Beruflich Dokumente
Kultur Dokumente
1 . Core SIP
2. Wireshark
3. SIP and the PSTN
4. SIP, VVoIP and QoS
5. SIP Security and Identity
6. Firewalls, NAT and Session Border Controllers
7. SIP trunking
8. Testing, Troubleshooting and Interoperability
9. ENUM, Peering and Interconnect
10. SIP in the Cloud, LTE, the IMS and VoLTE
1 1 . SIP and Fax over IP
12. SIP with Unified Communications
NOTE:
th
This program was last updated on March 11 2019. All new / edited sections are
shown in a bold, blue font.
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The SSCA® SIP How long will it take to work through?
training program Running times for this program are approximate as 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.
NOTE: Timings do not include the additional Skype for Business material and labs.
The time it takes to ‘Play’ all of the slides and Videos (also known as the
‘running’ time) plus complete all of the quizzes is = 13hours 31mins.
The TOTAL time will be more than this and dependent on factors such as
slides being replayed, note taking, working on Labs (some of which can take a
few hours), also doing some ‘extra’ work with the Software tools provided for
the labs which we believe is a great idea as it increases student skills.
Further study time for the SSCA® and the taking of the SSCA® final test itself
should also be accounted for.
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.
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SSCA® SIP
Core SIP
Description
1 Core SIP topics
SIP
SIP (The Session Why SIP?
Initiation Protocol) What is SIP?
is described in SIP ‘from the RFC’
this module along What are ‘Requests for Comments’ – RFCs?
with the many More than just 3261
New RFCs
other components IETF Working groups
and Services that Based on HTTP
will be Where does SIP fit in?
encountered on a SIP Clients and Servers
SIP based SIP User Agents
network SIP Dialog - INVITE
SIP System Architecture
The URI - Unique Resource Identifier
Module times
SIP Addressing
SIP Addressing Examples
Running time
85 minutes SIP Servers and Operation
Registration
Quizzes Re-Registration
10 minutes SIP Proxy servers and why we need them
Proxy Server ‘State’ types
DHCP and SIP
Total SIP Proxy – Trapezoid Model
95 minutes 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
o Components
o Information Sources
o Example
SIP Messaging
Request Methods
Response Codes
SIP Headers
INVITE – Example
RESPONSE (200 OK) – Example
More on Headers
Support and Require Headers
o Timer (Session Times)
o 100rel (PRACK)
Short form ‘compact’ Headers
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SSCA® SIP SDP – the Session Description Protocol
Core SIP 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
MIME
MIME
Multiple MIME parts
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SSCA® SIP
Wireshark
2 Wireshark topics
Description
This module on
Wireshark
What is Wireshark?
Wireshark is an Initial Setup
introduction that is Free SIP Account options
intended to get Free @thesipschool.com SIP account / address
students setup Desktop clients
quickly so that o Jitsi client for testing
o Blink client for testing
they can capture o X-lite client for testing
traffic to analyze o PhonerLite client for testing
during the Core Mobile clients
SIP module and o MizuPhone for testing
o Linphone for testing
the rest of the o WeePhone SIP for testing
course. More Social Study directory
advanced Free DID and Credit
Wireshark training SIP test numbers
can be found in Download Wireshark
Wireshark
the o Introduction
Troubleshooting, o Menus, Screens and Views
Testing and o Capturing traffic
Interoperability o Profiles
o Display Filters
module of this o Capture Filters
course. o SIP Packet Analysis
o SIP ladders and Audio Playback
Module times
o Other Menu options
o SIP INVITE Analysis
o Follow a UDP Stream
o Frame Relationships
Running time o Colouring Rules
41 minutes o RTP Streams
LAB Exercises
Quizzes What are the codes?
1 minute
Labs equate to
exercises
suggested within
the module
Total
122 minutes
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SSCA® SIP
SIP and the PSTN
3 SIP and the PSTN topics
Description
SIP Networks will
SIP and the PSTN
SIP to PSTN Overview
of course have to SIP to PSTN Call Flow
allow connections SIP to PSTN Detail
to and from the PSTN to SIP Call Flow
PSTN. This SIP to PSTN Call Failure
module works SIP Codes and the PSTN
through SIP and
PSTN Early Media
connectivity Early Media explained
Early Media - SIP to PSTN Call
Module times
Early Offer and Delayed Offer
Early Offer / Delayed Offer
Running time
26 minutes
Gateways
Quizzes Default Gateway?
