Cisco.com: Networkers 2004 Presentations

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VVT-1001: Understanding ENUM
ENUM is a protocol that translates an E.164 number (also known as a "telephone number") into a list of URIs (Universal Resource Indicators). These URIs can be anything from other telephone numbers to SIP and e-mail addresses. This session explores the ENUM protocol in detail, use of the Domain Name System (DNS), and namespace definitions. This session concludes with an example deployment of SIP in a VoIP environment in which Cisco IOS Software-based gateways are used between Internet and PSTN. Also discussed will be gateways that use ENUM to translate from E.164 numbers to SIP URIs.

VVT-1N00: Introduction to IP Telephony or Voice over IP (VoIP)
This high-level introductory session reviews the basic concepts of IP telephony, or voice over IP (VoIP). The session includes a review of traditional telephony technologies, how voice services and markets have changed to enable VoIP, the basics of VoIP technology (including information on signaling and bearer technologies), and summary with examples of IP telephony networks.

VVT-1N10: IP Contact Centers: Intro to IPCC Technologies, Concepts and Terminology
This session provides an introduction to Cisco IP Contact Center (IPCC) technology, including core solution components, call routing strategies, agent and supervisor tools, reporting, and third-party business application integration concepts. Industry-leading practices and examples will illustrate how today's organizations are using contact centers concepts. Topics include contact center terminology, contact center evolution, multichannel contact centers, call handling strategies, self-service, and the future of call centers.

VVT-2000: Choosing the Correct Voice/Video Signaling Strategy: H.323
This session examines the H.323 architecture and its use in building a multimedia infrastructure. The International Telecommunications Union (ITU) standard H.323 is the most widely deployed VoIP signaling standard in the world today. In this session the elements of the H.323 standard are briefly reviewed, and used to develop a VoIP network design. The inherent H.323 capabilities are examined, and best common practices to avoid potential problem areas are discussed. Security and address translation issues in an H.323 signaling architecture are highlighted. Logical designs for intercommunication between H.323 and other common VoIP signaling protocols such as MGCP, SIP, and Cisco's SCCP are developed to complete the view of signaling in today's multimedia network infrastructure.

VVT-2001: Choosing the Correct Voice/Video Signaling Strategy: MGCP/SIP
This session will describe the history, efforts, and status of the Session Initiation Protocol (SIP) and Media Gateway Control Protocol (MGCP), prefaced with an introduction to the Internet Engineering Task Force (IETF) Standards process to understand what is and is not an IETF Standard. Once that section is complete, a thorough examination of the core features and capabilities of SIP and MGCP is presented, including what the elements are for each protocol, what the call flows look like and what they reveal, and how the two protocols work together when a user of one protocol attempts to communicate with a user of the other protocol.

VVT-2002: Deploying Unified Communications in the Enterprise
This session gives network engineers designing or deploying Unified Communications (using Cisco Unity software) in the enterprise environment direction to the planning process, and identifies clear and methodical ways for such design and implementation to occur. Supporting resources will also be discussed, as will some of the common pitfalls and the ways they can be avoided.

VVT-2003: IP Telephony Security
This session will cover the technical details of securing voice over IP. This session will outline the methods and technologies for hardening of the infrastructure, operating systems, and endpoints, and will include an extensive discussion of certificate-based authentication and encryption of signaling and media streams.

VVT-2004: Designing Voice Enabled IPSec VPNs
This session covers planning and design issues as they relate to voice over IP, QoS, IPSec, and service provider implementations using Internet T1s, Frame Relay, and broadband (DSL and cable) links. The advantages and issues of deploying IPSec only, IPSec with GRE (a tunneling technology), GRE with dynamic crypto maps, and DMVPN (Dynamic Multipoint VPN) will be reviewed, as well as the performance characteristics of each option. The session will also address designing IPSec VPNs for high availability using dial-backup and multiple broadband links. Head-end redundancy and traffic load-balancing will also be reviewed. This solution covers details on the technology for deploying At Home Agent using Cisco IP Contact Center (IPCC), and design rules and best practices for an IPCC application will be also be discussed.

