TCT_2

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TCT-2 
PCM - TDM Principles 
PCM Principles 
Chapter 1 
Types of signals 
 Information can be in analog form or it can be in digital 
form. 
 
 The telephone instrument generates the information 
(i.e. voice) in the form of analog. 
 
 The computer generates the information (i.e. data) in 
the digital form 
 
 When this information (either analog or digital) is 
transmitted over the medium, then this is called signal. 
 
 Thus the signal can be either Analog or Digital. 
Analog Signal 
 Analog is a continuous electrical signal that varies 
continuously in amplitude and frequency 
 
 Analog signal can have infinite number of amplitude 
values or states within a specified range 
 
 In analogue system, it is difficult to remove noise 
and wave distortions during transmissions. 
 
 For this reason analog signal cannot perform high 
quality data transmission 
Analog Signal 
Digital signal is an electrical signal which posses 
two distinct states, On/off or positive / negative. 
 
Widely used form of digital signals are binary 
signals, in which one amplitude condition 
represents a binary digit 1, and another amplitude 
condition represents a binary digit 0. 
 
Noise and distortions have little effect, making high 
quality data transmission possible 
Digital Signal 
Digital Signal 
Digital Signal 
Multiplexing 
 Multiplex is a technique of transmission of 
information from more than one source to more 
than one destination on the same medium or 
facility. 
Advantages: 
 Many signals can share an existing channel and 
make better use of the channel capacity 
 Allow several different signal to be clustered into a 
single group, for easy handling and maintenance 
Dividing a link into channels 
Types of multiplexing 
Frequency-division multiplexing (FDM) 
FDM is an analog multiplexing technique 
that combines analog signals in 
frequency domain. 
Frequency domain 
 In frequency domain a given bandwidth of 
frequencies is divided into a number of frequency 
slots having a bandwidth of 4 KHz. 
 
 
This is called Frequency Division Multiplexing 
FDM Multiplexing Process 
FDM De-multiplexing Process 
Example of FDM Technique 
TDM is a digital multiplexing technique that 
combines digital signals in time domain. 
 In time domain a given time period is divided 
into a number of time intervals of equal duration 
called time slots 
TDM is compatible with digital signals and makes 
good use of digital circuitry for these signal 
Simplistically, TDM physically switches from 
originator to originator to share the time available, 
and the receiving unit does the same in 
synchronism. 
Time-division multiplexing (TDM) 
Source 3
Multiplexer
Source 1
Source 2 2
3
1
1 2 3 1 2 3 1 2 3 1 2 3 1 2 3 Multiplexer
TIME DOMAIN 
 
TDM Process 
WDM is an analog multiplexing technique that 
combines signals in optical domain. 
 
 In Optical domain a given band of wavelengths is 
divided among a number of information signals with 
suitable spacing. 
 
 Each information signal is assigned a specific 
wavelength 
 
 This is called Wavelength Division Multiplexing 
Wave length Division Multiplexing (WDM) 
Wavelength-division multiplexing (WDM) 
Prisms in WDM 
Basic Principle of (WDM) 
TX-A 
ITU Ch.1 
TX-A 
ITU Ch.2 
TX-A 
ITU Ch.3 
TX-A 
ITU Ch.4 
 RX-A 
ITU Ch.1 
 RX-A 
ITU Ch.2 
RX-A 
ITU Ch.3 
RX-A 
ITU Ch.4 
4 CH 
WD
M 
MUX 
4 CH 
 WDM 
MUX 
Optic fiber 
EDFA 
25 db gain 
(λ1). 
(λ2 ) 
Data in 
(λ3 ) 
(λ4 ) 
 (λ1). 
 (λ2 ) 
Data Out 
 (λ3 ) 
(λ4 ) 
Interleaving 
The process of taking a group of bits or bytes from 
each input line for multiplexing is called 
interleaving. 
We interleave bits or bytes from each input onto 
one output. 
Two types of interleaving methods are adopted 
Bit Interleaving 
Byte Interleaving 
Bit Interleaving is used in PDH systems & Byte 
interleaving is used in SDH systems. 
Interleaving 
Need of digital transmission 
 A digital signal is superior to an analog signal 
because it is more robust to noise and can 
easily be recovered, corrected and amplified 
 
 For this reason, the tendency today is to 
change an analog signal to digital data 
 
 The most common technique to change an 
analog signal into data signal (digitization) is 
called Pulse Code Modulation . 
PCM (Pulse Code Modulation) 
 PCM is the most frequently used analogue-to-digital 
conversion technique. 
 
 Developed by Mr. A .H . Reaves of U.S.A 
 
 It is defined in the ITU-T G.711 specification. 
 
 The main parts of a conversion system are 
 
 The encoder (the analogue-to-digital converter) 
 
 The decoder (the digital-to-analogue converter) 
 
 The combined encoder/decoder is known as a codec 
The PCM generation involves the following steps 
 
Filtering 
 
Sampling 
 
Quantization 
 
Encoding 
 
TDM (Time Division Multiplexing) 
 
Line coding 
PCM Generation 
PCM Generation 
 Filtering is the first step in the PCM process 
 
 
 
 Low pass filters are used to limit the incoming voice signal 
(information) to the frequency band of „0‟ to „4000‟Hz. 
 
 
 
 This band is called as voice band (0 - 4 KHz) 
 
 
 
 This filtering is to avoid aliasing 
Filtering 
Sampling 
 Sampling is the second step in the PCM process 
 
 It is a process of periodically slicing ( sampling) the 
continually changing incoming analog signal to a 
serious of constant amplitude pulses. 
 
 After sampling the generated signal is called PAM 
(pulse amplitude modulated signal) 
 
 Sampling will easier the process of Converting to 
digital (PCM) signal. 
T1 T2
 T3 
time 
T4 T5 T6
 T7 
Incoming audio (analog) 
Signal 
Sampled PAM (analog) 
Output Signal 
T1 T2
 T3 
time 
T4 T5 T6
 T7 
Sampling Representation 
Sampling Signal 
Information Theory shows that full restoration of a 
signal at the reception end can be obtained by 
transmitting the value of the signal sampled at 
regular intervals. 
St 
St 
St 
Signal Sampling Pulses Sampled Signal Recovered Signal 
In the PCM process the sampling (slicing) for incoming 
audio (analog) signal is done to convert into digital 
signal. 
 
The rate at which the sampling to be done,for recovery 
of the original signal is defined by the Nyquist theorem. 
 
