Introducing Voice Gateways Part 2

Traditional Voice
Versus Unified Voice
Lecture 1 & 2: Traditional Voice Versus Unified Voice
Josh Lowe Winter 2016
Topics
Where It All Began: Analog Connections
The Evolution: Digital Connections
Understanding the PSTN
The New Yet Not-So-New Frontier: VoIP
Replica of Edisons Phonograph


The point of that introduction is to show you an
example of an analog signal
In todays world analog signals are captured
electronically rather than using tinfoil, but the principle
is the same
The volume and pitch of your voice results in different
variations of electrical current, voltage, frequency,
charge, etc.
Example voice waveform using electronic signals


Analog phone lines use electrical properties (frequency,
amplitude, etc.) to convey the changes in your voice
over cabling
Other signals than just voice are conveyed across a
phone line:
Dial tone
Dialed digits
Busy signals
Etc.
More on these in later lectures

Each analog circuit is composed of a pair of wires, one
ground (or positive, or tip) and one battery (or
negative, or ring)
These wires power the analog phone and allow it to
function

The jagged line over the analog phone below
represents a broken circuit
When the phone is on hook, the phone separates the
two wires preventing the electric signal from flowing
through the phone.

When the phone is lifted off hook the wires are
connected causing an electrical signal to flow from the
phone company central office (CO) into the phone
This is known as loop start signaling

Loop start signaling is the typical signaling type used in
a home environment.
Loop start is susceptible to a problem known as glare
Glare occurs when you pick up the phone to make a
call at exactly the same time as a call comes in to the
phone before the phone has a chance to ring
Uhhh...Oh! Hello, Bob! Im sorry, I didnt know you
were on the phone!
Usually not a problem in home environments because
the chances of it happening are slim, and if it does
happen the call is usually meant for your house anyway

In business environments glare can be a significant
problem because of the large number of employees
and high call volume
For example, a corporation may have a key system (a
type of internal phone system) with five analog links to
the telephone company
10

If a call comes in for extension 5002 at the same time
as extension 5000 picks up the phone the key system
will connect the two signals causing x5000 to receive
the call for x5002
This happens because the loop start signal from x5000
seizes the outgoing telephone line at the same time as
the incoming call is received on the same line.
11

Because of glare, most modern internal telephone
systems designed for larger corporate environments
use ground start signaling
Ground start signaling was originally used in pay
phones systems
When the handset was lifted off of the phone there was
no dial tone until a coin is dropped in
The coin brushes up against the ring and tip wires,
briefly grounding them
This signaled the phone company to send dial tone on
the line
12

Analog signals pose problems
Analog signals fade over long distances and have to be
regenerated by repeaters
Repeaters cant distinguish between the audio and
noise on the line, so each time the voice is amplified by
the repeaters, so is the noise
The farther the signal travelled the more it has to be
repeated and the more distorted the original signal
becomes
13

The second problem with analog is the number of wires
needed to support a large area or business
Each phone requires two wires and the bundles of wire
become massive and difficult to maintain
A solution that allows multiple calls to be sent over a
single wire is needed.
The solution should also allow the calls to be sent for
longer distances without being badly distorted
Digital connections are that solution
14
Moving from Analog to Digital

Digital signals use numbers to represent levels of
voice instead of a combination of electrical signals
(waveform)
Digitizing voice is the process of changing analog
signals (waves) into a series of numbers that can be
used to reconstruct the original signal at the other end
of the line
15

Each number sent represents a sound that someone
made while speaking
Todays network technology can send numeric signals
(in binary format) for vast distances without any
degradation or loss of data
Even with a significant amount of noise on the line, its
still relatively easy to tell the difference between a 1
and 0 in an electrical signal
This solves the problem of sending the signal long
distances, but what about the huge number of cables
required to send the voice?
16

Digital signaling can also solve the issue of requiring a
large number of cables
Digital voice uses time-division multiplexing (TDM)
TDM allows voice networks to carry multiple
conversations at the same time over a single path
(multiplexing)
The digitized voice is transmitted in specific time slots
(thus, the time-division) that differentiate the separate
conversations
17

Based on the time each voice data was sent the
telephone carrier is able to distinguish and reassemble
the voice conversations
18

