Session Initiation Protocol - SIP Training and Certification Course

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

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:

Core SIP
Wireshark
SIP, the PSTN and SIP-T
SIP, VVoIP and QoS
SIP Security
Firewalls, NAT and Session Border Controllers
SIP Trunking
Testing, Troubleshooting and Interoperability
ENUM and DNS and Peering
SIP in the Cloud, LTE, the IMS and VoLTE
SIP and Fax over IP
SIP with Unified Communications
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.

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SIP Training and Certification Self Study Online

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Payment in advance is requested for access to the self-study course.

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