As per the Nyquist theorem “If a band limited signal is 
sampled at regular intervals of time and at a rate equal 
to or more than twice the highest signal frequency in the 
band, then the sample contains all the information of 
the original signal” 
Sampling Rate 
As per the Nyquist theorem the sampling frequency is 
two times (twice) the incoming audio signal band. 
fs  2fh, 
Where, fs is the sampling frequency; 
 fh is the highest frequency in the audio band. 
The highest frequency in the incoming analog (audio 
band) is 4KHz, then the sampling frequency is twice of 
this which is equal to 8KHz or 8000 samples per second. 
The time period of sampling or “sampling rate” is 
denoted with Ts 
Ts = 1/sampling frequency = 1/8000 = 125 µ seconds 
Sampling Rate contd………. 
Quantization is the third step in the PCM process 
 
With the sampling the incoming analog signal is 
converted into PAM signal. 
 
This PAM signal is still an analog signal and it will be 
converted into digital form by the process called 
quantization. 
 
Quantization is a process of breaking down the 
incoming discretelevel of sample signal into quantified 
finite number of amplitude values or steps. 
Quantization 
Quantization contd……. 
 A sampled signal (PAM) exists only at discrete times. 
 
 This sampled signal‟s (PAM) amplitude is drawn from 
a continuous range of amplitudes of an analogue 
signal. The discrete value of a sample (PAM) is measured 
by comparing it with a scale, having a finite number 
of intervals. 
 
 These intervals are called the 'quantizing intervals' 
and identifying the interval in which the sample lies. 
Quantization levels 
 In the above signal we have five samples “a, b , c, d, e “ 
To quantize these five samples, the total amplitude may 
be divided into eight ranges or intervals. 
Sample „a‟ lies in the range 5, the quantizing process will 
assign a binary code corresponding to 5, i.e., 101. 
Similarly, codes are assigned for other samples also. 
Here the quantizing intervals are of the same size, 
hence, it is called Linear Quantization. 
Quantizing has to be done for both +ve and –ve swings. 
 
For example, a 0.1 V signal can be divided into 10 mV 
ranges like 0 - 10 mV, 10 - 20 mV, 20 - 30 mV, 30 - 40 mV 
and so on. 
 
The interval 0 - 10 mV may be designated as level 0, 10 - 
20 mV as level 1, 20 - 30 mV as level 2 etc. 
Quantization levels contd……. 
If a sample has an amplitude, say 23 mV or 28 mV, it 
will be assigned the level 2, in either case. 
This is represented in binary code as 1010. When 
these are decoded at the receiving end, the decoder 
will convert them into analogue signals of amplitude 
25mV each. 
Thus, the process of quantization leads to an 
approximation of the input signal with some deviations 
in amplitude. 
These deviations, between the amplitudes of samples, 
of actual value at the “tx” end and the reconstructed 
value at “rx” end gives rise to quantization error. 
Quantization levels contd……. 
Analogue Signal 
amplitude Range 
Quantizing 
Level 
Binary 
Code 
Decoded 
O/P 
Maximum 
Error 
0 – 10 mV 0 1000 5mV  5mV 
10 – 20 mV 1 1001 15 mV  5mV 
20 – 30 mV 2 1010 25 mV  5mV 
30 – 40 mV 3 1011 35 mV  5mV 
40 – 50 mV 4 1100 45 mV  5mV 
 In quantization, the lower value of each interval is assigned to 
a sample falling in that particular interval. 
At the receiving end, the mid value of the interval is assigned, 
while decoding. 
Quantization Error 
Quantization error contd……. 
One way to reduce quantization error is to increase 
the amount of quantization intervals. 
 
 The difference between the input signal amplitude 
height and the quantization interval decreases as 
the quantization intervals are increased (increases 
in the intervals decrease the quantization error). 
 
However, the amount of code words or bandwidth 
also need to be increased in proportion to the 
increase in quantization intervals. 
 There are two types of quantization methods are adopted. 
Linear quantization & Non-liner quantization 
 In linear quantization, equal step size results in equal error 
for all amplitudes. Thus, the signal to noise ratio for 
weaker signals will be poorer in comparison with signal to 
noise ratio for stronger signals. 
 To reduce this error, it is, therefore, it is necessary to 
increase the number of steps in the given amplitude range. 
 This would however, increase the transmission bandwidth 
because bandwidth is B = fh log N 
 Where “N” is the number of quantum steps and “fh” is the 
highest signal frequency. 
Linear-quantization 
 As per the speech statistics, the probability of 
occurrence of small amplitude is much greater than 
that a large one. 
 It is appropriate to provide more quantum levels in 
the small amplitude region and only a fewer 
quantum levels in the region of higher amplitudes. 
 In this case, no increase in transmission bandwidth 
will be required, provided that the total number of 
specified levels remains unchanged. 
 This will also bring about uniformity in signal to noise 
ratio at all levels of input signal. This type of 
quantization is called Non-linear quantization 
Non- linear quantization 
In practice, non linear quantization is achieved using 
segmented quantization. 
 
There are equal number of segments for both 
positive and negative excursions. 
 
In order to specify the location of a sample value it is 
necessary to know three things. 
1. Sign of the sample , 1 bit=positive or negative 
2. Segment number, 3 bits=8 segments 
3. Quantum level within the segment, 4 bits=16 levels 
Non-linear-quantization Contd….. 
As seen from Fig. the first two segments in either polarity 
are collinear, i.e., the slope is the same and hence, they 
may be considered as one segment. Thus, the total 
number of segments appears to be 13 
Non-linear-quantization Contd….. 
 The non linearity introduced at the transmitting end 
by the non-linear quantizing can be neutralized at 
the receiving end by a reverse procedure. 
 
 
 As the non linearity, before the transmission, is 
achieved by 'compressing' the signal, it can be 
neutralized by 'expanding' the received signal. 
 
 
 Hence, the procedure is called “companding” in 
short. 
Companding 
There are two types of companding schemes are used in PCM 
 
-Law Companding (also called log-PCM) 
 This is used in North America and Japan. It uses a 
logarithmic compression curve which is ideal in the sense 
that quantization intervals and hence quantization noise is 
directly proportional to signal level 
 
A- Law Companding 
 This is the ITU-T standard. It is used in Europe and most of 
the rest of the world. It is very similar to the -Law coding. It 
is represented by straight line segments to facilitate digital 
companding. 
Companding 
 Both are linear approximations of a logarithmic input/output 
relationship 
 Both are implemented using 8-bit code words (256 levels, 
one for each quantization interval). This allows for a bit rate 
of 64 kbps 
 Both break the dynamic range into 16 segments (8 positive 
and 8 negative) - each segment is twice the length of the 
preceding one, and uniform quantization is used within each 
segment 
 Both use similar encoding techniques for the 8-bit word - the 
first (most significant bit) identifies polarity, bits 2, 3 and 4 
identify the segment, and the last four bits identify the 
quantization level within the segment 
Similarities between A-law and µ-law: 
 Different linear approximations lead to different lengths 
and slopes 
 Numerical assignment of the bit positions in the 8-bit code 
word to segments and to quantization levels within 
segments are different 
 A-law provides a greater dynamic range 
 µ-law provides better signal/distortion performance for 
low level signals 
 A-law requires 13 bits for a uniform PCM equivalent, 
whereas m-law requires 14 bits 
 International connections should use A-law (µ to A 
conversion is the responsibility of the µ-law country) 
Differences between A-law and µ-law: 
Companding curve 
Encoding is the fourth step in the PCM process 
 
With the quantization (non-linear quantization) the 
incoming PAM samples are assigned with 
quantified levels. 
 
