Effective 1-1-2009

University


SIP 2009 Planning Guide

SIP-Session Initiation Protocol

& Special Introduction to OCS-Office Communications Server & RP-Response Point  

2-5 Days Onsite, Via Webseminar or ~32 hours online.  


For more information and scheduling, please call Tom Cross 303-594-1694

or cross@gocross.com

Volume discounts are available.  All major credit cards are accepted.  Special SIP Forum & Microsoft Partner discounts are available.  Additional discounts for multiple sessions are available. 

Reference Promotional Code - SIPF8203


What Providers, Agents and Users are saying about these courses:  

According to Matt Jolly IT Consultant, “VoIP Business Executive course VoIP training to a new higher level. For example, there is nothing like the tutorials SIP available anywhere or from anyone. For the channel partner or customer, this course provides critical insights for successful implementation and management. The new user interface speeds learning allowing viewers to grasp complex concepts faster than ever before. With this course, VoIP providers can rapidly accelerate the learning process for their channel partners which in turn accelerate revenues. Now is the time for users and providers alike to make this course an integral part of their business operations.”  Matt Jolly - Consultant
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“I am writing regarding Tom Cross and the online training and VoIP/SIP course. By far this is the best program in place, as Tom is one of the most recognized trainers in the United States for VoIP/SIP. It might be worth to add this course to your sign-up package for agents around the country not only as a profit center but something that you could co-brand and have a "leg-up" on your competition.”  Bill Bowyer - CEO – VoIP in America

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“The SIP/VoIP courses are more than a superb primer on VoIP/SIP technology; they are an indepth business plan for a VoIP/SIP implementation. In addition, the VoIP/SIP diagnostic and troubleshooting guide is the most thorough approach to VoIP QoS available. I need information that informs but does not overwhelm. Information that guides but not drives you away. The courses provide insights and actionable information that has helped me get the technology we need sooner but saved me a considerable amount in understanding what we didn't need. The SIP course especially is a valuable one which provides much needed information in a readily understandable format.”  Paul Daubitz - President – ATI-TeleManagement


Who Should Attend:  - This online course is designed for enterprise executive and technical managers, channel partners, VAR-Value-Added Resellers, SI-Systems Integrators, telephone interconnect, agents, master agents and consultants.  In addition, this course will benefit corporate technical, staff marketing, training business development, sales, channel managers, operations, engineering, support and other corporate managers for SIP-VoIP providers, carriers, software developers and hardware manufacturers. 

What You Will Learn:

·         Review the fundamentals of IP-Internet Protocol and platforms required for high performance SIP-VoIP systems.  This includes soft switches, gateways, routers, services and other critical components.

·         Explore business applications and opportunities.  Review what customers are buying today and why they are buying.  In addition, emerging “killer applications” will be explained in depth.

·         Quickly grasp complex subjects such as H.323, MGCP and SIP.  As SIP-Session Initiation Protocol emerges are the key VoIP communications protocol, discover how this technology will impact all voice communications systems from key, PBX, IP-PBX, hosted, managed and other systems.

·         Understand basic and advanced SIP-VoIP concepts features.  From hosted, managed, IAS, and IP-PBX, quickly understands “what’s-what” for different customer applications and business models.

·          Probe the issues behind Integrated and Converged Access.  Understand when and why organizations need a converged access solution.

·         Understand why “network assessment” is critical to any SIP-VoIP implementation and why this step cannot be overlooked.

·          Address the issue of QoS-Quality of Service by overcoming jitter, echo, noise and other network problems.  Review the role of RTCP and other tools to monitor and maintain high performance VoIP networks.

·         Understand the functions of the new communications “toolbar.”  See how the benefits of “unified communications” as they improve business operations.

·         Assess the Top-10 issues why SIP trunking and hosted VoIP is more than “dial-tone,” and how it can represent change in the business and business model of even the smallest enterprises.  Discuss and explore new ways to improve fundamental business processes.

·         Explore how a SIP-VoIP call is processed and review potential security attacks.   Discover how SPIT, VOMIT, DOS and other terrorist attacks can target not just data, but voice packets.

·         Review SIP and SIP Trunking and all the implications and applications from TCO-Total Cost of Ownership to QoS-Quality of Service.  SIP Trunking is the most profound new form of telecommunications since POTS. 

