| VVT-2000 |
Choosing the Correct Voice/Video Signaling Strategy for Your Organization Part 1 |
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Tuesday 1:30 pm - 3:30 pm; Wednesday 1:30 pm - 3:30 pm
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These two sessions examine the dominant signaling and control protocols for voice-over-IP (VoIP) communications and VoIP application services. The functional aspects, features, and signal flows, and the strengths and weaknesses of each architecture are presented.This session focuses on the International Telecommunications Union (ITU) protocols H.323 and H.248. The ITU protocols are deployed in many enterprise and service provider networks today. This session examines the H.323 architecture, signal flows within the architecture, and interconnection to applications services. A brief Signaling System 7 (SS7) tutorial is presented to lay the foundation, as the session shifts focus to the H.248 client/server protocol. The session is summarized with an overview of interoperability with other VoIP architectures.
This session is designed to be particularly useful for attendees working in service provider public networks or enterprise networks, who are planning to deploy voice and related services across their IP backbones.
Associated Sessions:
VVT-2003 ''Emergency Services and IP Telephony''
VVT-2020 ''Designing Service Provider Hosted IP Telephony Networks''
VVT-2021 ''Designing and Deploying Managed Voice/Data services for Enterprise and SMB subscribers''
VVT-2022 ''Designing Voice Infrastructure & Applications for PSTN Interconnect''
VVT-2030 ''H.323 Videoconferencing and Video Telephony Design''
VVT-3010 ''Troubleshooting IP Telephony Networks in Campus Environments''
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| VVT-2001 |
Choosing the Correct Voice/Video Enabled SOHO/Telecommuter Solutions |
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Tuesday 4:00 pm - 6:00 pm; Wednesday 4:00 pm - 6:00 pm
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This session focuses on the Internet Engineering Task Force (IETF) protocols Session Initiation Protocol (SIP) and Media Gateway Control Protocol (MGCP). The IETF protocols are rapidly emerging, with new applications being developed every day. The session provides an in-depth understanding of the SIP elements and architecture, and detailed real-world examples of SIP signal flows, including interoperability with the ITU's H.323 protocol. The session then shifts focus to the MGCP, and the client/server architecture. The session concludes by bringing together the peer-to-peer and client/server protocols in an interoperable architecture.
This session is designed to be particularly useful for attendees working in service provider public networks or enterprise networks, who are planning to deploy voice and related services across their IP backbones.
Associated Sessions:
VVT-2003 ''Emergency Services and IP Telephony''
VVT-2020 ''Designing Service Provider Hosted IP Telephony Networks''
VVT-2021 ''Designing and Deploying Managed Voice/Data services for Enterprise and SMB subscribers''
VVT-2022 ''Designing Voice Infrastructure & Applications for PSTN Interconnect''
VVT-2030 ''H.323 Videoconferencing and Video Telephony Design''
VVT-3010 ''Troubleshooting IP Telephony Networks in Campus Environments''
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| VVT-2003 |
Emergency Services and IP Telephony Applications |
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Tuesday 4:00 pm - 6:00 pm; Wednesday 1:30 pm - 3:30 pm
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This session provides a summary of the design issues relating to the proper routing of emergency calls within a Cisco AVVID (Architecture for Voice, Video and Integrated Data) network. It will discuss emergency call considerations covering the interaction between the telephony topology, the dial plan, user and device mobility, and the choice of gateways and their interface point into the telephony network. This presentation is applicable for any enterprise with a footprint spanning more than one public safety jurisdiction area.
This session is for Network Architects, Engineers and Operators interested in emergency telephony services in an IP telephony environment.
Associated Sessions:
VVT-2000 Choosing the Correct Voice and Video Signaling Strategy for Your Organization Part 1
VVT-2001 Choosing the Correct Voice and Video Signaling Strategy for Your Organization Part 2
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| VVT-2010 |
Designing and Deploying IP Telephony Applications |
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Wednesday 10:00 am - 12:00 pm; Thursday 10:00 am - 12:00 pm
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This session focuses on the design and deployment of both Cisco and third-party IP telephony applications that interface with Cisco CallManager software. This session provides a systems design approach for integrating existing IP telephony applications over a Cisco IP telephony infrastructure. Cisco CallManager Computer Telephony Integration (CTI) scalability and redundancy design considerations will be addressed for Cisco IP telephony applications such as the Cisco Customer Response Solutions (CRS) platform, the Cisco IP SoftPhone, and Cisco IP Phone services. General provisioning guidelines will also be given for third-party applications. An example of an enterprise applications design and deployment scenario will tie all of the session concepts together.
Attendees must have previous experience with Cisco CallManager administration. Familiarity with Cisco CTI applications is helpful but not required.