For transmission, these quantified levels are given 
a binary code. This process is called encoding. 
 
 In practical systems, quantizing and encoding 
are a combined process 
Encoding 
P(1 bit) ABC(3 bits) WXYZ(4 bits) 
Polarity bit 
„1‟ for +ve 
„0‟ for –ve 
Segment code Step number in 
the 
segment 
Encoding Contd…. 
 The MSB indicates the sign of the sample. 
 
 Next 3 bits indicate one out of eight segment numbers. 
. 
 Last 4 bits indicate one out of 16 positions in the 
segment. 
 
 A voltage 'Vc' will be encoded as 11110101. 
 
 The quantizing and encoding are done by a circuit 
called coder. This coder converts PAM signals, into an 
8 bit binary code 
Encoding contd …. 
Using encoding each sample is assigned with 8 
bits. 
 
Hence for 8000 samples, the no of bits are 8000 x 
8 bits = 64000 bits or 64 Kbits or 64 Kbps. 
 
 
 After encoding each analog voice channel (4 KHz) 
will be converted into64 Kbps digital channel. 
Encoding Contd…. 
As per CCITT recommendation each sample is 
characterized by means of 8 bits. 
Each voice channel requires a bit rate equal to:8000 samples X 8 bits = 64 Kbps. 
Time between two samples(Ts) 
Sample duration (∆t) 
One bit duration 
: 125 µs. 
: 3.91 µs. 
: 0.488 µs. 
Sample
s 
Ts=125 µs 
 ∆t Bit duration 
0.488µs 
St 
t 
8 bit word 
Encoding Contd…. 
Sample duration 
3.91µs 
TDM is the fifth step in the PCM process. 
 
The encoded output of the PCM is 64Kbps, this is the 
data rate of a single channel. 
 
For better utilization of media bandwidth & to 
have cost effective solution, the number of channels 
are multiplexed using TDM and the multiplexed 
output is send on the media as a frame. 
 
In TDM multiplexing process the available time period 
of 125 µs is divided into number of time slots 
Time Division Multiplexing (TDM) 
Time slot is nothing but a channel,we can send
 either voice or data (64Kbps ) on this channel /time 
slot. 
 
In T1 system 24 time slots are multiplexed as a PCM 
frame 
 
In E1 system 32 time slots are multiplexed as a PCM 
frame 
 
After multiplexing this is called PCM-TDM frame or 
simply PCM frame. 
Time Division Multiplexing (TDM) 
Line Coding is the sixth and last step in the PCM process. 
The multiplexed (PCM-TDM) output, which is in the form of 
PCM frame, to be inter connected to either VF cable pair or 
unbalanced wire for further transmission over the medium. 
On these interconnecting cables the signal is likely to 
 undergo high frequency attenuation, distortion and cross 
talk. 
The signal has a strong dc content and thus prevents the 
use of ac-coupled circuits. 
For distortion free transmission, the PCM-TDM output 
should be converted into a suitable code which will match 
the characteristics of the medium. This coding is called the 
Line Coding. 
Line Coding 
Characteristics of Line Coding 
The total bandwidth of the signal should be as small 
as possible 
 
The energy in the upper part of the signal spectrum 
should be low so that the attenuation distortion is 
low 
 
The energy in the in the lower part of the spectrum 
should also be low to reduce interference from and 
to VF circuits in the same cable 
Characteristics of Line Coding 
No dc component. When the voltage level in a digital 
signal is constant for a while, the spectrum creates 
very low frequencies (results of Fourier analysis) 
 
These frequencies around zero, called DC components, 
present problems for a system that cannot pass 
frequencies below 200Hz. Also a long distance link 
may use one or more transformers to isolate different 
parts of the line electrically 
 
Hence to over come this problem the line coding should 
not have any DC components. So that transformers 
can be used for coupling purposes 
Characteristics of Line Coding 
 Contain adequate timing information to correctly interpret 
the signals received from sender. 
 
 The receiver‟s bit intervals must correspond exactly to the 
sender‟s bit intervals. 
 
 If the receiver clock is faster or slower, the bit intervals are 
not matched and receiver might interpret the signals 
wrongly 
 
 Have an in built error monitoring capability 
 
 Immunity to Noise and Interference 
Types of Line Codings 
Non-return-to-zero (NRZ) - Unipolar 
 
 
Return to Zero (RZ) – Unipolar 
 
 
Alternate Mark Inversion (AMI) – Bipolar 
 
 
High Density Bipolar order of 3 (HDB-3) - Bipolar 
 The output of PCM-TDM is in NRZ (Non-Return to Zero) 
line coded format. 
 This NRZ format cannot be effectively transmitted 
directly on a transmission line because the signal 
contains a DC component and lacks timing information. 
 An additional line coding is necessary, which converts 
this NRZ line code to a pseudo ternary code suitable for 
transmission. 
 There are different types of additional line coding 
schemes such as AMI & HDB-3 are used. 
 These additional line coding schemes eliminate the DC 
component of NRZ code. 
Line Coding 
RZ (Return-to-Zero) Line Coding 
 Return-to-zero (RZ) describes a line code, in which the signal 
drops (returns) to zero between each pulse. 
 This takes place even if a number of consecutive 0's or 1's 
 occur in the signal. The signal is self-clocking, a separate 
clock does not need to be sent along with the signal. 
 But this line coding requires twice the bandwidth to achieve 
the same data-rate as compared to non-return-to-zero 
format. 
NRZ (Non-Return-to-Zero) Line Coding 
Non-Return-to-Zero (NRZ) describes a line code, it is a binary 
 code in which one‟s (1's) are represented by a positive voltage. 
Zero‟s (0's) are represented by a negative voltage), with no 
other neutral or rest condition. 
The pulses have more energy than a RZ code. Unlike RZ, NRZ 
does not have a rest state. NRZ is not inherently a self-
synchronizing code 
This line coding NRZ requires less bandwidth to achieve the 
same data-rate as compared to return-to-zero format 
AMI (Alternate Mark Inversion) Code 
 AMI code was first devised by Barker 
 AMI code is often termed as bipolar signal. 
 In this code, successive marks (bit 1‟s) are alternatively of 
positive and negative polarity and equal in amplitude 
 In this AMI code spaces (bit 0‟s) is of zero amplitude. 
 The disadvantage of the AMI code is the absence 
of significant timing information for long sequence 
of zeros 
 
 The realization of code is also simple. 
 
