Multiplexing for Telecommunications

TELX 2411
Course Notes

R 1,7 June 2018

2
History of Tl Facilities
The evolution of the telephone network, since the first transmission of intelligible
speech in 1876, has involved many technologies. The interface to the subscriber
handset has changed little. However, once out of the local loop, the telephone
exchange and facilities between exchanges have changed dramatically.

AT&T developed Tl technology for the PSTN in 1962. It was made commercially
available in 1977, and at a very high dollar rate. The term "Tl" refers to a specific
digital method of transmitting voice and data over 24 channels at 1,544,000 bps.
Prior to 1977, carrier systems (analog) were employed exclusively by telephone
companies and a few early, high-bandwidth users. The services were provided on
a Special need basis, designed and priced individually for each application.

In 1982, AT&T tariffed Tl (Tariff No. 270), renamed it "High Capacity Terrestrial
Digital Service" (HCTDS), and made it available to commercial customers. Tariff
No. 270 allowed users to lease Tl facilities as a standard offering, and established
prices that were significantly less expensive than the same number of individual
lines (24).

Deregulation was also taking place in the early 80's, and in 1983, the separated
Regional Bell Holding Companies (RHBC) began to tariff Tl services, making them
widely available. The other carriers soon followed with similar offerings.
In May of 1985, AT&T tariffed Fractional Tl (FTl) service. Fractional Tl provides
for the transportation of subsets (or fractions) of a 1,544,000 bps Tl data stream.
The service allows users to purchase only the number of channels required
without paying the cost of a full Tl span. Fractional Tl allows the user to buy
individual channels rather than Tl spans of 24-channels. This is ideal for the user
who wants Tl speed and capability but does not need 24 channels.
Years ago Tl transmission was used almost exclusively by telephone companies to
multiplex voice circuits between central offices. It is important to understand the
evolution from the early analog telephone systems to the widely available Tl
digital services of today, as the terminology and implementation of modern
networks are derived from some of the early systems.

3
ANALOG SYSTEMS

Prior to the implementation of Tl, the telephone network was built entirely with
analog components. The analog systems used Frequency Division Multiplexing
(FDM) to carry multiple voice conversations over the network between central
offices. The purpose of the multiplexer is to accept many low speed inputs and
combine them into a single higher speed link called an aggregate. The voice
frequency spectrum for individual telephone lines (300 to 3400 Hz= 300 to 3,400
cycles per second) was carried from the customer to the central office over a pair
of copper wires called the local loop. At the central office, the multiplexer shifted
the voice frequencies to one of 24 different channel frequencies and mixed all 24
conversations onto a single pair of wires. These groups of channels were further
miultiplexed into super-groups and mastergroups for transmission over
microwave facilities.

The objective was to make more efficient use of the available pairs of wires. The
resulting pair gain allowed the network to grow substantially without a build out
in wiring. As the technology advanced, digital techniques were developed for
multiplexing the signals that were less sensitive to noise and cross-talk.

DIGITAL SYSTEMS

In the early 1960's, Tl digital systems, called T-carrier, replaced the older analog
transmission facilities. The implementation of these digital systems took place
over a long period of time due to the huge installed base of equipment. The new
equipment, therefore, had to maintain some level of compatibility with the old. It
had to operate with existing local loops and telephones, and with the wiring
frames and other equipment at the Central Office.

Tl digital equipment is installed almost everywhere in the network today. The

analog voice signal from a telephone is carried to the central office over the local
loop and is digitized, at the central office by a coder-decoder, or CODEC.
The CODEC samples the analog voice signal 8,000 times per second and produces
an eight bit, digital representation of each sample called Pulse Code Modulation
(PCM). The result is a 64,000 bit per second stream of digital pulses called a DS0,
or a digital signal level zero. The existing world-wide standard for digital voice is
Pulse Code Modulation. It is referred to as "Voice Quality" because it has long

4
been used on interoffice tool trunks, which have the strictest quality
requirements. PCM digitized voice channels will also support 64kbps modem
operation.

The Tl standard for North America settled to 24 PCM DS0 voice circuits
multiplexed over a single circuit. The major characteristic of Tl was that it was all
digital, not analog in nature. Voice was digitized and transmitted at 1,544,000 bps
(24 channels at 64 Kbps per channel plus 8,000 framing bits). A standard 22-gauge
local loop circuit could be used to carry 1,544 Mbps data (over short distances)
instead of the standard 3.4 KHz {thousand cycles per second) voice band.

Tl "spans" utilize Time Division Multiplexing (TDM) to combine multiple voice,

data or video channels for transmission over a single Tl span as well as divide the
aggregate into 24 separate channels {channelized Tl) at the receiving end. Time
Division Multiplexers deal with digital signals only, thereby eliminating the
cumbersome and expensive filters found in analog systems. Time Division
Multiplexers are smaller, more reliable and less expensive, and also help
eliminate (well almost) cross-talk or amplifier noise in the multiplexer. The TDM
equipment used in a Tl span is referred to as a channel bank.

Newer advancements provide further multiplexing to increase the density of calls

much like the super-groups and master-groups in the analog system. For long haul
circuits, a multiplexing hierarchy is used for transmission over microwave or fiber
optic cables.

Tl LOCAL LOOPS
As circuit requirements increased, Tl was also implemented to consolidate the
local loops (pair gain introduced). Some channel banks are designed for use in
extreme weather and are installed in outdoor metal wiring cabinets, which serve
a local neighborhood. Business customers with multiple voice lines
and/or data requirements have installed channel banks on their premises.
Tl is a multi-channel facility that will support both voice and data transmission. Tl
is supported universally by both equipment vendors and common carriers. The
ISDN {Integrated Services Digital Network) primary rate service {PRI = 23 B-
Channels + 1 D-Channel) is based on a Tl transmission scheme. Therefore, it is the
only transmission scheme that can evolve into future ISDN standards.

5
Putting T-1 circuits into a network can:

• Save large amounts of money

• Give the user the flexibility to reconfigure connections quickly without
waiting for the service from the local phone company
• Improve voice quality compared to transmission over analog lines
• Improve reliability by simplifying the network and its maintenance
• Provide the bandwidth needed by growing businesses

In the simplest terms a T-1 service offers a two way connection at 1.544 million
bits per second. That is, one T-1 will carry the equivalent of 160 computer ports
running at 9600 bps each.

The most common use for T-1 circuits early on was phone companies using them
as trunks between the CO's. In a phone company environment, one T-1 circuit on
two normal pairs of copper wires carries 24 phone conversations in digital form.
The rate of 1.544 Mbits/s is a digital signal level one, DS-1. The T-1 refers to a
system of copper wire cables and regenerators that reinforce the digital signal at
intervals of about one mile.

T-1 circuits are still new to many end users. The first widely tariffed T-1 service
was offered in 1983. In the voice world, within the Telco's T-1 has been around
since the mid 1960's. The very first telephone service sold to the public connected
only a few hundred phones wired to one central office. Each subscriber had a pair
of wires that linked their telephone to the switchboard in the CO. The two wires
were twisted to reduce crosstalk and extend the range.

To establish a call, the operator manually inserted plugs in jacks to set up one
continuous loop of copper wire from the battery in the CO, to one subscriber,
back through the CO to the second subscriber and returning to the battery. That
portion of the circuit between each customer and the CO is still called the "local
loop". The path was solid copper all the way.

6
 Analog Voice

Copper Copper
co

Isolated central offices now needed to be interconnected. Interoffice trunks were

run. Again they were solid copper pairs. The call connections were still set up by
establishing a continuous copper loop between subscribers. Only now the loop
could be extended between central offices over the trunks. Each interoffice
connection required a separate pair.

Within a few decades, the number of local loops and interoffice trunks increased
rapidly. The technology then was aerial wire on glass knob insulators. Soon the
skies above the cities were filed with poles, cross arms, and countless wires. We
ran out of room for more wires.

Frequency division multiplexing, FDM, was invented. FDM's were in wide use up
to the 1980's for long haul circuits. It has now been replaced with digital facilities.
The main reason FDM gave way to digital signal processing is the susceptibility of
analog paths to noise. Additional features of digital transmission avoid the
difficulties of switching, controlling, and maintaining analog channels.

7
OVERVIEW OF A Tl SPAN
A typical T-carrier system consists of a transmission component, a user interface,
and termination equipment, as well as multiplexing schemes and signal
hierarchies.

Termination Equipment: Several types of terminal equipment other than the

basic switch provide digital connectivity. The equipment can be grouped into
three general categories: terminals (channel banks and transcoders), digital cross-
connect systems (DCSs), and multiplexers. Terminals take analog input and
transform it into a digital stream. Digital multiplexers and transcoders are the
interfaces between the different bit rates in the digital network. They "stack"
blocks of data on top of each other before they enter a high-capacity transmission
media. Digital cross-connects are the interconnection points for terminals,
multiplexers, and transmission facilities.

CHANNEL BANKS:
A channel bank 'performs the first stop of call handling. Channel banks are a
bridge between the analog and digital worlds, and have two basic functions. They
convert analog voice to digital code, and vice-versa, and they combine, or
multiplex, the resulting digital streams from several active sessions (voice or data)
onto a single stream. They are a combination of the digital terminals and the
digital multiplex function. A channel bank is a Time Division Multiplexer used in Tl
networks, which also includes the CODECS (Coders/Decoders) for each channel.
The channel bank accepts 24 analog voice connections, digitizes the signals, and
combines them into a 1,544,000 bps Tl aggregate called a DSl, or digital signal
level 1. The CODEC samples each analog voice signal 8,000 times per second and
produces an eight bit digital representation of each sample called Pulse Code
Modulation (PCM. The result is a 64,000 bps stream of digital signals called a DS0,
or digital signal level zero.

COOEC

8
A Time Division Multiplexer (TDM} divides the T-carrier into equal time slices, and
assigns one time slot to each channel. The TDM takes the information from each
channel, in sequence, and places it into a time slot on the T-carrier. Another TDM
at the other end of the link receives the aggregate stream and sorts it out into the
original channels.

The Tl multiplexer consists of fundamental elements:

• Channel Interface Units: These ports provide the physical level connection of
equipment, either voice or data, to the Tl multiplexer. Channel interface units are
available for synchronous data and asynchronous data using several types of
interfaces or connectors.
• Buffer: There is a bulk data memory in the Tl multiplexer. It provides data bit
storage which can be read and written between the channels and the aggregate
Time Division Multiplexed link.
• Frame Builder/Multiplexer: The frame builder, sometimes called the Time Slot
Interchanger (TSI}, multiplexes the information from the channel interface units
into an aggregate for transmission over the Tl link. It also de-multiplexes the
received aggregate into separate channels. The most common framing format is
ESF format.
• Tl Line Interface: The primary function of the Tl line interface is to convert the
aggregate stream, from the frame builder/multiplexer, to a format suitable for Tl
transmission. This includes converting the signals from unipolar to bipolar format,
controlling link transmit and receive functions, framing control, and ensuring
sufficient ones density.
• Other Elements: Other elements usually include a timing reference and a
controller module which sets up the configurations and routing tables on user
command. The controller module may be attached to a supervisory terminal
which can be located at the same site or at a remote site.
• Full Duplex: Tl links always operate in full duplex mode, where data traffic flows
in both directions, as shown in the above point-to-point multiplexer
configuration.

