Voice over IP Foundations — A 5 Day Course

Synopsis

The aim of this course is for delegates to gain essential data networking and Voice over IP (VoIP) knowledge in a single, week-long class. In this course, you will learn how VoIP works, why VoIP works, and how to use VoIP.

Course Objectives

On completion, delegates will understand:

Core concepts of how Internet Protocol (IP) carries a VoIP packet
Configure DHCP and DNS to support IP telephony
Real-Time Transport Protocol (RTP)
Session Initiation Protocol (SIP) - Call set up, Instant Messaging, Presence
Session Description Protocol (SDP)
The H.323 protocol suite, including H.225, RAS, and H.245
The role of endpoints, gatekeepers, gateways, and MCU in an H.323 network
SIP proxy, Session Border Controller (SBC), and SIP softswitch
Media Gateway Control Protocol (MGCP) analysis
MGCP architecture
A technical comparison of H.323, SIP, and MGCP
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 to use Wireshark to decode and troubleshoot RTP, SIP, MGCP, and H.323 call flows
Configure the trixbox Softswitch and SIP proxy
Configure SIP gateways and softphones
Security issues to consider when setting up VoIP

Intended Audience

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.

Pre-Requisites

The skills and knowledge required for a delegate to sit this course are as follows

TCP/IP Networking
Telecommunications Fundamentals

Publicly scheduled dates, locations, and prices

A schedule of dates for this course is not currently available. Please call 0800 651 0338 or use our contact form to enquire about places and availability.

Packetizing Voice

Key architectural VoIP components
End-to-end voice transmission
Packetizing voice (encapsulation)
Transmission time allocation
QoS and capacity considerations
Sources of delay
Coder processing delay (Think Time)
Algorithmic delay (Look Ahead)
Packetization delay
Serialization delay
Queuing delay
Jitter buffer function
VoIP QoS requirements: Packet Loss, Latency, Jitter

VoIP in the LAN

MAC address
IP address and ARP
Ethernet switching
Logical and physical segmentation
VLAN - 802.1Q/P
802.3af - Power over Ethernet (POE)

IP Networking

IP addressing
Static routing
OSPF
EIGRP

TCP/IP Review

Transmission Control Protocol (TCP)
VoIP protocols that use TCP
User Datagram Protocol (UDP)
VoIP protocols that use UDP

SIP-Related IP Services

DNS
How SIP uses DNS
DHCP
How IP telephony uses DHCP

Voice Compression

G.711 u-law and a-law
G.729
G.723.1

Real-Time Transport Protocol (RTP)

Dealing with packet Loss, latency, jitter
How various protocols define the RTP session
Session Description Protocol
H.245 terminal capabilities
The RTP profile
The RTP payload type field
RTP telephony events (RFC 2833)
How RTP removes jitter
How RTP handles packet loss
How RTP identifies the talking party
How RTP handles silence suppression
How RTP is used to mix voice (conference calls)
The RTP header
RTP Control Protocol (RTCP)
SDES
Sender/receiver reports
Bye reports

SIP Architecture

SIP architecture
Proxy: stateful, stateless, call stateful, Session Border Controller
SIP methods: INVITE, ACK, BYE, CANCEL, REGISTER, INFO, PRACK, etc.
SIP response codes: 1xx, 2xx, 3xx, 4xx, 5xx, 6xx
SIP headers (To:, From:, Call-ID:, Allows: Required, Via)
Session Description Protocol (SDP)
SIP Addressing, Session Control, and Call Setup

SIP Uniform Resource Indicators (URIs)

Understand the format of SIP URIs and how URIs interoperate with
PSTN dialing plans, e-mail systems, and web pages
Generic URI information (RFC 2396)
Direct or Proxy
PSTN number (RFC 2808)
Instant messaging
Presence
In registrations
Content-Type Header

SIP Call Flow Examples

Review how SIP calls are set up for applications like PSTN, instant messaging,
VoIP, and more in this technical, in-depth analysis of the protocol.
Call attempt - unsuccessful
Presence subscription
Registration
Presence notification
Instant Message Exchange
Call setup - successful
Cancel
Vacant number
100rel
www authenticate

SIP Syntax

Request Message
Response Message
The Start Line
Via Header
SIP Dialog
From Header
To Header
Call-ID Header
Dialog State
CSeq Header
Max-Forwards Header
Proxy-Authenticate Header
Contact Header
Expires Header
User-Agent Header
Content-Length Header
Allow Header
Supported Header

Session Description Protocol

v= Header
o= Header
s= Header
C= Header
t= Header
m= Header
a= Header
Offer/Answer Model
SDP Offer/Answer Rules
UPDATE Method
RTP SEND and RECV Defined
Media Direction and RTCP
How RTCP Works
Placing a Call on HOLD

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

Media Gateway Control Protocol (MGCP)

Architecture
Verbs: CRCX, MDCX
Responses
Packages (DTMF, Line, Trunk, Generic, etc.)
Parameter lines
Sample call flow protocol analysis

H.323

ASN.1 primer
H.323 architecture
Gatekeeper
Gateway
MCU
Terminal
H.323 versions
H.323 gatekeeper-controlled call flow example

Queuing

Priority queuing
Weighted fair queuing
Weighted precedence
Traffic policing and traffic shaping
Low latency queuing
The effects of data traffic and fair queuing on VoIP
Mixing voice and data traffic effectively
Determining bandwidth needs for voice traffic
Assessing the impact of voice on data networks
Low speed links
High speed links

QoS-Related Networking Protocols

Differentiated Services (DiffServ)
Call Admission Control