Associated Sessions:
VVT-4011, ''Designing and Deploying Advanced IP Phone services''
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| VVT-2013 |
Deploying Unified Communications in the Enterprise |
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Wednesday 1:30 pm - 3:30 pm; Thursday 1:30 pm - 3:30 pm
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Making a business and technical decision to deploy Unified Communications in the enterprise requires considering many deployment and implementation options. This session discusses the most important features and factors involved in a successful Cisco Unity deployment. An overview will be provided of the most critical features, functions, interoperability, redundancy, and deployment options. The subject areas covered in the session will include:
-A Cisco Unified Communications approach to message access
-Cisco Unified Communications architecture
-Features that Cisco Unified Communications delivers to your enterprise
-Important features of Cisco Unity Version 4.X
-Administration and subscriber options
-Cisco Unity technology for Domino and DUCS (Domino Unified Communications Services)
-Cisco Unity and Personal Assistant design and deployment guidelines
-Microsoft Exchange/Active Directory and Cisco Unity technology
-Voice-mail-only requirements
-Cisco Unity technology for the Domino architecture
-Message store sizing, supported hardware platforms, and deployment models
-Cisco Unity networking and failover
-Cisco Unity Message Repository functions
-Traditional voice-mail interoperability with the Cisco Unity application
-The Cisco Unity bridge
-Helpful Cisco Unity and Personal Assistant links
This session is designed to be particularly useful for Cisco Unity and voice system designers with architectural responsibility during the planning and design phases of an enterprise-wide telephony infrastructure/unified communications deployment, as well as groupware-specific messaging architects and engineers. Attendees should have a working knowledge of telephony dial plans and telephony routing concepts, of traditional voice-mail systems and administration, and of groupware (Microsoft Exchange or IBM/Lotus Domino).
Associated Sessions:
VVT-3010 Troubleshooting IP Telephony Networks Part 1
VVT-3011 Troubleshooting IP Telephony Networks Part 2
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| VVT-2014 |
Centralized and Distributed Deployment Models for IP Contact Centers |
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Tuesday 1:30 pm - 3:30 pm; Wednesday 10:00 am - 12:00 pm
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This session focuses on planning and deploying the Cisco IP Contact Center (IPCC) solution (Express and Enterprise) for small, medium-sized, and large deployments. Specific centralized and distributed deployment models and configurations will be presented. Attendees will gain an in-depth understanding of Cisco IPCC interworking components, their functions, and the decisions needed while planning a Cisco IPCC deployment. Topics include configuration overview, scripting overview (business logic), call flows, call routing, call transfers, Cisco interactive voice response (IVR) queuing, and single- and multi-site deployments. Attendees will receive recommendations for a successful deployment and implementation.
This session is designed to be particularly useful for enterprise network managers, engineers and call-center teams that need to plan and deploy the Cisco IP Contact Center (IPCC) solution. Attendees should have a basic understanding of IPCC concepts prior to attending this session.
Associated Sessions:
VVT-2015 Designing IP Contact Centers: Resources, Servers, and Bandwidth Provisioning
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| VVT-2015 |
Designing IP Contact Centers: Resources, Servers, and Bandwidth Provisioning |
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Tuesday 4:00 pm - 6:00 pm; Wednesday 4:00 pm - 6:00 pm
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The session reviews traffic engineering principles and the use of traffic Erlang models and other methods to determine required call center resources, server capacities, and bandwidth requirements. Attendees will learn how to determine the required number of agents (inbound and outbound), interactive voice response (IVR) ports, and gateway ports (PSTN trunks) using traffic calculators to meet required service levels. Topics include sizing capacity rules for determining the number of Cisco CallManager and Cisco IP Contact Center (IPCC) servers required. This session will highlight the Cisco IPCC application/real-time traffic flows; network quality of service (QoS), and bandwidth requirements between remote Cisco IPCC components deployed over a WAN.
This session is designed to be particularly useful for enterprise network managers, engineers and call-center teams that need to plan and deploy the Cisco IP Contact Center (IPCC) solution. Attendees should have a basic understanding of IPCC interworking concepts and deployment models prior to attending this session.
Associated Sessions:
VVT-2014 Centralized and Distributed Deployment Models for IP Contact Centers
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| VVT-2020 |
Designing Service Provider Hosted IP Telephony Networks |
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Wednesday 10:00 am - 12:00 pm; Thursday 1:30 pm - 3:30 pm
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Current economic conditions have forced many companies to figure out how to increase revenue and lower operational expenditures. Enterprises have begun to look at outsourcing various services, and service providers have been looking at how to manage those outsourced services. One of those services is Hosted IP Telephony, commonly referred to as IP Centrex or Virtual IP PBX.This session covers the design considerations and configuration of this type of solution. Topics include security, the use of ALGs or border controllers in the solution, quality of service (QoS) both on the LAN and WAN, and survivability on the customer premises equipment (CPE). The session will also discuss network availability and protocol considerations.
