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Introduction to IP Telephony (VVT-100) Are you finding an increasing requirement to use your data network to transport voice? This session provides introductory information on how voice networking is accomplished using packet technologies, with an emphasis on voice over IP (VoIP). The session also reviews voice technology basics, including information on voice network transport technologies such as voice over Frame Relay (VoFR) and voice over ATM (VoATM). In addition, the fundamental signaling protocols for voice-over-packet networks are introduced and some sample configurations are presented. The following sessions cover information you may find useful if you register for this session: VVT-210 Designing and Deploying IP Telephony Applications, VVT-213 IP Telephony Interoperability with PBX/Voicemail Systems and VVT-310 Troubleshooting IP Telephony Networks in Campus Environments. This session is a comprehensive overview for networking engineers and planners with little or no voice experience. Back to top Introduction to IP-Based Videoconferencing (VVT-130) This session introduces IP videoconferencing based on the H.323 standard. It covers industry standards, architecture, and H.323 video network components, including endpoints, multipoint control units (MCUs), gateways and gatekeepers, and the functions of each. The session also describes T.120 data collaboration and the multimedia aspects of audio- and video-encoding techniques. The following sessions cover information you may find useful if you register for this session: VVT-230 Deploying IP Video: Case Study and VVT-330 Troubleshooting IP Videoconferencing Networks. This session is useful for attendees working in the following areas: Any technical staff wishing to get an introduction to IP videoconferencing technologies. Back to top Building a Voice and Video Signaling Strategy for your Organization (VVT-200) This session examines the dominant Signaling and Control Protocols for Voice over IP communications, and VoIP application services. The functional aspects, features, signal flows, and the strengths and weaknesses of each architecture are presented here in detail. The session focuses on the Peer-to-Peer Signaling Protocols H.323 and SIP, as well as the Client/Server Control Protocols MGCP, MEGACO/H.248, and Cisco's Skinny Station Control Protocol. A brief SS7 tutorial/refresher is presented to provide a more in depth understanding of the interaction with the Call Agent Architectures. In addition to the above objectives, this session analyzes the interaction between the Peer-to-Peer and Client/Server Signaling Architectures, and traces end-to-end communications in these architectures. This session provides an understanding of how each of the protocols are used within larger infrastructures to deliver advanced services for Voice communications. The following sessions cover information you may find useful if you register for this session: VVT-100 Introduction to IP Telephony, VVT-210 Designing and Deploying IP Telephony Applications, VVT -222 Designing & Deploying IP-based Signaling Gateways, VVT-220 Deploying Wholesale IP Telephony Using SIP and H.323,VVT-223 Designing IP-Based Managed Voice and Video Services, VVT-320 Troubleshooting Service Provider IP Telephony Networks and VVT-420 Deploying Advanced Softswitch Applications. This session is useful for attendees working in the following areas: attendees working in Service provider-Public networks or Enterprise Networks planning to deploy voice and related services across their IP backbone. Back to top Designing Voice/Video Enabled SOHO IPsec VPN Telecommuter Solutions (VVT-201) It is common for today's enterprise customers to transport voice over IP (VoIP) and video traffic over private WANs. One emerging application provides telecommuters or small office/home office (SOHO) workers the same network connectivity and services that they would have in traditional "brick and mortar" offices including telephony services. This session covers best-practice design guidelines for successful IP telephony deployments in telecommuter environments. Important aspects such as QoS on telecommuter/SOHO customer premises equipment (CPE), as well as the service provider network, will be covered. Proper enabling of various QoS tools such as Low Latency Queuing (LLQ), link fragmentation and interleaving (LFI), and traffic shaping needs to occur on both traditional WAN media for site-to-site deployments and on SOHO access media such as DSL and cable. The session focuses on typical telecommuter/SOHO access technologies. Also discussed will be service provider SLAs: what they mean, and letting service providers know what to enable in order to achieve a desired class of service. In most cases, when extending enterprise IP telephony to telecommuters/SOHOs, a virtual private network (VPN) will be deployed to create transparent connectivity. With IP Security (IPSec) VPN