·      Explore Microsoft’s OCS-Office Communications Server features, concepts, call flows, configurations and other issues for evaluation and implementation.  


Detailed Course Outline

Day One - Fundamentals of Data/Internet Telecommunications

NOTE: Day one is recommended for those wanting an update in internet technologies, data networking and network services (bandwidth).

-  Fundamental Network and IP Technologies – the IP in SIP

·         1 – Voice-to-Digital-to-Packet Transmission

·         2 – Back To Basics – Cabling, Conduit and Electrical Systems

·         3 – Transmission Concepts – DSL, T-1/E-1, ISDN-PRI, SIP Trunking, GIG-E

·         4 – Optical Fiber & Bandwidth

·         5 – Integrated Access Services – Dynamic Bandwidth Allocation – BOD-Bandwidth On Demand

·         6 – Introduction to IP-Internet Protocol and VoIP-SIP , MPLS-Multi-Protocol Label Switching, DiffServ-Differentiated Services, DSCP Differentiated Services Code Points and Packet Priority Classifications, TOS-Type of Service, EF-Express Forwarding, MPLS Uniform mode, MPLS Pipe and Short-Pipe modes, WRR-Weighted Round Robin, TCB-Traffic Conditioning Blocks - Marker, Meter, Shaping, Droppers and PHB-Per Hob Behavior.

·         7 – TCP/IP and other Protocols and Layers – RTP, RTCP, SDP, SOAP, SALT

          -  Call processing with Route, Image, DHCP, DNS, Image, Configuration servers

·         8 – Hardware – Routers, Switches – MAC-Media Access Control, WiFi-VLANS-VPNS

·         9 – Protocols “Rules of the Road” – H.228, H.323, MGCP, SIP, and Desktop “Softphones,” “Toolbars” and other end points (desksets)

·         10 – IP-PBX and Hosted VoIP/SIP – Integrated/Unified/Homogenized

- Top-10 Critical Technologies to SIP

1 –  IP protocol, IP networking and a VPN

2 - The difference between IAS-Integrated Access Service versus Converged Access Service

- Enhanced IAS with MPPP-Multi-link Point-to-Point Protocol, PPP Multilink Protocol (MP), L2TP-Layer 2 Tunneling Protocol  

- VPLS-Virtual Private LAN Service - new name for metro-ethernet

     - Switching Versus Routing - key benefits of VPLS

3 –  SIP-Session Initiation Protocol Trunking -

-        SIGTRAN (Signaling Transport)

-        SCTP-Stream Control Transmission Protocol

4 – Justification for an IP PBX – options and approaches

5 – Technical difference between IAS-Integrated Access Service, Hosted and Managed VoIP

-  Call processing with Route, Image, DHCP, DNS, Configuration servers  

-  Media Gateways replace PBXs -  the following tutorials are some examples of customer applications of MG-Media Gateways:

      - Connection of IP-PBX to PSTN
- Connection of IP-PBX to PSTN & SIP trunk provider
- Survivable connection to SIP trunk provider
- Connection of PBX & IP-PBX to PSTN & SIP trunk provider
- Connection of IP-PBX to Hosted VoIP provider
- Connection of IP-PBX & PSTN to Microsoft OCS Server

-  SC-Session Controllers or SBC-Session Border Controllers are access devices operate at Layer 5 Session Layer, where as routers operate at Layer 3 Network.   Some of the key SBC/SC functions are:

- Secure network peering - private and public to enhance performance

- Topology hiding - using various types of inter-AS-Autonomous System features as well as separating media (voice) and hide signaling (IP addresses) and data streams (traffic)

- Border call routing - routing at AS level rather than with interior protocols

- Interoperability - access/restrict to reduce voice spam

- QoS & Call Admission Control - load/jitter correction

- Billing systems interoperability - reduce billing errors

- NAT-Network Address Translation - routing for maximum performance

- CALEA-Communications Assistance for Law Enforcement Act 

- Compatibility with billing

- Dialect conversion 

- Protocol conversion

- Codec conversion

- Firewall restrictions 

- Wholesale and Transit peering 

6 – “Open Source” PBX options

7 – QoS-Quality of Service importance - how to measure it and fix it

8 –  Softphones – Where they make sense - user benefits

9 -   The difference between IPT-Internet Protocol Telephony and VoIP - Cisco, Broadsoft, Sylantro and other platforms