This session is intended for audiences with a working knowledge of SIP and MGCP.
Associated Sessions:
VVT-2000 Choosing the Correct Voice and Video Signaling Strategy for Your Organization Part 1
VVT-2001 Choosing the Correct Voice and Video Signaling Strategy for Your Organization Part 2
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| VVT-2021 |
Designing and Deploying Managed Voice/Data Services for Enterprise and SMB Subscribers |
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Wednesday 1:30 pm - 3:30 pm; Thursday 10:00 am - 12:00 pm
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Managed voice and data services enable service providers to offer business customers a cost-effective, outsourced alternative to legacy PBX systems and time-division multiplexing (TDM) Centrex services. Now service providers can sell, deploy, and provide ongoing management services for their customers' IP Telephony solutions.This session covers various H.323-based enterprise and small and medium-sized business (SMB) deployment models, including design considerations for dial plan and call routing design, call flows, scalability, and redundancy. Other areas of focus include a closer look at how the multiple deployment options interconnect in a unified architecture, operations support system (OSS) overlay, Multiprotocol Label Switching (MPLS)-based network advantages and specifically, how they apply to voice-over-IP (VoIP) applications.
This session will be of particular interest to Service Provider and Enterprise Network Architects, Engineers and Operators interested in deploying managed Voice Services.
Associated Sessions:
VVT-2000 Choosing the Correct Voice and Video Signaling Strategy for Your Organization Part 1
VVT-2001 Choosing the Correct Voice and Video Signaling Strategy for Your Organization Part 2
VVT-2022 Designing Voice Infrastructure and Applications for PSTN Interconnect
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| VVT-2022 |
Designing Voice Infrastructure and Applications for PSTN Interconnect |
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Wednesday 4:00 pm - 6:00 pm; Thursday 4:00 pm - 6:00 pm
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Both Service Providers and Enterprise customers can benefit from IP-based voice services and applications for increased return-on-investment, capacity, next-generation services, and reduced expenditures. Interconnecting a TDM-based voice network (PSTN) with an IP-based voice infrastructure provides these opportunities while maintaining interoperability across multiple types of signaling networks. This session will cover architecture choices and design issues important for interconnecting packet voice infrastructures & applications to the PSTN. These issues include: - network analysis and dimensioning - network availability & redundancy - PSTN interconnect architectures (voice gateways, signaling controllers, softswitches) - signaling interworking (SS7, ISDN, R2/CAS to/from VoIP) - billing & management issues - network security - numerous case studies. It is assumed that the attendee will have have some knowledge of PSTN and packet voice technologies.
This session will be of value to anyone interested in learning how designing a VoIP infrastructure for PSTN interconnect, PSTN gateways, long distance transit VoIP, wholesale voice, or managed business voice. Attendees should have a basic background in PSTN and VoIP call control (SS7, ISDN, H.323, and SIP).
Associated Sessions:
VVT-2000 ''Choosing The Correct Voice/Video Signaling Strategy For Your Organization, Part 1''
VVT-20001 ''Choosing The Correct Voice/Video Signaling Strategy For Your Organization, Part 2''
VVT-2020 ''Designing Service Provider Hosted IP Telephony Networks''
VVT-2021 ''Designing and Deploying Managed Voice/Data services for Enterprise and SMB subscribers
VVT-2023 ''Designing Voice over Broadband / Local Services''
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| VVT-2023 |
Designing Voice over Broadband/Local Services |
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Wednesday 1:30 pm - 3:30 pm
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This session covers the central call control architecture and the service features that are common to all access technologies, including billing, network management systems (NMSs), and operations support. Various broadband access technologies will be discussed, including core features that are specific to the access technology.
In this context, The Cisco Broadband Local Integrated Services Solution (BLISS) architecture, which provides a comprehensive unified solution for providing residential voice-over-IP (VoIP) services over many different broadband access technologies, will be discussed.
Attendees should have a solid background in VoIP concepts. An understanding of signaling and call control is preferred.
Associated Sessions:
VVT-2000 ''Choosing The Correct Voice/Video Signaling Strategy For Your Organization Part 1''
VVT-2021 ''Designing and Deploying Managed Voice/Data Services for Enterprise and SMB subscribers''
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| VVT-3010 |
Troubleshooting IP Telephony Networks Part 1 |
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Tuesday 1:30 pm - 3:30 pm; Thursday 1:30 pm - 3:30 pm
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This session focuses on a selection of tools such as Cisco CallManager-based traces, to present the low-level signaling flows and their associated entries in the trace files. Case studies of various types of calls are used, and the associated expected results are introduced. Line-by-line analysis of the trace file entries is supported with call flow diagrams; allowing the audience to become familiar with fundamental call flow concepts required as the foundation of any troubleshooting effort.