solutions replacing WANs, customers expect to transport VoIP over their IPSec VPN networks as well. This session highlights best-practice IPSec VPN design considerations unique to VoIP, with such aspects as performance, QoS for IPSec tunnels, and platform selection criteria to guide participants in making the best platform selection when designing a VoIP over IPSec network. Finally, a case study on the Cisco internal telecommuter/SOHO deployment using VoIP over IPSec VPNs will be featured. Today's Cisco deployments will be covered as well, providing insight into lessons learned and future deployment plans. The following sessions cover information you may find useful if you register for this session: VVT-100 Introduction to IP Telephony, VVT-200 Choosing The Correct Signaling Strategy For Your Organization and VVT-211 Designing and Deploying Multi-site IP Telephony Networks. This session is useful for attendees working in the following areas: Technical staff for Enterprise and Service Provider customers. Back to top Designing and Deploying IP Telephony Applications (VVT-210) This extended breakout session focuses on the design, development, and deployment of both Cisco and third-party IP telephony applications that interface with the Cisco CallManager software. This session provides a systems design approach for integrating existing IP telephony applications over a Cisco IP telephony infrastructure or programmatically extending existing enterprise data applications to fit into a voice/data convergence solution. Cisco CallManager scalability and redundancy design considerations will be addressed for Cisco IP telephony applications such as the IP Interactive Voice Response (IP IVR), IP Integrated Contact Distribution (IP ICD) Softphone, Cisco Personal Assistant, and Cisco Conference Connection. 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. The following sessions cover information you may find useful if you register for this session: VVT-200 Building a Voice and Video Signaling Strategy for your Organization, VVT-214 Designing, Sizing, and Deploying IP Contact Centers and VVT-310 Troubleshooting IP Telephony Networks in Campus Environments. This session is useful for systems engineers or software architects. Back to top Designing and Deploying Multi-site IP Telephony Networks (VVT-211) This session will focus on design and deployment of IP telephony in enterprise environments. It covers important design and planning elements to consider and includes configuration examples. Attendees will apply design concepts to three distinct deployment models: a large campus site, a centralized call processing model and a distributed call processing model. Critical issues covered during this session include phone connectivity options, powering options, IP addressing, network infrastructure, QoS strategies, admission control, dial plans, gateway selection, and voice messaging. The following sessions cover information you may find useful if you register for this session: VVT-210 Designing and Deploying IP Telephony Applications, VVT-212 Designing and Deploying Enterprise Unified Messaging Systems, VVT-213 IP Telephony Interoperability with PBX/Voicemail Systems and VVT-310 Troubleshooting IP Telephony Networks in Campus Environments. This session is for network managers and engineers involved in the planning and deployment of IP telephony in enterprise networks. Back to top Designing and Deploying Enterprise Unified Messaging Systems (VVT-212) This session describes how to deploy unified communications across an enterprise-scale organization in both mixed and IP-only environments. In addition to providing an overview of major deployment models, this session also addresses best practices for deploying unified communications systems. Topics covered include: Cisco Unity™ Unified Messaging; Interoperability - How to integrate existing legacy PBX and voice-mail systems with unified communications solutions; Failover and fault tolerance - How to safeguard message integrity if the system goes offline; Sizing - How to determine the number of ports/sessions an enterprise requires; and Groupware integration - How to integrate unified communications with existing groupware. The following sessions cover information you may find useful if you register for this session: Any VVT sessions that cover IP telephony. This session is useful for attendees who work in enterprise-scale organizations, internal IT, hosted applications and decision makers interested in implementing Unified Messaging Systems. Back to top IP Telephony Interoperability with PBX/Voicemail Systems (VVT-213) This session focuses on how to make the transition from an existing legacy voice network (such as PBX and/or voice mail) to an IP-based network. Attendees will learn how the capabilities of both Cisco CallManager and Cisco Unity™ unified messaging can simplify and facilitate migration. This session also provides a general overview of Cisco CallManager Q Signaling (QSIG) configurations, situation-specific deployment scenarios, and current implementations, as well as a brief roadmap for future supplementary services and additional network features. The following sessions cover information you may find useful if you register for this session: VVT-100 Introduction to IP Telephony, VVT-210 Designing and Deploying IP Telephony Applications, VVT-211 Designing and Deploying Multi-site IP Telephony Networks and VVT-310 Troubleshooting IP Telephony Networks in Campus Environments. This session is useful for attendees working in the following areas: Enterprise network technical staff. Back to top Designing and Sizing IP Contact Centers (VVT-214) This session focuses on designing and sizing Cisco's IP Contact Center (IPCC) solution. The first half will focus on planning and designing. Attendees will gain an in-depth understanding of the interworkings of the IPCC components, their functions and decisions needed while planning your IPCC deployment. Topics include, call routing, call transfers, IVR Queuing, Voice Gateways, single and Multi-Site Deployments. The second half will focus on sizing and provisioning call center resources. Attendees will learn how to size resources to optimize and maximize the value of the contact center. Topics will include determining required number of agents, IVR ports, and gateway (PSTN) ports using traffic calculators to meet required service levels; determining number of CallManager and IPCC servers required. This session will briefly highlight the IPCC application/real-time data flows, network QoS and bandwidth requirements between IPCC Peripheral Gateways (PGs) and the ICM Central Controller in remote IPCC deployments over a WAN. The following sessions cover information you may find useful if you register for this session: VVT-210 Designing and Deploying IP Telephony Applications and VVT-212 Designing and Deploying Enterprise Unified Messaging Systems. This session is designed for Enterprise Network Managers, Engineers and Call Center Teams that need to plan and design the Cisco IP Contact Center (IPCC) solution. Attendees should have a basic understanding of IPCC prior to attending this session. Back to top Design Guidelines for Voice over IPsec VPN Deployments (VVT-216) It is common for today's enterprise customers to transport voice-over-IP (VoIP) traffic over private WANs. With IP Security (IPSec) virtual private networking (VPN) solutions replacing WANs, customers expect to transport VoIP over their IPSec VPN networks as well. This session will cover best-practice design guidelines for a successful VoIP-over-IPSec VPN deployment. Important topics include QoS on the edge of the enterprise network and in the service provider network. Proper enabling of various QoS tools, such as LLQ, LFI, and traffic shaping, needs to occur on traditional WAN media for site-to-site deployments and on SOHO access media such as DSL and cable. Also discussed will be service provider SLAs what they mean, and information service providers need in order to move to a desired class of service. Best-practice IPSec VPN design considerations, unique to VoIP, will be highlighted, with particular emphasis on performance, dealing with MTU sizes, and fragmentation and bandwidth provisioning. Platform selection criteria will guide attendees in making the best platform selection when designing a VoIP over IPSec network between a Cisco IOS Software- or appliance-based platform. Finally, a case study on Cisco's internal deployment using VoIP over IPSec VPNs between various sales offices, as well as to the SOHO users in the field, will be featured. Attendees will learn what Cisco is deploying today, and will gain an understanding of the lessons learned and future deployment plans. The following sessions cover information you may find useful if you register for this session: VVT-100 Introduction to IP Telephony, VVT-210 Designing and Deploying IP Telephony Applications, VVT-211 Designing and Deploying Multi-site IP Telephony Networks and VVT-310 Troubleshooting IP Telephony Networks in Campus Environments. This session is useful for attendees working in the following areas: Enterprise network technical staff. Back to top Deploying Wholesale IP Telephony Using H.323 and SIP (VVT-220) This session covers deployment considerations when using H.323 and the coexistence of SIP in a wholesale SP VoIP network. The session will review H.323 Cisco Gateway, Gatekeeper and Directory Gatekeeper topology design and configuration; billing using AAA/RADIUS, redundancy, security and interconnection methods. Interconnecting SIP-based networks into the H.323 network will be discussed, both in terms of coexistence and interoperability. The possibilities of SIP as a wholesale core protocol will also be discussed. Students should have an understanding of the H.323 protocol and SIP protocol. The following sessions cover information you may find useful if you register for this session: VVT-200 