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

Overview

In this 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 13 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. Wireshark
  3. SIP, the PSTN and SIP-T
  4. SIP, VVoIP and QoS
  5. SIP Security
  6. Firewalls, NAT and Session Border Controllers
  7. SIP Trunking
  8. Testing, Troubleshooting and Interoperability
  9. ENUM and DNS and Peering
  10. SIP in the Cloud, LTE, the IMS and VoLTE
  11. SIP and Fax over IP
  12. SIP with Unified Communications
  13. WAN Broadband Assessment for SIP trunking - Recommended, Optional Module

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 = 84minutes, Quizzes = 7 minutes, Total = 91minutes
  • SIP – Who Benefits
    • Why SIP?
    • What is SIP?
    • SIP from the RFC
    • What are 'Requests for Comments' - RFCs?
    • More that just 3261
    • New RFCs
    • IETF Working Groups
    • Based on HTTP
    • 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
    • Proxy Server 'State' types
    • 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 Client Configuration
    • Configuration scenarios
    • Some basic elements needed to configure a client
  • SIP Messaging
    • Request Methods
    • Response Codes
    • SIP Headers
    • INVITE-Example
    • RESPONSE (200 OK)-Example
    • More on Headers
    • Support and Require Headers
      • Timer (Session Times)
    • SDP Example - Call Hold Trace
    • Call Hold - Old and New Methods
    • Music on Hold example
    • INVITE and reINVITE
  • SIP Mobility
    • SIP Call Forking - Parallel
    • SIP Call Forking - Sequential
    • Call legs, dialogs and Call IDs
    • Dialog trace example
    • Dialogs and Transactions
    • Branch Ids
    • Call Forward to Voicemail
    • Call Forward - No Answer
    • Replaces Header
    • Diversion Headers
  • 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
    • Loose and Strict Routing
    • 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
    • The Call Process

2: Wireshark
Running time = 40 minutes, Quizzes = 1minutes, Lab approx. 80 minutes, Total = 121 minutes
  • Wireshark
    • What is Wireshark
    • Initial Setup
    • Free SIP Account options
    • Desktop clients
      • Jitsi client for testing
      • Blink client for testing
      • X-lite client for testing
      • PhonerLite client for testing
    • Mobile clients
      • Media5 for testing
      • Linphone for testing
      • WeePhone SIP for testing
    • Social Study directory
    • Free DID and credit
    • SIP test numbers
    • Download Wireshark
    • Wireshark
      • Introduction
      • Menus, screens and views
      • Capturing traffic
      • Profiles
      • Display filters
      • Capture filters
      • SIP Packet Analysis
      • SIP ladders and audio playback
      • Other menu options
      • SIP INVITE analysis
      • Follow a UDP stream
      • Frame relationships
      • Colouring rules
      • RTP streams
    • View Captures in the 'Cloud'
    • LAB Exercises
    • What are the codes?

3: SIP-T and the PSTN
Running time = 26 minutes, Quizzes = 7 minutes, Total = 33 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 and SIP-1
    • 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
    • RFC 4733 replaces 2833
    • RFC 4734
    • SIP INFO 'Trace' example
4:  SIP, VVoIP and QoS
Running time = 81minutes , Quizzes = 7 minutes, Lab = 10 minutes, Total = 98minutes
  • 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
    • Signaling paths
    • Speech paths
    • IP PBX
  • Voice Sampling and Codec
    • Encoding
    • Codecs for Voice
    • Try the Codec Test
    • High Definition (HD) Voice
    • Sound tests
    • Wideband (HD) codecs
    • Opus codec
    • Opus audio examples
    • Codec choices and MOS - Mean Opinion scores
    • Packet rate/ packets per second
  • 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
  • SIP, SDP and VoIP
    • SIP in the TCP/IP Model
    • SIP and SDP Messages
    • SIP and SDP Codec mapping
  • Video over IP
    • Streaming Voice and Video-1 way transmission
    • Two-way conferencing with RTP
    • Codec and bandwidth considerations
    • Video bitrate calculator
    • Setting video codecs on devices
    • Audio and video in the SDP body
  • Assured SIP Services
    • Service provider architecture
    • Proxy and access router funtions
    • Resource-priority
    • Video example
    • Reason header for pre-emption events
    • More proxy details
    • Multi level pre-emption and precedence (MLPP)
    • Summary
5: SIP Security
Running time = 41 minutes, Quizzes = 7 minutes, Lab = 120 minutes, Total = 168 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
    • sdes and the Crypto attribute
    • Crypto attribute example
    • SRTP Call Example
    • SRTP with ZRTP
    • RFC 4474 for Caller Identity
    • caller identity
    • DTLS/SRTP
    • Ongoing developments for Identity
    • S/MIME and SIP
    • MIME and ISUP
    • SIP Trunking and Security
    • Enhancing SIP Trunk Security
  • Attacks and Responses
    • Types of Attack on a VoIP/SIP Network
    • FBI network examples
    • Responses and Protection
    • Response Identity - A Problem!
