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Conquest Communications has teamed up with the SIP School to offer you this great course. If your have an IT background you will easily handle the terminolgy and concepts in this course. If you do not have a strong background in IT please complete the LANs WANs and VoIP course first. SSCA SIP Training alone is $375 USD or with Certification Test is $451.25 USD. Consider buying the two courses together, LANs WANs and VoIP plus SSCA SIP Training and Certification Test for $773.10 USD as a bundle and save some money. Your 5% discount still aplies so contact me for your discount code. OverviewThe SIP School™ is ‘the’ place to learn all about the Session Initiation Protocol also known as SIP. There is so much information on the internet about SIP that is both hard to read and poorly presented making it difficult for people to learn about this most important protocol. So The SIP School™ with its lively, clear and fully animated eLearning program has become the only place to enroll to learn about SIP. Who would benefit from the SSCA® SIP training program?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 and Carriers, Network Design specialists, Sales and Marketing personnel working with VoIP equipment and services; all of these will benefit from this program. What’s in the SSCA® SIP training program?Once you’ve enrolled with you’ll see nine modules. You can work through the modules in order or simply choose the ones you are most interested in. The modules are listed here but for more detail, please look further into this document. Click on a topic to see the details.
Total Running time for this program (including time taken to work on all the labs) is approximately 17¾ hours from the start to finish though the time will vary based on the student’s own experience and of course, how much time they want to spend on the material and if they want to replay some modules. This does not include study time for the SSCA® or the taking of the SSCA® final test itself. Become a ‘SIP School Certified Associate’ or SSCA®You can gain access to the test separately or with a ‘bundle’ license – check license ‘purchase’ options carefully.
The SSCA® certification is recognized in the Telecommunications world as the only certification on SIP to strive for ‘Globally’. It is endorsed and supported by the TIA (Telecoms Industry Association) along with BICSI and a rapidly growing number of Manufacturers, Service providers and Carriers. Module times Running time = 66 minutes Quizzes = 7 minutes Total = 73 minutes SIP (The Session Initiation
Protocol) is described in this module along with the many other Components and
Services that will be encountered on a SIP based network Topics: SIP – Who Benefits 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 Registration Re-Registration 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 INVITE – Example RESPONSE – Example
SIP Request
Methods SIP Response Codes
SIP Headers SIP HEADER -
INVITE SIP HEADER - 200 Response SDP – the Session Description
Protocol 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 – The Session
Description Protocol SDP Component in a
SIP Message SDP Example Extending SDP 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 - 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 MIME Multiple MIME
parts SIP and the PSTN 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 Module times Running time = 10
minutes Quizzes = 1
minutes Lab – ‘Various’ ~
approx 80 minutes Total = 91
minutes Note: If the
student wishes to take more time over the module – this could run to a few
hours of learning. It’s completely dependent on the students desire to learn Wireshark This module on Wireshark is an introduction and is intended to get
students setup quickly so that they can capture traffic to analyze
during the Core module and the rest of the course. More advanced Wireshark training can be found in the Troubleshooting,
Testing and Interoperability module of this course. Topics: Wireshark What is Wireshark? Your Initial Setup
SIP account with
Voipuser.org X-lite client for testing Configure X-Lite Download Wireshark Wireshark – Basic Layout Wireshark icons Using Wireshark – Capturing Using Wireshark – Simple Filters Using Wireshark – More SIP statistics Using Wireshark – RTP Statistics Saving Captures Over to you! What are the
codes? Link to
Troubleshooting module for Advanced Wireshark Module times Running time = 25
minutes Quizzes = 7
minutes Total = 32
minutes SIP Networks will
of course have to allow connections to and from the PSTN. This module works
through SIP and PSTN connectivity Topics: SIP-T 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 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 RFC 4733 replaces
2833 SIP INFO ‘Trace’
example Top of Page Module times Running time = 37
minutes Quizzes = 7
minutes Lab – ‘Various’ ~
approx 10 minutes Total = 54
minutes This module is a
refresher module on the basics of Voice over IP and also focuses on
components that are important to a SIP based Network Topics: 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 IP PBX Voice Sampling and Codec Encoding Codecs for Voice Try the Codec Test
High Definition
(HD) Voice Sound tests Wideband (HD) codecs 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 Module times Running time = 38
minutes Quizzes = 7
minutes Lab – ‘Various’ ~
approx 120 minutes Total = 165
minutes SIP Security is
a complex issue and this modules covers many SIP Security problems along with
possible solutions Topics: 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 Crypto multiple
streams RFC 4474 for
Caller Identity 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 Firewalls, NAT and Session Border Controllers Module times Running time = 32
minutes Quizzes = 7
minutes Total = 39
minutes Inevitably, all
IP traffic comes across a Firewall / NAT device and in the case of SIP they can
stop the flow of SIP message. This module looks at the problems and the
solutions including Session border controllers. Topics: 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 (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 Recommended
Session Border Controller features SBCs in Action! Module times Running time = 40
minutes Quizzes = 7
minutes Lab – ‘Setting up
SIP Trunks’ ~ approx 120 minutes o (If student has access to a SIP server) Total = 167
minutes This module
teaches the theory of connecting a SIP based PBX to the PSTN and it is the
foundation of vendor specific Trunking modules.
