SIP Training and Certification Course
Online self-study and certification exam, $495
Dates, Locations and Registration

Overview

In this 14.5 hour self study course, VoIP and telephony network engineers and professionals will learn all about Session Initiation Protocol (SIP) while preparing for a certification exam to earn the internationally recognized SSCA certification. The course consists of 9 modules each of which has it's own quiz session at the end to help participants prepare for the certification exam. The training is delivered online in a lively, easy-to-follow, fully animated format.

Course Topics:

  1. Core SIP
  2. SIP Trunking
  3. SIP-T and the PSTN
  4. Firewalls, NAT and Session Border Controllers
  5. SIP Security
  6. SIP and VoIP
  7. Testing and Troubleshooting
  8. SIP and Unified Communications
  9. ENUM and DNS

Who Should Attend

This course is ideal for service engineers, network design specialists, product managers, manufacturers of IP PBX and IP phone equipment, SIP security equipment manufacturers, SIP Trunk service providers and Carriers, sales engineers, and Marketing personnel working with VoIP equipment and services.

Agenda

1: Core SIP
Running time = 63 minutes, Quizzes = 7 minutes, Total = 70 minutes
  • SIP – Introduction
    • SIP – The Session Initiation Protocol
    • SIP ‘Official Summary’
    • Based on HTML
    • Where does SIP fit in?
      • SIP Clients and Servers
      • SIP User Agents
      • Simple Call Session Setup
      • SIP System Architecture
      • The URI - Unique Resource Identifier
    • SIP Addressing
      • SIP Addressing Examples
  • SIP Servers and Operation
    • Registration
    • Re-Registration
    • SIP Proxy servers and why we need them
      • SIP Server – Proxy Mode
      • SIP Server – Re-Direct Mode
      • Proxy Server ‘State’ types
    • Location Services
    • DHCP and SIP
    • SIP Proxy – Trapezoid Model
    • SIP Server in Proxy Mode
    • SIP Server in Proxy Redirect Mode
    • Stateful and Stateless Proxies
    • Location Server
      • Location Server – Components
      • Location Server – Information Sources
      • Location Server – Example
  • SIP Messaging
    • Request Methods
    • Response Codes
    • SIP Headers
    • SIP Request Methods
    • SIP Response Codes
    • SIP Headers
    • SIP HEADER - INVITE
    • SIP HEADER - 200 Response
  • SDP – the Session Description Protocol
    • SDP in a SIP Message
      • An SDP Example
      • Extending SDP
      • Changing Session Parameters
      • Call Hold example
      • Multiple ‘m’ lines
    • SDP Component in a SIP Message
      • SDP Example - Put a call on Hold
      • SDP Example - Call Hold Trace
    • INVITE and reINVITE
  • SIP Mobility
    • SIP Call Forking - Parallel
    • SIP Call Forking - Sequential
    • How do we keep track?
      • Call leg and Call ID
      • Tag and Branch ID
    • Call Forward - No Answer
    • Call Forward to Voicemail
  • More on Proxies and SIP Routing
    • Stateless Proxy
    • Stateful Proxy
    • More Proxy information
    • VIA and Record Route
      • VIA Details
      • Record-Route Defined
      • Record Route Example
    • Session Policies
  • MIME
    • Multiple MIME parts
  • SIP and the PSTN
    • SIP to PSTN Call Flow
    • SIP to PSTN Detail
    • SIP Codes and the PSTN
  • SIP and B2BUA
    • B2BUA - Back to Back User Agent
    • B2BUA Example
    • B2BUA Benefits and Features
  • SIP Summary
    • Request for Comments
    • New RFCs
    • SIPIT
    • The Call Process
2: SIP Trunks
Running time = 36 minutes, Quizzes = 7 minutes, Lab 20 minutes, Total = 163 minutes
  • SIP Trunks - Introduction
    • What is a SIP Trunk
    • Alternative to TDM
    • Separate Data and Voice connections
    • Converging the network
    • SIP Trunks and Codec
    • SIP Trunk Benefits
  • SIP Trunking - In More Depth
    • SIP Trunk Capabilities