7 minutes Gateways and expectations
Telephony routing over IP (TRIP)
TRIP Examples
Total
SIP-T and PSTN Bridging
33 minutes
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
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SSCA® SIP
SIP, VVoIP and
QoS
4 SIP , VVoIP and QoS topics
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SSCA® SIP SIP, SDP and VoIP
SIP, VVoIP and SIP in the TCP/IP Model
QoS
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
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SSCA® SIP
SIP Security and
Identity SIP Security and
5 topics
Description
SIP Security and
Identity
Identity are
complex issues Authentication and Authorization
and this module SIP Proxy Authentication – in detail
covers many 401 and 407 Authorization
Security and SIP Authorization
Identity problems PROXY Authentication
along with Hashing Algorithms [MD5, SHA etc.]
possible
solutions. Encryption
Why Encrypt SIP?
Module times
Encryption types (Symmetric / Asymmetric)
Keying and Hashing
Caller Identity
RFC 4474 for Caller Identity
Caller Identity
DTLS/SRTP
Ongoing developments for Identity
Enterprise PSTN Identities
P-Preferred and P-Asserted
CNAM
STIR/SHAKEN
The PSTN Caller ID Spoofing Problem
Types of Fraudulent calls
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
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SSCA® SIP Caller Identity (continued)
SIP Security and RFC 4PASSporT Token from Example
Identity
PASSporT Token in JSON
PASSporT Token Protected Header
PASSporT Token Payload
Fetching Certificate
Success Call Flow
Failure Call Flow – Missing Identity Header
Failure Call Flow – Bad Identity Header
Next steps and references
NIST Recommendations
NIST Recommendations on securing VoIP
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SSCA® SIP
Firewalls, NAT and Firewalls, NAT and Session
Session Border
6 topics
Controllers
Border Controllers
Description
Overview
Inevitably, all IP Issues to address
traffic traverses a
Firewall / NAT
device and in the
Firewalls
What does a Firewall do?
case of SIP these Are Firewalls effective?
devices can stop
the flow of SIP
messages. This
NAT or Network Address Translation
What is NAT?
module looks at NAT Request
the problems and NAT Response
the solutions UDP Hole punching
including a focus NAT Hairpinning
on Session Media Hairpinning/Tromboning
Multiple NATs
Border
Controllers.
NAT in more detail
Module times
Types of NAT
NAT – Full Cone
NAT – Restricted Cone
Running time NAT – Port Restricted Cone
67 minutes NAT – Symmetric
New Terminologies
o Mapping and Filtering
Quizzes Endpoint Independent Mapping
10 minutes Address Dependent Mapping
Address and Port Dependent Mapping
Total NAT Filtering Rules
77 minutes
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 5389
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)
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SSCA® SIP The Solutions (continued)
Firewalls, NAT and ‘Near end’ NAT
‘Far end’ NAT
Session Border GRUU (Globally Routable User Agent)
Controllers
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
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SSCA® SIP
SIP trunking
7 SIP trunking topics
Description
+ ‘optional’ Skype for Business access
This module
teaches the
theory of
SIP Trunks
What is a SIP Trunk
connecting a SIP Alternative to TDM
based PBX into Separate Data and Voice connections
an ITSPs own Converging the network
network and also SIP Trunks and Codecs
focuses on SIP Trunk Benefits
Network
technologies, SIP Trunking – In More Depth
Security, SIP Trunk Capabilities
Troubleshooting SIP Trunking Network Examples
as well as offering SIP Peering
Peering problems?
advice on how to Least Cost routing (LCR)
select an ITSP for Disaster Recovery
your company or Disaster Recovery ‘Expanded detail’
clients. Disaster Recovery – Last resort?