VVT-2005: Implementing Voice Enabled IPsec VPNs
This session is for enterprise customers interested in implementing a successful voice over IP (VoIP) over IPSec VPN deployment. Configuration examples will be reviewed for the typical deployment models (site to site, small office, and home office)using access methods of Frame Relay, Internet T1s, cable, and DSL. Basic Rate ISDN (BRI) and ASYNC dial backup configurations for small office, home office (SOHO) deployments will also be included. Performance data from internal testing will help guide the attendee on the selecting the appropriate product for the desired link speed and number of users. Troubleshooting techniques including use of Service Assurance Agent (SAA) and Netflow will be illustrated. A review of common problems and lessons learned from supporting customer and internal Cisco deployments will also be provided. Case studies of SOHO and site-to-site deployments over MPLS and IP Internet service providers will be reviewed.

VVT-2006: Emergency Services and IP Telephony
Emergency services calling (9-1-1 in the United States and Canada) can be an important part of your IP telephony system. The mobility of an IP phone can cause problems for the emergency responders locating the caller. This session will provide an overview of the 9-1-1 network, connection options for 9-1-1 calling, and the effects on design considerations for your IP telephony implementation. Standards that are developing around emergency services will also be discussed.

VVT-2010: Applied Scripting for IP IVR/IPCC Express
This session will provide attendees examples of how to use the advanced interactive voice response (IVR) capabilities of the Cisco IPIVR and IPCC Express solutions. Advanced IVR capabilities covered include database integration, Automatic Speech Recognition (ASR), Text to Speech (TTS), Voice XML, Java, HTTP triggers, and e-mail generation.

VVT-2011: Internet Service Node (ISN) for IP Contact Centers Design Session
This session familiarizes attendees with the principles of designing Cisco Internet Service Node (ISN) solutions for voice XML-based self-service interactive voice response (IVR), queuing, and call control applications. ISN deployment models, network design, call transfer methods, and call flows will be discussed, and attendees will review a sample ISN sizing exercise.

VVT-2012: Troubleshooting IP Contact Centers
This session will include problem analysis and troubleshooting methodologies associated with the Cisco IPCC (Internet Protocol Contact Center) product suite. The attendee will experience step-by-step methodologies and tools that will help in understanding how to manage the IPCC environment. Highlights will include the new 5.0 ICM (Intelligent Contact Manager) Support Dashboard Tools.

VVT-2013: Designing IP Contact Centers: Resources, Servers, and Bandwidth Provisioning
This session reviews traffic engineering principles and the use of Cisco authored traffic Erlang models (such as the IPC Resource Calculator) and other tools to determine required call center resources, server capacities, and bandwidth requirements. Attendees will learn how to determine the required number of agents, interactive voice response (IVR) ports, and gateway ports (PSTN trunks) using traffic calculators to meet required service levels. Topics include capacity sizing rules for determining the number of Cisco CallManager and Cisco IP Contact Center (IPCC) servers required. The 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.

VVT-2014: Centralized and Distributed Deployment Models for IP Contact Centers
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 IPCC deployment.

VVT-2015: IP Contact Centers: Clustering Over the WAN (High Availability and Resiliency)
This session will provide attendees with knowledge of the model for Cisco IPCC Enterprise high-availability clustering core components over the WAN. The session examines deployment model specifics including: layout and configuration of components and communication paths, component interaction across the WAN, network/bandwidth requirements and provisioning, call/messaging flows in normal and failure scenarios, design rules and best practices, and caveats and safe deployment rules.