Digital connections to the telephone network in
Canada, the US, and Japan are T1 circuits
A T1 circuit is built from 24 separate 64-kbps channels
known as digital signal 0 (DS0)
Each of these channels can support a single voice call,
so theoretically a T1 can support up to 24 calls at a
time
Outside of Canada, the US, and Japan corporations
use E1 circuits which allows you to use up to 30 DS0s
for voice calls
19

Using digital technology solves the problems mentioned
earlier, but introduces a new one: signaling
Analog circuits use specific variations of the frequency
of the electrical waveform to pass supervisory signaling
Digital signals dont have this ability, they only send 1s
and 0s
To solve this, two primary styles of signaling were
created for digital circuits:
Channel associated signaling (CAS)
Common channel signaling (CCS)
20
Channel Associated Signaling (CAS)

CAS steals bits that would typically have been used to
communicate voice information, and uses them for
signaling
Because T1 CAS steals bits from the voice channel, its
often called robbed-bit signaling (RBS).
Since some of the voice information is no longer being
sent the voice quality does technically drop a little
The number of bits stolen is small enough that the
change in voice quality is not noticeable
21
Channel Associated Signaling (CAS)

For a T1 the eighth bit on every sixth sample in each
DS0 is stolen for signaling
Each column is a DS0

(unique voice call)
22
Common Channel Signaling (CCS)

CCS dedicates one of the DS0 channels from a T1 or
E1 link for transmitting signaling information
Referred to as out-of-band signaling because the
signaling traffic is sent separately from the voice traffic
T1 connections using CCS only have 23 usable DS0s
for voice (as opposed to all 24 with CAS)
A signaling protocol used on the common channel
sends the necessary information for all voice channels
The most popular signaling protocol for CCS is Q.931
which is used for ISDN circuits
23
Common Channel Signaling (CCS)

CCS is the most popular connection used between
voice systems in the world
It offers more flexibility with signaling messages, more
bandwidth for the voice bearer channels, and higher
security (because of the out-of-band signaling)
Also allows hardware vendors to communicate
proprietary messages and features between their
systems using ISDN signaling (CAS does not)
24
Understanding the PSTN

The Public Switched Telephony Network (PSTN) is
to voice networks what the Internet is to data networks
If you have ever made a call from a home telephone,
you have connected to the PSTN
Its primary purpose is to establish worldwide telephony
connectivity
25
Pieces of the PSTN

When the phone system was originally created
individual phones were wired directly together
If you wanted to connect with more than one person
you needed multiple phones
This was obviously not scalable
26
Pieces of the PSTN

The modern PSTN is now a worldwide network built
from the following pieces:
Analog telephone: Able to connect directly to the PSTN
and is the most common device on the PSTN. Converts
audio into electrical signals
27
Pieces of the PSTN

Local loop: The link between the customer premises (such
as a home or business) and the telecommunications service
provider.
28
Pieces of the PSTN

CO switch: Provides services to the devices on the local
loop. These services include signaling, digit collection, call
routing, setup, and teardown.
29
Central Office (CO)
http://www.telcodata.us/search-area-code-exchange-detail
30
Central Office (CO)
Photograph taken by Andrew Filer (afiler.com). Used under creativeJosh

commons
license
Lowe Winter 2016
31
Pieces of the PSTN

Trunk: Provides a connection between switches. These
switches could be CO or private.
32
Pieces of the PSTN

Private switch: Allows a business to operate a miniature
PSTN inside its company. This provides efficiency and cost
savings because each phone in the company does not
require a direct connection to the CO switch.
33
Pieces of the PSTN

Digital telephone: Typically connects to a PBX system.
Converts audio into binary 1s and 0s, which allows more
efficient communication than analog.
34
Understanding PBX and Key Systems

Many businesses have hundreds or even thousands of
phones they support in the organization
If the company purchases a direct PSTN connection for
each one of these phones, the cost would be
astronomical
Instead, most organizations choose to use a PBX or
key system internally to manage in-house phones
These systems allow internal users to make phone
calls inside the office without using any PSTN
resources
Calls to the PSTN forward out the companys PSTN
trunk link
35
36