    • SIP Trunking Network Examples
    • SIP Peering
    • Peering problems?
    • Least Cost routing (LCR)
    • Disaster Recovery
  • Some PBX Requirements
    • Enterprise PSTN Identities
    • P-Preferred and P-Asserted
    • Call Progress Tones
  • Trunking Variations
    • Single Site, TDM PBX
    • Single Site, No 'Forklift'
    • Single Site, Converged
    • Converged - SIP/IP PBX
    • Multiple Site, 'Converged'
  • Media Gateways
    • SIP PBX to Non-SIP PBX
    • SIP PBX to Non-SIP PBX, Call Flow
  • SIP Trunk Performance
    • The ADSL issue
    • Codecs, Voice and Data
    • Symmetric DSL (SDSL)
    • Bandwidth Calculator
    • Testing your link
  • Security and SIP Trunking
    • SIP Trunk Security - Overview
    • Session Border Controllers
  • Setting up a SIP Trunk
    • Add a VoIP Provider
    • Provider SIP Servers
    • Authentication
    • Stun and the Firewall test
    • Add a Dialling Rule
    • Trunk setup complete
    • Registration Trace
    • Call out Trace
  • Next Generation Networks
    • What are NGNs?
    • An Example - British Telecom
  • Troubleshooting and Interops
    • SIP Trunks and Common Problems
    • The SIP Forum
    • SIPits
    • SIPit Results
    • SIP Connect Document.
  • Choosing an ITSP
    • Understanding ITSP Offerings
3: SIP-T and the PSTN
Running time = 25 minutes, Quizzes = 7 minutes, Total = 32 minutes
  • SIP-T and the PSTN - Introduction
    • SIP to PSTN Overview
    • SIP to PSTN Call Flow
    • SIP to PSTN Detail
    • PSTN to SIP Call Flow
    • SIP to PSTN Call Failure
    • SIP to PSTN Call trace
  • Early Media
    • Early Media - SIP to PSTN Call
  • Early Offer and Delayed Offer
    • Early Offer / Delayed Offer
  • Gateways
    • Default Gateway?
    • Gateway Location and Routing with TRIP
    • TRIP Examples
  • SIP-T and PSTN Bridging
    • SIP-T
    • SS7, ISDN and SIP
    • ISUP and SIP Messages
    • ISDN User Part (ISUP) to SIP Codes
    • PSTN to PSTN via SIP
    • ISUP Encapsulation
    • ISUP Encapsulation / SDP
    • Addressing Notes
  • SIP and DTMF
    • DTMF - Quick Re-Cap
    • What is DTMF?
    • DTMF Transport methods
    • DTMF 'Inband'
    • RFC 2833 'Trace' example
    • SIP INFO 'Trace' example
4: Firewalls, NAT and Session Border Controllers
Running time = 24 minutes, Quizzes = 7 minutes, Total = 31 minutes
  • Overview
    • Issues to address
  • Firewalls
    • What does a Firewall do?
    • Are Firewalls effective?
  • NAT or Network Address Translation
    • What is NAT?
    • NAT Request
    • NAT Response
    • Multiple NATs
    • The NAT Problem
  • Types of NAT
    • Types of NAT
    • NAT - Full Cone
    • NAT - Restricted Cone
    • NAT - Port Restricted Cone
    • NAT - Symmetric
    • The NAPT or (PAT) Problem
    • Problems with NAT, Firewalls and SIP
  • The Solutions
    • STUN (Simple Traversal of UDP)
    • STUN and rport
    • Problems with STUN
    • TURN (Traversal Using Relay NAT)
    • Interactive Connectivity Establishment (ICE)
    • How ICE works - Simplified!
    • More on ICE
    • Universal Plug and Play (UPnP)
  • The RTP Problem
    • The Firewall Problem
    • Solving the RTP Problem
    • Symmetric RTP
    • Media Proxy
    • Application Level Gateway
    • SIP Aware Firewalls - Incoming
    • SIP Aware Firewalls - Outgoing
  • Session Border Controllers
    • SBC for the Enterprise
    • SBC for the ITSP
    • Enterprise SBC - in Action!
5: SIP Security
Running time = 35 minutes, Quizzes = 7 minutes, Lab = 120 minutes, Total = 162 minutes
  • Authentication and Authorization
    • SIP Proxy Authentication
    • 401 and 407 Authorization
    • SIP Authorization
    • PROXY Authentication
    • SSL with MD5 Cracked!
    • MD5 v SHA
  • Encryption
    • Why Encrypt SIP?
    • Certificates and HTTPS
      • Certificate Authorities
      • Certificate Example
      • Self-Signed Certificates