9
Let's take a look at the three most common digital network configurations. They
are:
• Light density
• Medium density
• Heavy density

10
 Light Density System ·

Central Office PCM CXR DSX1

Mak;,g
Switcil
1 Channel · (DS1}
Bank ...,._____...,
OTS
/ Regen ______
l,...
24

Ctmtr:al Office ,PCMCXR OSX1

Analog 1 . Channel lDS1J OTS
Switch Bank Regen

24

11
Light Density System

This is the lowest order of digital network configuration. It is referred to as an

analog to digital system. The main components are:

• Central Office
• PCM Carrier {Channel Bank}
• DSX-1
• Span Line
• Office Terminating Shelf {OTS)
• DS-1 {1.544 Mb/s}
• 24 channels

12
Medium Density System

Digital DSX1 (l)St} OMX (DSJ} DSX3

1 ~

Offices , LTS
...,,-=
·-= ~ 1;3
(Ds:3)
-
:f ~
r...., __., 2!! .....:::1' '-----

B: _J
·<
Optical
Cable •

Digital osx, , DMX OSX3 .---

Offices - ;

f ~
LTS
-
1:3 ·- -,

~ ~.

i=:-
..

r-,.. _....,, 2S __.., ~

13
Medium Density System

This network is the most common configuration in use throughout North America.
The main components are:

• Digital Office
• Digital Multiplexer (OMX)
• DSX-3
• Light Guide Transmission System (LTS)
• Digital Cross-connect Switch (DCS)
• DS-3 (44.736 Mb/s)
• 672 Channels

14
 Heavy Density System
(DS1) OMX (DS3J {0$3) (DS3A)

Digital
Office
1.
. DSX3 Radio
Switch
Digital
Radio

Er
- 7
.(DS1) OMX (DS3] (DS:JA)
PCM CH DSX.3 Radio Digital

I
I
L
Bank.

n Switch• Radio

.- ...,.,
(DS1J .(DSJ)

PCM CH
Bank

Er
[
- -· - - - - - .:._!

15
Heavy Density System

This network configuration is use were longer distances are involved, such as
between cities and across the country. The terrain will also dictate the use of this
network for example: mountains or water. The main components are:

• Radio switch
• Digital radio
• Digital echo canceller (DEC)
• DS-3A
• 1344 channels
• Redundancy

16
 Digital Multiplex Hierarchy
Analogue
Voic:e
Channels DS-1
1,544
Mbltls •1
••
•
24 •
DS-0
••
64 kbitts OS-1 , •
Digital 1 1.5-44
Mbitls
•
i------_.,..,
1,,-__.~·n .
Voice 12'---..._.l
DM.01 ......-----1....i DM-12
Channel • 1
: OM-23
, DS-4
24
• • . · • 274.176
4
......•..,..~---.., : OM-34 ,_.M_bi!lll_
. ......,...,..
DS.A 7
56 kbitts
High 1.
OS-1
1.5,,1.4
•
Speed Mbitl&

Data
DM-A1 DS =. Digital Signal.
OM = Digital MuffiplQQf
24 eg. DM-23 muffiplexs10S-,:2to 1 0&3 signal

17
 Digital Multiplex Hierarchy

The digital hierarchy was developed as a North American standard in a joint

recommendation from Canada and the United States to the CCITT.

The development of the hierarchy started with the introduction of the DS-1 digital
signal which permitted the transmission of 24 coded VF two-way channels,
interleaved in time, over two copper VF pairs. The other rates of the hierarchy
were developed as higher capacity line systems were projected.

The digital hierarchy transmission rates currently used in the North American
network are set out in the following pages. Also showing us how the various
Digital Multiplex equipment relates in the hierarchy. The multiplex equipment is
denoted by the channel and aggregate level. For example, a DM 23 multiplexes
seven DS-2 signals into a DS-3 signal. The transmission lines for the various rates
are designated Tl, T2, T3 and T4.

18
Digital Signal Standards

North American

Digital Signal Carrier Transmission Equivalent number -

(OS} Designation System Rate of Voice Channels

* OS-A None 56 k/bits 1 Data Channels

* DS-0 None o4 klbns 1 Digital Voice Channels
OS-1 T-1 1.544 M/bits 24 Voice Channels
DS"'.2 T-2 6.312 M/bits 96 Voice Channels
DS·3 T-3 44. 736 M/bits 672 Voice Channels
* DS-4 T-:4 274.176 M/bits 4032 Voice Channels
* * DS-4x7 rol~ 564.992 M/bits 8064Voice Channels
* proposed North American standard levels
'
* * Rockwell Optical

CEPT
CEPT Carrier · Transmission Equivalent number
Designation System Rate of Voice Channels

CEPT--0 None Q4 k/bits 1

CEPT-1 E-1 2s048 M/bitS'- 30
CEPT-2 E-2 8.448 M/bits 120
CEPT-3 E.;.3 34.368 M/bits 480
CEPT-4 E-4 139.264 M/bits 1920
CEPT-5 E-5 565.148 M/bits 7680

19
The North American digital system basic bit rates are shown on the previous page.
This system uses digital signals called DS and are currently used in North America,
Korea, Republic of China and Japan.

The rest of the world has adopted a system developed by the European
Conference of Postal and Telecommunications Administration, CEPT. The signals
are called CEPT signals and are also shown on the previous page.

Due to sampling and quantizing differences, slightly different digital rates can
result from different DS-1 input streams, and because each D51 is not
synchronized with the next, synchronized bit streams are achieved using bit
stuffing. The bit stream is raised to a common speed of 1.544Mbs.

DSX·l DSI-3
TZ T4
-----,Tl
Channel DfyflZ DM23
............ T3

Bank tvtux
Byte EHt facility
lnterleaved lnterl<Hxved
24,DS-0, or 4, DS-1 's Bit 7, DS·2's Bit Bit
OS-A Channels Stulfedto Stuffed antl i-------1 Interleaved'
@ 1.54.4 Mbfs 1.546 Mbls lvll.Jltiplexed to
Then mult1plexec1 44J36Mbfs 6, DS-3''$ Bit $1µ1fed
to a 6.132 Mb!s 08-3 and Multiplexed to
PS-2 a274.17BMbfs
DS-4
DSX Cross DSXCross
Connect Connect

The next stage of the hierarchy, DS-2, is developed by asynchronously bit

interleaving four, DS-1 signals into a 6.312 Mbs signal.

In order to extract information at the receive end, control bits must be included
so that the multiplexer knows the difference between stuff bits and information
bits.

20
A DS-3 is created by bit interleaving seven, asynchronous DS-2's and adding
framing, parity, alarm and justification bits, producing a 44.736 MB/S signal, as
with the DS-2 multiplexing, pulse stuffing is required for clock synchronization.

The latest DS-1 to DS-3 multiplexers provide DS-1 to DS-3 multiplexing in one
package. The DS-1 to DS-2, and DS-2 to DS-3 multiplexing steps take place
separately, as an internal function of the equipment. The multiplexer is divided
into a low speed and high speed stage.

21
Introduction to PCM

The evolution from electromechanical switches and analog carrier to more

efficient digital switches and carriers is one of the most important historical
transitions in the telephone industry.

In 1938, Alexa Reeves received a patent for pulse code modulation. Technical
improvements in time division multiplexing, TDM (PCM) has made it possible to
apply digital concepts to telephone transmission systems. The first TDM, PCM
system was installed in Akron Ohio, in 1962.

Millions of voice channels today are digitized and multiplexed in PCM voice
channel banks. The least expensive devices separate the three-step conversion
process for economy, eliminating multiple coders-decoders. When receiving, a
channel banks codec converts the T-1 bit stream to pulses interleaved on the
analog bus. Then through the TDM process individual ports pick off their own
pulses by watching a specific, repetitive timeslot. The pulse amplitude modulated
signal is then filtered at the port to re-create the original analog signal. A device
designed for voice and based on an analog PAM process must be modified to
handle digital signals. More recent channel banks digitized voice channels
individually or in pairs then multiplexed onto a digital bus. Current channel banks
are also much more sophisticated handling data. They offer many options, in the
form of hardware plugs or plug-in modules to handle specific forms of data.
Traditional channel banks offer more than 50 different plug-in modules, each
specialized for particular function. For example, the two-wire and four-wire E&M
interfaces would be on different modules. There are, however, many devoted
applications with limited need in private networks.

A knowledge of the basic concepts of analog and digital signals is necessary to

understand the PCM process that happens in our channel bank.

Consider an analog waveform representing voice. The amplitude is dependent on

the sender and can vary widely within equipment limitations. Frequency can also
depend on the limitations of the equipment. The analog signal is continuously
varying in an infinite number steps over a wide range of amplitude and frequency.

22
In comparison to the analog waveform, a digital signal is a series of pulses, all
having fixed amplitudes and fixed durations in time. Because of these
characteristics, the pulses of a digital bit stream can easily be detected and
regenerated.

The success of Alexander Graham Bell transmitting speech using an analog

electrical signal was so overwhelming that an immense telephone communication
industry evolved around analog transmission. Because of the outstanding success
of analog transmission techniques, such as FDM, many years passed before
serious attention was given to other methods of transmission speed signals.

Analog Transmission

Signal
Amplified
Plus
Signal
Noise
and Noise

23
In analog transmission, the signal is applied to a two wire facility. As the analog
signal is transmitted over distance, it is reduced in amplitude and must be
amplified at certain intervals. In addition to the loss of power, the analog signal
has accumulated transmission noise. Since amplifiers cannot distinguish between
the unwanted noise and the wanted signal, both are amplified. Amplifying both
the noise and the signal does not improve the quality of the analog signal.

As the distance increases, the transmission noises amplified along with the signal
at each amplification point. Eventually the noise power can outweigh the signal
power, making the voice intelligible. Analog transmission is always haunted by
noise and crosstalk.

24
Digital Transmission

Digital transmission is similar to analog in many ways. The signal is applied to

copper wires. Transmitting the digital signal over distance reduces the amplitude
and adds noise. At this point, however, the similarity between digital and analog
transmissions ends.

The digital signal, with its reduced amplitude and added noise, is applied to a
regenerator. This device regenerates and it pulse edits the output for each pulse
received. Unlike the amplifier that amplifies both the signal and noise, the
regenerator generates a new signal and prevents the noise distortion from being
transmitted through the regenerator.

This regeneration of the digital signal occurs at every repeater point. A clean
signal is then transmitted over the remainder of the transmission path.

