VoiceCon San Francisco 2008 Executive Wine Tasting Event - Business Is Getting Done

Procuring a new enterprise communications system through a RFP process can be a harrowing experience, because IP telephony system design, capabilities, and attributes have undergone a series of dynamic changes the past few years. The RFP you issue will reflect how well your understanding of current generation solutions can satisfy your evolving communications requirements. This workshop will review and analyze how to prepare the core RFP requirements for your next IP telephony system by focusing on the following:
* System design and topology, including hardware, network, and power requirements
* Redundancy and resiliency attributes
* Port interface and traffic handling
* E911 and security issues
* Standards support, including SIP and SOA
* Voice terminals: analog, digital, and IP; desktop telephone instruments; mobile communications devices; soft client options
* Generic software features
* Systems management and administration
* Pricing guidelines, including potential hidden fees The workshop will also address network consolidation considerations, i.e. single system solution replacing an existing network of multiple systems, and installed system upgrade options. Allan Sulkin, is President of TEQConsult Group, and is a leading industry analyst and consultant focusing on enterprise communications. He created and hosts the annual VoiceCon Orlando RFP tutorial session.

SIP (Session Initiation Protocol) has become the dominant protocol for IP communications. This tutorial explains what SIP is, how it works, what the major issues for SIP deployments are, and how SIP will evolve in the future. The session focuses on the technical aspects of SIP and how it is used. It analyzes in detail the major components of SIP architecture, SIP addressing and registration, session establishment, SIP message routing and connecting SIP across the PSTN. You will learn about SIP extensions and how SIMPLE works for IM/presence. The tutorial also examines some of the challenges SIP faces, including NAT traversal (and the tools developed to cope with it: STUN, TURN and ICE) and security. The tutorial concludes with an assessment of how SIP may evolve and its role in peer-to-peer environments. You will receive an inventory of SIP resources?books, papers and organizations. David Bryan is a leading expert in SIP and P2PSIP. In addition to his role as CEO at SIPeerior Technologies, he is active in the IETF, where serves as co-chair of the P2PSIP working group. David has published numerous IETF drafts, academic papers and industry trade articles. David heads p2psip.org, the leading community site for P2PSIP. Prior to SIPeerior, David co-founded Jasomi Networks (sold to Ditech), and worked for Cisco and Vovida.

Whether an enterprise is early in its migration to IP Telephony or further along and now evaluating the concept of Unified Communications, it’s not easy to build a credible business case. IP Telephony and, in particular, UC, are complex , and involve numerous technologies, decision-makers and equipment, software and service providers. This tutorial will be divided into two segments: The first is devoted to IP Telephony businesses cases, the second will cover the UC business case. It will be based on IP Telephony cost data that has been gathered over the past five years from more than 800 companies that have implemented the technology, as well as real-world data from hundreds of IT decision makers on the newer UC products and technologies. This tutorial will cover the following questions:
* How do companies build a business case around IP Telephony and Unified Communications? What are the key metrics?
* What does it really cost to implement IP Telephony? What are the cost components of a UC business case?
* What resources (internal and external) companies must devote to their VOIP and UC rollouts, per end-unit, per year segmented by rollout size and vendor.
* What are some of the key pitfalls? Where did companies go wrong?
* What are some compelling business case models for both IP Telephony and UC? Robin Gareiss is Executive Vice President and Senior Founding Partner for Nemertes Research, where she oversees research projects and direction, conducts strategic seminars, develops cost models, and advises leading enterprises, vendors, and carriers. She currently serves as chief financial officer, as well. For the past 17 years, Robin Gareiss has worked closely with hundreds of senior IT executives, analyzing their use of technology and capturing best practices. Ms. Gareiss is a widely recognized expert in voice over IP, convergence, collaboration, carrier services, IP networking, and branch-office technologies. Before joining Nemertes, Ms. Gareiss served as Senior News Editor of InformationWeek, and prior to that, she worked at Data Communications magazine. Ms. Gareiss has a BS in journalism and a minor in education, with honors, from the University of Illinois, Urbana-Champaign. Irwin Lazar is the Principal Analyst and Program Director for Unified Communications and Collaboration at Nemertes Research, where he develops and manages research projects, develops cost models, conducts strategic seminars and advises clients. His background is in network operations, network engineering, voice-data convergence, and IP telephony. Mr. Lazar is responsible for benchmarking the adoption and use of emerging technologies in the enterprise in areas including VOIP, unified communications, Web 2.0 initiatives, social networking and collaboration. A Certified Information Systems Security Professional (CISSP) and sought-after speaker and author, Mr. Lazar is a columnist for No Jitter and Collaboration Loop and the late Business Communications Review magazine. He is a frequent resource for the business and trade press. He is regular speaker at events such as Interop, VoiceCon, and Enterprise 2.0. Mr. Lazar serves as the conference director for FutureNet (formerly MPLScon), the chair for Network World IT Roadmap Web 2.0 track, and is on the advisory board for the Enterprise 2.0 conference.

