VML Gateway to ASR Server; all must be G.711, no silence suppression. WAN vs. LAN The use of WAN vs. LAN is determined by the geographic model that you have chosen. See the discussion in the "Geographic Models" section of Chapter 3, "Choosing a Deployment Model" (page 27). 69

76 High Availability VLANs Chapter 8: - Planning Network Topology High Availability VLANs Two Layer 2 switches should comprise a single VLAN. There are two reasons for this design: 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. A Content Services Switch (CSS) and its redundant partner must reside on the same VLAN in order to send keep-alive messages to each other. If one of the Layer 2 switches fails, one CSS is still functional. ASR and TTS The VoiceML gateway sends MRCP requests to the ASR/TTS servers in order to perform voice recognition and text-to-speech instructions that are defined in a VoiceML document. The ASR/TTS high-availability configuration and behavior differ between Standalone and ICM-integrated deployments, Quality of Service (QoS) Unified CVP implements Layer 3 QoS defaults on all relevant network paths, and provides a management interface via the Operations Console to modify QoS settings at each end of specifically designated data paths. Changes in QoS settings require restart or reboot to take effect. Security Best Practices Refer to the "Enabling Security" chapter in the Configuration and Administration Guide for Cisco Unified Customer Voice Portal, as well as to the information provided in the Cisco Unified Customer Voice Portal (CVP) Release 4.0 Solution Reference Network Design (SRND) document. 70

77 Chapter 9 Creating Dial Plans Creating dial plans in a SIP environment differs from creating dial plans in an H.323 environment. Please see the Cisco Unified Customer Voice Portal (CVP) Release 4.0 Solution Reference Network Design (SRND) document for additional information. 71

78 72 Chapter 9: - Creating Dial Plans

79 Chapter 10 Scripting Alternatives: VoiceML vs. Unified ICME Scripting Two service creation environments (script editors) are available in the Unified CVP solution. Sophisticated IVR applications can be developed using Call Studio. This is an Eclipse-based service creation environment whose output is an intermediary file which describes the application flow. That file gets loaded onto the VML Server machines for execution. To invoke a VML Server application, the script writer includes a special micro-application in his Unified ICME routing script. This micro-application actually instructs the VML Gateway to interact with the VML Server directly to execute the application. Final results are then passed back to Unified ICME. Among its many features, the VoiceML scripting environment has a drag-and-drop interface with a palette of IVR functions, can do database queries, and can even be extended with Java code written to perform any task a Java application could perform. The ICM Script Editor is used to develop agent routing scripts, and to invoke Unified CVP micro-applications basic building blocks of a voice interaction design. The Unified CVP micro-applications are Play Media, Get Speech, Get Digits, Menu, Play Data, and Capture. They are combined and customized in the Unified ICME routing script to produce a viable voice interaction with the caller. While it is possible to develop full scale IVR applications using micro-applications, it is not recommended. Micro-application-based scripts are primarily used for initial prompt and collect operations, as well as for directing the playing of.wav files while calls are in queue. Note: The capability of using Unified ICME scripting for anything other than simple functions has been kept in support of legacy deployments. New customers are strongly advised to use the VoiceML scripting environment. In the "typical" 2-script implementation for Unified ICME-integrated models, which is what is under discussion here, the ICM script remains in control (and receives control back), even while it "delegates" the more complex self service activity to the VML Server script. Data can be passed from one script to the other and back through ECC variables. 73

80 Chapter 10: - Scripting Alternatives: VoiceML vs. Unified ICME Scripting Note: The same special micro-application that is used to invoke VML Server applications can also be used to invoke arbitrary "External VML" pages from a Media Server or other customer-provided source. This capability should only be used for very simple VML needs, however, because Cisco has no way to verify that customer-provided VML documents are compatible with the IOS Voice Browser (as opposed to VML documents that are generated by the VML Server, which are guaranteed by Cisco to be compatible with the IOS Voice Browser). Though the capability has not been removed from the Unified CVP 4.0 and later offerings, customers are discouraged from using it directly. Additionally, all the VML Gateway sizing metrics that Cisco provides are based on the specific VML documents that are generated using either VML Server applications or micro-applications. Using VML from another source will require the customer to perform his own empirical performance and capacity testing in order to determine how to size the VML Gateways. 74

81 Chapter 11 Developer Services Questions and/or support issues related to such items as Call Studio scripting or ASR grammar ARE NOT covered by Cisco Technical Support. Note: Cisco Technical Support is limited to standard Cisco product installation/configuration, and Cisco-developed applications it does not include services or support for items such as those just mentioned. A separate service agreement and subscription fee is required to participate in the Developer Services Program. For more details on how to subscribe, go to Getting Started! on the Developer Support Web site at Developers using Call Studio scripts, ASR, and the like may be interested in joining the Cisco Developer Services Program. This fee-based subscription program was created to provide you with a consistent level of Services that you can depend on while leveraging Cisco interfaces in your development projects. The Developer Services Program provides formalized services for Cisco Systems interfaces to enable developers, customers, and partners in the Cisco Technology Developer Program to accelerate their delivery of compatible solutions. The Developer Services Engineers are an extension of the product technology engineering teams. They have direct access to the resources necessary to provide expert support in a timely manner. For additional information, refer to Frequently asked Questions about the Program and Support under Q&A on the Developer Support Web site at 75

82 76 Chapter 11: - Developer Services

83 Glossary ACD Automatic Call Distributor. A programmable device at a call center that routes incoming calls to targets within that call center. Advanced Speech Model See VRU-Only Model. AIN Advanced Intelligent Network. A broad term encompassing a carrier s interface to adjunct computing devices like the NAM. ANI Automatic Number Identification. The number from which a call originated. Application A specific set of customer call-processing business rules as captured in the customer s custom NAM/ICM scripts, VoiceML scripts, the customer s custom prompts, and any database lookups/api interactions defined to the NAM/ICM. Generally these rules will apply to a specific customer function. Application Developer The person who designs and writes the applications. Application Prompts The customer s custom prompts for use with their NAM/ICM scripts. 77

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