Digital Transmission

Original Signal

Attenuated Signal

Regenerated Signal
(Voltage)

Regenerated Signal
(Pulse Width)

25
Analog Multiplexing

Up until about 1960, telephone transmission was accomplished almost entirely

within an analog carrier wave using the technique of frequency division
multiplexing, FDM.

Because of the success of the analog transmission techniques, such as frequency

division multiplexing, years passed before serious attention was given to other
methods of transmitting voice signals.

Frequency division multiplexing, FDM, is an analog carrier system which combines

the voice band signal ( 300-3400 Hz), with a number of carrier frequencies to
separate each conversation. This allows a single transmission medium, such as a
pair of copper wires, to carrier multiple voice signal simultaneously.

With FDM, each channel is translated to a different frequency. In this example,

channel 4 is translated from 0-4Khz to 8-12 Khz.

However, such analog carrier systems have always been haunted by noise and
crosstalk. This problem, as well as the rising complexity and cost of electrical
filters that founded FTM systems, created a search for more practical and efficient
transmission method.

One of the methods under investigation then was time division multiplexing,
TDM.

26
 Frequency·Division ·Multiplex·ing

28
Common
Transmission
24 - - Medium .------,

Frequency
Translation
16
Frequency
. (kHz) 12

Individual Individual
Voice Circuits Voice Circuits

27
TDM Technique

Output.

E E
.E

Output H
Value A
D H D

Tirrie - - - - - - - -

28
Time Division Multiplexing

In TDM systems, a number of circuits share a common transmission path but at

separate time intervals. Time division multiplexing in its simplest may be thought
of being like the action of the commutator shown in the previous page.

The arm of the commutator is used to sample the output of each devices or
signals. The samples are interleaved to form a stream for each sampled signal
occupies a certain time slot. To separate the signals, a similar commutator is used
at the receiving end. The receiving commutator must be accurately synchronized
with the transmitting commutator. This time slicing process is carried out
electronically by sampling gates at a very high speed.

It was demonstrated experimentally even before the development of FDM that

time division multiplexing could be used to transmit many speech messages
simultaneously over the same circuit. Such techniques could not be put into
practical use at the time due to the limitations of mechanical devices used for
high-speed switching during that era. The invention of the vacuum tube and the
electric wave filter made frequency division multiplexing much more attractive for
use in telephony transmission systems. However, researchers continued to
investigate time division methods.

29
Pulse Amplitude Modulation

n·
I I.
II
n 11 n
-,r-----~L _____ JL-----,r-~---~L
II . · u
I •1-4-125 µ/s _.j
11 . - I
u
- , r-s-Micro seconds

--,
t I
II I
11 /
u.

30
Pulse Amplitude Modulation PAM

Time division systems usually employ some type of pulse modulation in contrast
to the more familiar amplitude and frequency techniques used in FDM.

The most popular type of pulse code modulation has been pulse amplitude
modulation, PAM. Using this method, a continuous signal, such as speech is
represented by a series of pulses called samples. The amplitude of each sample is
directly proportional to the instantaneous voltage amplitude of the signal at the
time of the sampling.

Since amplitudes of the samples are continuously variable, the problems of

cumulative noise and distortion associated with the analog signal are still present
in the PAM systems. Although this cumulative noise problem prevents PAM from
being used for interoffice trunking, some PABX's and channel banks use it
internally over short distances.

It was Alex Reeves who resolved that the problems of cumulative noise and
distortion could not be overcome in PAM systems. His investigation resulted in
the invention of a radically different approach to transmitting speech signals
known as pulse code modulation, PCM.

31
Simplified Time Division Multiplex
PCM System

32
Pulse Code Modulation (PCM)

The application of TDM multiplexers in digital transmission begins with pulse code
modulation 1 PCM. In PCM 1 continuously variable voice signals are transferred into
a series of digital encoder pulses. The process is then reversed to recover the
original analog signals. PCM has three basic steps:

• sampling
• quantizing
• encoding

This procedure can be carried out in three successive operations by individual

pieces of equipment 1 or by a single electronic chip called a codec 1 coder-decoder.

Sampling

The incoming analog signal 1 representing the variations in the voice 1 is sampled
8000 times a second. The sampling circuit uses the sample to send a narrow
square wave pulse whose voltage i.e. height 1 is the same as the analog signals at
that point. This process is called pulse amplitude modulation, PAM.

The rate of 8000 samples per second comes from the Nyquist theorem. This
theorem shows an analog reconstruction from digital data can contain all the
information of the original analog signal if the sampling rate is at least twice the
highest frequency in the original signal.

In telephoning, the human voice is said to have a range of 300 -3400 Hz, or
approximately a 4 kHz bandwidth i.e. the VF channel. Therefore, a sampling rate
of 8000 times per second or one sample every 125 µs, will provide for an
adequate reconstruction of the original signal.

33
Sampling

5 Micro seconds

.·t2s~
Micro
Seconds

34
Quantizing

The amplitude of the PAM pulse can assume an infinite number of possible
values. In PCM, the pulse amplitude is assigned to the nearest value of a set of
discrete voltages or levels. These discrete levels or voltages known as quantum
steps are used to represent any level within the speech range. This is
accomplished by using the quantum step nearest to the actual amplitude value of
the pulse sample.

For example, an actual sample with the value of 28.24, would be represented by a
quantum step 28. A sample of 28.61 would be represented by quantum step 29,
and so on. Since the quantum step only approximates the actual value, there is
always some error. In speech signals, such errors are random in nature and are
referred to as quantizing noise or quantizing error.

Quantizing noise is the major source of signal distortion in PCM systems. The
degree of quantizing noise is mainly a function of the number of quantum steps
used. The more quantum steps used, the less quantizing noise. However,
increasing the number of steps increases the bandwidth required to transmit the
coded signals.

35
Quantizing

f\ 2SIS
• Non-Unifom1
~rrtizecj
St~
V
Loud
Speech

_'!""*" _ _ _ _ _ _ _ _ _ .........__. 29th Step

28.61

1st Sample ·-·- - - - - - -_-_-_-_-___, 28th Step

27th Step

36
Companding

It is, of course, necessary that the quantizing process detect all of the positive and
negative amplitude levels within the dynamics speech range. Experiments have
shown that approximately 2048 uniform size quantum steps are required to cover
the speech range and to provide sufficient signal fidelity. An excessively large
bandwidth is required to transmit such a large number of uniform quantum steps,
2048 equals 11 binary bits.

One way of reducing the number of quantum steps without sacrificing quality is to
make the size of the steps non-uniform, thereby taking advantage of the
statistical distribution of speech amplitudes. Most of the information in speech
signals, 85%, is concentrated at low amplitude levels. If the quantum steps are all
equal in size, the low-level or weak signals suffer the greatest amount of
quantizing error.

Current techniques for overcoming this are to use non-linear coding process or
companding. Companding is a contraction of the words compressing and
expanding. This process was first used in analog transmission systems to reduce
the effort of noise. The low-level signals are increased and the high level signals
are reduce. After transmission, during which noise is accumulated, the signal is
expanded back to its original dynamic range.

The compounding effect is achieved in PCM systems by effectively using more

quantizing levels for the lower amplitude samples and fewer for the higher
amplitude samples. In this way, the quantizing distortion is reduced. The result is
a constant signal-to-noise over a wide range of PAM signal amplitudes.

There are two types of compounding used on digital transmission links. Each uses
a different non-linear scale against which the samples are compared. One is called
Mu-law and the other is called A-law. Mu-law is used in T-1 carrier applications
and the A-law in El carrier applications.

37
Companding
- - - - -- -- - - - - - - -

1010

I
l
1011
I t .t
f
f I
11ttl t
I I
l t

1101
I I I
t

111D l
I f
I f
,n,
I
15al ~

116..l) 2'» PAM Sample mV

38
Encoding

The final step in the PCM process is to code the quantum steps into digital form. If
each quantum step is numbered in decimal form, then some type of digital code
can be developed to represent each of the numbers. Ordinarily, a binary code is
used that consists of a combination of binary ones and zeros, each group
representing a decimal number. Once the code is established, a series of on-off
binary pulses representing the code group can be used for transmission.

The number of quantum steps that can be represented with the binary code is 2
to the nth power, where "n" is the number of binary bits positions required for
each coded group. Current PCM coding uses an eight bit code with companding.
This requires 256 quantum steps. Only 255 combinations are used. The all "O"
encode is not used. The bit positions are assigned a value, as shown in the next
page. This is called the folded binary code. Note, a zero voltage signal is
represented by all ones.

39
Encoding

Code Level . SitNumber

Level Description 1 2 3 4 5 6 7 8

+0 Peak positive level 1 0 0 ·O 0 0 D 0

...1 1 0 a 0 0 0 0 .1
+2 1 0 0 0 0 0 1 0
• •
• •
.
• •
+126 , 1 1 1 1 1 1 :1 0
+127 Centre levels 1 1 1 1 1 1 1 1
-127 Nominal
. '
zero 0 1 1 1 1• 1 1 1
..:126 0 1 1 1 1 1 1 0
• •
• •
• •
~2 0 0 0 0 0 0 1 0
~1 0 0 0 0 a 0 0 1
a Peak negative level 0 0 0 0 0 0 1· 0
• Note: an alt.zero code wiU be suppressed.

40
PCM Frame Format

Standard Frame

In the digital transmission systems used in North America, 24 channels are

multiplexed into their 125 µs time period provided by the 8000 times per second
sampling rate. The standard format has one 8-bit word from each of the 24 voice
channels in series, plus one extra bit each time period. This is referred to as a
frame.

This establishes the basic transmission rate of 1.544 Mb per second as the
fundamental level in the digital hierarchy.

1.544 Mb/s = (8 bits/ channel X 24 channels/ frame +1 frame bit) X 8000 samples
per second

In the D4 format, 24 channels are transmitted with 8 bits per channel, plus one
framing bit for synchronization every 125 microseconds (one frame every 125
microseconds). This is repeated 8,000 times per second (1 second = 8,000 125-
microsecond intervals) to produce 1,544,000 total bps (193 bits x 8,000 times per
second= 1,544,000 bps).

41
St· an d·ard. .·.·. .F.rame. .· · ... · . ··at
·.. · · . ·. ·F······· orm . . ..

Framing Bit 7
F

~---------Frame= 125J1,ec-----------ii!i,,;1

42
Signalling Information

The 24 8-bit words of a frame represent voice samples as well as signalling

information. The signalling that has to be transmitted between two telephone
offices can take two forms:

• pulsing, dial pulse

• supervision, on-hook and off-hook

This information is sent in bit eight, the least significant bit, {LSB), of every
channel in every six frame using the technique called bit robbing.

SA and SB are two types of inter-office signalling. The SA signalling bit is use for
the on-hook/off-hook status and dial pulse information, and is sent in every odd
six frame. The SB signalling bit is used only for certain types of foreign exchange
office (FXO) to supply extra control for such things as specialized ringing, and is
sent in every 12th frame. Collectively these functions are called telephone
signalling.