When you first look at a PBX system, it looks like a
large box full of cards (which it essentially is!). Each
card has a specific function:
Line cards: Provide the connection between telephone
handsets and the PBX system.
Trunk cards: Provide connections from the PBX system to

the PSTN or other PBX systems.
Control complex: Provides the intelligence behind the PBX
system; all call setup, routing, and management functions
are contained in the control complex.
37

PBX systems, while technically a single point of failure,
typically offer 99.999% uptime (five 9s) with a lifespan
of 7 to 10 years!
PBX systems are well known to be reliable pieces of
equipment and this is often one of the reasons that
companies are sometimes hesitant to move to a VoIP
solution
http://cityinfrastructure.com/Voice/Voice.html
38

Key systems are similar to PBXs but are geared
towards small business environments (typically less
than 50 users)
As technology has advanced, the line between key
systems and PBXs has begun to blur
Key systems usually have fewer features than a PBX
and typically make use of shared lines
For example, if an small office using a key system had
four telephone lines, each phone would show the same
four lines
If someone picks up a call on line one, it appears as in
use on every phone on the system
39
Nortel CICS
40
Connections to and Between the PSTN

Home users and small offices typically connect using
one or more analog connections.
Medium-to-large size businesses often use one or more
digital connections (T1 or E1) to connect to the PSTN
41
Connections to and Between the PSTN

In order for all of the telephony service providers of the
world to communicate a common signaling protocol
must be used similar to TCP/IP for the Internet
The protocol generally used is called Signaling
System 7 (SS7)
SS7 is an out-of-band signaling method used to
communicate call setup, routing, billing, and
informational signaling through the PSTN to the
destination.
This is primarily a telephony service provider
technology that you typically do not use from a
customer standpoint
42
PSTN Numbering Plans

Just like data networks use IP addressing to organize
and locate resources, voice networks use a numbering
plan to organize and locate telephones all around the
world
Organizations managing their own internal telephony
systems can develop any internal number scheme that
best fits the company needs (similar to private IP
addressing)
When connecting to the PSTN however, you must use

a valid, E.164 standard address for your telephone
system (similar to a public IP address)
43

E.164 is an international numbering plan created by the
International Telecommunication Union (ITU)
Each number contains the following components:
Country code
National destination code
Subscriber number
44

For example, the North American Numbering Plan
(NANP) uses E.164 to break numbers down into the
following components:
Country code
Area code
Central office or exchange code

Station code
45

For example, the NANP number 1-602-555-1212
breaks down like this:
NANP
46
The New Yet Not-So-New Frontier: VoIP

What is VoIP really?
With digitized voice, weve got half the puzzle. The
voice has been translated from an analog signal into a
series of 1s and 0s
VoIP is taking the 1s and 0s of digitized voice and
placing them into a data packet with IP addressing
information in the headers.
You can then take that VoIP packet and send it across
the data network instead of a traditional telephony
network
So whats so great about sending voice over a data
network instead of a telephony network?
47
VoIP: Why It Is a Big Deal for Businesses

The business benefits of VoIP include the following:
Reduced cost of communicating: Instead of relying on
expensive tie lines (private trunks) or toll charges to
communicate between offices, 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
instead of both voice and data cables.
Seamless voice networks: The voice traffic is crossing your
network (relatively speaking) rather than exiting to the PSTN.
This provides centralized control of all voice devices attached to
the network and a consistent dial-plan.
48

Take your phone with you: With VoIP phone systems, the
cost of making changes is virtually eliminated. In addition, little
to no reconfiguration of the voice network is needed to add new
phones. When combined with a VPN configuration, users can
take IP phones home with them and retain their work extension
IP SoftPhones: Users can now plug a headset into their laptop

or desktop and allow it to act as their phone. SoftPhones are
becoming increasingly more integrated with other applications
such as e-mail contact lists, instant messenger, and video
telephony
Unified e-mail, voicemail, fax: All messaging can be sent to a
users e-mail inbox. This allows users to get all messages in
one place and easily reply, forward, or archive messages.
49

Increased productivity: VoIP extensions can forward to ring
multiple devices before forwarding to voicemail. This eliminates
the phone tag game.
Feature-rich communications: Because voice, data, and
video networks have combined, users can initiate phone calls
that communicate with or invoke other applications from the
voice or data network to add additional benefits to a VoIP call.
Open, compatible standards: You can now connect devices
from different telephony vendors together. This will allow
businesses to choose the best equipment for their network,
regardless of the manufacturer.
50
The Process of Converting Voice to

