SSCA® 'Elite' SIP training + on-line access to the SSCA® 'Elite' Certification test

Everyone…! This training is designed to suit anyone working with SIP such as: 

• Manufacturers of IP PBX and IP Phone equipment
• SIP Security equipment manufacturers
• SIP Trunk service providers
• Hosted/Cloud service providers
• Carriers
• Mobile Network Operators
• Network Design specialists
• Sales and Marketing personnel working with Voice and Video over IP equipment and services

All of these will benefit from this program.

This program assumes the student has a ‘good’ understanding of Data networking technologies along with the ‘basics’ of Voice and Video over IP.

This could be gained through long-term working experience, other certifications such as Cisco’s CCENT/CCNA/CCNP, even The SIP School’s own ‘Networking for VVoIP program’ also available via the website.

Please check carefully as having the skills required will make the SIP learning experience a more productive one.

Once you have enrolled, you will see 14 modules. You can work through the modules in order or simply choose the ones you are most interested in.

1. Core SIP

SIP

• Why SIP?

• What is SIP?

• SIP ‘from the RFC’

• What are ‘Requests for Comments’ – RFCs?

• More than just 3261

• New RFCs

• IETF Working groups

• Based on HTTP 

• Where does SIP fit in?

• SIP Clients and Servers

• SIP User Agents

• SIP Dialog - INVITE

• 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 – Proxy Mode 

• SIP Server – Re-Direct Mode 

• Location Services

• SIP Server in Proxy Mode

• SIP Server in Proxy Redirect Mode

• Stateful and Stateless Proxies

• Location Server

- Components

- Information Sources

- 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)

- 100rel (PRACK)

• Short form ‘compact’ Headers

SDP – the Session Description Protocol

• SDP – The Session Description Protocol 

• SDP in a SIP Message 

• An SDP Example 

• Extending SDP 

• Multiple ‘m’ lines

• Changing Session Parameters 

• SDP Example - Put a call on Hold 

• SDP Example - Call Hold Trace

• Call Hold – Old and New Methods

• Music on Hold example

• INVITE and reINVITE 

SIP Mobility

• 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

• History-info

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

• MIME

• Multiple MIME parts

SIP and B2BUA

• B2BUA - Back to Back User Agent

• B2BUA Example

• B2BUA Benefits and Features

SIP ‘Call Process’ Summary

• The Call Process

2. Wireshark

• What is Wireshark?

• Initial Setup

• Free SIP Account options

• Free @thesipschool.com SIP account / address

• Test Numbers

• Desktop clients

- Jitsi client for testing

- Blink client for testing

- Bria Solo client for testing

- PhonerLite client for testing

• Mobile clients

- Bria Solo for testing

- MizuPhone for testing

- Linphone for testing

- WeePhone SIP for testing

• SIP phone in a Browser

• SIP Browser clients

• Free DID and Credit

• Security and SIP in Wireshark

• Social Study directory

• Security and SIP in Wireshark

• 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

• Use the Cloud

• PCAPs from ‘other’ places

• LAB Exercises

• What are the codes? 

3. SIP and the PSTN

SIP and the PSTN

• SIP to PSTN Overview 

• SIP to PSTN Call Flow 

• SIP to PSTN Detail 

• PSTN to SIP Call Flow 

• SIP to PSTN Call Failure 

• SIP Codes and the PSTN

Early Media

• Early Media explained

• Early Media - SIP to PSTN Call

Early Offer and Delayed Offer

• Early Offer / Delayed Offer

Gateways

• Default Gateway? 

• Gateways and expectations

SIP-T and PSTN Bridging

• SIP-T and SIP-I 

• 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? 

• Inband vs Out-of-band

• RFC 2833 ‘Trace’ example 

• RFC 4733 replaces 2833

• RFC 4734

• SIP INFO 6086

• RFC 2833 ‘Trace’ example 

• SIP INFO ‘Trace’ example

4. SIP, VVoIP and QoS

What is VoIP or Voice over IP?

• 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 Codecs

• Encoding 

• Codecs for Voice 

• Dynamic [RTP Payload type]

• The ‘Codec Test’

• MOS, R-Factor and High Definition (HD) Voice

• Sound tests

• Codecs and Bandwidth

• Packet Rate / Packets per second

• Variable bit rate / Constant bit rate codecs

• Wideband (HD) codecs

• Opus codec

• Opus audio examples

The Real Time Protocol or RTP

• RTP Intro

• RTP Encapsulation 

• RTP Header Trace 

• Real Time Control Protocol (RTCP)

• RTCP-XR (Extended Reports) 

