Description

In this lecture-only course, you will learn core concepts of how the Internet Protocol (IP) carries a Voice over IP (VoIP) packet. You will learn the fundamentals of Session Initiation Protocol (SIP) architecture, SIP-related IP services, the advantages and disadvantages of SIP Trunking as well as Quality of Service (QoS)-Related Protocol.

What You'll Learn

Core concepts of how Internet Protocol (IP) carries a VoIP packet
Advantages and disadvantages of SIP Trunking
Understand how DHCP and DNS supports IP telephony
Real-Time Transport Protocol (RTP)
Session Initiation Protocol (SIP) – Call set up, Instant Messaging, Presence
Session Description Protocol (SDP)
SIP proxy, Session Border Controller (SBC), and SIP softswitch
Media Gateway Control Protocol (MGCP) analysis
MGCP architecture
How to implement QoS to ensure the highest voice quality over your IP networks
The impact of jitter, latency, and packet loss on VoIP networks
How Wireshark can decode and troubleshoot RTP, SIP, and MGCP call flows
Discuss trixbox Softswitch and SIP proxy
Discuss SIP gateways and softphones

Who Should Attend

This class is for people who need to understand VoIP technology. IT managers, technical sales/marketing personnel, consultants, network designers and engineers, product design engineers developing integrated-services products, telecom technicians and managers integrating PBX services within data networks, and systems administrators who will manage a converged network would benefit from this course.

Course Overview

Module 1 Packetizing Voice

Telephony Architecture
Introduction to the VoIP Standards
Connecting VoIP to PSTN
Traffic Engineering
PSTN to VoIP Using Magic
Voice Digitization
Companding Mu-Law vs. A-Law
Time Division Circuit Switching
Voice Packet
The 20-Millisecond Voice Packet
The 60-Millisecond Voice Packet
The Voice Packet Header
Other Voice Packet Sample Sizes
Voice Packet Analysis
Voice Packet Analysis: Other Voice Packet Sample Sizes
QoS Overview
Latency
Packet Loss
Jitter
Controlling Delay
Sources of Delay
The First Voice Packet
The Second Voice Packet
The Third Voice Packet
Jitter Buffer Under Perfect Conditions
An Adaptive Jitter Buffer

Module 2 SIP Trunking

The Legacy Circuit Switch
VoIP Phases
VoIP Phase 1: LAN Connect the Line Side
VoIP Phase 2: Decompose the Switch Cabinet
VoIP Phase 3: Shrink the MGs and Add Survivability
VoIP Phase 4: Add SIP Trunking
VoIP Phase 5: Eliminate the Old MGs
VoIP Phase 6: Add EMUN
VoIP Phase 7: Mass Acceptance of SIP Trunking with ENUM?
SIP Trunking Costs
Other Means of Connection
The “Old PBX” can do SIP Trunking if the Vendor Offers the Software
SIP Trunking Protocols
Peer-to-Peer RTP
Hairpin RTP
Disadvantages and Advantages of SIP Trunking
Disadvantages
Advantages
ITSPs
SIP Trunking Examples
SIP Trunk Outbound Call
Public VoIP

Module 3 VoIP in the LAN

IP and Ethernet
A Sample Ethernet Switched Network
MAC Addresses
IP MAC Address Learning
known Destination MAC Addresses
Flood the Broadcast
Response to Flooded Packet
Learning Port Information
Switching
MAC Table Aging
Ethernet Communications Limits
Virtual LANs
VLAN Trunk
VLAN Tags
Untagged Frames
Port-Based VLANs
Broadcast Frame in VLAN 10
VLAN Trunking for VoIP Phones
IEEE 802.3af Device Detection
IEEE 802.3af Power Classifications
QoS at Layer 2
VLAN Tagging Process
IEEE 802.1q Frame Tagging

Module 4 IP Networking

One-Way vs. Both-Way Routing
Static Routing
Subnet Masks and Routing
Routing and Switching
Routing Protocols
Distance Vector Routing
Link-State Routing
TCP/IP Review
Transmission Control Protocol (TCP) vs. User Datagram Protocol (UDP)
Connection-Oriented Protocol (TCP)
TCP/IP Packet Format and Operation
Connectionless Protocols (UDP)
UDP Packet
DNS
Basic Method of DNS
Dial Plan Essentials
Dial Plan Example
Digit Map
Enbloc vs. Overlap
Common Modifications to REGEX
Symbols
Regular Expressions
Metacharacters
Matching
Normalization Examples

Module 4 SIP-Related IP Services

DHCP Option for SIP
DHCP Discover
DHCP Offer
Root-Level Domain Registration
Basic Method of DNS
Why Start with ENUM?
ENUM: NAPTR Query
ENUM: NAPTR Response
Locating SIP Servers: An Example
NAPTR Response
SRV Query
SRV Response
A Record Query
Regular Expressions
The Metacharacters