Packets
Harry Nyquist created a process to convert analog
signals into digital format (called the Nyquist Theorem)
Nyquist found that he could accurately reconstruct
audio streams by taking samples that numbered twice
the highest audio frequency used in the audio
So if the highest frequency was 2000Hz you would
need to take 4000 samples in the same period in order
to accurately reconstruct the original audio
51

Packets
Audio frequencies vary based on volume, pitch, and so
on:
The average human ear is able to hear frequencies from 20
20,000 Hz
Human speech uses frequencies from 200 9,000 Hz
Telephone channels typically transmit frequencies from 300

3,400 Hz
The Nyquist theorem is able to reproduce frequencies from 300
4,000 Hz
52

Packets
If human speech uses frequencies between 2009,000
Hz and the normal telephone channel only transmits
frequencies from 3003,400 Hz, how can you
understand human conversation over the phone?
Studies have found that the telephone channel
frequency gives you enough sound quality to identify
the caller and sense their mood
The telephone channel frequency range does not send
the full spectrum of human voice inflection and lowers
the actual quality of the audio
53

Packets
According to the Nyquist Theorem you should sample
at twice the highest frequency, so to sample
frequencies up to 4000 Hz you would need to sample
8000 times every second
A sample is a numeric value that consumes a single
byte of information and describes the audio signal at
the time the sample was taken
During the process of sampling,
the sampling device puts an
analog waveform against a Y-axis
lined with numeric values.
54

Packets
The process of converting the analog wave into digital,
numeric values is known as quantization
Because 1 byte of information can represent only
values 0255, the quantization of the voice scale is
limited to values measuring a maximum peak of +127
and a maximum low of 127
55

Packets
Notice in the diagram that the 127 positive and negative
values are not evenly spaced
To achieve a more accurate numeric value the
amplitude more common to voice are tightly packed
with numeric values, whereas the fringe amplitudes
on the high and low end of the spectrum are more
spaced apart
56

Packets
The sampling device breaks the 8 binary bits in each
byte into two components: a positive/negative indicator
and the numeric representation
The first bit indicates positive or negative, and the
remaining seven bits represent the actual numeric
value
Because the first bit in the diagram is a 1, you read the
number as positive. The remaining seven bits represent
the number 52
This is the digital value
used for one voice
sample.
57

Packets
8000 of these 1-byte samples are taken every second
8000 samples per second times 8 bits in each sample
gives you 64,000 bits per second
Its no coincidence that uncompressed audio (including
the G.711 audio codec) consumes 64 kbps. Its also no
coincidence that each channel of a T1 or E1 can carry
64 kbps of data
Once the sampling device assigns numeric values to all
these analog signals, a router can place them into a
packet and send them across a network
58

Packets
Codec = Coder/Decoder; Codecs are the formulas
used to convert analog to digital and vice versa (the
sampling, quantization, etc.)
Important note! There are two forms of the G.711
codec mentioned previously: -law (used primary in the
United States, Canada, and Japan) and a-law (used
everywhere else)
The previous description represented G.711 a-law.
G.711 -law codes in exactly the opposite way (all of
the 1s would be 0s and all of the 0s would be 1s)
When a country using -law is communicating with a
country using a-law it is the responsibility of the -law
side to do the conversion to a-law
59

Packets
The last and optional step in the digitization process is
compression
Advanced codecs, such as G.729, allow you to
compress the number of samples sent and thus use
less bandwidth
This is possible because sampling human voice 8,000
times a second produces many samples that are similar
or identical
For example, say the word cow out loud to yourself,
which takes about a second to say
There are really only three sounds, the k, the
ahhhhh, and the wuh
60

Packets
Since most of these 8000 samples sound exactly the
same (or pretty close) theres no need to send them all!
Compressed codecs typically send a sound sample
once and simply tell the remote device to continue
playing that sound for a certain amount of time
This is often described as building a codebook of the
human voice
Using this process the G.729 codec is able to reduce
the bandwidth from 64 kbps down to just 8 kbps for
each call!
61