• RTP / RTCP and UDP Ports

Quality of Service

• QoS described

• QoS Issues 

• Measuring Delay 

• Jitter and Packet Loss 

• General VoIP Acceptance Criteria 

• QoS across all Networks 

• 802.1Q – VLANs 

• 802.1Q/P Tagging 

• 802.1P - L2 Classification 

• TOS and DiffServ

• Layer 3 Classification

• DSCP with Assured forwarding (AF) 

• Bandwidth decisions

• Link options – Symmetric DSL (SDSL) 

• Bandwidth (kbps) vs. Packet per Second (pps)

• Network Behavior Analysis

• Issues that can affect QoS

• QoS Summary

• Testing your link

SIP, SDP and VoIP

• SIP in the TCP/IP Model

• SIP and SDP Messages (e.g. Invite and 200OK)

• SIP and SDP Codec mapping

Video over IP

• What is 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

• Assured SIP intro

• Service Provider Architecture

• Proxy and Access Router functions

• Resource-Priority

• Video ‘example’

• Reason Header for Pre-emption Events

• More Proxy details

• Multi-Level Pre-emption and Precedence (MLPP)

• Summary

5. SIP and Media Security

Authentication and Authorization

• SIP Proxy Authentication – in detail 

• 401 and 407 Authorization 

• SIP Authorization

• PROXY Authentication

• Hashing Algorithms [MD5, SHA etc.]

Encryption

• Why Encrypt SIP? 

• Encryption types (Symmetric / Asymmetric)

• Keying and Hashing

• Certificate Authorities 

• Certificate Example

• The Certificate application process

• Installing your new Certificate

• Backup your Private key

• Self-Signed Certificates

• Public Key Infrastructure - PKI

TLS – Transport Layer Security

• TLS in Action

• TLS 1.2 Capture example

• TLS 1.3

• SSL/TLS checking

Securing SIP signaling

• Securing SIP Signalling and then the media

• ‘SIPS’ addressing

• TLS and SIP in action

• Combinations of what you may see…

Securing the Media Stream

• 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 ‘showing’ Crypto

• Crypto – multiple streams

• DTLS/SRTP 

• SRTP with ZRTP

• Encryption summary

SIP trunks and Security 

• SIP trunks 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 ‘example’ software tools

NIST Recommendations

• NIST Recommendations on securing VoIP

3rd party training to extend your knowledge

• The SANS institute

6. STIR/SHAKEN and the ‘identity’ problem

STIR/SHAKEN

• Introduction and topics

Who’s calling?

• The PSTN Caller ID Spoofing problem

• The ‘scale’ of the problem (USA)

Caller Identity

• Caller Identity 

• Enterprise Identities 

• P-Preferred and P-Asserted 

• CNAM/eCNAM

Spoofing

• Spoofing a number - Video

STIR/SHAKEN

• Robocalling and more

• Why this is a problem

• A First Step: STIR/SHAKEN

• STIR/SHAKEN in a Nutshell

• What is a PASSporT?

• Haven’t I Heard of SIP Identity Already?

• STIR/SHAKEN Architecture

• Signed INVITE Example

• PASSporT Token from Example

• PASSporT Token in JSON

• PASSporT Token Protected Header

• PASSporT Token Payload

• The ‘digital signature’

• Fetching Certificate

• Success Call Flow

• Failure Call Flow – Missing Identity Header

• Failure Call Flow – Bad Identity Header

• Certificate management for STIR/SHAKEN

• Partner system

• STI Certificate for Authentication

• Attestation

• The SIP School ‘test system’

• Verstat

• STIR/SHAKEN in action

• Video - Authentication to Verification

• Service providers with SHAKEN

Enterprises and the ‘A’

• The ‘Attestation gap’

• How to ‘fix’ the gap – some options

• Delegate Certificates

• Delegate Certificates base PASSporT

• Delegate Certificates for OTT providers

• Enterprise Certificates

• TN Databases

• Distributed Ledger

• Trust

• Getting ‘Creative’

• Which option is best?

Rich Call Data

• What is Rich Call Data?

• Rich Call data location

• Adding Rich Call Data

• Rich Call Data in the token

• RCD jCard / rcdi

• RCD and Delegated certs

International STIR/SHAKEN

• International Attestation

• ATIS and International calls – Bilateral

• ATIS and International calls – Central Registry

Out of Band STIR/SHAKEN

• Why is this a problem?

• Out of Band (OOB) STIR with TDM

• Another OOB example

Call Diversion

• Diverted call flow

• “div” in a SIP INVITE

• “div-o”

Call Analytics

• An overview

What’s happening now

• The Traced Act

• Where are we now?

• ‘Other Services and Techniques

• Bringing it all together

• Possible extensions

• FCC mandate

• Robocall mitigation

• Find the call originator

• Industry Traceback Group (ITG)

Resources

• ‘Some’ other companies offering STIR/SHAKEN

• ATIS testbed

• STIR and SHAKEN references

• STIR/SHAKEN conference

• Best practices.