Module 5 Voice Compression

Voice Compression Hardware
ASICs
DSPs
Mean Opinion Scores
Codecs
G.711, G.723.1, G.726
G.728 and G.729
Voice Compression
Formants
The Predictor
PCM Sampling
Voice Compression Algorithms
ADPCM Compression
Vocoder
G.729 Example
Codec Comparison Exercise
Zero Packet Loss
Ten Percent Packet Loss
Twenty Percent Packet Loss
T.38 Fax Spoofing
Call Setup
Discovering the Fax Tone
T.30 Negotiation
Shifting to 9.6 Kbps
T.38 Phase

Module 6 Real-Time Transport Protocol (RTP)

RTP Architecture
RTP and RTP Control Protocol
Encapsulating the Voice Packet
RTP Ports
RTP Profile
Payload Types
Mapping Payload Type to Codec Type
How H.323 Identifies the Payload Type
NTP vs. RTP Timestamp
RTP Timestamps
RTP Timestamps and Silence Suppression
RTP Timestamps and Jitter Calculation
Controlling Jitter
Jitter Buffer Delay
Mixers
Synchronization Source
Conference Bridge Adds CSRC
RTP Header
UDP Packet with RTP Header and Voice
Required Fields
Version
Padding Bit
Extension Bit
CSRC
Market Bit
Payload Type
Sequence Number
Timestamp
SSRC
The Format-Specific Parameter (fmtp) Attribute
RFC 2833 Example: A Dialing Event
Transmitter Processing
Receiver Processing
Controlling Serialization Delay
Perfect Candidate for LFI and RTP Header Compression
RTP Header Compression Process (RFC 2508)
RTP Header Compression Format
RTCP
RTCP QoS: Round-Trip Delay Calculation
Sender Reports
Receiver Reports
Source Descriptions
Source Description Items
Other RTCP Packets

Module 7 SIP Architecture

SIP User Agents
SIP Requests (Methods)
SIP Response Codes
SIP Proxy
SIP Back-to-Back UA
Session Border Controller
Forking Proxy
SIP Redirect Proxy
Global SIP Architecture
Overview of Operation
Classic SIP Trapezoid
INVITE Request
Session Description Protocol
Proxy Function
180 Response
200 Final Response
BYE
INVITE and ACK
SIP Functional Stack
SIP Core Documents and Extensions

Module 8 SIP Call Flow Examples

SIP Call Analysis
SIP Registration with Authentication
SIP Call without INVITE Authentication
The 100rel Process
Busy Number
Abandoned Call (Cancel)
SIP Redirect (Call Forward)
Call Transfer
E&M Tie Trunk
See a Problem
Solution: SIP 183 Response

Module 9 Session Description Protocol

Session Description Protocol
v= Header
o= Header
s= Header
c= Header
t= Header
m= Header
a= Header
Offer/Answer Model
Offer/Answer: Example 1
Offer/Answer: Example 2
SDP Offer/Answer Rules
UPDATE Method
RTP SEND and RECV Defined
Media Direction and RTCP
How RTCP Works
Placing a Call on HOLD

Module 10 SIP NAT Traversal

SIP NAT Traversal
One-Way Voice Results
Full Cone NAT
IP Address Restricted NAT
Port Restricted NAT
Symmetric NAT
Simple Traversal of UDP through NATs
Traversal Using Relay NAT
NAT with Embedded SIP Proxy
Public VoIP Example

Module 11 Media Gateway Control Protocol (MGCP)

Protocol Comparison
MGCP Call Model
Hairpin Call Example
Defined Endpoints
MGCP Commands
MGCP Syntax Example
Return Codes
Return Code Table
Parameter Lines
DTMF Package
Line Package
Digit Maps
MGCP Trace Procedure
MGCP Trace (Steps 1-8)
MGCP Trace (Steps 9-14)
MGCP Trace (Steps 15-22)
MGCP Trace (Steps 23-28)
MGCP Established Call
MGCP Trace (Steps 29-36)
MGCP Trace (Steps 37-40)

Module 12 Queuing

CoS vs. QoS
Leaky Bucket
First In, First Out
Type Classification
Session ID Classification (Fair Queuing)
Dequeuing
16. QoS-Related Protocol
Sources of Delay
Packetization Delay
Algorithmic Delay (Look Ahead)
Coder Processing Delay (Think Time)
Queuing Delay
Serialization Delay
Low-Speed Link
How 56-Kbps Links Cause Jitter
Upgrade to T1/E1 and Prioritize Voice
QoS Technology Solutions: Differentiated Services (DiffServ)
Supporting a VoIP Call with DiffServ
ToS Field
DiffServ Process at the Edge Router
DiffServ Process in the Core
DiffServ Highlights
Traffic Engineering: An Art Form
Measuring Engineering
Grade of Service
Appendix A: Glossary
Appendix B: H.323

Voice Over IP Foundations Virtual Classroom Live March 17, 2025

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