Choosing The Correct Signaling Strategy For Your Organization and VVT-320 Troubleshooting Service Provider IP Telephony Networks. This session is useful for attendees working in the following areas: Service Provider voice technical staff. Back to top Designing & Deploying IP-based Signaling Gateways (VVT-222) The multitude of circuit-switched network (SCN) signaling protocols offer many challenges for converged networking. New application servers (ASPs) have been created to interwork these signaling protocols between legacy and IP networks. Application protocols such as MAP, ISUP, and INAP can use the facilities of the SIGTRAN protocol suite for transfer between PSTN signaling and IP networks. This session discusses how transitioning legacy signaling networks to an all-IP infrastructure provides for signaling traffic expansion and greater transport efficiencies while maintaining interoperability with multiple types of networks. Learn how to design and deploy an IP-based signaling network that can still offer the scalability and reliability of a traditional PSTN network. We will explore the use of the IP-based Signaling Gateway to bridge legacy and IP signaling networks, design criteria, and the merits and trade-offs of various designs and deployments. Session topics include: an overview and comparison of the SIGTRAN protocol suite: transport protocol SCTP and adaptation layer protocols M2PA, M2UA, M3UA, SUA, IUA; how ASPs (Application Server Processes) can best utilize these protocols, signaling gateway deployment considerations such as security, redundancy, routing key, and multi-homing; and what solutions offer SIGTRAN as a strategy for transitioning hybrid networks. The following sessions cover information you may find useful if you register for this session: VVT-200 Choosing The Correct Signaling Strategy For Your Organization, VVT-220 Deploying Wholesale IP Telephony Using SIP and H.323 and VVT-320 Troubleshooting Service Provider IP Telephony Networks. This session is useful for attendees working in the following areas: Service Provider environments where the technical staff is required to understand and implement signaling gateways bridging IP and PSTN networks. It is recommended that the attendee have some knowledge of PSTN protocols such as SS7 and of transport protocols such as TCP. Back to top Designing IP-Based Managed Voice and Video Services (VVT-223) As enterprises and small- and midsized business (SMB) customers move towards outsourced services, the need for various service provider -managed services will arise. The business customer can reap the rewards of increased productivity, profitability and investment protection, while the service provider will be able to increase revenue as well as increase ability to add new value-added services. This course covers design considerations when creating service provider-managed services for SMB and enterprise customers. Managed service infrastructures will include the design characteristics of VoIP long distance interconnect, voice VPNs, IP-based Centrex services and managed IP PBX. Topics will review considerations with billing, security, service quality, and the incremental steps that a service provider can take in deployments. Attendees should have an understanding of the H.323 protocol, and SIP protocol and an understanding of designing these types of networks. The following sessions cover information you may find useful if you register for this session: VVT-200 Choosing The Correct Signaling Strategy For Your Organization, VVT-220 Deploying Wholesale IP Telephony Using SIP and H.323 and VVT-320 Troubleshooting Service Provider IP Telephony Networks. This session is useful for attendees working in the following areas: Technical staff from the service provider and enterprise environments. Back to top Deploying IP Video: Case Study (VVT-230) This session will cover the IP videoconferencing deployment of a large New York financial institution, which has a large videoconferencing network, based on ISDN-based H.320 and proprietary videoconferencing units. This customer has started to move current videoconferencing sites to H.323, while making use of the current IP infrastructure. In this session, attendees will learn why this customer decided to make the move from current videoconferencing technologies to H.323, how the IP network is configured, how the gatekeeper infrastructure is configured, how multipoint and gateway calls are handled, and what dial plans and call-routing mechanisms are being used. The following sessions cover information you may find useful if you register for this session: VVT- 130 Introduction to IP-Based Videoconferencing and VVT-330 Troubleshooting IP Videoconferencing Networks. This session is useful for attendees working in the following areas: Any enterprise technical staff working on an IP videoconferencing deployment. Back to top Troubleshooting IP Telephony Networks in Campus Environments (VVT-310) This session focuses