 Bipolar violation technique is used to detect errors 
in the line signal. 
 
 It is used in 24 channel PCM (T1) system. 
AMI (Alternate Mark Inversion) Code 
To overcome the shortcomings of AMI code, HDB-3 code has 
been devised. It makes a substitution on binary formations 
containing more than 3 zeros. 
This substitution must obey the following rules 
The 4th zero is converted to 1 (mark) with the same polarity 
as immediately preceding mark, thus violation is introduced. 
This bit is known as „V‟ (Violation) bit. 
The „V‟ bit, i.e., placed in place of 4th zero, must be of 
opposite polarity to the previous „V‟ bit. 
1st zero of four consecutive zeros will be made as 'B' bit 
(Bipolar/ Balance bit) 
This „B‟ bit takes a value of 1 (B00V) if the number of 1s‟ 
between two „V‟ bits is even, and takes a value of 0 (000V) if 
the number of 1s‟ between two „V‟ bits is odd. 
HDB-3 (High Density Bipolar of Order 3) 
 In the 1st set of four zeros, the first zero bit i.e. „B‟ bit assumes „1‟ (B00V) if the 
preceding 1s are even or „B‟ bit assumes „0‟ (000V) if the preceding 1s are odd. 
 Every 4th zero is converted to „V‟ bit and assumes same polarity as its proceeding 
mark. 
 From the 2nd set of four zeros, the first zero bit i.e. „B‟ bit assumes „1‟ (B00V) if the 
no. of 1s between two „V‟ bits are even and assumes „0‟ if the no. of 1s between 
two „V‟ bits are odd. 
 „B‟ bit follows AMI and „V‟ bit doesn‟t follow the AMI. 
 As the long sequence of zero is avoided, more timing information is available in 
the signal. 
 Code violation technique is employed to detect errors. 
Steps for conversion of a unipolar 
binary signal into an HDB - 3 code 
Line Coding Techniques 
Information can be in the form of two types of signals 
 
Voice 
Data 
 
 Data Signals are required to be structured into 
eight bit signals. 
 
 No need of filtering, sampling, quantizing & 
encoding 
Types of information Signals 
Digital Transmission systems 
 PDH (1975) 
 Plesio-synchronous Digital Hierarchy 
 
 SDH (1990) 
 Synchronous Digital Hierarchy 
 
 OTH (2000) 
 Optical Transport Hierarchy 
Plesio-synchronous Digital Hierarchy 
Plesiochronous is a Greek word meaning Almost 
Synchronous, but not fully Synchronous.Each PDH circuit has its own (autonomous)clocks 
that are nominally of the same frequency but are 
not locked; resulting an unsynchronized network 
called Plesiochronous. 
 
In PDH technology, ITU-T has recommended two 
systems under G.702 standard. They are 
 E1 System with 30 Voice channels 
 T1 System with 24 Voice channels 
 In E1 / PDH system the information from source to 
destination is transmitted in the form of a Frame with 30 
channels/32 time slots 
 
The duration of the E1 frame is 125µs with 32 time slots 
 
The duration of each time slot is 32 / 125µs = 3.9µs 
 
Each time slot carries 8 bits, the bit duration 3.9 µs /8 = 
488ns 
 
Each E1 frame has 32 time slots and each time slot 
carries 8 bits 
E1 / PDH System 
Total number of bits per frame is 32 x 8 = 256 bits 
 
These 256 bits are transmitted with in the time period of 
125µs. 
 
The bit rate or data rate is always mentioned as bits / 
second (bps) 
 
The number of bits sent per second is 256 x 8000 = 2048000 
bits/sec or bps = 2048 Kbps = 2.048 Mbps 
 
Hence, this 30 channel / 32 time slot PCM / PDH system 
is popularly known as 2.048 Mbps system / E1 system 
E1 / PDH System 
In E1 frame there are 32 time slots are available 
These time slots are numbered as TS0 to TS31 
The TS0 carries the synchronization signal. 
The TS16 carries the signalling information. 
TS1 to TS15 carry speech samples of channel 1 to 15. 
TS17 to TS31 carry speech samples of channels 16 to 30. 
For data channel,the signaling information is not necessary 
Data can be carried from TS1 to TS31 (31 channels) 
E1 Frame 
encoded voice / data 
signals 
encoded voice / data signals 
Signalling 
information 
TS0 TS1 
Synchronization 
Signal 
TS31 TS16 TS15 TS17 
Synchronization in E1 Frame 
 The out put of PCM terminal is continuous stream of bits 
 At the receive end, the receiver has to discriminate 
between frames and channels 
 So, it has to recognize the start of each frame correctly 
 This operation is called frame alignment or 
synchronization and achieved with the help of a fixed 
digital pattern called Frame Alignment word (FAW) 
 The FAW is inserted into the transmitted bit stream at 
regular intervals 
 The receiver looks for FAW and once detected, it knows 
that the next time slot contains the information for 
channel 1 followed by 2 and so on 
The signaling information is transmitted in the form of 
DC pulses 
The signaling levels retain their constant amplitudes 
for much longer periods than the speech 
The signalling is slow varying component
 compared to the speech signal. 
Therefore, a signalling can be digitized with lesser 
number of bits. 
Only 4 bits are enough for representing signalling 
information 
Signalling in E1 Frame 
Each speech channel / time slot should also have its 
corresponding signaling channel / time slot. 
Signaling channel requires only 4 bits and it doesn‟t 
requires all the 8 bits available in time slot. 
In each frame time slot no. 16 is used for sending the 
signaling information pertaining to two voice channels. 
To send the signaling information pertaining to all 30 
voice channels, you require to send 15 frames. 
One extra frame is required to accommodate the 
information pertaining to these frames, hence a total of 16 
frames are send, called as multi frame. 
Signaling in E1 Frame 
In E1 / PDH system the information from source to 
destination is transmitted in the form of multi frame 
 
 
Each multi frame is composed of 16 frames. 
 