    • Rogue SIP Proxy
    • Phishing and SIP exploit
    • More Examples RFC 4475
    • Try for yourself with recommended software tools
6: Firewalls, NAT and Session Border Controllers
Running time = 62 minutes, Quizzes = 7 minutes, Total = 69 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
    • UDP Hole punching
    • Hairpinning
    • Multiple NATs
    • The NAT Problem
  • 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
    • BEHAVE
  • The Solutions
    • STUN
    • STUN and rport
    • Problems with Classic STUN
    • TURN
    • STUN RFC 5389
    • Interactive connectivity establishment (ICE)
    • ICE 'in theory'
    • Candidate information and other 'ICE stuff'
    • ICE in practice
    • ICE tags
    • ICE-Lite and Trickle- ICE
    • ICE client settings
    • More in ICE
    • Universal plug and Play
    • 'Near end' NAT
    • 'Far end' NAT
    • GRUU
  • The RTP Problem
    • The firewall problem
    • Solving the RTP problem
    • SIP 'Refer' problems
    • SBC 'Interop' example
    • SBC Manufacturers examples
    • From SIP to WebRTC (and back)
7.  SIP Trunking
Running time = 70 minutes, Quizzes = 7, minutes, Lab - 120 minutes, Total = 197 minutes
  • SIP Trunks
    • What is a SIP trunk
    • Alternative to TDM
    • Separate data and voice connections
    • Converging the network
    • SIP trunks and codecs
    • 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
    • Disaster recovery 'expanded detail'
    • Disaster recovery - last resort?
    • Number consolidation
    • Virtual presences
  • Trunking Variations
    • Single site, no forklift
    • Single site, TDM PBX
    • Single site, converged
    • Converged, SIP/IP PBX
    • Multiple site, converged
    • Multiple site, converged +central SBC
    • Multiple site, converged + Multiple SBCs
  • Media Gateways
    • SIP PBX to Non-SIP PBX
    • SIP PBX to Non-SIP PBX, call flow
  • SIP Trunk Performance
    • Connection types
    • The ADSL issue
    • Codecs, Voice and Data
    • Symmetric DSL (SDSL)
    • Bandwidth calculator
    • Testing your link
    • ADSL developments
    • Fibre options
    • WAN Optimization, hybrids and SD-WAN
    • Software defined WANs explained
  • Security and SIP Trunking
    • SIP trunk security - overview
    • Session border controllers
  • More on SBCs
    • The corporate SBC
    • SIP Refer issues
  • Setting up a SIP Trunk
    • Add a VoIP provider
    • Provider SIP services
    • Authentication
    • Add a dialing rule
    • Trunk setup complete
    • Call out trace
    • Comparing SIP packets from two ITSP providers
    • Skype for business and SIP trunks
  • Some PBX Requirements
    • Enterprise PSTN identities
    • P-preferred and P-asserted
    • Call progress tones
  • Troubleshooting and Interops
    • SIP trunks and common problems
    • The SIP forum
    • SIPits
    • SIPit results
    • SIP connect
    • SIP connect 1.1 onto 2.0
  • Choosing and ITSP
    • Understanding ITSP Offerings
    • Sticking points
    • What you may need in the future
    • SIP trunk connectivity
    • Finding an ITSP
    • SIP trunking checklist for ITSP evaluation
  • Working together
    • SIP trunk connectivity items from the field
8.  Testing Troubleshooting and Interoperability
Running time = 54 minutes, Quizzes = 7 minutes, Labs = 240 minutes, Total = 295 minutes
  • Setting up your test environment
    • Your setup
    • Using SIP IP phones and softphones
    • Jitsi, Blink, X-lite and phonerLite setup reminder
    • Linphone, media 5 and WeePhone SIP setup reminder
    • Choosing a trial/test ITSP
    • Get another SIP account
    • SIP2SIP account