10 -  Unified Communications – Mobility Applications


Day Two - Introduction to SIP-Session Initiation Protocol 

- SIP Planning - SIP Introduction and Overview

      SIP Definition – IETF (RFC-3261) and Manufacturers

        - CPL-Call Processing Language

        - AOR-Address Of Record – q-values

        - Location Service - DNS-Domain Name Service

        - CPL-Call Processing Language

        - B2BUA-Back-2-Back User Agent

 -     Session Initiation Protocol for Telephones (SIP-T): RFC 3372

-     SIP-SS7-Signaling System 7 call processing including – IAM-Initial Address Message, Routing label, CIC-Circuit Identification Code and Message Type Code. Examples of Message Type Codes include: Called Number, Calling Number, DPC-Destination Point Code, OPC-Origination Point Code, SS7-ISUP ACM-Address Complete Message, ANM-ANswer Message, CPG-Call ProGress Message, COT-COTinuity Message, SUS-SUSpend Message, RES-RESume Message, FOT-FOrward message Transfer, INR-INformation Request message, INF-INFormation Message, RELease and other messages.

      SIP – Applications Layer 7 Protocol – Peer-to-Peer protocol

      SIP – Before and After

      SIP and Hosted – Better or Worse or Both

      SIP Signaling – Introduction, URI-Uniform Resource

-       SIP & SBC-Session Border Controllers, servers, gateways,

-       SIP with and without IADs-Integrated Access Devices

-       SIP and SIP Phones, Softphones, Mobility,

-       SIP Signaling Basics – Inbound/Outbound calling  

-       UC-SIP Bandwidth Planning - Critical Concepts for PC Video, data and voice

-       SIP Trunking - Four types and counting of SIP Trunking offerings  

-       SIP Trunking – Incremental “Slope” Growth

-       CODECS-COmpression-DECompression signal processors – issues and answers

 - SIP Trunk Replacement & Disaster Planning

-       SIP & Open Standards

-       SIP and Trunk Replacement – same or different thing

-       SIP and Proxy ARP-Address Resolution Protocol

-       SIP and HSRP-Hot Standby Routing Protocol

-       SIP and MPLS-Multi-Protocol Label Switching – COS and QoS

      SIP QoS – oxymoron or critical concept

-       SIP on-net and off-net issues – overflow call processing

      SIP TCO-Total Cost of Ownership – Top-10 Benefits

 - SIP Technology - Indepth

-       SIP – OSI-Open Systems Interconnection - "If you do not know where you are going, what difference does it make which path you take".....Cheshire Cat (Alice in Wonderland)

-       SIP “Methods” – Writing call processing as easy as email – invite, ACK, bye, etc.

-       SIP Signaling “commands” – 1xx-6xx

-       SIP Inbound and Outbound call processing  

    SDP-Session Description Protocol - headers, Via, Max-Forwards, To:, URL-Uniform Resource Locator, URI-Uniform Resource Identifier, call-ID, Cseq, Contact, Content-Type, Content-Length, Security and others

    Session Description Protocol Security Descriptions (SDES)

-       SIP Features - Forks, SIP Proxy, Redirect, Presence, Forking – parallel-sequential-mixed, loops, spirals

      SIP Network devices - UA-User Agent, UAC-User Agent Client, UAS-User Agent Server

-      Proxy Server, Redirect Server, Registrar Server, B2BUA-Back-to-Back User Agent

-      SRTP-Secure Real-time Transport Protocol (RFC-3711)

  -  Authentication Tag and the Master Key Identifier

   -  Encryption  


Day Three - Advanced SIP Planning and Security 

- SIP Security – “Best Practices” – Reality Check

-       SIP Security “Best Practices” – overview

-        SIP Firewalls and Security – SPIT-SPam over Internet Telephony, DOS-Denial Of Service, VOMIT-Voice Over Misconfigured Internet Telephony and other emerging problems

-       SIP Security and “Access Policy” – Stateful IP Filtering and Inspection, Static and Stateless IP Filters, TLS-Transport Layer Security, NAT-Network Address Translation, Persistent connection, Multi-homed hosts, etc.