The no. of bits in each multi frame are 16 x 256 = 4096 
bits 
 
The duration of a multi-frame is16 x 125µs = 2000µs or 
2ms. 
 
These 16 frames are numbered as F0 to F15 
Multi frame in E1 System 
Each frame is having 32 time slots numbered as TS0 to TS31 
 
The TS0 of F0 carries the synchronization information called as 
Frame Alignment Word (FAW) 
 
The TS16 of F0 carries the Multi Frame Alignment Word (MFAW) 
 
The TS1 to TS15 & TS17 to TS31of F0 will carry 1st samples of 
voice / data pertaining to CH1 to CH30 
 
The TS0 of F1 carries supervisory & alarm signals or NFAW 
 
 The TS16 of F1 carries the signalling information pertaining to 
Channel no.1 & Channel no.16 (Ch N & Ch N+15) 
Multi frame in E1 System 
The TS1 to TS15 & TS17 to TS31 of F1 will carry 2nd samples of 
voice / data pertaining to CH1 to CH30 
 
The TS0 of F2 carries Frame Alignment Word (FAW) 
 
The TS16 of F2 carries the signaling information pertaining to 
Channel no.2 & Channel no.17(Ch N & Ch N+15) 
 
The TS1 to TS15 & TS17 to TS31of F2 will carry 3rd samples of 
voice / data pertaining to CH1 to CH30 
 
TS0 of all even frames carries Frame Alignment Word (FAW) 
 
TS0 of all odd frames carries supervisory & alarm signals or 
NFAW 
Multi frame in E1 System 
TS0 TS1 TS2 TS16 TS30 TS31 
TS0 TS1 TS2 TS16 TS30 TS31 
TS0 TS1 TS2 TS16 TS30 TS31 
F-0 
F-1 
F-2 
TS0 TS1 TS2 TS16 TS30 TS31 
F-15 
. 
. 
. 
. 
. 
. 
. 
Multi frame Structure 
TS17 
TS17 
TS17 
TS17 
TS15 
TS15 
TS15 
TS15 
Frame Alignment word (FAW) 
 FAW is transmitted in the TS0 of all even frames 
 
 This FAW will have 8 bits 
 
 General convention is that if bit value is 1, it is an 
alarming condition & if we are not using any bits , 
these bit value goes to „1‟. 
 
 but if bit value is 0, it is un alarming condition & if we 
are using any bit, these bit value goes to „0‟. 
Frame Alignment word (FAW) 
X 0 0 1 1 0 1 1 
 The FAW bit values are always fixed, they are as 
 The 1st bit (B1) „X‟ is reserved for international use 
and it is normally set to 1, because this PDH mux is 
not globally compatible 
 
 After considering the 1st bit (B1) value as „1‟, the 
FAW becomes 
1 0 0 1 1 0 1 1 
X 1 A Sa4 Sa5 Sa6 Sa7 Sa8 
 Supervisory & Alarm signals or NFAW is transmitted in the 
TS0 of all odd frames 
 This Supervisory & Alarm signals or NFAW will have 8 bits 
 To distinguish between the TS0 of frames carrying 
supervisory & alarm signals or NFAW , from those carrying 
FAW, the 2nd bit (B2) value is fixed as „1‟, 
 
 
 1st bit (B1) : „X‟ reserved for international use. It is 
 normally set to 1 
 2nd bit (B2) : is set to 1 to prevent simulation of the FAW 
 3rd bit (B3) : „A‟ shows the remote alarm indication 
 4th to 8th bits (B4 to B8): „Sa4 to Sa8‟ are additional spare 
 bits 
Supervisory & Alarm Signals or NFAW 
Supervisory & Alarm Signals or NFAW 
1 1 0 0 1 1 1 1 
1 1 1 0 1 1 1 1 
 4th to 8th bits (B4 to B8) or „Sa4 to Sa8‟ : these bits are 
additional spare bits, can be used in specific point to point 
applications within national borders. When these bits are not 
used, these bits value should be set to 1 
 Generally 4th bit (B4) or „sa4‟is used for NMS purpose, this 
 NMS bit is used for operations, maintenance and 
performance monitoring, this 4th bit value becomes „0‟ 
 When there is no alarm, the NFAW bit value is 
 
 
 When there is an alarm is generated, the NFAW bit value is 
Multi Frame Alignment word (MFAW) 
0 0 0 0 X Y X X 
 MFAW is transmitted in the TS16 of F0 
 This MFAW will have 8 bits, Out of this 8 bits, first 4 bits will 
carry MFAW & next 4 bits will carry NMFAW. 
 
 
MFAW NMFAW 
 1st to 4th bits (B1 to B4) :„MFAW , the bit values are fixed & they are 
 normally set to „0‟ 
 5th to 8th bits (B5 to B8) :„NMFAW , the bits of 5th, 7th & 8th are 
reserved ,represented as „X‟ and these bits value are normally set to 
„1‟ 
 6th bit (B6) : used for indicating distant multi frame alarm, 
 represented as „Y‟ and this 6th bit value will be „0‟ if there is no any 
alarm and if there is an alarm, 
Multi Frame Alignment word (MFAW) 
 The bit values of MFAW ( i.e. first 4 bits) is always fixed as 
„0 0 0 0‟ 
 The bit values of NMFAW (i.e.next 4 bits) only will change, 
 when there is no any alarm & no fiber cut the MFAW & 
NMFAW bit values are 
 
 
 
 when there is Remote multi frame alarm (RMA) is 
generated due to fiber cut, the MFAW & NMFAW bit 
values are 
0 0 0 0 1 0 1 1 
0 0 0 0 1 1 1 1 
PDH multi frame, frame, time slot & bit representation 
2.048 mbps/E1 frame (summary) 
Basic Binary rate 
Line coding 
: 2048kbps ±50 ppm 
: HDB3 
Nominal amplitude : 2.37V(for co-axial cable) 
 Impedance 
Frame length 
Available bits/slot 
Multiplexing type 
Frame Rate 
: 3.00V (for balanced cable) 
: 75Ώ (for coaxial cable) 
: 120Ώ (for balanced cable) 
: 256 bits 
: 8 bits 
: Byte Interleaving 
: 8000 frames/sec 
 2Mbps frame is the most commonly used basic frame 
All the European networks support this type of frame. 
ITU-T Standards 
G.703 
G.704 
G.7 11 
G.7 12 
G.714 
G.713 
G.732 
G.735 
G.823 
: Digital interfaces 
: Basic frame structure 
: PCM coding law 
: Characteristics of a speech channel 4-wire 
interface 
: Separate characteristics of a 4-wire interface 
 of the transmit and receive directions 
: Characteristics of a speech channel 2-wire 
interface 
: PCM multiplex equipment 
: PCM multiplex equipment with a facility for a 
synchronous data interface 
: Jitters and Wanders 
Primary MUX Equipment 
Chapter 2 
Primary MUX Equipment 
 The 30 channel PCM mux equipment has been 
designed to convert speech, signaling and data 
information at the transmit end into a digital output 
bit stream of 2048 Kbps. 
 