As with any other information on the digital line, signalling over T-1 is done in bits.
The presence of a specific signalling condition must be coded and multiplexed
with the voice signal. PCM channel banks have a standard way to transmit
signalling. Since the data stream is redundant, small portions of it can be taken for
signalling with no apparent effects on voice quality. In North America, the least
significant bit in every six sample in each voice channel is devoted to Rob Bit
Signalling. (RBS) These bit positions are not available for voice information.

More important to users of voice and data networks, the least significant bit is not
available for data either. The network can change the bit to meet signalling needs.
Therefore the bit is completely unreliable. To avoid dealing with the signalling bit,
the practice for data in North America is to ignore its position entirely. In a 64,000
bit per second voice channel one bit in eight is discarded by being forced to the 1
state. One result is 56,000 bits per second DDS. The practice also guarantees
l'sdensity at 12 1/2% even if the data is all zeros. This does not apply to point-
point data circuits, as RBS would be turned off.

43
Newer transmission equipment and switches that signal on an external channel,
CCS, rather than in band support, CAS, support clear channel service that allows
full use of the DS-0. This kind of transmission has been deployed by carriers in the
1990s as ISDN, commonly known as PRI in telephone PBXs.

Signalling lnformatio.n

1~ 125 µs ·(193 bits) Frame alignment

signal (101010) .
in odd"'flllmbered
:frames
(1~3,5,7,9, l 1)
,,.. Channe1·1 . ._ J ... Channet 2 . ·. ~ j,.. Channer2~ .·..,,
Multiframe
alignment signal
81 . 82 ••• 88 81 ·s2 ... BB . (001'11 O) in even-
numbered frames
Frame (2;4,6,8,1 12)o.

1 2 3 4 7 8

Multiframe

Channel A signaling: Channel B signaling:

brtB of each bit 8 of each
channel time. slot channel time slot
in frameo < inframe12

44
 . l

- :f-?~,u
1:~AM~\Uf
OF1Ff~~ Signalling
CQNC>JttO,N:lO
C.HANJ4£t. UNfi
1
BtfS
TODS1
(NEARE;NO)
S'lRE.AM
A S
FXS (TO DRAW DT)
LOOP START
IDLE
ltJ~Y
(i;BOUNC.~!~~~ ·
HING OPEN~ tOOP O.P£M (ltl:LI)
. R1NC. GROUND, L.OOP OPEN (SElZUAE)
RtNG OPEN, LOOP CLOSED (BUSY)
FXO (TO RING PHONE)
LOOP START
IDLE 1
· BOS¥·· 1J
G:RQU·Nfl llt81L
TIP 012,m..NClwtii~tiG.
'TIP Aftll\t,Jil.:l8 l.:IA E'U~;£1t.lP-
···· .··· "1if'J,W,VJMW1 ..~ tl'ff!\lk~tmM
TlP •GRGWMlr .
4 w1RE:eau··· tntll Al
M·GROUND OR]Qlllif t1'ttG)
M·BATTERY .{SUSY\~*.. . . ..
4WlREE&M CTYPJS•Bt
E OPEN QR BATTERY flD·LE)
e,~ND (BU.SY} ..

45
Frame synchronization

At this point, we need to establish two very important requirements of the

system. If we are transmitting between any two points, and the transmitter sends
the first bit of the first channel, how does the receiver know what it's receiving?
What about signalling information, how does the receiver know which frame is
the sixth frame?

It was stated earlier that a frame consisted of 24, eight bit words plus a
synchronization or overhead bit for total of 193 bits. This synchronization bit,
which is called the S-bit, will contain two separate and distinct patterns of ones
and zeros that will synchronize the transmit and receive terminals.

These two patterns are called:

• terminal synchronization
• frame synchronization

Terminal synchronization-TS

Identifies the next bit is the first bit of the first channel. This is required to align
the receiver timing with the incoming bit stream from the far end transmitter, so
that the channel word decoding can be accomplished in the proper sequence. To
do this, a TS framing pattern is sent as a repetitive pattern of 101010. This is a
unique pattern that no other tone, voice, or data will duplicate on a continuing
basis.

Frame synchronization-FS

Identifies that the next 24 channels will contain signalling information in the least
significant bit, bit 8 of each channel. Since signalling is transmitted only once
every six frame, a pattern is needed for the receiver to identify which of the
frames is the 6th frame. It is again, a unique repetitive pattern consisting of
001110, interleaved with the TS pattern, and placed in bit 193 of every other
frame.

46
Frame Synchronization

Terminal Syhc
101010 - - - - - -

'S' haredBit
Combiner Cct.
Transmit
Corntrion
(193rdbit)
1 o o o 11 0.111 a o

001110 _ _ _ ___._....
Frame Sync

47
Superframe Format

The two TS and FS patterns are combined to time-share the 193rd bit in each
frame. A complete sequence of these time-shared bits requires 12 frames, and is
called a superframe.

With these patterns, the two signalling conditions SA and SB can be sent in
alternating six frames. Signalling A is sent in the 8th bit of all 24 channels in the
frame following the frame where the FS pattern changes from Oto a 1. Signalling
Bis sent in the 8th bit of all 24 channels in the frame following the frame where
that FS pattern changes from a lto a 0.

48
Super Frame Format

'"""'"----,----2315
Multiframe (Superlrame)
bits-----------~
• I
Signalling S+gnalling Signallil"lQ

Frame Frame Frame Frame Frame Frame Frame FrameJFrame·. Frame Frame. Frame Frame Frame
12 1 2 3 4 S 6 7 . ·~··.. 9 10 11 12 1

t t t
'
1
J
0
0
J..
0
1 ) 1
0
J ' 'J
1
J
1
·o
0 .

I
t---- Frames _ ____,.,.+••-- Frame 6 ---J
~I

,r~-,~,:,~,~~,~,~,:,~,~~,
Frame bit
lnfoonly

Fr.me bit
Info & Signaling

Fname bit

• Each frame of 24 channels information bit$ plus Frame bit is

repeated 12 times to form the Multiframe (M-frame) ·
• Signalling is inserted in 6th and 12th Frame of each M-frame

49
Extended Super Frame-ESF

The extended superframe, ESF, extends the multiframe structure from 12 to 24

frames and redefines the 8 kilobit per second framing overhead.

The Extended Superframe format is an extension of the D4 format that was

announced in 1981. As old AT&T equipment is replaced, new equipment
supporting the ESF is installed, and users are offered the ESF. In addition to
benefiting end users indirectly by offering a more reliable digital service, ESF
allows the use of the bandwidth to provide more advanced services.

Unlike the D4 Superframe, which requires 8,000 bps for housekeeping, ESF
requires only 2,000 bps. This implies that the other 6,000 bps become available
for other service-related purposes.
Among these services might be the user's ability to reconfigure their networks in
realtime from a data terminal.

The ESF has 24 frames in its definition of a Superframe, but only six bits in its
framing pattern. Rather than re synchronize every 1.5 milliseconds in the regular
format, it only needs to re synchronize every 3 milliseconds.

The framing bits in ESF consists of three separate patterns:

Framing

Beginning with frame four, the framing bit of every fourth frame forms a 2kbps
pattern of 001011, which is used to obtain frame and multiframe synchronization.
Frame synchronization is used to identify each 64 kbs PCM channel of each frame.
Multiframe synchronization is used to locate each frame within the multiframe in
order to extract the cyclic redundancy check, CRC, and data link information,FDL,
and identify the frames that contain signalling. Signalling bits carried by frames 6,
12, 18 and 24 are called A, B, C and D respectively.

Cyclic Redundancy Check-CRC

This is contained within the framing bit positions of frames 2, 6, 10, 14, 18 and 22
of every multiframe. The check bits (CB) that make up the CRC are used to detect
errors within the extended multiframe.

50
Data Link

Beginning with frame one of every multiframe, every other 193rd bit is part of a 4
kbs data link commonly called facilities data link, FOL, which can be used for:

• transmission of alarms
• status signals
• supervisory signalling
• network configurations
• performance indicators
• maintenance indicators

51
 Extended Superirame Format

Bit Use in Each Signalling Bit

Framing Bits Channe!Ttma Slot Us.a Options

Frame
Number F D C Traffic Signalling T 2 4 16

Framing Bits 1 - D -· Bit$1-8

F: Framing 8® C2 Kbp$) 2 - - C ea., 1--8 ;

D: Data Link Bits (4 Kbps) 3 - D -- Bits ,-a

4 a - Bits1-8
C: CRC {Cyciical Redoodancy Check) s - D C- Bits1-8
-
Bits (2 Kbp&) 6 - D- - 8its1•7 Bit8
,,
A A A
7 - Bits1'-8
Option 1:
a a - - 8ibc 1-8
Tr.anspanmt (Bi1 8 is for traffic) 9 - D -C Bits 1-8
Optioo 2: 10 - 0- - Bits1-8
2 State Signalling (Signalling Bit Al
Optioo 4:
11
12 ,- - - 131t111-8
Bits 1-7 Bit 8 - A B 8
13 - 0 - BnsH!
.4 State Signalling (Signalflng Bits 14 - -0 C Bits Hi
A and 8) 15 - - BitsH!
Option 16; 16 0 -0 - Bits1-8
16 State Signalling {Signalling Bits 17 - - BitsH!
-
A. 8~ C, and D) 18 - D- C Bits 1-7 Bit8 A A C
19 - - Bit$ 1-8
20 1 - - Bits1-8
21 - D C- ·13rn,, ,..a
2Z - - Bits 1-8
1-s ·
23 -1 D - -
Bib
- .A
24 - Bits 1-7 Bit 8 . B D

52
CRC

On the transmit end:

Each ESF M frame (4632 bits) is divided by a Fixed Polynomial x + X+l (10001 in
binary) Control bits are set to "1" for this calculation

The remainder is inserted in check bits E1-E6 of the next frame

On the receive end:

The ESF M frame (4632 bits} is divided by the same fixed polynomial x +X+l as per
the transmit end

The remainder is compared with the remainder sent from the transmit end in
check bits E1-E6 of the next frame.

If the remainder is the same this indicates no errors have occurred, if the
remainder is not the same this indicates one or more errors have occurred. 98.4%
probability of detecting errors

Each time the remainder's are different a 1 is added to a counter which calculates
the error rate.

53
Line Coding Techniques

When ordering a Tl span, the user must specify the coding method to be used on
the span. In the early 1960s, considerable study was given to the choice of a
coding method for placing the information signal onto the transmission carrier. A
bipolar encoding scheme was selected to prepare the signal for direct application
to a Tl copper facility. The choice of bipolar was largely determined by the
characteristics of the copper medium. In bipolar coding, alternating
positive and negative pulses represent one state (a binary 1). Absence of bipolar
coding pulses represents the other state (a binary 0).

Tl information is transmitted as positive and negative marks (ones} which

alternate. A logical O is coded as a "zero excursion." This coding technique is
called bipolar return to zero, or Alternate Mark Inversion {AMI).