7. Firewalls, NAT and Session Border Controllers

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

• NAT Hairpinning

• Media Hairpinning/Tromboning

• Multiple NATs 

NAT in more detail

• Types of NAT 

• NAT – Full Cone 

• NAT – Restricted Cone 

• NAT – Port Restricted Cone 

• NAT – Symmetric 

• New Terminologies

- Mapping and Filtering

• Endpoint Independent Mapping

• Address Dependent Mapping

• Address and Port Dependent Mapping

• NAT Filtering Rules

The NAT & Firewall ‘problem’

• The NAT problem

• The NAPT or (PAT) Problem

• The Firewall Problem 

The Solutions

• Interactive Connectivity Establishment (ICE)

• ‘Classic STUN’ (Session Traversal Utilities for NAT) 

• VIA received parameter

• VIA rport parameter

• Problems with ‘Classic’ STUN 

• Symmetric RTP 

• STUN RFC 8489

• Request and Response example

• TURN (Traversal Using Relays around NAT) 

• ICE ‘In Theory’

• Candidate information and other ‘ICE stuff’.

• ICE ‘In action’

• ICE tags

• ICE-Lite and Trickle-ICE

• ICE Client settings

• More on ICE

• Media Proxy

• Application Level Gateway 

• SIP Aware Firewalls - Incoming 

• SIP Aware Firewalls - Outgoing 

• Universal Plug and Play (UPnP)

• ‘Near end’ NAT

• ‘Far end’ NAT

• GRUU (Globally Routable User Agent)

Session Border Controllers

• SBC for the Enterprise and SBC for the ITSP 

• Recommended Session Border Controller features

• SBCs in Action! 

• SBCs and message manipulation / normalization

• SIP ‘Refer’ problems

• SBC ‘Interop’ example

• SBC Manufacturers – examples

• SBCs in the Cloud / as a Service

8. SIP Trunking

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

• Trunk ‘bursting’

• Elastic SIP

SIP Trunks, MPLS and SD-WAN

• MPLS, basic explanation

• MPLS Label format

• MPLS in a MAC frame

• MPLS example network

• MPLS benefits

• Your own private WAN

• but ‘Not the only client’

• Separate MPLS networks

• VPLS explained

• WAN Optimization, Hybrids and SD-WAN 

• Software Defined WANs explained

- Orchestrator

- Policies

- SD-WAN device capabilities

Setting up a SIP Trunk

• SIP trunk configuration on ‘sample’ PBX

• Outbound ‘Dialling’ Rule 

• Calling across the trunk

• Call analysis with Wireshark

- Call Flow

- SIP ladder

Modes of Operation

• Registration Mode

• Static Mode

Security and SIP Trunks

• SIP Trunk Security - Overview 

Microsoft (a little)

• Skype for Business and SIP Trunks

• Servers and Protocols

• Microsoft Teams and Calling plans

• Microsoft Teams and Direct Routing

Troubleshooting and Interops

• SIP Trunks and Common Problems 

• The SIP Forum 

• SIP Connect 

• SIP Connect 1.1 onto 2.0

• Interoperability testing

Choosing an ITSP

• Understanding ITSP Offerings 

• 'Sticking points’?

• What you may need in the future

• SIP trunk ‘connectivity’ 

- Things to watch out for when connecting to your ITSP

• ‘Finding’ an ITSP

• SIP trunking Checklist for ITSP evaluation

9. Testing, Troubleshooting and Interoperability

Setting up your test environment

• Your Setup

• Using SIP IP Phones and Softphones

• Jitsi, Blink, Bria Solo and PhonerLite 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 ‘Revisited’

• Colours and the Intelligent Scrollbar

• Packet ‘Marking’ and ‘Comments’

• New Packet Window

• Exporting ‘Specified’ Frames

• RTP Streams

• TShark (Terminal-based Wireshark)

• 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

• SIPIT 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

More SIP Testing Tools

• SIP Workbench

• SIP Scan

• Visualware for testing

• HoverIP

• NSLookup 

• Voip-info for more tools!

• Using the NET to find answers

• Other SIP Resources

10. ENUM, Peering and Interconnect

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 ‘Mirrors’

• ‘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

• NAPTR and DNS records

• 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 and Interconnect (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’

• Who’s involved?