 At the receiving end all the original information will 
be extracted by proper De multiplexing operations 
from the incoming digital bit stream. 
 
 This system provides the local /trunk exchanges 
with various signaling capabilities for different 
types of exchange equipment 
 The performance of 30 channel PCM multiplexing 
equipment confirms to the ITU (T) Recommendations 
G 703, G 711, G 712 and G 732. 
 
 Nowadays, all Primary MUXes are being used as 
Programmable Drop-Insert MUXes only. 
 
 The same drop-insert can be configured as terminal 
mux by disabling one of the aggregate 2048 Kbps 
links. 
 
 This terminal is generally used at Head Quarter 
station and at last terminal station. 
 
Primary MUX Equipment 
Terminal Multiplexer 
A terminal multiplexer multiplexes all the 30 
Voice / Data circuits into a standard PCM 
signal of 2.048 Mbps 
 
Terminal multiplexer interfaces one direction 
for transmission and reception. 
 
This type of multiplexing system is used at 
the end stations or Head quarter stations. 
 
 
MUX 2.048 Mbps 
Internal bus 
 
User Interfaces 
Terminal Multiplexer 
Drop-Insert Multiplexer 
 
A drop-Insert multiplexer multiplexes all 
the 30 voice/data circuits into two 
standard PCM signals of 2.048 Mbps 
 
Drop-insert interfaces two directions for 
transmission and reception. 
 
This type of multiplexing system is used 
at the intermediate stations or way 
stations. 
 
MUX 2.048 Mbps 2.048 Mbps 
Aggregate Aggregate 
30 analog or 
digital or a 
combination 
of both 
channels 
Drop-Insert Multiplexer 
User 
Interfaces 
Internal bus 
Drop-Insert MUX has two aggregate 2048 kbps links. 
 
 Each link has transmit and receive paths. 
 
 The 30 channels on the channel side can be mapped 
to either of the two aggregate links. 
 
Mapping of channels from one aggregate link to the 
other aggregate link is possible. 
 
 These mapping functions are otherwise known as 
“Cross Connections”. 
Drop-Insert Multiplexer 
PDH HIERARCHY 
Chapter 3 
The first digital multiplexingsystems were 
introduced at the beginning of the 1970s. 
 
The introduction of digital exchanges for 64 Kbps 
channels increased the pressure to bunch 
together great numbers of channels for digital 
transmissions. 
 
Three international multiplex hierarchies arose. 
The bit rates for these hierarchies were 
standardized gradually. 
Digital Multiplexing 
PDH Hierarchy 
In PDH, there are three different hierarchy systems are 
existing over the globe. They are 
 
Japanese PDH System 
 
European PDH System 
 
North American PDH System 
 
These hierarchy systems are meant for accommodating 
more number of voice and data channels for the users. 
2048 kbit/s 
64 kbit/s 
x 
4 
x 
30/31 
x 
24 
x 
3 
x 
7 
x 
5 
x 
3 
Japan USA Europe 
primary rate 
2. order 
3. 
4. 
5. 
32064 kbit/s 
x 
3 
97728 kbit/s 
397200 kbit/s 
x 
4 
139264 kbit/s 
x 
4 
564992 
kbit/s 
x 
4 
34368 kbit/s 
x 4 
8448 kbit/s 
44736 kbit/s 
274176 kbit/s 
x 
6 
1544 kbit/s 
6312 kbit/s 
x 
4 
PDH Systems Worldwide 
PDH Hierarchy 
 In PDH at each hierarchical level, the tributaries are not 
clocked by the standard reference clock. 
 These PDH systems are clocked by their own internal 
clock, hence PDH is not fully synchronous, it is 
plesiochronous. 
 The lower level tributaries with variation in clock are 
multiplexed to form digital stream of next hierarchical 
level. 
 PDH streams are 
E1 2.048 Mbps+/- 50ppm 
E2 8.448 Mbps+/- 30ppm 
E3 34.368 Mbps+/- 20ppm 
E4 139.264 Mbps+/- 15ppm 
 ITU-T recommended two PDH two systems under G.702 
 
 Based on the different first level of hierarchy bit rate they are 
 called E1 system or T1 system 
 
 In E1 system the first level is 2048 Kbps, In T1 system the 
first level is 1544 Kbps 
 
 The Internationally agreed maximum level is 4 for 
 international interconnections 
 
 Levels higher than 4th level are not mentioned in the 
recommendation. 
PDH Hierarchy 
Synchronous Digital Hierarchy (SDH) 
 In SDH at each hierarchical level, synchronous 
transport module is formed with information pay 
load and overhead bits and a synchronizing 
mechanism is in-built with a standard reference 
master clock. 
 
 In this SDH hierarchy the data rate of next stage is 
exact multiple of previous stage data rate. 
 
 Generally SDH transmission is used on OFC links 
and in a very limited way on digital Radios 
 
Optical Transport Hierarchy (OTH) 
 In OTH, optical data units and optical transport units 
are formed as data frames 
 
 These data frames are transported on different 
wavelength of the Wave Division Multiplexing (WDM) 
or Wave length Division Multiplexing on optical fiber. 
 
 Nowadays still more no . of wavelengths are 
multiplexed together called as Dense Wave length 
Division Multiplexing (DWDM) and data frames are 
transported on optical fiber. 
 Higher order PCM systems are designed for the trunk network, 
by assembling primary blocks of 30 channels of 2.048 M b/s in a 
hierarchical fashion similar to analogue groups, subgroups and 
super groups of FDM. 
 
 India adopts European system basing on first level bit stream of 
2048 Kbps , which can go up to fifth level bit stream of 564992 
Kbps & 7680 speech channels. 
 