Number Consolidation
Virtual Presences
Module times
Trunking Variations
Running time Single Site, No ‘Forklift’
87 minutes Single Site, TDM PBX
Single Site, Converged
(not including the Converged – SIP/IP PBX
‘optional’ Skype Multiple Site, ‘Converged’
Multiple Site, ‘Converged’ + central SBC
material) Multiple Site, ‘Converged’ + Multiple SBCs
Media Gateways
Quizzes
7 minutes
SIP PBX to Non-SIP PBX
SIP PBX to Non-SIP PBX, Call Flow
Labs equate to
multiple exercises
‘suggested’ within SIP Trunk Performance
the module such Connection types
as SIP PBX and The ADSL issue
Codecs, Voice and Data
trunk Symmetric DSL (SDSL)
configuration. Bandwidth Calculator
Testing your link
Approx. lab time ADSL Developments
120 minutes Fibre Options
Trunk ‘bursting’
Elastic SIP
Total
214 minutes
SIP Trunking, MPLS and SD-WAN
NOTE: Total MPLS, basic explanation
MPLS Label format
including the
MPLS in a MAC frame
‘optional’ Skype MPLS example network
material is MPLS benefits
approx. 274 Your own private WAN
minutes but ‘Not the only client’
Separate MPLS networks
VPLS explained
WAN Optimization, Hybrids and SD-WAN
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SSCA® SIP SIP Trunking, MPLS and SD-WAN (continued)
SIP trunking Software Defined WANs explained
o Orchestrator
o Policies
o SD-WAN device capabilities
Modes of Operation
Registration Mode
Static Mode
More on SBCs
The ‘corporate’ SBC
SIP REFER issues
Microsoft Teams
Microsoft Teams and SIP connectivity
Choosing an ITSP
Understanding ITSP Offerings
'Sticking points’?
What you may need in the future
SIP trunk ‘connectivity’
o Things to watch out for when connecting to your ITSP
‘Finding’ an ITSP
SIP trunking Checklist for ITSP evaluation
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SSCA® SIP
Testing, Testing, Troubleshooting
troubleshooting
8 topics
and Interop.
and Interoperability
Description
Setting up your test environment
Learn how to Your Setup
‘monitor’ and Test Using SIP IP Phones and Softphones
SIP devices and Jitsi, Blink, X-Lite and PhonerLite setup – reminder.
services using Linphone, MizuPhone and WeePhone SIP setup - reminder
Wireshark. This Choosing a ‘Trial/Test’ ITSP
Get ‘another’ SIP account
tool enables SIP2SIP account
delegates to Configure Blink and Jitsi on the same PC for testing
analyze call Using ‘Test Numbers’
control messages
to establish where Wireshark
a fault may lie in a Where to ‘capture’
SIP infrastructure. More options for Packet Capturing
Full examples are Wireshark ‘Revisited’
provided and Colours and the Intelligent Scrollbar
delegates are Packet ‘Marking’ and ‘Comments’
New Packet Window
encouraged to
Exporting ‘Specified’ Frames
follow the RTP Streams
exercises to try TShark (Terminal-based Wireshark)
for themselves. PCAP-ng and PCAP formats
Alternatives to Wireshark
Module times
You try!
Interoperability Testing
Running time Interop Testing and why Interop can be tough
56 minutes Different interpretations in the RFC 3261
Interop Test Scenario
Quizzes Interop Test Operations
7 minutes Sample Interop Traces with Wireshark
Wireshark example videos to help understand interop issues
More Sample captures
Labs equate to Video call testing
various exercises Video tests with Wireshark trace analysis
suggested within ‘Basic’ Interop Test List
the module SIPIT events
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SSCA® SIP
ENUM, Peering and
Interconnect ENUM, Peering and
9 topics
Interconnect
Description
ENUM (along with
DNS) is
ENUM Explained
developing into an
essential protocol
on SIP networks What is E.164?
What is ENUM?
and its purpose is Why ENUM?
to assist in finding Call Routing and ENUM - Example
destination SIP
devices from a
single SIP
Enum, DNS and Domains
Why are we using DNS?
address. Peering DNS Operation
is also discussed DNS Root Server ‘Mirrors’
as more and more ‘Finding’ Domain name servers using NSLookup
services providers The e164.arpa Domain
are ‘connecting’ Approved ENUM Delegations (RIPE)
TIERS 0, 1, 2 and 3
together to allow
e164.arpa Domain ‘in action‘
a full IP to IP ENUM Delegations
experience. Address of Record
Inclusion of the PSTN to SIP UA – Example
IP-NNI The ENUM Query
recommendation DNS Response to an ENUM query
NAPTR and DNS records
builds on Finding SIP servers using the tool - DIG
‘Peering’ to IP to PSTN (Simplified)
enable ITSPs to RFC 6140
‘Peer’ in a more
effective manner. Types of ENUM
Different ‘Types’ of ENUM
Module times
The Problems with ‘Public’ ENUM
Example – ‘Private’ ENUM
‘Carrier’ ENUM and e164enum.net
Running time
63 minutes
Peering and Interconnect (for VoIP and Video)
Stay ‘On-Net
Quizzes From ITSP to PSTN and Back…!