    • Configure Blink and Jitsi on the same PC for testing
    • Using test numbers
  • Wireshark
    • Where to capture
    • More options for packet capturing
    • Wireshark revisted
    • Colours and the intelligent scrollbar
    • Packet marking and comments
    • New packet window
    • exporting 'specified' frames
    • RTP streams
    • TShark
    • PCAP-ng and PCAP formats
    • Alternatives to wireshark
    • You try
  • Interoperability Testing
    • Interop testing and why interop can be tough
    • Different interpretations in the RFC 3261
    • Interop test scenario
    • Interop test operations
    • Sample interop traces with wireshark
    • Wireshark example videos to help understand interop issues
    • More sample captures
    • Video call testing
    • Video tests with wireshark trace analysis
    • Basic interop test list
    • SIP IT events
  • Common SIP problems
    • Will it ever work
    • Where can you start checking
    • What else can you do
    • Common SIP VoIP problems
    • Troubleshooting SIP trunks
    • 4xx client failure responses
    • 5xx server failure responses
    • 6xx global failure responses
9.  ENUM, DNS and Peering
Running time = 43minutes, Quizzes = 7 minutes, Lab - 20 minutes, Total = 70 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 Operation
    • DNS root server morrors
    • Finding domain name servers using NSLookup
    • The e164.arpa domain
    • Approved ENUM delegations (RIPE)
    • TIERS 0,1,2 , and 3
    • e164 arpa domain in action
    • ENUM delegations
    • Address of record
    • PSTN to SIP UA - example
    • The ENUM query
    • DNS response to an ENUM query
    • PSTN to SIP UA  - example (2)
    • NAPTR query - a different view
    • Finding SIP servers using the tool - DIG
    • IP to PSTN (simplified)
    • RFC 6140
  • Types of ENUM
    • Different 'Types' of ENUM
    • The Problems with 'Public' ENUM
    • Example - 'Private' ENUM
    • Carrier ENUM and e164enum.net
  • Peering for VoIP and Video
    • Stay on-net
    • From ITSP to PSTN and back
    • Loss of features with the PSTN
    • Peering profiles and agreements
    • Bi-lateral peering
    • Multi-lateral peering
    • Back to ENUM
    • A complete infrastructure
  • ViPR
    • Register your number
    • Testing ENUM
    • DIG and NAPTR records
10.  SIP in the Cloud, LTE, the IMS and VoLTE
Running time = 53minutes, Quizzes = 7 minutes, Total = 60 minutes
  • Hosted SIP
    • What hosted SIP service is
    • Hosted functions and features
    • Example network including fallover
    • Hosted clients in action
    • Why hosted - benefits and things to consider
    • Why on-site PBX - benefits and things to consider
  • Auto Provisioning
    • Auto provisioning example
    • Boot server
    • Client config
    • Client boot sequence
    • Client config download
    • RFC 6011
    • Benefits of hosted SIP service
    • Benefits of onsite PBX and SIP trunks
  • SIP, LTE, the IMS and VoLTE
    • Network overview
    • RAN, eNodeB, EPC, IP core and 3GPP
    • 4G, LTE, LTE Advanced, WiMAX2
    • The RAN and EPC
    • Default bearer setup
    • Introduction to the servers and functions in the IMS
    • Device registration (with SIP)
    • SIP registration packet example
    • SIP in the IMS-call flow explained
    • Introduction to VoLTE and the threat of OTT services
    • Making VoLTE work
    • SIP call flow for VoLTE
    • Quality settings recap
    • VoLTE media flow
    • More on VoLTE
    • The IMS
    • Layers architecture
11.  SIP and Fax over IP
Running time = 33minutes, Quizzes = 7 minutes, Total = 40 minutes
  • Faxing Basics
    • Faxing background