IP-NNI

• Network-to-Network interface [NNI]

• ATIS and the SIP Forum for NNI

• Benefits of SIP NNI

• History of IP NNI Effort

• Layers of Interconnection

- IP Interconnection Profile

- IP Interconnection Routing

• IP NNI Profile

• IP NNI Trust Model

• Identities

• Codecs

• DTMF and Fax

• Fault Isolation and Troubleshooting

• QoS

• SIP-Specific Details of IP NNI

• IP Interconnection Routing

• Aggregate Approach

• Per-Telephone Number (TN) Approach

• What’s Next for NNI

11. SIP in the Cloud

‘Types' of ‘Cloud’

• Public, Private and Hybrid

Hosted SIP

• What Hosted SIP service is

• Hosted functions and features

• Example Network including ‘failover’

• ‘Hosted’ clients in action

• Why Hosted – Benefits and things to consider

• Why on-site PBX – Benefits and things to consider

The Cloud and ‘Anything as a Service’

• Pizza as a Service

• IaaS / PaaS / SaaS

• SaaS in ‘reality’

- What is Virtualization?

- Virtual Machines

- Emulation

- Virtual Machines (contd.)

- Network Functions Virtualization (and VNF)

- SBCs in the Cloud / as a Service

• Virtualization of the PBX

• Our own Network examples

• Moving to the Cloud

- Example with - AWS / Azure and Twilio

- Call flow in the example ‘Cloud based’ system

Video demos of ‘Cloud systems’

• Visualising the migration to the cloud

• Cloud marketplaces

• Azure – Anyone VM SBC

• RDP connection to the SBC

• Anynode configuration

• SBC and Twilio 

• AWS instances

• AWS and 3CX (PBX)

• SBC, PBX and Twilio

• Capturing the ‘Cloud call’

Auto Provisioning 

• Auto Provisioning Example

• Boot Server

• Client Config

• Client boot sequence

• Client config download

• RFC 6011

• Zero-Touch Provisioning

• Zero-touch example

• Benefits of Hosted SIP Service

• Benefits of Onsite PBX and SIP trunks

Troubleshooting

• Troubleshooting a cloud service

• What to look for – Dashboards

• What to look for – PCAP files

• Monitoring across cloud-based services

12. SIP in Cellular networks

SIP in Cellular networks

• Network Overview

• RAN, eNodeB, EPC, IP Core and 3GPP

• 4G, LTE, LTE Advanced LTA-Pro, WiMAX2

• The RAN and EPC

• Default Bearer Setup

• Introduction to the Servers and Functions in the IMS

- CSCF

- S-CSCF

- P-CSCF

- I-CSCF

- Home Subscriber Server HSS

- Application Server

- TAS

- PSCF

- DNS and ENUM

• 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 Preconditions in Action

- With Codec examples within SDP

• SIP Call flow for VoLTE

• Quality settings ‘recap’

• VoLTE media flow

• More on VoLTE

• The IMS

• Layers architecture

- Application

- IMS / Session Control

- Access and Transport

- 3GPP

• Multiple access devices 

- RCS and OTT

• Who provides IMS solutions?

• IPX and Peering for Security, QoS and SLAs

• GSMA and IR.92

• HD Voice News

• VoLTE media flow

• More on VoLTE

• The IMS

• Layers architecture

- Application

- IMS / Session Control

- Access and Transport

- 3GPP

• Multiple access devices 

- RCS and OTT

• Who provides IMS solutions?

• IPX and Peering for Security, QoS and SLAs

• GSMA and IR.92

5G

• Benefits of 5G

• 5G service examples

• Voice over 5G

• 5G – NSA Option 3x (and more)

• Mandatory Codecs

• SIP in 5G

• Summarizing the state of 5G 

• Resources

• Coverage Checker

13. SIP and Fax over IP

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?

14. SIP in UC, UCaaS and CPaaS 

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 

• Clients and UC providers

• More IM Clients

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) 

• 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 

• Inter-Domain Federation

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 

• Creating a Conference

• Creating a Conference: Details

• Adding a participant

• Adding a participant: Details

• Alternative INVITE with REFER

• IETF work and Conferencing 

Unified Communications

• What’s 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

• UCasS, SIP and the WAN

CPaaS and APIs

• Introduction to CPaaS and APIs

• What is an API?

• Communications API examples

• REST APIs and more...

• More examples of API use with your PBX

• Creating CPaaS applications

• Creating an IVR

• UCaaS v CPaaS

You can gain access to the certification test separately or with a ‘bundle’ license – check license ‘purchase’ options carefully.

The SSCA® ‘Elite’ certification is recognized in the Telecommunications world as the only certification on SIP to strive for ‘Globally’.

It is endorsed and supported by USTelecom, Incompas, the Comms Council UK along with BICSI and an extensive number of Manufacturers, Service providers, Cloud providers, Carriers and Mobile Network Operators.

To prepare for the certification test, each SIP training module has its own ‘mini’ quiz at the end to help delegates ‘gauge’ how well they are doing.

NOTE: An access license for any training course and certification test is for 12 months from the date of purchase.