This session is designed to be particularly useful for networking professionals with technical hands-on responsibilities over a Cisco CallManager environment. Attendees should be familiar with the general configuration and operation of Cisco CallManager and associated applications, but do not need to know how to troubleshoot the problems.
Associated Sessions:
VVT-2000 ''Choosing The Correct Voice/Video Signaling Strategy For Your Organization, Part 1''
VVT-20001 ''Choosing The Correct Voice/Video Signaling Strategy For Your Organization, Part 2''
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| VVT-3011 |
Troubleshooting IP Telephony Networks Part 2 |
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Tuesday 4:00 pm - 6:00 pm; Thursday 4:00 pm - 6:00 pm
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The session discusses the following 3 IP Telephony components which must be well understood in order to facilitate troubleshooting: Digit Analysis, Echo and Directory Integration.
Digit Analysis: In a Cisco IP Telephony network call setup is handled by Call Manager (CCM). CCM collects and analyzes the dialed digits in order to route the call as required. This session provides insight into how CCM does digit analysis and how it is influenced by translation and route patterns, partitions, calling search spaces, etc. Sample CCM traces are discussed in detail in order to show how common digit
analysis problems can be identified and corrected.
Echo: Echo is a natural part of any telephony network. This session helps the attendee understand why echo occurs, and what can be done to troubleshoot and minimize echo. G.168 echo canceling enhancements first introduced in IOS 12.2(11)T are also discussed.
Directory Integration: Call Manager relies on the integrated DC Directory or an external directory such as iPlanet or Active Directory
(AD). This session looks under the covers and reveals see how CCM interacts with the directory. At the same time common directory
integration problems and useful troubleshooting techniques are discussed.
This session is designed to be particularly useful for networking professionals with technical, hands-on responsibilities over a Cisco CallManager environment. Attendees should be familiar with the general configuration and operation of Cisco CallManager and associated applications, but do not need to know how to troubleshoot the problems.
Associated Sessions:
VVT-2000 Choosing the Correct Voice/Video Signaling Strategy For Your Organization Parts 1
VVT-2001 Choosing the Correct Voice/Video Signaling Strategy For Your Organization Parts 2
VVT-2010 Designing and Deploying IP Telephony Applications
VVT-2011 Designing Voice- and Video-Enabled IPSec VPNs
VVT-2013 Deploying Cisco Unified Communications in the Enterprise
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| VVT-4010 |
Advanced Dial Plan Design for IP Telephony Networks |
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Wednesday 10:00 am - 12:00 pm; Thursday 4:00 pm - 6:00 pm
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The dial plan is one of the most important and complex elements in IP telephony design. In IP telephony networks, the dial plan provides reachability of dialed destinations, flexibility to select alternative routes based on route availability or cost where digit manipulation is required, and calling policies based on users or groups. This session provides an in-depth view of the Cisco CallManager dial plan construct and operation, and detailed dial-plan design guidelines for each of the IP telephony deployment models with recommended best practices to ensure successful, scalable deployments. This session also covers the various dial plan tools available in Cisco CallManager, such as route patterns; translation patterns for digit manipulation; dial plan interaction with PTSN gateways; and remote IP WAN calls through the H.323 gatekeeper. The session also discusses how to ensure that the primary path for voice calls is the IP WAN. Depending on the deployment model, overlapping dial plans are typically required. This will be covered, along with dial plan considerations for voice mail and other IP-based voice applications.
This session is designed to be particularly useful for system designers with architectural responsibility during the planning and design phases of an enterprise-wide telephony infrastructure, and networking professionals with technical and/or operational responsibilities over a Cisco CallManager-based system. Working knowledge of dial plans and telephony routing concepts (Cisco CallManager-based or not) is highly desirable.
Associated Sessions:
VVT-2000 Choosing the Correct Voice/Video Signaling Strategy For Your Organization Parts 1
VVT-2001 Choosing the Correct Voice/Video Signaling Strategy For Your Organization Parts 2
VVT-2011 Designing Voice- and Video-Enabled IPSec VPNs
VVT-2013 Deploying Cisco Unified Communications in the Enterprise
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| VVT-4011 |
Designing and Developing Advanced IP Phone Services |
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Wednesday 1:30 pm - 3:30 pm; Thursday 4:00 pm - 6:00 pm
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This session discusses how to combine voice- and Web-based transactions by learning how to develop IP phone services. Intended for systems architects and programmers, this session will show you how to build powerful voice and data applications for the enterprise using the Cisco API for its SCCP-based IP phones. The session focuses on programming IP phone services using Cisco XML objects for the Cisco IP Phone as well as its available software developer kit (SDK) tools.
Network Architects, Engineers, and programmers looking for more information on how to build powerful voice and data applications for the enterprise.
Associated Sessions:
VVT-2010 ''Designing and Deploying IP Telephony Applications''
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