      • Format type
    • Securing SIP and VoIP
    • SSL and TLS
    • SIP and TLS
    • TLS Thoughts
      • TLS and SIP in Action
    • SIPS and SIP Addressing
    • Secure RTP (SRTP)
      • Setting SRTP on SIP Devices
      • Secure RTP (SRTP) - Example
      • SRTP and SRTCP
      • Caller Identity
      • DTLS/SRTP
    • S/MIME and SIP
      • MIME and ISUP
    • SIP Trunking and Security
    • Enhancing SIP Trunk Security
    • Alternatives - IPSec, ZRTP
  • Attacks and Responses
    • Types of Attack on a VoIP/SIP Network
    • Responses and Protection
    • TLS v SSL
    • Response Identity - A Problem!
    • Rogue SIP Proxy
    • Phishing and SIP exploit
    • More Examples RFC 4475
    • Try for yourself with recommended software tools
  • NIST Recommendations
    • NIST Recommendations on securing VoIP
6:  SIP and VoIP
Running time = 34 minutes , Quizzes = 7 minutes, Total = 41 minutes
  • What is VoIP or Voice over IP? An Introduction
    • What is VoIP?
    • What is Voice over IP?
    • VoIP - 'A Basic Call'
    • VoIP and TCP / UDP
    • VoIP over the Internet
    • Branch to Branch VoIP
    • IP PBX
  • Voice Sampling and Codec
    • Encoding
    • Codecs for Voice
    • MOS - Mean Opinion scores
  • The Real time Protocol or RTP
    • RTP Encapsulation
    • RTP Header Trace
    • Real Time Control Protocol
    • RTCP-XR (Extended Reports)
    • RTP / RTCP and UDP Ports
  • Quality of Service
    • QoS Issues
    • Measuring Delay
    • Jitter and Packet Loss
    • General VoIP Acceptance Criteria
    • QoS on the Network
    • 802.1Q - VLANs
    • 802.1Q/P Tagging
    • 802.1P - L2 Classification
    • TOS and DiffServe
    • Layer 3 Classification
    • Codecs and Bandwidth
    • Symmetric DSL (SDSL)
    • Testing your link
  • SIP, SDP and VoIP
    • SIP in the TCP/IP Model
    • SIP and SDP Messages
    • SIP and SDP Codec mapping
    • Where does SIP fit in?
    • SIP, SDP and VoIP INVITE
    • Audio and Video in the SDP body
7.  Testing and Troubleshooting
Running time = 40 minutes, Quizzes = 7, minutes, Lab - 240 minutes, Total = 287 minutes
  • Setting up your test environment
    • Using SIP IP Phones
    • Using SIP Softphones
    • SIP Communicator
    • Choosing a 'Trial/Test' ITSP
    • Get a SIP URI of your own
    • Using 'Test Numbers'
    • Multiple Setup options for you to try
    • Getting Free ITSP Accounts
    • Configuring your Softphone
    • Even more SIP Softphones
    • Example - The SIP Phones @ The SIP Place
  • Wireshark
    • Loading Wireshark
    • Network interface setup for capture
    • Wireshark - Basic Layout
    • Understanding Wireshark Icons
    • Using Wireshark - Capturing
    • Using Wireshark - Simple Filters
    • Using Wireshark - SIP Statistics
    • Using Wireshark - SIP ladders
    • Using Wireshark - RTP Statistics
    • Saving Captures
    • Where to Capture?
  • Interoperability Testing
  • Interop Testing
  • Why Interop can be tough
  • Different interpretations in the RFC 3261
  • Interop Test Scenario
  • Interop Test operations
  • Sample Interop Traces
  • Wireshark example videos to help understand interop issues
  • SIPIT events
  • Common SIP problems
    • Will it ever work?
    • What else can you do?
    • Common SIP/VoIP Problems
    • Troubleshooting SIP Trunks
    • 4xx - Client Failure Responses
    • 5xx - Server Failure Responses
    • 6xx - Global Failure Responses
  • More SIP Testing Tools
    • SIP Scenario
    • SIP Scan
    • TestYourVoIP.com
    • HoverIP
    • NSLookup
    • Using the NET to find answers
    • The SIP Wiki
8.  Sip and Unified Communications
Running time = 45 minutes, Quizzes = 7 minutes, Total = 52 minutes
  • Communication Breakdown
    • Playing Voicemail tag
    • Can't find people
    • Available but not Available..!
    • More Examples of communication problems
  • IM Clients
    • IM Client Features