-       SIP and MIM-Man-In-the-Middle attacks – Understanding wireline and WiFi wireless attacks

-     Telephone Numbers – North American Numbering Plan and International ENUM-E.164

- SIP Security Architectures – Building Blocks

-          SIP Security Architectures – eight different VoIP configurations evaluating SIP-Aware Firewalls and other security options  -

                        - Type 1 – Dedicated IP Pipe for VoIP

                        - Type 2 – Merged MPLS-Pipe with LER Tagging VoIP

                        - Type 3 – Merged IP pipe with SIP-Aware Firewall (SAFW)

                        - Type 4 – Separate IP Pipe for VoIP with Existing Non-SIP Firewall and SIP-Aware Firewall (SOFW)

                        - Type 5 – Merged IP Pipe with Incumbent Non-SIP-aware Firewall, No DMZ Port and SIP-aware Firewall (SAFW)

                        - Type 6 – Looks like Type 5 but Merged IP Pipe with Incumbent Non-SIP-aware Firewall, No DMZ Port and SIP-aware Firewall

                        - Type 7 – Merged IP Pipe with Incumbent Non-SIP-aware Firewall with a DMZ Port

                        -  Type 8 – Merged IP Pipe with Incumbent Non-SIP-aware Firewall

-       Other approaches to SIP Security - Proxy/Gateway Inside the Firewall, Proxy/Gateway in Co-Edge Mode and Proxy/Gateway Outside the Firewall, how Firewalls add time delays to TCP/IP  

- 50-Point Comprehensive SIP Security Checklist - more than 50 different security concepts to review and include in the implementation and ongoing network management program

-       SIP Security-Privacy Lifecycle Management - key planning for capturing, storage, users, and disposition (archiving/destruction)

 - SIP Class of Service & Quality of Service

-       SIP COS-Class Of Service and QoS-Quality of Service – ethernet meets “smart” IP

-       Managing “real-time” voice with RTCP-Real-Time Control Protocol – MRB-Metrics Report Blocks

-       Inside MRB – what’s what with all the info

- SIP Applications and Future Outlook

-        SIP Applications

        – IM-Instant Messaging call screening

        - SIP Presence Communicated by IM-Instant Messaging

        - Click-to-call and others

-       SIP for Call Centers – calling options and pricing benefits

        - Event Notification

        - Ondemand Conferencing

-       Integration of additional "third-party" developed SIP-enhanced services provides additional business and enterprise justification for SIP trunking.

-       UDDI-Universal Description, Discovery and Integration uses standards-based services such as XML, HTTP, SOAP, TCP/IP (define above) uniform service description and service discovery protocol. Discovery services provide a consistent publishing interface and allow programmatic discovery (registration) of services.  Description services provide the means for internet registration - to be found but not advertisement or placement on search engine listings. UDDI file structures are designed using a "publish-once-read-by-many" concept.  That is, web site URL-Uniform Resource Locator can be designed using UDDI standard file structures which can be published to the UDDI server network.  The UDDI network can be accessed (discovered) by search engines, customers and other list builders in a standard published (register) format.  UDDI Registries and protocol servers with:

- White Pages - Names, Address, Contact and Vcard information

- Yellow Pages - Industry categorizations and taxonomies

- Green Pages - Technical information including internal URL file discovery structures

-       UDDI is also designed to replace the robot.txt search engine web site document structure concept. Here are some of the web site description-discovery-registry information retrieved by search engine spiders/bots and other retrieval programs.

-       Voice-driven yellow pages - SALT-Speech Applications Language Tags adds voice commands to web applications.  SALT is an extended set of markup (meta) tags based on XML-eXtensible Markup Language though compatible with HTML-Hyper-Text Markup Language and others.

-       SIP – exciting new applications

-        SIP Total Tutorial with Future Outlook – IMS-IP Multi-media Systems – content servers, wireless integration, media gateways, etc.  

- Top-10 Steps to a Successful SIP Implementation

1 - User Needs Assessment

2 - Network Assessment

3 - Systems Upgrade

- Indepth POE-Power Over Ethernet & Comprehensive Disaster Planning Tutorial

4 - Pre–Installation Planning

5 - Data Systems Integration - VLANs, VoWLANS, Planning for WiFi, WiFi and IP Wireless "Roaming," WiFi Security and more

6 - Installation and Cutover

7 -  Managing Change - Training

8 - Ongoing Use and Expectations

9 - Billing

10 - Managed Services - TCO-total cost of ownership, monitoring, remote support, training, business development and others & Future Applications

- Diagnosing & Tools for Troubleshooting SIP Networks

1 - Problems: 