 As per the ITU-T recommendations under G.702, the fifth level of 
hierarchy is not standardized 
 
 Digital hierarchy recommended by ITU-T under G.702 is up to 
fourth level of bit stream of 139268 Kbps & 1920 channels is 
standardized. 
Higher order PCM Systems used in INDIA 
Higher order PCM Systems used in INDIA 
From other 
1st order 
Mux 
 
From other 
2nd order 
Mux 
 
From other 
3rd order 
Mux 
 
PDH multiplexing from 2nd order onwards involves two basic 
operations irrespective of hierarchical level.They are 
Bit interleaving 
Justification 
 
A digital multiplexer can be considered as parallel to 
serial converter 
 
A digital multiplexer accepts a set of inputs (tributaries) 
applied in parallel and interfaces the inputs in to a single 
output signal having specific time intervals allocated to 
each message serially 
Basics of PDH Multiplexing 
 The multiplexing of severaltributaries can be achieved 
by either bit by bit multiplexing (bit interleaving) or word 
by word multiplexing (byte interleaving) 
 In 30 channel PCM, the signal E1 is formed by byte 
interleaving. 
 But in higher order multiplexing, i.e. forming E2 out of 
4E1s or E3 out of 4E2s or E4 out of 4E3s, multiplexed 
signal is formed by bit-interleaving. 
 In bit-interleaved multiplexing, one bit is taken at a time 
from each tributary to produce a multiplexed signal. 
Interleaving 
 There are four bit streams to be multiplexed. One bit is 
sequentially taken from each tributary so that the resulting 
multiplexed bit stream has every fifth bit coming from the same 
tributary. 
 Four incoming tributaries are shown in fig. (a) and multiplexed 
signal formed by bit-interleaving is shown in fig. (b) 
fig. (a) 
 
 
fig. (b) 
Bit Interleaving 
Bit interleaving 
 In Byte interleaving, one byte (8 bits) is taken 
sequentially taken from each time slot, so that the 
resulting multiplexed stream has every thirty three byte 
coming from the same time slot 
 
 Byte interleaving sets some restraints on the frame 
structure of the tributaries and require great amount of 
memory capacity. 
 
 Bit interleaving is much simpler because it is 
independent of frame structure and also requires less 
memory capacity. 
Byte Interleaving 
Byte Interleaving 
Byte by Byte or Word by Word MULTIPLEXING 
Justification 
 This process is to enable the multiplexer and de-multiplexer to 
 maintain correct operation. 
 In PDH every tributary has its own clock & every tributary is timed 
 with plesiochronous frequency 
 This plesiochronous frequency is a nominal frequency and there 
is always a variation of frequency . For example, the primary 
multiplexer (E1) output is 2.048 Mbps +/- 50ppm. 
 To account for this small variations of the tributary frequencies 
when multiplexing to the next hierarchy level, a process known 
as Justification is used. 
 Justification is of two types 
Positive Justification 
Negative Justification 
 In PDH we use only positive justification, no negative justification 
Positive justification or Pulse stuffing 
Positive justification or Pulse stuffing involves 
intentionally making the output bit rate of a channel 
higher than the input rate. 
The output channel therefore contains all the input 
data plus a variable number of “stuffed bits‟ that are 
not part of the incoming subscriber information. 
The stuffed bits are inserted at the specific locations, 
to pad the input bit stream to the higher output bit 
rate. 
 This stuffed bits must be identified at the receiving 
end so that “de-stuffing” can be done to recover the 
original bit stream. 
Justification 
 In positive justification or stuffing there are two things 
are involved. 
Justification opportunity bits (R bits) 
 These bits are nothing but stuffing bits. These bits are 
available as extra bits that can be used when the rate of 
the incoming tributaries is higher than its nominal value. 
These bits are at Tx side. 
Justification control bits (J bits) 
 In order for the device that receives the multiplexed 
signal to be able to determine whether a justification 
opportunity bit or stuffing bit contains useful 
information. These bits are at Rx side 
Plesiochronous Multiplexing 
Bit Stuffing 
 In PDH hierarchy, level-1 / E1 / 2 Mbps uses certain frame 
 structure and it uses Byte interleaving. 
At higher order levels also the frame begins with Frame 
Alignment Word (FAW), with the difference that, at these levels, 
multiplexing is carried out by bit interleaving. 
 E1 follows byte interleaving and all other higher order levels E2, 
E3 & E4 follows bit interleaving, thus making it impossible to 
identify the lower level frames inside a higher level frame. 
Recovering the tributary frames requires the signal to be de- 
 multiplexed 
 In the higher order tributaries the transmission rate is more than 
4 times that of lower order, to leave room for the recovery of 
justification, FAW & Alarm bits, after de-multiplexing. 
PDH, higher hierarchical levels 
PDH, higher hierarchical levels 
Start with Frame Alignment Word (FAW) . 
Multiplexing is carried out bit by bit . 
The higher hierarchical levels are obtained by 
multiplexing 4 lower level frames . 
Nominal transmission rate is more than 4 times 
that of the lower level. 
Provision for over head bits, in order to leave room 
for the permitted variations in rate (justification 
bits), FAW, alarm and spare bits. 
De-multiplexing for recovering the tributary frames 
2nd order MUX – 8 Mbps 
2nd order digital multiplex and higher order multiplex 
equipments usually working on a non synchronous mode 
even though the tributary signals may be constrained to 
be within the same timing tolerance limits and operate at 
the same nominal bit rate in the plesiochronous net work. 
 
 
2.048 Mbps 
 8.448 Mbps Output 
2.048 Mbps 2nd ORDER MUX 
 
2.048 Mbps 
 
2.048 Mbps 
F
A 
Country code 
Justification 
Control bits Justification 
bits 
100.4μ seconds 
212bits 212bits 
Set-1 Set-2 
212bits 
Set-3 
212bits 
Set-4 
10 
bits 
 
AL 
M 1 
 
 
1 
200bits 
Inform. 
 
4 
208bits 
Inform. 
 
4 
208bits 
Inform. 
 