VVT-2021: Designing and Deploying Business (Hosted or Managed) IP Voice/Data Services
The Cisco Business Voice Services Solution enables service providers to offer a portfolio of voice and data services over a common IP transport core to small and medium-sized business (SMB) and enterprise customers. This session discusses deployment models, VPN design for private and overlapping dial plans, routing logic examples, call flows, and scaling, billing, and high-availability considerations for service providers to manage or host core services such as business phone, site-to-site voice/data connectivity , centralized PSTN and Internet access. Other areas of focus include how the multiple deployment options interconnect in a unified architecture, the operations support system (OSS) overlay, Multiprotocol Label Switching (MPLS)-based network advantages and voice-over-IP (VoIP) applications. This session also includes a discussion of component options (e.g. call agents, gateways, Cisco Call Manager and Call Manager Express for H.323 IP phones, managed IADs and SIP IP phones), hosted versus managed service considerations, planning for end-to-end voice quality and security, and customer case studies.

VVT-2030: Understanding IP Video Telephony and Audio and Data Conferencing Solutions
This session will provide an overview of the technical components required for rich-media conferencing over an IP network with specific information about Cisco® CallManager 4.0 and Cisco MeetingPlace and how they operate in a Cisco IP Communications environment. Attendees at this session will be able to ascertain how these products might be beneficial to their organization, and can then follow the appropriate technology tracks to learn more about designing, deploying, and troubleshooting these products. Cisco CallManager 4.0 and Cisco MeetingPlace bring audio, video and Web-based data collaboration to the desktop. Cisco MeetingPlace integrates with popular messaging applications such as Microsoft Outlook and Lotus Notes, while also offering future integration with instant messaging and presence applications, and video communications solutions.

VVT-2031: Designing and Deploying IP Video Telephony Networks
Video communication capabilities have been integrated into Cisco® CallManger 4.0. These capabilities extend a host of voice features to video, including hold, transfer, conference, call forwarding, integrated directories, and extensible markup language (XML) services. This session covers Cisco CallManager 4.0 video-related capabilities in depth, including a detailed explanation of how quality of service (QoS) should be deployed, which protocols are used (such as H.323, Skinny Control Client Protocol [SCCP], and Sessions Initiation Protocol [SIP]), how to integrate existing videoconferencing equipment, what monitoring and reporting capabilities are available, and which applications and endpoint solutions offer integrated video features. This session will also include a detailed discussion of the public branch exchange (PBX) system-like call routing functions in Cisco CallManager; including least cost routing, automatic alternate routing, and calling permission per user and device.

VVT-2032: Designing and Deploying IP-Based Audio and Web Conferencing Solutions
This session will provide a detailed overview of the Cisco® MeetingPlace solution and what each component does. Other topics include how Cisco MeetingPlace integrates with the public switched telephone network (PSTN), traditional public branch exchange (PBX) systems, Cisco CallManager, Microsoft Outlook and Lotus Notes messaging and calendaring clients, and how to provide Web collaboration capabilities to your users.

VVT-3020: Troubleshooting IP Telephony Networks: Elements of Dial Plan Functionality
The dial plan is at the very core of any telephony implementation and as such, is one of the most important and complex elements in IP telephony design. The role of the dial plan in IP telephony networks is to help users reach dialed destinations, provide the flexibility to select alternative routes based on route availability or cost where digit manipulation is required, and set calling policies based on users or groups. This session will provide an in-depth look at the functional primitives of dial plan functionality in the Cisco® CallManager, Cisco IOS® Software-based H.323 gateways, and Cisco IOS Software-based gatekeepers, such as: best match routing logic, dial plan wildcards, calling search spaces, partitions, route lists and route groups, digit manipulations, transformation masks, and other Cisco CallManager-based dial plan entries. This session will also address the dial peers matching process (inbound and outbound), POTS and VoIP dial peers configuration, Cisco IOS Software dial plan wildcards, matching on calling/called number, and other H.323 dial plan entries including gatekeeper address resolution involving tech prefixes, zone prefixes, and other dial plan entries.