Topics: SIP Trunks 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 Disaster Recovery
‘Expanded detail’ Disaster
Recovery – Last resort?
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 SIP Trunking
and MPLS MPLS, basic
explanation Your own VPLS but ‘Not the only client’ Separate MPLS
networks 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 Some PBX Requirements Enterprise PSTN
Identities P-Preferred and
P-Asserted Call Progress
Tones 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 SIP Connect
‘Missing pieces’ SIP Connect 1.1 Choosing an ITSP Understanding ITSP
Offerings Resource Websites TMCnet – Sip trunking
Siptrunk.org No Jitter - Hotzone Testing, Troubleshooting and Interoperability Module times Running time = 65
minutes Quizzes = 7
minutes Lab – ‘Various’ ~
approx 240 minutes Total = 312
minutes Learn how to Monitor and Test SIP devices and services using Wireshark. This tool enables delegates to analyze call control messages to establish where a fault
may lie in your SIP infrastructure. Full examples are provided and delegates
are encouraged to follow the exercises to try for themselves. Setting up your
test environment Using SIP IP
Phones Using SIP Softphones Even more SIP Softphones SIP Communicator Choosing a
‘Trial/Test’ ITSP Getting Free ITSP
Accounts Configuring your Softphone Get a SIP URI of
your own SIP2SIP accounts Configuring SIP
Communicator with a SIP2SIP account Using ‘Test
Numbers’ Multiple Setup
options for you to try Configure X-Lite and SIP Communicator on the same PC for testing Example - The SIP
Phones @ The SIP School™ 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 Saving Captures Wireshark in more depth! SIP Statistics RTP / VoIP Capture
and Playback More ‘SIP ladder’
analysis Coloring rules More ‘filter
expressions’ More Help on Wireshark if you need it You try Where to Capture? Interoperability Testing Interop Testing Why Interop can be tough Different
interpretations in the RFC 3261 BLISS – Basic
Interoperability for SIP Services 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 Workbench SIP Monitoring
example app SIP Scan TestYourVoIP.com HoverIP NSLookup SIP Center and Voip-info for more
tools! Using the NET to
find answers The SIP Wiki Module times Running time = 32
minutes Quizzes = 7
minutes Lab – ‘Registering
/ Testing ENUM’ ~ approx 20 minutes Total = 59
minutes ENUM (along with DNS) is developing
into an essential protocol on SIP networks and its purpose is to assist in
finding destination SIP devices from a single SIP address. Topics: 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) MARTINI Types of ENUM Different ‘Types’
of ENUM The Problems with
‘Public’ ENUM Example –
‘Private’ ENUM Example –
‘Operator’ ENUM Stay ‘On-Net From ITSP to PSTN
and Back…! Peering Profiles
and Agreements A few providers ViPR Verification
Involving PSTN Reachability (ViPR). What is ViPR ViPR and P2P Initial PSTN Call ViPR Call Record Query the DHT DHT query and
Validation The Next call is a
SIP Call ViPR Summary Try for yourself Register your
number Testing ENUM ENUM and the future How is ENUM moving
forward? Useful Links SIP and Unified Communications Module times Running time = 48
minutes Quizzes = 7
minutes Total = 55
minutes SIP and Unified
Communications shows you how SIP underpins all the elements of Unified
Communications to realize efficiencies that a successful implementation
promises to business. Topics Include 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 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 UC in a SIP
network UCI Forum The UCI Forum -
Challenges UCI Forum goals UCIF website Relevant RFCs RFCs Galore Top of Page Please call me on 07-41596950 for an obligation free discussion. I am currently doing this training and am finding it to be good indeed. It is filling in many gaps I have as a self taught, self read student of the IP. In fact I bought the bundle training offer of Session Initiated Protocol and LANS/WANS and VoIP. I strongly recommend this training to you, regards Trevor. |
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