7 minutes Loss of features with the PSTN
Peering Profiles and Agreements
Total Bi-lateral Peering
70 minutes 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
o IP Interconnection Profile
o IP Interconnection Routing
IP NNI Profile
IP NNI Trust Model
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SSCA® SIP IP-NNI (continued)
ENUM, Peering and Identities
Interconnect
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
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SSCA® SIP
SIP in the Cloud,
LTE, the IMS and SIP in the Cloud, LTE,
VoLTE 10 topics
Description the IMS and VoLTE
SIP is becoming
critical to phones ‘ Types' of ‘ Cloud’
and servers Public, Private and Hybrid
involved in a
Hosted SIP
hosted setup.
SIP is also critical
when used by What Hosted SIP service is
Hosted functions and features
VoLTE calls in Example Network including ‘failover’
order to make ‘Hosted’ clients in action
voice sound great Why Hosted – Benefits and things to consider
on a mobile Why on-site PBX – Benefits and things to consider
device. This
module aims to Auto Provisioning
make students Auto Provisioning Example
aware of SIPs Boot Server
role in all of these Client Config
environments. Client boot sequence
Client config download
RFC 6011
Module times Zero-Touch Provisioning
Zero-touch example
Benefits of Hosted SIP Service
Running time Benefits of Onsite PBX and SIP trunks
68 minutes
Quizzes
PBX in the Cloud with SIP Trunks
Cloud and SIP trunk Config overview
7 minutes Configuring a SIP trunk on the ‘Hosted’ PBX (in the Cloud)
E.164 Outbound routing example
Total Calling from Softphone via Cloud PBX
75 minutes PCAP for analysis
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SSCA® SIP SIP, LTE, the IMS and VoLTE (continued)
SIP in the Cloud, VoLTE media flow
LTE, the IMS and
More on VoLTE
The IMS
VoLTE Layers architecture
o Application
o IMS / Session Control
o Access and Transport
o 3GPP
Multiple access devices
o RCS and OTT
Who provides IMS solutions?
IPX and Peering for Security, QoS and SLAs
GSMA and IR.92
HD Voice News
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SSCA® SIP
SIP and Fax over IP
Description
11 SIP and Fax over IP topics
A lot of
Faxing Basics
Faxing background
companies are T.30 Fax signaling
now trying to run Associated tones and protocols
Fax services The ITU and TIA standards
across SIP trunks
and finding it’s not Fax over IP
an easy service to Fax over IP benefits
get working From the old to the new
successfully. This Intro to FoIP
module intends to FoIP and SIP trunks
describe the Protocol conversions
various flavors of
Fax over IP along Fax Protocols
what should be G.711 Pass-through
focused on in T.37 Store and Forward
order to T.38 Relay
Where does SIP fit in?
troubleshoot any
UDPTL
issues. Protocol options for the future
Total
Bandwidth
T.38 and G.711 network traffic
40 minutes
Troubleshooting
The basics
More complex issues to watch out for
Ongoing Efforts
RFC 6913 and sip.fax tag
Use DTMF events instead?
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SSCA® SIP
SIP and Unified
Communications SIP and Unified
12 topics
Description
SIP and Unified
Communications
Communications
shows you how
SIP underpins all
Communication Breakdown
Playing Voicemail tag
the elements of Can’t find people
Unified Available but not Available..!
Communications More Examples of communication problems
to realize
efficiencies that a IM Clients
successful IM Client Examples and Features
implementation More in IM Clients
promises to
business.
The Background Stuff
The IMPP working group
Module times IMPP and CPP
More IMPP work
SIMPLE
Running time
How it all works
42 minutes
Quizzes Presentity
A Basic SIP subscription
7 minutes Multiple Presence States
Presence and P2P
Total A Presence Network
49 minutes 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
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SSCA® SIP Conferencing (continued)
SIP and Unified INVITE other participants
Communications
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
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