DAY 1

Introduction
Welcome to the world of CCNA Voice! As technology continues to evolve, the realm of
voice, which was traditionally kept completely separate from data, has now begun to
merge with the data network. This brings together two different worlds of people: data
Technicians and voice technicians . One of the primary goals of the new CCNA Voice
certification is to bridge these two worlds together.
This course help you to pass the
ICOMM 8.0 exam and make you much more knowledgeable .

ICONS USED IN THIS BOOK

640-461 ICOMM v8.0

Introducing Cisco Voice and Unified
Communications Administration
Exam Number:

640-461 ICOMM

Associated Certifications:

CCNA Voice

Duration:

90 minutes (60-70 questions)

Available Languages:

English and Japanese

 Where It All Began: Analog Connections

The Evolution: Digital Connections
Understanding the PSTN
The New Yet Not-So-New Frontier: VoIP

Where It All Began: Analog Connections

In 1877, Thomas Edison created a brilliant device known as a phonograph

SoundCollecting Horn

Cylinder Coated
with Tinfoil

Electrical signals to capture the properties of voice.

L L

ON-HOOK MODE

OFF-HOOK MODE

MAKING A CALL

How the analog phone works ?

The connections of the tip and ring wire to

your analog phone.

LOOP START SIGNALING

PROBLEM WITH LOOP START SIGNALING

Loop start signaling is susceptible to a problem known as GLARE .

GROUND START SIGNALING

Ground start signaling originated from its implementation in pay phone systems.

Types Of Signaling

The types of signaling in the analog world include :

1. Supervisory signaling (on hook, off hook, ringing)
2. Informational signaling (dial tone, busy, ringback and so on)
3. Address signaling (dual-tone multifrequency (DTMF) and Pulse).

The Evolution: Digital Connections

Analog signaling was a massive improvement over tin cans and string, but still posed
plenty of problems of their own.

Because Of Some Disadvantages Analog is Moving to Digital

1.Distance Limitation
2.Wiring Requirements
1.Distance Limitation
To increase the distance the analog signal could travel,the phone company had to
install repeaters to regenerate the signal as it became weak.

2.Wiring Requirements
The second difficulty encountered with analog connections was the sheer number of
wires the phone company had to run to support a large geographical area or a business
with a large number of phones.

Moving from Analog to Digital

ANALOG

DIGITAL

Time-division multiplexing (TDM)

Digital voice uses a technology known as time-division multiplexing (TDM).

TDM (contd.)
Digital voice connections to the PSTN as
1. T1 circuits in the United States, Canada, and Japan. A T1 circuit is built from 24
separate 64-kbps channels known as a digital signal 0 (DS0).
2. Each one of these channels is able to support a single voice call.
3. Areas outside the United States, Canada, and Japan use E1 circuits, which
allow you to use up to 30 DS0s for voice calls.

TWO BASIC METHODS FOR VOICE OVER IP

Signaling for digital circuits:

Channel associated signaling (CAS): Signaling information is transmitted using
the same bandwidth as the voice.(Robbed Bit Signaling)
Common channel signaling (CCS): Signaling information is transmitted using a
separate, dedicated signaling channel.(Out-of-Band Signaling)

Channel Associated Signaling

COMMON CHANNEL SIGNALING

CCS is the most popular connection used between voice systems worldwide
because
1. More flexibility with signaling messages
2. More bandwidth for the voice bearer channels,
3. Higher security

Tip: When using CCS configurations with T1 lines, the 24th time slot is always the
signaling channel. When using CCS configurations with E1 lines, the 17th time slot is
always the signaling channel.

Understanding the PSTN

The connection to one massive voice network, known as the PSTN.
Its primary purpose is to establish worldwide pathways to allow people to easily
connect, converse, and disconnect.

Pieces of the PSTN:

Analog telephone
Local loop
CO switch
Trunk
Private switch
Digital telephone

Understanding the PSTN

Understanding PBX and Key Systems

Private branch exchange system looks like a large box full of cards. Each card has a specific
function:
Line cards
Trunk cards
Control complex

Understanding PBX and Key Systems

Connections to and Between the PSTN

Each two-wire analog connection has the capability to support a single call.

Connections to and Between the PSTN

VoIP
The amazing process of converting spoken voice into packets.
The packet gets to its destination

1. In time (QoS)
2. Choosing the proper coding and decoding (codec) methods
3. Making sure that the VoIP packet doesnt fall into the wrong hands (encryption),

Benefits of VoIP for Business

Reduced cost of Communicating: VoIP allows you to forward calls over WAN connections.

Reduced cost of cabling: VoIP deployments typically cut cabling costs in half by running a
single Ethernet connection

Take your phone with you: Users can take IP phones home with them and retain their
work extension.
IP Softphones
Unified e-mail, voice mail and fax: All messaging can be sent to a users e-mail inbox.
Open, compatible standards: Connect devices from different telephony vendors
together.

The Process of Converting Voice to Packets

Nyquist found that he could accurately reconstruct audio streams by taking samples
that numbered twice the highest audio frequency used in the audio.
The Nyquist theorem is able to reproduce frequencies from 3004,000 Hz.

Nyquist Theorem

Digitizing Analog Signals

1.

Sample the analog signal regularly.

2.

Quantize the sample.

3.

Encode the value into a binary expression.

4.

Compress the samples to reduce bandwidth, optional step.

So, whats a sample?

A sample is a numeric value that consumes a single byte of information.
sampling 8,000 times (2 * 4000) every second.

Quantization

This process of converting the analog wave into digital, numeric values is known as quantization.

Encoding
Positive/negative

segments

intervals

8,000 samples a second times the 8 bits in each sample, and you get 64,000 bits per second .
G.711 audio codec consumes 64 kbps.
Note: There are two forms of the G.711 codec also called as PCM method: -law (used
primary in the United States and Japan) and a-law (used everywhere else)makes more sense .

Compression Bandwidth Requirements

Using this process, G.729 is able to reduce bandwidth down to 8 kbps for each call.
a measurement system known as a Mean Opinion Score (MOS) to rate the quality of the
various voice codecs.

Role of Digital Signal Processors

Cisco designed its routers with one primary purpose in mind: routing.
A DSP is a chip that performs all the sampling, encoding, and compression functions on audio
coming into your router.

DSP REQUIREMENT
Cisco bundles these DSP chips into packet voice DSP modules (PVDM), which resemble the
Old memory SIMMs.