Digital line signals need conditioning for transmission. A simple pulse signal may
be suitable for modulating a carrier but it is generally not a good form of applying
directly to metallic lines.

Other factors which encourages the use of conditioning or further encoding are:

• the need to maintain timing information

• the need to provide error-checking capability
• the need to remove unwanted signal energy

The line coding techniques we will look at are:

• Unipolar Non-return to Zero-NRZ

• Alternate Mark Inversion-AMI
• 15 Zero Suppression
• Bipolar 8 Zero Substitution-88ZS

54
Line Coding Techniques .

• Unipolar Non-.return to Zero (NRZ)

• ·Alternate Mark (nversion (AMI)

• 15 Zero Suppression ·

• Bipolar8 Zero Substitution (B8ZS)

55
Unipolar Non-Return To Zero-NRZ

The signals internal to PCM equipment are, in general, unipolar non-return to

zero, NRZ, a format to simplify equipment design.

The unipolar NRZ has a pulse for a 1 and no pulse for a 0, with the level
maintained by the du ration of the signal interval. If the number of ones and zeros
are approximately equal, no DC component exists. However, if long strings of
ones or zeros occur, significant DC wonder may result, increasing the likelihood of
errors.

NRZ-L - Non-Return to Zero-Level

1
I
I
1 1 0 1 I 0
I
0 1 f
I
D
I
. Binary
information

I . I I
I
I

~-__L___I __I___/ __;_____J ~--L___

Transition with level change

56
Alternate Mark Inversion-AMI

For transmission over digital lines, the AMI format is used. It has a 50% duty cycle
were the ones are represented by pulses and zeros are a zero voltage level.
However, each alternate one is reversed in polarity. This process is also referred
to as alternate Mark inversion-AMI.

The advantages of bipolar format over unipolar format are:

Line Frequency Halved

If we transmit a unipolar stream, the line frequency rate would be equal to the
transmission bit rate of 1.544 Mb per second {l.544 MHz). However, when we
transmit in the bipolar form, the frequency is effectively halved to 772 kHz.

DC Component Removed

Because the bipolar bitstream uses both positive and negative pulses, it is AC in
nature. This allows us to transformer coupled DC voltages to the line to power
line repeaters that are required to regenerate the signal over any distance.

Error Detection

With the bipolar bitstream, every other logical one is inverted. If at the far end
you received two consecutive ones of the same polarity, then you have what is
called a bipolar violation or. BPV. A BPV is in an error, a bit has been added or
deleted,. The system will count the errors and upon exceeding a predetermined
value, will give you an alarm.

57
Alternate Mark Inversion (AMI)

Binary
information
1 =mark.
... 1 1 0 1 0 0 1 1
0=5pace I I I
.I I 1
+ I i
I I

0 ---+--
1 I I
l I I
I . I I
J l
. I

Advantages • In Attemate Marie lnwrs ion each

mark (1}~nteredcausesa
1. ·Roducad bandwidth bY halving AC frequency
reversal in .the polarity of the signal
2. Allows transtoonar coupling (q~asi ~ Aq - (commonly known as Bipolar) .

3. Error detection (nextpulse m!Jft ba. opposite polarity) • DSX standard .

58
 I

Ones Density

Repeaters and multiplexers must track the data bit pattern through rapidly
changing phase shifts of many bit times. They cannot track on O's only l's. A long
string of zeros increases the uncertainty in timing recovery. How many bit periods
have passed since the last 1 was received? If the period of all O's is long enough,
and the jitter large enough, uncertainty may exceed one bit time. The receiver
cannot tell where one bit position ends and the next starts.

If the line repeaters loses synchronization, it may oscillate, which interferes with
T-1 signals in adjacent cables. This problem can be very hard to isolate.

Therefore, a certain ones density must be present on the line to ensure proper
timing. The specification for T-1 transmission facilities, AT&T compatibility bulletin
119, states that no more than 15 zeros shall be sent in a row, AND there shall be
at least N one 1 s in every 8 (N+l bits) for an average ones density of 12 1/2%. This
number of ones is sufficient to keep any type of T-1 circuit synchronize. Ones
density is particularly important to the standard line repeaters used in common T-
carrier spans, (span lines). They are not in central offices where they could receive
accurate clocking.

In 1987 the FCC recognize that the newer equipment which now dominates the
public network is not as sensitive to strings of zeros. Nor are new repeaters
subject to oscillation. Therefore the FCC relaxed the ones density requirement. As
far as they are now concerned, no harm will be done to the equipment by up to
80 zeros in a row as long as the average density is still at least 12.5%. However 1
many central offices set their equipment to raise an alarm after 15 zeros. Carriers
naturally want to avoid alarms and so still want users to maintain the old pulse
density.

59
15 Zero Suppression

To maintain the line repeater clocking, digital line signals have the restriction
whereby the maximum number of consecutive zeros that can be transmitted is
15, a minimum average ones density of 12 1/2% is required for clock recovery.

To ensure this, a PCM codec is constructed so that the "00000000" code is never
transmitted. Instead, the second least significant bit is set to one so that
"00000010" is transmitted. In effect, an encoder errors is purposely produced to
avoid the all-zero code. Fortunately, maximum negative amplitude samples are
extremely unlikely, so no significant degradation should occur.

If the least significant bit were forced to a 1, a smaller decoding error would
result. However, in every six frame this bit is used for signalling purposes and
therefore, is occasionally set to zero independent of the coding. Remember,
signalling has precedence over the encode. Hence, the choice of the second least
significant bit.

60
15 Zero Suppression

......- - - - - - - ~ - - 1 5 Zeros ----------...-i

I1 0 0 l O IO I IO IO O I 1.0 I
O
Input
O IO IO l O O I O j

- - - - - - - - - 1 3 Zeros --------......-i
If IO ·IO IO I oj .o IO i
IO O IO IO IO l~ IO 1 IO I
Output

• PCM CODEC constucted such that any ·00000000· byte

becomes ·0000001 o·
• Maximum consecutive number of zeros in output is 13
.(15 consecutive zeros prevented) ,

61
88ZS

The ability to send any data pattern, including indefinite period of all zeros,
defines a Clear Channel.

Bipolar violation underlies one method to keep the ones density up when
transmitting consecutive zeros on a T-1 carrier. The method substitutes a known
pattern of ones, with bipolar violations, for a group of zeros. A bipolar violation
occurs when two one's pulses in a row have the same polarity.

It is not possible to signal control functions with normal bipolar patterns, because
all of them can be valid data. The technique of binary 8-zero suppression, or 88ZS,
violates the bipolar pattern by sending consecutive pulses of the same polarity.
The violations distinguish a byte substituted for all zeros from a normal byte of
data containing legal ones. If the receiver sees the known pattern, it restores the
eight zeros. 88ZS or something similar is needed to transmit 64 kb per second
clear data channels with absolute assurance and maintaining one's density to
AT&T standards.

This 88ZS pattern was selected to preserve the balance of plus and minus voltages
to prevent DC components on the wire, so it could be sent to transformers and
coupling capacitors as an AC signal. 88ZS is limited to T spans, typically local loops
that access a public network, and is not carried into and through the higher order
multiplexers like the M13. A M13 converts all pulses to a single polarity, and so.
loses track of the bipolar violations unless specifically designed for 88ZS

62
Bipolar 8 Zero Substitution

.B3ZS example
1 0 0 0 0 0 0 0 0
,... Original Word
--1
input

w
8 bits

f O IO IO lo 10 i Ofo j O!
as

I I l I
1 0
j;
I
I.
0 r.
I
0 v! a: 0t
v: B
I J I
I
I
fojo1ojv1sjo1vjsl output
I
I

I
I
0 = 0 I

V= Bipolar Violation
B= Bipolar Pulse

• Bipolar with 8 zeros substitution is the new DS-1 Standard for transparent 05-0 signals

• B consecutive zeros are replaced with ·ooovaovs· code

" Multiplexer substitutes and Demultiplex removes sud'l that the l'Tl8duim ·is transparent to ttie user

63
Clocking

All the digital nodes in the central office including multiplexer switches, DAC's,
share station clock. From the days when digital switches were introduced up to
the 1990s there was a single precise timing source, the basic system reference
frequency. If provided master clock for all of the Bell System in the US and other
carriers as well, such as in Canada. It was based on a cesium atomic clock, the
primary time reference. Carriers have since migrated to independent local
clocking sources for each CO, most often based on the global positioning system.
A GPS receiver in the central office can generate a clock to within 10 minus 11.
This is close enough to having identical clock that the expenses of distributing
clocking from one source no longer make sense.

The cost of cesium oscillators have come down considerably. It is economically

practical to make such a clock part of a large switch, like the 4ESS, and even a T-1
multiplexer. Connecting all switches in the network to the switches single clock
ensures that they all send and receive bits at the same rate. There are no bits lost
because one switch sends them faster than the next which can receive them.
Likewise, all of our multiplexers within the network are tied to the same clock as
well. Normally, our clock source in our multiplexers is set to derived. The
multiplexers also have the ability to generate their own clock via the internal or
holdover clock, and also accept an external clock source.

Temperature changes and other influences cause very minor variations in bit rate.
Buffer memory or elastic stores hold extra bits temporarily until they can be
processed. If the rate changes too much, the buffer overflows and there is a
frame slip. The buffer is cleared and operation continues, but data, 193 bits in one
frame, are lost or repeated.

Having one clock for all our nodes reduces the amount of buffering needed to
minimize the periodic errors due to frame slips. Small buffers also introduce
minimal transit delay.

64
Demarcation Hardware

Regardless of the vendor, a dedicated T-1 line will appear at the customers end as
two wire pairs, one for data send and one for data receive. The physical connector
used could be a DB-15 type connector or a RJ-48C. The RJ-48X is the same as the
RJ-48C except it has shorting pins that create a loopback when nothing is inserted
into the jack. For larger installations a DSX-1 panel may be present.

T-1 Termination Facility

T-1
CSU
- DSIJ T-1 Mux
V
+
Line

- D

L~:~~~~k ·.
(unframed
serviceJ
! OSX-1
Interface
L-----1 r
Various Interfaces
And Connectors V.35, RS-422
lnlemal to Mux
Connector
0815 or RJ46C

~~;:-11---?--- Present
- - - Customer Supplied--

Demark

DB-15 PIN PIN RJ48-C

g
., ....
••
1
11
3
15
9(Receive !Tam N!!lwmk}R
A(Ret:eive from Netwulk)T
N Connection
1
2
3,6

••••
9 B nsmltto Netwmk R1 4
1 A Tl'llllfflllt In Network T1 5
•• N.A. Optional Shield .7,8
••
15 ••
5,6,10,13 Reserwd forTelco use N.A .
8

65
A DSX-1 Panel

---····,,.,.,_, BATT
-----RTN
,,,,=~~-SGND

T
R Out
""---tT NE

Front Cross-Connect
---~R fn

TL.t--"""-'-4?"-
0ut T
R , ..,,_,,,,•~M~><·•=·-,...,,_,_,
T ~w-·c,,--,w-•••--w_,_..,
In R ,----,,,.__,,.,~,..--......"--

A digital signal crossconnect, DSX, is a central terminal for digital equipment at a

particular digital signal bit rate, providing both permanent and temporary
connections. DSX test ports provide bridged and series access for test or patching.