    • T.30 fax signaling
    • Associated tones and protocols
    • The ITU and TIA standards
  • Fax over IP
    • Fax over IP benefits
    • From the old to the new
    • Intro to FoIP
    • FoIP and SIP trunks
    • Protocol conversions
  • Fax Protocols
    • G.711 Pass-through
    • T.37 store and forward
    • T.38 relay
    • Where does SIP fit in
    • UDPTL
    • Protocol options for the future
  • FoIP in action
    • SIP in FoIP - call flow
    • SIP INVITE
    • INVITE for T.38
    • The INVITE SDP body
    • Wireshark FoIP example
    • SIP T.38 Call flows-IETF draft document
  • Bandwidth
    • T.38 and G.711 network traffic
  • Troubleshooting
    • The basics
    • More complex issues to watch out for
  • Ongoing Efforts
    • RFC 6913 and sip.fax tag
    • Use DTMF events instead
12.  SIP and Unified Communications
Running time = 42 minutes, Quizzes = 7 minutes, Total = 49 minutes
  • Communication Breakdown
    • Playing voicemail tag
    • Can't find people
    • Available but not available
    • More examples of communication problems
  • IM Clients
    • IM client examples and features
    • More in IM clients
  • The Backround 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)
    • The methods in action
    • Publish
    • Subscribe
    • Notify
    • Message
    • is-composing
    • Rich presence
    • 2 places at the same time
  • Presence Federations
    • What is federation
    • Multiple presence sources
    • Super-aggregation
    • Control a conference
    • Why SIP
    • Centralized conferencing
    • Centralized signaling
    • Centralized mixiing
    • Centralized authentication
    • B2BUA
    • Conference components
    • The focus
    • More than one focus
    • Creating a conference
    • Creating a conference: details
    • Adding a participant
    • Adding a participant: details
    • Alternative INVITE with REFER
    • IETF work and conferencing
  • Unified Communications
    • Whats all the fuss
    • Unified confusion
    • What is unified communications
    • From UC to UCaaS
    • Components involved
    • What should UC do
    • 21st century dial tone
    • The unified inbox
    • Unified aware applications
    • Find me-Follow me
    • Device awareness
    • Unified comms for business
    • Do your homework
    • Humans and UC
    • Migrating to UCaaS
    • UCaaS, SIP and the WAN
13.  WAN Broadband Assessment for SIP Trunking 'Optional'
Running time = 54 minutes, Quizzes = 7 minutes, Total = 61minutes
  • Planning Steps
    • Servers and operation
    • Do you have BYOB or OTT
    • Your ISP and ITSP-do they support QoS
    • What is a good time to run assessment tests
    • Assess the components and configuration of your WAN
    • Set up teh WAN assessment network
  • The Assessment System
    • Logging onto the assessment system
    • VoIP assessment items
      • SIP call flows
      • Call metrics
      • Jitter
      • Lost packets
      • MOS estimates
      • Traffic flows
      • Reporting
    • Starting a test
      • Test types
      • Codec selection
      • Number of trunks
      • Number of calls
      • Assessment period
      • Traceroute test
      • ToS and QoS requirements
    • Analyzing a test
      • Call metrics
    • Generating reports
    • Analyzing reports
      • Analyzing MOS
      • Analyzing jitter/delay/packet loss
      • Router RTT responses
      • How to spot problems in the network from the report
    • Advice on how to configure a WAN network

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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Last modified February 20, 2018