on debugging and troubleshooting telephony applications designed to operate with Cisco AVVID (Architecture for Voice, Video and Integrated Data) for enterprise networks. It focuses on troubleshooting and monitoring the IP telephony components using different tools and utilities. Some common problems faced in deploying IP Telephony will be explained using a case study. Participants should know the configuration and operation of Cisco IP telephony components including Cisco CallManager, Windows 2000,Cisco IOS Voice Gateways, Catalyst Voice Gateways and telephony signaling protocols like MGCP, H.323. Participants will acquire new skills in debugging and troubleshooting IP telephony networks using different utilities. The following sessions cover information you may find useful if you register for this session: VVT-320 Troubleshooting Service Provider IP Telephony Networks and VVT-211 Designing and Deploying Multi-site IP Telephony Networks. This session is useful for attendees working in the following areas: The session is for enterprise customers who have deployed IP telephony. Back to top Troubleshooting Service Provider IP Telephony Networks (VVT-320) This session will focus on troubleshooting Service Provider IP Telephony Networks including enhanced services such as voice applications (calling card), dial-plan manipulation, accounting and fax transmission. A basic understanding of VoIP, telephony and fax signaling is recommended in order to gain most value from this session. The following sessions cover information you may find useful if you register for this session: VVT-220 Deploying Wholesale IP Telephony Using SIP and H.323, VVT-221 Moving to IP/ATM-based Solutions for Class 4 Network Replacement and VVT-223 Designing IP-Based Managed Voice and Video Services. This session is useful for attendees working in the following areas: Service Provider voice network technical staff. Back to top Troubleshooting IP Videoconferencing Networks (VVT-330) This session takes the audience past the introductory and deployment phases of IP videoconferencing and deals with topics related to managing and troubleshooting a deployed network. In this session the audience will learn how to troubleshoot common error scenarios in an H.323 videoconferencing network, such as: debugging H.323 reliability, availability and serviceability; (RAS) messages and troubleshooting endpoint registration problems; debugging H.225 and H.245 call-signaling messages and troubleshooting call failures; troubleshooting gatekeepers, MCUs, and H.323/H.320 gateways; troubleshooting poor video quality, QoS and RSVP; troubleshooting videoconferencing through firewalls and NAT. The following sessions cover information you may find useful if you register for this session: VVT-130 Introduction to IP-Based Videoconferencing, VVT-230 Deploying IP Video: Case Study and NSC-215 Deploying QoS in an Enterprise Environment. This session is useful for attendees working in the following areas: Enterprise network IP videoconferencing technical staff. Back to top Advanced Dial Plan Design for IP Telephony Networks (VVT-410) The dial plan 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 provide 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: If the IP WAN is not available, the call will take the PSTN transparently to the called and calling party. Depending on the deployment model, overlapping dial plans are typically required and this will be covered along with dial plan considerations for voice mail and other IP-based voice applications. The following sessions cover information you may find useful if you register for this session: VVT-211 Designing and Deploying Multi-site IP Telephony Networks and VVT-310 Troubleshooting IP Telephony Networks in Campus Environments. This session is useful for attendees working in the following areas: Enterprise network voice technical staff. Back to top Deploying Advanced Softswitch Applications (VVT-420) The Softswitch has been well known for providing packet voice applications such as tandem transit and PRI offload. This session introduces new Softswitch applications that provide new definitions of the Softswitch, which do much more than merely handling only MGCP and PSTN signaling protocols. The presentation will include values of these applications and their architecture, features, and system design challenges. At the end, the session will illustrate on how the Softswitch can be seamlessly integrated with other existing packet voice solutions so as to unify as one network of voice networks. Audience is recommended to attend the following sessions also: VVT-220, VVT-222, and VVT-223 This session is intended for service providers interested in choosing the Softswitch as their next generation packet voice solutions. The audience is assumed to have basic understanding about packet voice technologies, including H.323, MGCP, and SIP, and PSTN protocols. Back to top
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