- Delay
- Jitter
- Equipment Configurations
- Clipping & Dipping
- VAD-Voice Activity Detectors - 
- Connection Issues
- Echo
- Signal-Noise Level and & Loss
- Comfort Noise
- Packet Loss Concealment

      - Zero Insertion

      - Waveform substitution

      - Model-based methods

- Crosstalk - Nearend and Farend
- Serialization
- Packet Payload Delays
- Packet Sizing Problems - Take the "Vo-eye-P Test" 
- Transcoding Problems
- Asynchronous Transcoding Problems
- Electrical Interference - Surges, Sags, Shared Neutrals

2 - Testing for Problems

- RTCP-XR-MRB-eXtended Reports - Metrics Report Block

3 - More than 30 Problems & Solutions - like "CarTalk" bring your problems to "Nettalk"

4 – Best Practices - review of concepts such as Resiliency & Reliability – QoS in VoIP-SIP

5 – Vendors of Technical Solutions for VoIP Network Management

6 - Conclusions and the Bottom Line  

NOTE: Course contents are constantly being update, please inquire about special requirements.


Day Four - (with optional add-on day of labs) - Introduction and Demonstration of OCS-Office Communications Server and Introduction to RP-Response Point

Here are just some of the terms used in Microsoft OCS-Office Communications Server.  Each of these terms and others are explained in this OCS Special Edition:

  •         Architecture – Front End, AD-Active Directory, Conferencing, Perimeter and Applications Servers

  •          PBX Co-existence

  •          UC-Unified Communications

  •       Advanced Telephony Features

  •          PBX Integration Initiative – UC-Compatible, Non-UC-compatible

  •          Mediation Server – Forking

  •          SIP – Microsoft SIP “Standard” – Inbound/Outbound Calls, OnHold, Making a Call Via a PBX, Answering a Call Via a PBX, Call Forwarding – Always, DND-Do Not Disturb, Call Transfer, 3-Way Conferencing

  •          SIP-to-PBX

  •          SIP-to-PSTN  

  •          Microsoft Management Console (MMC)

  •          Administrative Tools "Snap-in"

  •          Call Detail Records

  •          Phone Management and Usage

  •          IM-Instant Messaging, Chat, Chatrooms, etc.

  •           Video Conferencing "Roundtable" - Planning and Compliance

  •          Saving of IM conversations for compliance

  •          Planning for OCS

  •          Troubleshooting

  •          Trunking – bandwidth, delay, echo

  •          Telephone deskset issues

  •          IP routing check

  •          PBX interoperability

  •          Third-party integration – open discussion

  •          LCS-Live Communications Server and OCS coexistence  

NOTE: This session is constantly changing due to new releases coming from Microsoft such as R2 expected for general release in 2/2009.   

  • Response Point Components

  • Base Unit - IP-PBX software, voice mail and VoIP gateway

  • SIP Trunking solutions

  • Management and Client Software - MAC functions, Outlook integration

  • PSTN gateway

In addition, there were will be "LIVE" demonstration of IM, video conferencing, call processing and other features from a live OCS system,

NOTE: This topics presented may not be the complete list of issues discussed at time permits.  An optional five-day or half-day is available for custom training.

Microsoft is a registered trademark of Microsoft Corporation.


Course Leader - Thomas B. Cross – CEO TECHtionary.com has three decades of experience in startups and consulting advisor with leading providers and venture capital companies in market planning and development, hardware/software design and development, project management, intellectual property in telecommunications, information technology, conferencing, teletraining, telecommuting, groupware, networks, call centers, internet, artificial intelligence and other fields. He has managed the successful development of more than 10 software, hardware and internet products to market and received industry awards for this work. He has authored 13 books, wrote, produced and directed 15 commercial videos and creator and producer of the World's Largest Animated Knowledge Source on Technology – http://www.techtionary.com – recipient of Web Hosting Magazine Editors Choice for Best Technical Help.  Tom is a columnist for many leading publications such as Internet Telephony where he is the Technology Editor and columnist on OCS-Office Communications Service Newsletter with a blog at http://blog.tmcnet.com/cross-talk/.  He is a member of the Technical Board of Advisors for the VoIPSA-VoIP Security Alliance. Tom holds CompTIA Certified Security Professional certification and Pearson Vue Certified Test Administrator.