Hardwire is direct cabling between the network elements. Such an arrangement

has the following disadvantages:

• difficult cable management

• inadequate access for testing and monitoring
• hard to add on or rearrange, affecting circuit integrity
• possibly difficult circuit back-up in case of failure, possible lengthy service
down-time

66
 r

Terminating digital network equipment at a DSX-1 has these advantages over

hard wire arrangements:

• the DSX-1 can handle a large number of terminations in a non-blocking

arrangement
• network equipment can be handled or coordinated efficiently, in spite of
the location at the site
• grooming-adding, removing, rearranging circuit connections is easier
• fast service recovery and alternate routing are possible in case of NE failure
• the DSX-1 provides quick access to circuits for testing and monitoring, both
intrusive or nonintrusive
• circuits can be rolled with minimal interruption circuit integrity

Remember when doing a crossconnect at the DSX-1 panel, a role must be made
with the patch cords or crossconnect wires. This enables the out to go to the in,
and the in to go to the out. This allows the equipment to have the correct
transmission and receive paths. The second reason our role is needed, is to allow
us to monitor from both directions.

When connecting our channel bank to the DSX -1 panel, out goes to out and in
goes to in. In the wiring of a tie cable between two DSX-1 panels, a roll is needed
in the tie cable.

DSX-1 Digital Crossconnect Specifications

Signal:

• bit rate -1.544 Mb per second +/- 77 bits per second

• line code-bipolar AMI
• line impedance-100 ohms +-5%

Pulse:

• amplitude-+/- 3.0 V +/-.6 V

• width at half amplitude-294 to 404 nanoseconds
• one's density-one in eight (12.5%) with no more than 15 consecutive zeros

67
 • power level O dbm at full power, can be adjusted with the LBO settings

Cable and patch lengths

• maximum length DSX-1 to equipment-650 feet/ 200 m

• maximum patch cord length-26 feet/ 8 m
• cable type-22 AWG shielded twisted pair
• insertion loss-0.15 dB at 772 kHz

CSU: Channel Service Unit

The first piece of equipment connected directly to the line normally is the CSU or
channel service unit. Traditionally before divestiture in the United States and
always outside the United States the CSU has been considered part of the
network. A CSU contains the last signal regenerator on the line before the data
terminal equipment, DTE, and a mechanism to put the line into loopback for
testing from the central office.

T-1 Urie lntefface CSU DSX--1 Interface

Send Data
Signal
Repeater
Monitor

Remote Test Remote Test

Control Remote Test Indication

A third section of the CSU monitors the signal to detect violations of the bipolar,
15 zeros and ones density rul.es and to look for loss of signal. Violations typically
produce warnings, via lights. During the development of the CSU's several
additional functions were acquired:

68
Loopback

Originally, in 1983, the loopback was switched in by a relay control with a

separate wire pair. These days, the loopback command is in band on the T-1 line.
There are three types defined, all of them for the carriers use only:

Line Loop Back LLB

On 04 or ESF framed lines operating 54016, loopback of the full T-1 is latched in
by a bit sequence of 10000 repeated at the T-1 rate for 5 seconds. Release of LLB
results from 100 repeated for 5 seconds. Under Tl.403, the pattern 00001110
11111111 in the FOL actuates the LLB, while 00111000 11111111 releases it.

Payload Loop Back PLB

The extended superframe format includes the facilities datalink, buried in the
framing bits. A CO uses this channel to send the simplified X.25 message to this
CSU commanding a loopback. Under Tl.403, the pattern 00010100 11111111 in
the FOL actuates the PLB, while 00110010 11111111 releases it.

OS-0 Loop Back

Though not usually handled by the CSU, there may be tests commands present on
the T-1 for loopback on individual OS-0 or subrate channels.

Another important function of the CSU is to generate a keep alive signal when the
attached terminal equipment fails or is disconnected. AT&T used to request one
of three types, depending on the equipment they used to supply the service:

• unframed, all l's

• 04 or ESF frame containing all ones in the data positions
• loopback of signal received from the network

69
A later version of 62411 specifies only the all ones type. This is also known as a
blue alarm or alarm indication signat AIS. In other words when the T-1
multiplexer fails or is disconnected 1 the CSU must send continuous unframed ones
to the network. The same applies to ESF lines from AT&T1 though other carriers
may request other types

Since the customer must now provide the CSU as well as the DSU 1 some vendors
are offering combined devices. These designs can save space 1 but when not built
into the DTE, may raise the question of where this CSU draws its power. When the
CSU belong to the network1 the carrier supply DC power through the signal leads.
Send and receive data pairs are balance and isolated from ground by
transformers. Each pair can therefore be used as one power lead. AT&T originally
thought it important to keep the CSU functioning during a local power failure at
the customer site. A working CSU would allow test personnel to distinguish that
type of power failure from a network problem.

ESF Statistics

The latest job commonly assigned to the CSU is to calculate and collect the ESF
error statistics there are two ways this is done.

AT&T started to deploy equipment that assumed the CSU would hold the latest
24-hour history of errors. Then 1 on a command from the central office over the
facilities data link, FDL1 the CSU sent the error history to the central office, also
over the FDL The line monitor unit, LMU 1 in the CO inserts a command and picks
off the statistics.

The second version is now the ANSI standard 1 adopted by the local exchange
carriers. In the ANSI version 1 this CSU sends a report to the CO every second,
describing the quality of the transmission during the previous 4 seconds. Only the
CO collects the statistics. The physical connection in the CO is the same, with the
LMU taking the reports off the FDL.

70
l's Density Enforcement

To meet FCC and AT&T standards for ones density, some CSU 1s can overwrite a
zero bit with a one when necessary to avoid transmitting long strings of zeros.
There are two methods or options available:

Part 68: the FCC allows up to 80 zeros in a row as long as the average density ones
is at least 12 1/2% over a short interval. That many zeros are often found in
datastreams from video codecs and data encryption devices. Some carrier
equipment will raise the alarm on finding as few as 16 consecutive zeros.

62411: AT&T limit in general is 15 zeros in a row, with a 12.5% average density.
However, for access to channelized or switch services the requirement is at least
one 1 in each byte. When set to this standard, the CSU will ensure that no carrier
alarms are raised. Unfortunately, substituting ones for zeros may cause problems
with the application. Some multiplexers will jam bit 7, or place a one in the
seventh bit of an all zero byte. The seventh position avoids altering signalling or
control bits.

71
 . · Central
Clients premlses

LMU - Line Monitor Unit, in the CO issue commands. to the CSU over
the FOL. ESF quality stats ar~.ret.urnedfrom the CSU to the LMU
also over the FOL The LMLf mm~tbe phsically inserted in to the line
andterrrlinatestheFDL. May also be used to.listen for pedodic
reports from'i:heCSU, typicaH~;:ever minutefromthe CSU: lnthis
case since it is only listening ot is tapped onto the FOL and does not
terminate it (ANSrStandard)
. . .

• OSS - Operation.Support Systems, used byTelco's to Provision,

Monitor and Maintain facilities. · ··

OGU - Office Channel Unit, just a CSU in the CO

72
Digital Service Unit- DSU

A Digital Service Unit, sometimes called a data service unit, is a piece of

telecommunications circuit terminating equipment that transforms digital data
between telephone company lines and local equipment. The device converts
bipolar digital signals coming ultimately from a digital circuit and directly from a
Channel service unit (CSU), into a format (e.g. V.35) compatible with the piece of
data terminal equipment (DTE) (e.g. a router) to which the data is sent. It gives
the customer a useable interface. They are also common on older PBX's or key
sets that do not have a T-1 interface. The DSU also performs a similar process in
reverse for data heading from the DTE toward the circuit. The
telecommunications service, a DSU supports can be a point-to-point or multipoint
operation in a digital data network.

A DSU is a two or more port device; one port is called the WAN (Wide Area
Network) port (our T-1 interface side) and the other is called a DTE port(s)
connects to our PABX or router. The purpose of the DSU is to transfer serial data
synchronously between the WAN port and the DTE ports. If more than one DTE
port is used, the DSU assigns the DTE data according to time slots (channels) on
the WAN side. This would be used in a fractional T-1 or where we would want to
split the T-1 signal amongst two or more pieces of equipment.

On the WAN side, the DSU, via a CSU, interfaces with a digital carrier such as DSl
or a low speed Digital Data Service. On the DTE side, the DSU provides control
lines, timing lines and appropriate physical and electrical interface. To maintain
the synchronous relationship between the ports, the DSU manages timing by
slaving ports to the bit rate of another or to its internal clock. Typically, the DTE
port provides timing to the data terminal equipment while the WAN port dictates
the rate. The customer in almost all cases will always derive the clocking from the
telco.

DSUs usually include some maintenance capabilities. At minimum, they can loop
data back at either the WAN or DTE ports, or at both. When only one port is
looped back, the data received at that port is simultaneously sent back toward
the port and passed in normal fashion to the other port. Most DSUs also allow
various data patterns to be generated and monitored to measure error rate of the
communication link. A DSU may be a separate piece of equipment, or may be
combined in a CSU/DSU.

73
 DSU·
f,1&"422
DSX-1 v-,~j--

J ~l~fil!§:JjQ§

DSU ~===.,\1~~~§
t""'='==-====--.,;t·-.,·, JOat~
_,.,,, ..

"Modem I t.Jn~Ofiv~r"

Smartjack

Several types of network interface device (NID) provide more than just a terminal
for the connection of wiring. Such NIDs are called smartjacks or possible network
interface unit (NIU) as an indication of their built-in "intelligence", as opposed to a
simple NID, which is just a wiring device. Smartjacks are typically used for
telecommunication service, such as Tl lines. POTS lines generally are not
equipped with smartjacks.

Despite the name, most smartjacks are much more than a simple telephone jack.
One common form for a smartjack is a circuit board with a face plate on one edge,
mounted in an enclosure. Normally we will see status indicators and a set of
bantam jacks to do intrusive and non-intrusive testing. The telco's will install
these devices at the demarc when they want to have remote loopback
capabilities.

74
A smartjack may provide signal conversion, converting codes and protocols (e.g.
framing types) to the type needed by the customer equipment. However, if we
want these features we will normally install a CSU.It may buffer and/or
regenerate the signal, to compensate for signal degradation from line
transmission, similar to what a repeater does.

A very common capability provided by a smartjack is loopback, such that the

signal from the telephone company signal is transmitted back to the telephone
company. This allows the telephone company to test the line from the central
office, without the need to have test equipment at the customer site. The
telephone company usually has the ability to remotely activate loopback, without
even needing personnel at the customer site. When looped back, the customer
equipment is disconnected from the line.