VVT-3021: Troubleshooting IP Telephony Networks: Elements of CallManager Functionality
This session will focus on troubleshooting techniques for Cisco® CallManager, with a strong emphasis on traces as generated by Cisco CallManager, Cisco IOS® gateways, and Cisco IOS gatekeepers. Analysis of traces allows elaboration of case studies describing the step-by-step progress of calls, including interactions between devices at call setup, voice conversation, and call tear-down. Protocol-level presentation of SCCP, H.323 and sub-protocols, Q.931, TFTP, DHCP will also be discussed. To augment the attendees' understanding of how to troubleshoot problems on an IP telephony network, case studies will be included to illustrate the function and protocol exchange on calls between SCCP Cisco IP phones, MGCP gateways, H.323 gateways, gatekeepers, and Cisco CallManager systems.

VVT-3030: Troubleshooting IP Video Telephony Networks
This session will discuss how to troubleshoot various aspects of an IP video telephony deployment, including SCCP Video endpoints, H.323 video endpoints, SCCP and H.323 Multipoint Conference Bridges, H.320 Gateways and H.323 Gatekeepers. We will review the troubleshooting tools available, including CallManager Trace files, Real-Time Monitoring Tool (RTMT), CDR Analysis and Reporting Tool (CAR), Sniffer traces, and much more. Finally, we will review many aspects of CallManager, Administration/Configuration and how those administrative choices you make impact the way things operate. This will include discussions of bandwidth control through CallManager Regions and Locations, Dial Plan configuration caveats, Automatic Alternate Routing configuration choices, endpoint, MCU and gateway configuration choices, how to integrate H.323 Gatekeepers into CallManager by using H.225 Trunks, and more.

VVT-4000: Advanced SIP Session
This session will delve deep into the Session Initiation Protocol (SIP) features and functionality. As an augmentation to VVT-2001 "Understanding SIP and MGCP, this session will more thoroughly cover such SIP functionality as signal flows for each SIP Request (14 total), capabilities such as Security considerations (involving IPsec, TLS, S/MIME, Digest, SIPFRAG and Asserted Identity), Preconditions (involving the use of RSVP), Firewall Traversal using STUN, Content Indirection, and look at the challenges involved in providing services such as Location Conveyance and an e911/112-like service.

VVT-4001: Advanced Dial Plan Design for IP Telephony Networks
This session provides detailed dial-plan design guidelines for each of the Cisco® IP telephony deployment models based on Cisco CallManager, with recommended best practices to help ensure successful, scalable deployments. The dial plan is one of the most important and complex elements in IP telephony design, providing a way to for users to reach dialed destinations, delivering the flexibility to select alternative routes based on route availability or cost, and establishing calling policies based on users or groups. This session will address the various dial plan tools available in Cisco CallManager and Cisco IOS® Software, such as: route patterns; translation patterns for digit manipulation; dial plan interaction with PTSN gateways; Class of Restriction (COR) dial peers; translation rules in Cisco IOS® Software; and intercluster calls through an H.323 gatekeeper. This session will also cover how to best use these tools to deal with real-world deployments. The main focus of this session is on system design, with some implementation aspects.

VVT-4002: Advanced Preferential IP Telephony Services for the Internet
Today's governments and military organizations use a telephone system service designed to prioritize calls. This capability is designed to help ensure that specific authorized people or command and control offices can place calls when the system is completely overwhelmed with calls. In the United States, these services area called the Government Emergency Telecommunications Service (GETS) and Multilevel Preemption and Precedence (MLPP), respectively. The services that are included in this effort extend beyond basic telephone and other real-time data services to include non-real-time (otherwise referred to as "elastic") data transmission services. In addition, similar services have been required on the Internet in response to the growing perception of the Internet as a critical communications infrastructure. The session includes discussion of what happened on the Internet on September 11, 2001 (and around other major outages), the kinds of services being called for, the issues that such services face in their design and deployment, and current recommendations on how to deploy them.


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