    • Enterprise Clients
    • More in IM Clients
    • IM and Mobile devices
  • The Background Stuff
    • The IMPP working group
    • IMPP and CPP
    • More IMPP work
    • SIMPLE
  • How it all works
    • Presentity
    • A Basic SIP subscription
    • Multiple Presence States
    • Presence and P2P
    • A Presence Network
    • Getting inside the SIP packets
    • Presentity and more!
    • A Basic SIP Subscription
    • Multiple Presence States
    • Presence and P2P
    • A Presence Network
    • Get inside the SIP packets
    • The Packet Structure
    • PIDF Message Body
    • XML
    • Tuples
    • Example Presence doc with Tuples (using a Mobile Phone)
    • Rich Presence
    • The METHODS in Action
    • PUBLISH STATE
    • PUBLISH and PIDF/XML body
    • SUBSCRIBE METHOD
    • 202 OK Response
    • NOTIFY
    • MESSAGE
    • Add A Buddy/Subscribe
    • is-composing
    • Alternative 'Presence States'
    • 2 Places at the same time
  • Conferencing
    • What SIP does in Conferencing
    • INITIATE a conference
    • JOIN a conference
    • LEAVE / EXIT a conference
    • INVITE other participants
    • REFER conference server to invite or others to join
    • EXPEL participants
    • CONFIGURE the media stream
    • CONTROL a conference
    • Why SIP?
    • Centralized conferencing
    • Centralized Signaling
    • Centralized Mixing (optional)
    • Centralized Authentication
    • B2BUA (Discussed in core module)
    • Conference Components
    • The Focus
    • More than one Focus
    • Conference Setup
    • iscomposing in Conference
    • MESSAGE in conference
    • BYE in conference
    • Alternative INVITE
    • SDP BODY OF INVITE
    • IETF work and Conferencing
  • XMPP v SIMPLE
    • What is XMPP?
    • SIMPLE and/or XMPP
    • Gateways
  • Federations
    • What is Federation?
    • Multiple Presence sources
    • Super-Aggregation
    • Inter-Domain Federation
  • Unified Communications
    • What's all the fuss?
    • Unified Confusion
    • Components involved
    • What should UC do?
    • 21st Centrury Dial tone
    • The Unified inbox
    • Unified aware applications
    • Find me - Follow me
    • Device awareness
    • Unified Comms for Business
    • Do your Homework
    • Humans and UC
    • UC in a SIP network
  • Relevant RFCs
    • RFCs Galore
9.  ENUM and DNS
Running time = 20 minutes, Quizzes = 7 minutes, Lab - 20 minutes, Total = 47 minutes
  • ENUM Explained
    • What is E.164?
    • What is ENUM?
    • Why ENUM?
    • Call Routing and ENUM - Example
  • Enum, DNS and Domains
    • Why are we using DNS?
    • DNS and the Web
    • The e164.arpa Domain
    • Approved ENUM Delegations
    • TIERS 0, 1, 2 and 3
    • TIERS and Registrars
    • DNS and AOR
    • e164.arpa Domain in action
    • Example - ENUM in the UK
    • Address of Record
    • Reseaux IP Europeens
    • PSTN to SIP UA - Example
    • The ENUM Query
    • NAPTR Records
    • DNS Response to an ENUM query
    • Calls Flows
    • PSTN to SIP UA - Example (2)
    • IP to PSTN (Simplified)
  • Types of ENUM
    • Different 'Types' of ENUM
    • The Problems with 'Public' ENUM
    • Example - 'Private' ENUM
    • Example - 'Operator' ENUM
    • A few providers
  • Try for yourself
    • Register your number
    • Testing ENUM
  • ENUM and the future
    • How is ENUM moving forward?
    • Useful Links

Registration and Training Fees

The per student registration fee for the online self-study seminar is $495 and includes certification exam fees.

Register securely online with confidence or please call (708) 246-0320.

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SIP Training and Certification Self Study Online

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Tel: (708) 246-0320   Fax: (708) 246-0251

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Last modified January 28, 2010