Additional smartjack diagnostic capabilities include alarm indication signal, which

reports trouble at one end of the line to the far end. This helps the telco company
know if trouble is present in the line, the smartjack, or customer equipment.
Indicator lights to show configuration, status, and alarms are also common.

Smartjacks typically derive their operating power from the telephone line, rather
than relying on premises electrical power, although this is not a universal rule.

REPEATERS
When a signal is transmitted down a cable pair, it is attenuated (loses strength)
and distorted (noise is introduced) by the characteristics of the cable. However, in
digital transmission, the signal intelligence is defined by the presence or absence
of pulses, not by the shape as in an analog carrier. Since the repeaters detect the
presence or absence of pulses and restore them to their original form and
amplitude (height), the signal arrives at the far end as a nearly exact replica of the
transmitted signal. Since noise and distortion are removed from the signal each
time it is restored (regenerated), the signal can be transmitted over long distances
and arrive at its destination in a zero-loss state.

The Tl span will typically have many repeaters up to a practical maximum of 40,
and they are '
I

75
separated by distances which are determined primarily by the cable attenuation
{typically 1 mile except 3000 feet between CSU and repeater) and the ability of
the repeaters to filter out pulse jitter {timing loss).

Regenerative Repeaters
Repeaters for digital signals are called "regenerative repeaters" since they create
brand new pulses to send along the line. The repeaters examine each incoming
"smeared" pulse and decide whether the pulse is binary one or zero. Based on the
decision made, either a brand new one or zero is created and sent along the line.
Noise or other impairments are not amplified and also sent along as with analog
transmissions. Repeaters are usually placed about one mile apart to be able to
reliably regenerate the 1,544,000 pulses per second.

Independent transmit and receive paths are provided to and from the central
office, using four wires {two for transmission in each direction). The repeaters are
bi-directional. Historically, power for the span repeaters has been provided by the
central office. This is slowly changing as fiber optics is being implemented. Fiber
cables cannot carry the power.

Office Repeater (The CSU)

The Tl span is terminated at the central office with an Office Repeater {OR). The
OR regenerates the signal for routing and switching within the central office. The
office repeater ensures that the Tl data stream complies with 1.544 Mbps pulse
shape requirements, as defined in AT&T Publication 62411.

This publication also defines the ones density and framing requirements. The OR
incorporates circuitry to enforce ones density and detect bipolar violations.

76
Test Level Points and Decibels

At test level point, transmission level point, is a point where the signal value may
be monitored, a non-invasive measurement. The nominal impedance at the TLP is
600 ohms for telephone circuits. The TLP reference value is the full value that a
full level test tone would be at that point. The decibel unit used at the TLP is
dBmO. Typical values at a channel are -16 dBmO (tx) and +7 dBmO (rx). 0 dBmO is
equal to O dBm, that is one milliWatt into 600 ohms.

A full level test tone is the maximum value of a signal that can be applied at that
point. This value is never used in practice, as it can saturate system components,
cause non-linear distortion and sometimes physically damaged components.

The standard test tone for telephone and data work is 1004 Hz, at 10 dB down
from the stated TLP value. Unless otherwise advised, all analog testing is done
with a 1004 Hz tone, -10 dBmO. This is often referred to as a 10 down tone.

• Typical data levels are 13 DB down from the stated TLP value, that is -13
dBmO
• voice levels are variable, with ranges from 4 to 6 dB below TLP typical.

The signal frequency of 2600 Hz is sent at a high level, usually -6 dB down, for a
short period, then at 20 dB below TLP, or -20 dBmO. SF levels are measured at the
lower value of 20 dB down.

Noise may be measured across a flat bandwidth of 4 kHz, in dBm units, or using
the C Message weighted frequency curve in dBrnC units. 0 dbrn= -9-dBrn and 0
dBrnC. C message filters account for the varying effects that noise at different
frequencies has on voice intelligence. Noise levels with respect to TLP levels vary
widely between companies and equipment.

77
This example uses a maximum noise level 37 dB down from TLP .

...
·

+13 TLP ~1§,!LP ·... · +7'fll? ____ .,., ¾

... · ..J'fLP
_________. , , ~ -~----·----- --·-·

Full Test Tone +13 demo ~1(l dB.mo

T-esl tone +:fdBm ~it£ d-8m

· Data level . . 0 d6m ~?fl <,H~m ·

SF Level -TdBm

f',loisef'lat 66 dBm (<l4 .girn) ~1 clBm (~ij~ ~am) 60 d1;1,rn (~30 dSm) . 50 dBm (..4.0dllm}

~c~ Noise ee d6rn (,,z4 r;if&m) 37 dirn (-Sl~) ® i1.1irf'I .(~ao ~m) {)Q d~r:m ,,a~1;11) <

You can configure the TLP's for any voice interface card or channel unit. The TLP's
specifie the receive and transmit levels in decibels, of the signal from a voice
circuit with respect to the digital trunk. The receive TLP refers to the digital to
analog level, the difference between a and b; the transmit TLP refers to the
analog to digital level, the difference between d and c

Voice drcuit

The TLP's are measured in dBm, which is the power of a signal relative to a signal
of one milliwatt.

A TLP is measured with respect to 0 dBm on the digital side. The 0 dBm value is
equal to 1 mW of power imposed on an impedance of 600 ohms at a frequency of

78
1004 Hz. On the digital side, 0 dBm is the digital signal power required to produce
0 dBm on the analog side of a standard digital-to-analog converter.

79
Tl Testing

Whether public or private, T-1 circuits and network equipment must be properly
tested and maintained to perform to maximum efficiency. Accordingly, all T-1
testing falls under one of two prescribed categories: out of service testing and in
service monitoring.

Out Of Service Testing

Out of service testing is so named because live traffic must be removed from the
T-1 link before testing can begin. In its place, test equipment transmits a specific
data pattern to a receiving test instrument that knows the sequence of the
pattern being sent. Any deviations from the transmitted pattern are then counted
as errors by the receiving instrument.

Out of service testing can be conducted on a point to point basis or by creating a

loop back. Point-to-point testing is a general practice, and requires two test
instruments, one at either end of the T-1 link.

By simultaneously generating a test pattern and analyzing the received data for
errors, the test instruments can analyze the performance of the link in both
directions.

Loopback testing is often used as a quick check of circuit performance or when

isolating faulty equipment. In loopback testing, a single test instrument sends a
link up code to the far end CSU before data is actually transmitted. The loop up

80
code causes all transmitted data to be loopback towards the test instrument. By
analyzing the received data for errors, the test instrument measures the
performance of the link up to and including the far end CSU.

Because loopback testing only requires a single test instrument, and therefore
only one operator, it is very convenient. However, loopback testing is limited in
that it can only analyze the combined performance of both directions of the link.
As such, it is extremely difficult to determine whether errors are originating on
the transmit or the receive side of the T-1 link at any given time.

T1 LINK

I: t1:~·.·····~-
LOOPBACK

TEST
INSTRUMENT

As out of service methods, both point to point and loopback test allowed detailed
measurements of any T-1 link. However, since all out of service testing requires
that live, revenue generating traffic be interrupted, it is impractical for long-term
testing. This type of testing is typically performed when the circuit is initially
installed or when errors are discovered when monitoring live data.

In Service Monitoring Of Live Data

The in-service method allows live data to be monitored at various access points
without disturbing revenue generating traffic. Since in service monitoring does
not disturb the transmission of live traffic, it is more suitable for routine
maintenance than out of service testing. Additionally, in service monitoring
indicates performance under actual operating conditions. But it's primary
disadvantage is that it's measurements may not be as precise as those available in
out of service testing.

81
Choosing in service monitoring or out of service testing at any given time often
depends on knowing which measurements are available in each method.

Measurement Description Available Prime Advantage Prime Disadvantage

Bit. Errors The basic Out of service only. Truest Live traffic must be
performance measurement of removed to allow
evaluator, counts a point-to-point transmission of a
number of logic circuit performance. known data pattern.
errors in the bit Permits stress
string, zero that testing to ensure
should be ones and that T-1 circuits and
vice versa. equipment are
Provide specific operating within
error applicable
measurements. standards.

BPVs A measure of the In service or out of Good indicator of Some Network

number of times service. circuit or repeater Equipment corrects
pulses of the same problem. BPVs making them
consecutive polarity Can be measured useful only when
were transmitted without disturbing testing metallic
across the T-1 live traffic. sections of the T-1
circuit, in violation circuit.
of the bipolar signal Satellite, fiber optic,
format. and microwave
Provides correct the bipolar
approximate bit format
error rate.
Frame Errors Measures the In service or out of When monitored for Only evaluates
number of times an service. a long.period can overhead bits not
incorrect value approximate actual data. Analysis only
appears in the bit error rate on an takes place on every
position reserved for in-service basis. 193rd bit.
framing, the 193rd Frame errors are
bit. often corrected by
Provides DCS and
approximate bit multiplexers. Some
error rate. frame errors are not
good indicators of
end-to-end
performance in
networks where this
equipment is
installed.
CRC Errors Detects one or more In-service or out of Very accurate in- Only available with
bit errors in a block service. service error ESF framing. DCS
of data. analysis detects bit and other Network
errors and 98.4% Equipment may
rate of accuracy recalculate the CRC.

82
Alarms and Error Events

Red Alarm-is a locally detected received failure. The red alarm occurs when an out
of frame {OOF) condition exists for more than 2.5 seconds. A red alarm means
that the local unit is not receiving a properly framed aligned signal from the
remote equipment.

Yellow Alarm-is a remotely detected failure generated when the remote end
detects a red alarm in superframe mode 1 the yellow alarm is generated by forcing
bit 2 of every channel in the superframe to a 0. In ESF mode the yellow alarm is
sent via the facility datalink.

Blue Alarm-is an unframed one signal, also known as the, keep alive signal.

Frequency Deviation-occurs when the 1.544 Mbps frequency deviates by more

than 50 bits per second.

Excess Zeros-are more than 15 consecutive zeros.

No Frame Sync-occurs when 2 out of 5 (for D1, D2,D3, &D4 framing) or 3 out of 5
(for ESF Framing) consecutive framing bits are incorrect.

No Signal-is a condition that occurs whenever 175 contiguous zeros appear on a

DSl line or when the input signal is lost for 150 ms or longer.

Severely Errored Seconds-is a 1 second interval having a bit error ratio worse than
1x10 to the minus 3.

Frame Slips-are caused by overflow or underflow of buffers due to extended clock

slippage. Back-to-back slips can be due to large periods of frame loss and buffers
unavailable to keep up.

Framing Errors-is an error in the 193rd framing bit position. A frame error is when
a frame bit does not conform to the 12 bit word using the framing bit in 12
consecutive frames. ESF framing errors occurs in the 24 bit framing pattern.