This tutorial is designed to help company CXOs/decision makers, IT and communications managers and technicians gain a good understanding of the architecture and functionality of both Microsoft Office Communications Server (OCS) and IBM Lotus Sametime, two Unified Communications solutions expected to see widespread deployment. Attendees will gain key insights into the significant differences in how IBM and Microsoft approach the UC market, where their solutions are similar and where they diverge. The tutorial covers telephony call-control capabilities within Sametime and Office Communications Server, describing the unique mechanisms each uses for integrating with enterprise telephony systems. You will leave this tutorial with a thorough overview of Microsoft Office Communications Server, IBM Lotus Sametime, and a foundation to know whether to pursue either or both of these products as part of your enterprise unified communications solution. Brent Kelly has written numerous articles and reports on unified communications and collaboration, focusing on Microsoft, IBM, and telephony vendors layering UC solutions on top of their PBXs. He has spoken and taught seminars on unified communications and on implementing IP Rich Media Communications in North America, Europe, Australia and South America. He leads the Unified Communications practice group at Wainhouse Research.

The cost of power and cooling the data center and communications closets is escalating, and Gartner predicts that by 2009, power and cooling costs will be second to salaries in the IT budget. This tutorial looks at the practical ways the enterprise can reduce electrical power and cooling costs. It won’t try to solve global climate change, but focuses instead on the energy conservation solutions now available and their impact on the enterprise bottom line. Methods for reducing the energy bill by controlling IT devices will be explored, and incentives from the government and power utilities will be presented. Quantitative information will be provided with additional resource websites for the attendee to learn how to calculate the energy cost reduction and cooling requirements. The session provides recommendations on how the lower the costs to support VoIP/IPT servers, PoE for the LAN, AC vs. DC powered devices, UPS and class 2 vs. class 3 phones. Gary Audin has more than 40 years of computer, communications and security experience. He has planned, designed, specified, implemented and operated data, LAN and telephone networks, and VoIP and IP converged networks all around the world, and he advises venture capital and investment bankers in communications technologies.

Legacy PBX systems are being retired and new functionality is essentially only available with IP-based systems. Migrating to Voice over IP (VoIP) and IP Telephony (IPT) has moved from “if” to “when.” But despite the growth of these new technologies, many enterprises have limited experience with IP Telephony and VoIP, and once the decision to migrate is made, there are a host of tough challenges to face - from inventorying the readiness of LANs and wiring closets, to WAN performance, IT organizational and staffing issues, security, utility costs and software support, patching and version control. This tutorial analyzes these issues with recommended actions and best practices that will lead to successful VoIP/IPT deployments. It cuts through the hype to the real advantages and presents how VoIP/IPT works and operates. This session will also guide the attendee through the rest of the Voicecon conference with suggested sessions, exhibits and other resources that will make the conference attendance experience as valuable as possible. Gary Audin has more than 40 years of computer, communications and security experience. He has planned, designed, specified, implemented and operated data, LAN and telephone networks, and VoIP and IP converged networks all around the world, and he advises venture capital and investment bankers in communications technologies.

As SIP moves from the Intranet to the Internet, security has gone from being a nice-to-have feature to a fundamental requirement. This session focuses on communications security aspects of SIP: How do you know you’re actually talking to the person you meant to call? How do you know who’s called you? How do you make sure that other people aren’t listening into your conversation? The IETF has developed (and is still developing) a variety of SIP and RTP-based protocol tools for providing these security services. Topics covered include:
* Introduction to communications security
* Security for signaling traffic
* User authentication
* TLS
* S/MIME
* SIP Identity
* Security for media
* SRTP
* SRTP key management (MIKEY, SDESCRIPTIONS, DTLS-SRTP, ZRTP)
* Privacy and anonymity The current state of the work at the IETF and other standards bodies is covered, as is the state of SIP Security implementations across the industry. Solutions for simultaneously providing identity and knowing who is calling, dealing with spam, allowing anonymous calls and providing appropriate wiretap access are described. The instructors will not only describe the various protocol components but explain how these work together as an integrated system that provides security for both signaling and media traffic. This session assumes some familiarity with SIP but no familiarity with cryptography or communications security. Dr. Cullen Jennings currently serves as IETF Real Time Applications Area Director. In that capacity, he has responsibility for the IETF’s activities in voice, video and instant messaging. At Cisco, Cullen focuses on conferencing, security and firewall and NAT traversal. He is responsible for helping set the direction for the technology that will make up the next generation of Cisco’s voice products, especially in conferencing, presence and rich media systems. Cullen is also a key contributor to all the SIP security work at IETF. He was the original designer of SIP certificate management system and the SIP Identity RFC. In addition, he has served as a chair and core member of the IETF IP Telephony (IPTEL), NAT Traversal (BEHAVE), and Web Distributed Authoring and Versioning (WEBDAV) working groups.