Packets
Unfortunately this reduction of bandwidth comes with a
price: quality
Quality was originally measured using a system known
as a Mean Opinion Score (MOS) to rate the quality of
various codecs.
A listener listens to a caller say Nowadays, a chicken
leg is a rare dish and rates the clarity of the sentence
on a scale of 1-5, where 1 is the worst and 5 is the best
62

Packets
63

Packets
You can use quite a few different audio codecs on your
network, each geared for different purposes and
environments
Some codecs sacrifice audio quality to achieve very
streamlined transmissions. Other codecs are designed
to meet the need for quality
Cisco IP phones have the ability to code using either
G.711 or G.729 by default.
G.711 is a baseline codec that pretty much every
vendor should support on all of their devices so that
phones from competing vendors can still communicate
64
Role of Digital Signal Processors

Cisco routers are designed to route data packets from
one location to another
They are not equipped with the kind of resources
needed to convert loads of voice into digitized,
packetized transmissions
This is where Digital Signal Processors (DSPs) come
into play. DSPs offload the processing responsibility for
voice-related tasks from the processor of the router
(similar to how a video card with a GPU offloads
graphics processing from your CPU)
65

A DSP chip performs all the sampling, encoding, and
compression functions on audio coming into the router
If you put voice interface cards into your router without
installing DSPs the cards would be useless. None of
the audio would be converted to digital packetized data
DSPs look a lot like memory chips you would put in
your computer
66

Make sure you add the number of DSPs to your router
to support the number of simultaneous voice call,
conference, and transcoding (converting from one
codec to another) sessions you plan to support
These chips come bundled into Packet Voice DSP
Modules (PVDM)
67

The PVDMs come in the following sizes:
PVDM2-8: Provides 0.5 DSP chip
PVDM2-16: Provides 1 DSP chip
PVDM2-32: Provides 2 DSP chips
68

Some codecs consume more DSP resources than
others
Generally speaking, the DSP resources are able to
handle roughly double the number of mediumcomplexity calls than high-complexity
69
Understanding RTP and RTCP

Real-time Transport Protocol (RTP) and Real-time
Transport Control Protocol (RTCP) are the protocols
of voice
RTP operates at the transport layer of the OSI model
on top of UDP, and RTP packets carry the actual audio
stream
Both RTP and UDP (which are both transport layer
protocols) are necessary and each serve different
functions
UDP provides port numbers to allow multiple
simultaneous sessions and header checksums to make
sure the header does not become corrupted
70

RTP adds time stamps and sequence numbers to allow
the remote device to put the packets back in order and
to use a buffer to remove jitter (variances in delay)
RTP can also be used for video purposes, hence the
Payload Type field in the RTP header
71

Once two devices attempt to establish an audio
session, RTP engages and chooses a random, even
UDP port number from 16,384 to 32,767 for each RTP
stream
RTP streams are one way so two RTP streams must be
established, one for each direction
At the time the devices establish the call, RTCP also
engages
72

RTCPs primary job is statistics reporting
These statistics include
Packet count
Packet delay
Packet loss
Jitter (delay variations)
Although this information is useful, it is not nearly as

critical as the actual RTP audio streams
Keep this in mind when you configure QoS settings
RTCP creates a separate UDP session using an oddnumbered port (one number higher than the RTP port)
73
74

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Traditional Voice

Versus Unified Voice

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Where It All Began: Analog Connections

Replica of Edisons Phonograph

Josh Lowe Winter 2016

Where It All Began: Analog Connections

Example voice waveform using electronic signals

Josh Lowe Winter 2016

Where It All Began: Analog Connections

More on these in later lectures

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Where It All Began: Analog Connections

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Where It All Began: Analog Connections

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Where It All Began: Analog Connections

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Where It All Began: Analog Connections

Josh Lowe Winter 2016

Where It All Began: Analog Connections

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Where It All Began: Analog Connections

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Where It All Began: Analog Connections

Josh Lowe Winter 2016

The Evolution: Digital Connections

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

The Evolution: Digital Connections

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Moving from Analog to Digital

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Moving from Analog to Digital

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Moving from Analog to Digital

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Moving from Analog to Digital

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Moving from Analog to Digital

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Moving from Analog to Digital

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Channel Associated Signaling (CAS)

Lecture 1 & 2: Traditional Voice Versus Unified Voice

Josh Lowe Winter 2016

Channel Associated Signaling (CAS)

Each column is a DS0