Bipolar Violations-is a bipolar signal, a one {Mark, Pulse) which has the same
polarity as its predecessor. Bipolar pulse violations are two consecutive one bits

83
of the same polarity. In other words, a bipolar pulse having the same voltage as
the preceding pulse in an AMI coding scheme.

Bert Pattern Descriptions

The QRSS, Quasi Random Signal Source, is the preferred test to measure T-1
performance. The QRSS has a frequency spectrum similar to live data by
containing every possible sequence or 20 bit combination while remaining within
the maximum zero constraint. Because the QRSS pattern stresses both timing and
equalization it should be used as the first stress pattern. When running this test in
15 min. intervals, that error time readings will measure proportionally to
performance objectives. Typically, acceptance or turn up testing is 3 out of 4 15
minute test having equal to or less than 20 errored seconds

The 3:24 pattern can be used to test any device that is out of tolerance in the
timing recovery circuit due to aging or loss of adjustment from damage. This is
done by generating 16 zeros in a row. The 3:24 pattern generates the n:,aximum
number of zeros the minimum ones density is produced.

The2 15-1 pattern can be used to test a repeater that is out of the tolerance and
the timing recovery circuit due to aging are lost of the adjustment from damage.
This test will stress the output of the automatic line build out (LBO)of the pre-
amplifier by testing a rapidly changing high and low density pattern that may
cause jitter in and out of tune repeater. 14 consecutive zeros are the maximum
produced by this pattern.

The 2 20-1 pattern can be used to test similar as the 2 15-1. This pattern will
stress slightly harder than the 2 15-1. Twenty two consecutive zeros are the
maximum produced by this pattern.

The 2 23-1 pattern can be used to test similar as the 2 15-1. This pattern will
stress slightly harder then the 2 20-1. Twenty two zeroes are the maximum
produced by this pattern.

84
The SPACE pattern will transmit all zeros and violate the ones density of the T-1
span line. The space pattern can be used to test CSU's, B8ZS for ones density
violation correcting capability. The space pattern can be used to test the Network
Equipment is not properly option for B8ZS encoding.

The MARK pattern can be used for continuity test, detect unwanted loops, and
will cause the repeater to consume the maximum CO power. When transmitted in
an unframed format it sends all one's similar to a blue pattern but is framed.

The 63 pattern is a random changing pattern used to simulate actual data.

The 511 pattern can stress slightly harder then the 63 pattern because it is a
larger random pattern. When changing, it simulates a larger variety of data than
the 63 pattern.

The 2047 pattern can stress harder then the 511 pattern. It is random changing
and simulates a larger variety of data than the 511 pattern.

The 1:1 or alternating ones pattern can stress the repeater's performance in
regard to DC current. By producing a pattern a 50% once density it will stress a
repeaters DC power consumption.

The 1:3 pattern will transmit an average ones density stressing repeaters.

The 1:7 can isolate or tests for timing recovery, one's density, and frame
alignment when running in a framed mode.

The 2:8 pattern will transmit minimum ones density stressing frame
synchronization.

The Yellow Alarm and Red Alarm patterns transmit simulated alarm condition.

85
Sub-Rate Multiplexing

Sub-rate multiplexing is the process of taking sub rate data, any data circuits
below 64 kb per second and multiplexing them onto a single DS-0. This is opposed
to super-rate multiplexing where we need to bond DS-0 together to transport
data speeds above 64 kb per second. Subrate switching is the process of switching
the subrate data from one DS-0 to another DS-0 at a hub location or terminal.
Special rate adaptation methods, or packaging methods are used to achieve
subrate multiplexing. The most common are:

• DDS-B
• Transparent (1.460)
• HCM

The term DDS is an acronym for digital data system, it describes the North
American digital transmission method that was initially deployed in the mid-1970s
and developed by Bell labs. It provides full-duplex point-to-point and multipoint.
transmission for subrate data circuits. Two major forms of DDS are available, DSO-
A and DSO-B. DSO-A is capable of transporting one subrate DS O circuit while DSO-
B is capable of subrate multiplexing. All subrate circuits must be of the same
speed. Speeds of 2.4, 4.8, 9.6, 19.2 and 56 kb per second are supported. All
circuits must be synchronous slave locked and propagation of signalling leads is
not supported.

86
 DDS DSO·A and DSO·B frame formats

• Framing bit • Standard CSU-to.CSU

• User data for MSB 8-bit DSO LSB communication channel
56 and 64 kb/s !II III!I! ·
DDS secondary channel, plus:
operation~ LJ ,,,,,,-- - end-to-end c.o. ntro. I lead propagation

~
·~ - continuity checking
• Switched 56 signalfing channel
• Primary channel • User data for 64 kb/s .operation
- user data
• inband
maintenance
codes

Frame 2.4 kb/s 4.8 kb/s 9.6 kbis 19,2 kb/s 56 or 64 kb/s

mm
1 i 2 3 4 1 2 1
2 5 6 7 8 3 4
3 9 10 11 12 5 6
4 13 14 15 16 7 8
5 17 18 19 20 9 10

5 frames, 5 frames, 5 frames, 5 frames, 1 frame,

ZO channels 10channels 5 channels 1 channel 1 channel

DSO-A: DSO-A: DSO-A: DSD-A: DSO-A:

• .user data.repeated • user data repeated • user data repeated SCH (17,9) OFF • 56 kb/s operation;
in all 20 cells in a!l 10 cells in all 5 cells • user data divided user data occupies
• majority vote 12/20 • majority vote 6/10 • majority vote 3/5 between cells framing bit plus
DS0-8: DS0-8; DSO-B: 2and 3 primary channel
• each cell • each cell • each cell • pad characters bits (7 bits total)
(numbers 1 to 20) (numbers i to 1O) (numbers 1 to 5) elsewhere • 64 kb/s operation:
represents a separate represents a rep resents a DSO•A_EC (using the OCU-DP
2A kb/s channel separate 4,8 kbfs separate 9,6 kb/s • user data divided channel unit) user
channel channel between cells data occupies
1 and2 framing bit, primary
• BCH code in cells channel bits, and
3and4 secondary channel
• framing in cell 5 bit (8 bits total)
DSO-B: • BCH (17, 9) code in
• each celf second DSO
(numbers 1 to 5)
represents a separate
9.6 kb/s channel
• 19.2 fits onto 2 and 3
and/or 4 and 5

87
Transparent rate adaptation support aggregate channels with bandwidths of: 8,
16, 24, 32, 40, 48, 56 and 64 kb per second. Transparent data channels carry data,
signalling and framing information through the multiplexer without adding any
system overhead. A common application of transparent rate adaptation is in the
use of transporting compressed voice circuits. Common bandwidths of
compressed voice are:

• pulse code modulation (PCM) of voice frequencies on a 64 kb per second

channel.
• 32 kb per second adaptive differential pulse code modulation (ADPCM),
sometimes known as M44 voice.
• coding of speech at 16 kb per second using low delay excited linear
prediction ld-celp
• coding of speech at each kilobits per second using conjugate structure
algebraic code excited linear prediction (CS_ACELP).
• High-capacity voice, HCV-8 and HCV-16

In reading about "G.xxx" voice compression standards please be aware that

although the voice channel are of the same size as above these standards
normally referred to VoIP transmissions that operate on a non-channelized T-1.
With the added overhead of UDP and IP there transmitted bandwidth is larger
than the stated voice compression rate.

High-capacity multiplexing, HCM, provides 98% bandwidth efficiency i.e. does not
support Clear Channel. HCM is a proprietary rate adaptation and subrate
multiplexing scheme that provides a bandwidth granularity of 800 bits per second
throughout the network. HCM provides additional functionality, such as:

• independent pass-through clocking, i.e. circuits can be synchronous non-

slave locked or asynchronous
• end-to-end signal propagation for up to eight control leads
• subrate multiplex circuits do not have to be the same speed however a
minimum speed of 2400 bits per second must be reserved.

88
 .
f,/
Ii

A 64 kb per second HCM frame with eighty 800 bits per second elements

87 86 85 B4 B3. 82 B1 BO
FO F s D D D D D D
F1 0 D D D D D
F2
F3
F4
10 rows
F5
F6
F7
F8
F9

8 columns (64 kb/s)

89
Primary Rate Interface

Primary rate interface, PRI, is actually a form of integrated services digital

network, ISDN. ISDN is delivered as basic rate (2B+D) and primary rate (23B+D).
Digital transmission of voice and data over copper wires, resulting in potentially
better voice quality than an analog phone can provide. It offers circuit-switched
connections.

In North America PRI service is delivered on one or more T-ls (sometimes

referred to as 23B+D) of 1.544 Mbs (24 channels). A Tl has 23 'B' channels and 1
'D' channel for signaling. In North America, Non-Facility Associated Signaling
allows two or more PRls to be controlled by a single D channel, and is sometimes
called 11 23B+D + n*24B 11 • D-channel backup allows for a second D channel in case
the primary fails.

The bearer channel (B) is a standard 64 kbs voice channel of 8 bits sampled at
8000 times per second with PCM encoding. B-Channels can also be used to carry
data, since they are nothing more than digital channels.

Each one of these channels is known as a DS0 that are clear channel. Most B
channels can carry a 64 kbs signal. The signaling channel (D) uses Q.931 for
signaling with the other side of the link.

A big advantage of the PRI over the T-1, PRI is capable of delivering Calling Line
Identification (CUD) or ANI in both directions so that the telephone number of an
extension, rather than a company's main number, can be sent. The PRl,(ISDN
protocol) delivers channelized, not-over-the-Internet service, powerful call setup
and routing features, faster setup and tear down, superior audio fidelity as
compared to POTS (plain old telephone service), lower delay and, at higher
densities, lower cost. PRI allows you to use the span bi-directional. For example
most Tl's require you too dedicate some channels for incoming and others for
outgoing unless it's a dedicated long distance Tl, out going only. This allows all 23
channels to be both outgoing and incoming rather than splitting them which is a
huge advantage.

90
Setting up the PR! is similar to setting up a T-1 however there are a few additional
steps that need to be set up on the customer PBX.

• Receive digit length and target lines. On a PRI one phone number is not
associated with the bearer channel. We must tell the PABX when it receives
the called party info on the D channel where to route it to.
• Ascending and descending order for the bearer channel use.
• Protocol to use on the D channel.
• Do we have licensing set up on our PABX to implement PRI

91
Tl Fail-Over (Tl Protection) Switch

A Tl Fail-Over (Tl Protection) Switch allows the user to connect a single Tl line
from the telephone company to an "active", as well as to a "standby" Tl terminal,
such as data server/ router etc. at the customer premises.

In the event of the failure of the data server/ router connected to the "A/ active 11
port of the Tl Protection (Fail-Over) Switch, it shall automatically switch and
connect the Tl line from the telephone company to the data server/ router on
the 11 B / standby" port without any customer or user intervention. This ensures
minimum downtime - which would have otherwise occurred due to equipment
failure connected to the 11 A / active" port. This equipment may be used to
enhance the reliability and the efficiency of the customer's data network.

92