4 
 
4 
204bits 
Inform. 
Frame Structure of 2nd order MUX - 8.448 
Mbps 
One frame of 848 bits is divided into 4 sets and 
the frame duration is 100.4μ seconds. 
 Stuffing rate is chosen as 64 Kbps per 30 
Channel block. 
 64 Kbps are added on to the primary bit stream 
by the multiplier for justification purposes. 
 Total no of stuffing bits in 120 Channel Mux 
system is 4x64 = 256 Kbps. 
Frame Structure of 2nd 
order MUX - 8.448 Mbps 
Frame Structure of 2nd 
order MUX - 8.448 Mbps 
Hence the clock for this system is = 4 (2048 Kbps + 
64 Kbps ) = (8192 Kbps+256 Kbps) = 8448 Kbps. 
Number of frames for a duration of one second 
= 8448 Kbps/848 = 9962 frames/seconds. 
Master FAW is 1111010000. 
 Justification bit = one per tributary. 
 Justification control bits = equal number for each 
tributary. 
8448Kbps 
2112 Kbps 
2112 Kbps 
2112 Kbps 
2112 Kbps 
2048 Kbps 
2048 Kbps 
2048 Kbps 
2048 Kbps 
+/-102 Hz 
Asynchron
ous PDH 
Inputs 
Synchronous 
Intermediate Inputs 
HigherOrder 
PDHOutput 
MUX 
Positive Justification of 2nd order MUX - 8.448 
Mbps 
2nd order MUX – 8 Mbps 
Nominal bit rate 
Tolerance 
Line code 
Frame length 
Frame rate 
 Bits per TI 
Multiplexing method 
Nominal justification ratio 
: 8448 Kbps 
:30 ppm 
: HDB3 
: 848 bits 
: 9962.264 frames/s 
: 206 bits 
: Bit-by-bit 
: 0.424 
2nd order MUX – 8 Mbps 
2nd order MUX – 8 Mbps 
3rd order MUX – 34 Mbps 
Nominal bit rate 
Tolerance 
Line code 
Frame length 
Frame rate 
Bits per TI 
Multiplexing method 
Nominal justification ratio 
: 34368 Kbps 
: 20 ppm 
: HDB3 
: 15368 bits 
: 22375 frames/s 
: 378 bits 
: Bit-by-bit 
: 0.436 
3rd order MUX – 34 Mbps 
PDH Higher order level 
characteristics 
Jitter & Wander in PDH 
Networks 
Chapter 4 
In any digital communication system error free 
transmission & avoiding cumulative noise-induced 
degradation is desirable. 
 
But in reality this is not possible due to mis-timing 
inside transmission equipment when data is 
regenerated. 
 
When mis-timing becomes large, errors are produced 
and the system can become unusable, even at low 
values of mis-timing sensitivity to amplitude and phase 
variations is increased and performance suffers. 
Jitter & Wander in PDH Networks 
Mis-timingmay be referred to as skew, wander or jitter 
depending on its frequency band 
 
Mis-timing may be the result of pattern dependency or 
due to noise sources such as thermal noise or 
crosstalk 
 
Mis-timing can also be inherent in the system design 
and caused by de-multiplexing (justification) in PDH 
systems or pointer movements in SDH systems 
 
Every PDH system will generate some degree of mis- 
timing and it is not possible to remove it completely. 
Jitter & Wander in PDH Networks 
Jitter & Wander in PDH Networks 
Jitter and wander are defined respectively as the short 
term and long term variations of the significant instant 
of a digital signal from their ideal positions in time 
The Significant instant may be taken as the midpoint 
or any fixed arbitrary point, which is clearly identifiable 
on each of the pulse. 
Jitter is an unwanted variation of one or more 
characteristics of a periodic signal. 
Jitter may be seen in characteristics such as the 
interval between successive pulses, or the amplitude, 
frequency, or phase of successive cycles 
Pictorial Representation Of Jitter And Its Effect On Digital Signal 
Jitter effect on digital signal 
Jitter is represented as a continuous time function with 
properties independent of the digital signal, which it 
affects. 
 
Jitter considered to be most significant occupies the 
frequency range from a few tens of Hz to several KHz. 
 
The unit of jitter is unit interval UI 
 
As per the ITU-T recommendations G.701, the UI is the 
nominal difference in time between the consecutive 
significant instants of an isochronous signal. 
Jitter effect on digital signal 
What is jitter? 
Jitter free 
clock 
(ideal) 
 
 
jittered 
clock 
phase- 
deviation 
time 
What is wander? 
amplitude / dB 
10 Hz 
w a n d e r r a n g e 
MHz 
frequency 
j i t t e r r a n g e 
Hz 
What is wander? 
Wander is also called as low frequency jitter 
Like jitter it can also endanger the error free 
transmission, because it has the property to 
accumulate in networks to higher levels 
Accumulation of wander is higher than the accumulation 
of jitter (PLL circuits). 
The more slower a phase variation is the harder it is to 
 detect. 
Jitter-reducing circuits are included in today's network 
elements, but they don‟t work very well with slow 
wanders. 
Extreme Exactly synchronization signals like PRC are 
 necessary to detect a wander. 
Which problems do jittered 
signals cause in networks? 
misinterpretation of information 
data bits 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 sampling 
point of time 
Jittered signals cause misinterpretation of information in digital 
networks. 
 If the sampling point of time runs out of the maximal 
 tolerable value, the bits will be misinterpreted either by 
sampling one bit twice (7,8) or by leaving one bit out (10,11). 
The consequence can be a immediate loss of the 
 synchronization of all following lower multiplex levels and 
 therefore a complete failure of the following transmission 
network. 
As long as the deviation don‟t trespass the critical level, 
mistakes do not occur. 
The higher the transmission rate is the higher is the jitter- 
sensitivity of a transmission system. 
Therefore higher quality of the transport signal is required. 
Which problems do jittered 
signals cause in networks? 
 Jitter is characterized by two main values: 
 Amplitude: deflection of signal edge deviation(“how far”) 
 Frequency: frequency of signal edge deviation (“how fast”) 
 The jitter unit of measuring the deviation of the signal edge is 
UI (unit interval). 
 The unit interval is a relative measurement unit referring to the 
length of a single bit and is therefore independent of signal 
type and bit rate. 
 This fact is very important to make signals from different 
 hierarchies comparable. 
Unit of Jitter (UI) 
For understanding the unit of jitter (UI), let us consider 
 
The instantaneous jitter amplitude is 1 µs In a 100 KHz 
square wave 
 
The period of frequency = 1 / 100 KHz = 10 µs. 
 
For a timing signal to differentiate between what is mark 
(bit „1‟) and space (bit „0‟) , the Unit Interval between 
the significant instants = 5 µs 
 
Jitter amplitude = 1µs /5 µs = 0.2 UI 
Unit of Jitter (UI) 
 Very low frequency jitter: 
 Inherent instabilities of clock sources. 
 
Noise induced jitter: 
 Phase noise in crystal controlled oscillator
 circuits used in clocks through out the system. 
 
Noise in logic circuits: 
 
 Variations in the propagation delays: 
 
 Slowly changing temperature delays: 
Sources of the jitter in 
transmission network 
 Mapping Jitter caused by justification processes. 
 Insertion and removal of justification bits and
 framing digits. 
 
 Jitter on the regenerated bit streams:Inter symbol 
 interference 
 Regenerator Jitter: Imperfect timing recovery at the 
 regenerators. 
 Pointer Jitter caused by pointer actions. 
 Jitter gain caused by accumulation of Jitter. 
 
 Stuffing and waiting time Jitter caused by stuffing 
techniques. 
Sources of the jitter in 
transmission network 
Thank you