This tutorial is designed to provide a comprehensive overview of the options available for Unified Communications (UC). It is intended for enterprise CXOs, decision makers, IT and Communications technical managers, and business managers, who are planning to install UC solutions — either via a new UC system or by adding UC to existing PBX and email systems and business applications. The session information will be based on a template that includes the major types of UC solutions — desktop/mobile productivity tools, audio/web/video conferencing, mobility support;, and communication-enabled business processes. The template, which will be easily adaptable to your enterprise procurements, bids and RFPs, includes:
* Diagrams of the required UC software and hardware.
* Estimates of the professional services required.
* Estimates of the total price for each solution. Marty Parker is principal of UniComm Consulting, offering Unified Communications (UC) consulting services to enterprises. Marty is an active leader in the Unified Communications community. He contributes to the VoiceCon Unified Communications eWeekly and blogs on No Jitter in addition to helping develop UC sessions at VoiceCon. Marty is a co-founder of UCStrategies.com, a UC industry resource site, and is the author and instructor of the BCR Training course, “Planning and Implementing VoIP Unified Communications.”

The migration to IP Telephony and converged networks has already had major impacts on how IT organizations are structured, as staff with backgrounds in voice and data have been brought together into integrated work groups. Now, with Unified Communications, just having voice and data people work together isn’t enough - applications, messaging and security professionals need to become actively involved. This creates challenges in terms of overcoming long-held stereotypes, and opportunities to create a much more dynamic and responsive IT organization. This tutorial will present case studies of different approaches that enterprises are taking to deal with this issue. In addition, the instructors will present a methodology for project management and recommend processes for procurement, integration, security and reliability. Attendees will gain a better understanding of their options for organizational restructuring, and concrete advice on how to achieve more effective implementation of IP Telephony, Unified Communications and Converged Network projects. Mark Berg is a senior consultant with PlanNet Consulting. He has 15 years of communication technology experience, primarily in higher education. He has set IT strategy, designed voice and data physical infrastructure, managed cost-recovery initiatives, and overseen IP telephony projects. Previous posts include senior management positions, including CIO, at two Southern California universities in telecommunications and networking. Mr. Berg received his degree in journalism from Biola University in La Mirada, California. David Stein is a principal with PlanNet Consulting and Director of Operations. He has more than 25 years of consulting, information systems and telecommunications experience, with a primary emphasis on voice, data and video communications and technology infrastructure projects. He has been a featured speaker at numerous conferences, and has authored several articles on IP Telephony that have been published in Business Communications Review and HIMSS. Mr. Stein graduated from Rensselaer Polytechnic Institute with a degree in Computer Science.

Information and Communications Technologies (ICT) can reduce energy, travel and related expenses for the entire enterprise through the judicious use of conferencing - audio, video, telepresence and web — and the emerging tools and technologies for collaboration. This tutorial the new products and services for video, telepresence and collaboration, and explores your options for enabling remote and mobile workers. It will be organized around four frames of reference:
* Global Enterprise: Wants to reduce travel and increase efficiency (Telepresence, video, web conferencing).
* Enterprise: Wants to tightly connect to customers/partners (Telepresence, video, VPN, presence federation)
* Small/remote offices and teleworkers (Video, desktop video, web conferencing, VPN, VoIP, presence)
* Road Warriors (desktop video, web conferencing, VPN, VoIP, presence) It will also address the networking requirements for these applications, with an emphasis on managing bandwidth and maintaining quality of service. John Bartlett is a leading authority on real-time traffic, application performance and Quality of Service (QoS) techniques. He specializes in helping enterprises manage voice, video and data application performance. John has engaged with over 50 enterprises and over 20 network vendors to analyze network performance problems, design network solutions, and support network solutions. John has 29 years of experience in the semiconductor, computer and communications fields in marketing, sales, engineering, manufacturing and consulting. He has contributed to microprocessor, computer and network equipment design for over 40 products. He has been consulting since 1996. John is a graduate of Dartmouth College, and Dartmouth’s Thayer School of Engineering.