Lecture 2
Lecture 2
Asma Abdurahman
Spring 2025
Lecture 3:
TDM is a technique used for transmitting several analog message signals over a
communication channel by dividing the time frame into slots, one slot for each meassage
signal. The importnant features of TDM are illustrated in the figure below.
Four input signals, all bandlimited to fx by the input filters, are sequentially sampled
at the transmitter by a rotary switch or commutator. The switch makes fs revolutions per
second and extracts one sample from each input during each revolution. The output of the
switch is a PAM or PCM (The most popular types of TDM, PAM and PCM) waveform
containing samples of the input signals periodically interlaced in time. The samples from
adjacent input meassage channels are separated by Ts/M, where M is the number of input
channels. A set of M pulses consisting of one sample from each of the M-input channels is
called a frame.
At the receiver, the samples from individual channels are separated and distributed
by another rotary switch called a distributor or decommutator. The samples from each
channel are filltered to reproduce the original message signal. The rotary switches at the
transmitter and receiver are usually electronic circuits that are carefully synchronized.
Synchronizating is perhaps the most critical aspect of TDM. There are two levles of
synchronization in TDM: frame synchronization and sample (or word) synchronization.
frame synchronization is necessary to estaplish when each group of samples begin and word
synchronization is necessary to properly separate the samples within each frame.
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The interlaced sequence of samples may transmitted by direct PAM or the sample
values may be quantized and transmitted using PCM. TDM-PCM is used in a varity of
applications, the most important one is PCM telephone systems where voice and other
signals are multiplexed and transmitted over a varity of transmission media including pairs
of wires, wave guies, and optical fibers.
• With PCM-TDM system, two or more voice channels are sampled, converted to PCM
codes, and then time-division multiplexed onto a single metallic or optical fiber
cable.
The above figure shows a block diagram for a PCM carrier system comprised of two
DS-0 channels that have been time-division multiplexed. Each channel’s input is alternately
sampled at an 8-kHz rate and converted to an eight-bit PCM code. While the PCM code for
channel-1 is being transmitted, channel-2 is sampled and converted to PCM code. When its
turn for channel-2’s PCM code to be transmitted, the next sample is taken from channel-1
and converted to PCM code. This is a continuous process.
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Sampling Rate:
There are two times to consider for a sampled signal, first, duration of the pulse, and
second, time between samples. It was found that time duration of the sample is not critical
and can be reduced without degrading the information. However, time between samples
must be chosen so that sampling is at least twice the highest frequency in the modulating
signal, in order to receive the modulating signal without distortion since if the sampling rate
is at frequency less than twice the highest, the spectrum of the sampled signal will show
overlapping of side bands and the original modulating signal spectrum.
= (2 x 4000 Hz)
125 𝜇𝑠𝑒𝑐 is time between samples and may be used by other channels of the system.
Bandwidth:
Generally, transmission bandwidth required for digital signal is larger than that
needed for analog signal. As the number of channels increases the transmission bandwidth
increases. This because pulse train spectrum consists of a fundamental ( f0=8000 Hz for
speech) and harmonics. Where all frequencies (fundamental and harmonics) will have side
bands resulting from the modulation process.
The multiplexer is simply an electronically controlled digital switch with two inputs and
one output. One eight-bit PCM code from each channel is called a TDM frame and the time it
takes to transmit one TDM frame is called frame time and it is equal to reciprocal of sample
rate. The figure below shows the frame allocation for a two channel PCM system. The PCM
code for each channel occupies a fixed time slot within the total TDM frame.
When sampling, quantizing and coding take place, a serial bit stream results which
requires some identification to where it begins as a scanning sequence.
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Framing is to identify to the far-end (receiver) when each full sampling sequency starts and
ends.
Frame: a full cycle of samples is called frame in PCM terminology.
Another bit is added to each full sequence of signaling (one cycle) i.e., a sample from
each channel. This is called framing bit.
Therefore, number of bits per frame in 24 channels:
= (7 + 1) x 24 + 1 = 193 bit.
Since sampling is at 8000 Hz = 8000 cycles/sec.
= 8000 frames/sec.
Then bit rate = 8000 X 193
= 1544000 bits/sec
=1.544 Mbps
b) The CEPT (Conference of European Postal and Telecommunications
Administrations) uses 32 channels instead of 24 channels. Where, 30 channels are
for speech, 2 channels: one for signaling and one for synchronization.
Bit rate = 8 X 32 X 8000
= 204800 bits/sec.
= 2.048 Mbps.
In these 32 channels, each frame is divided into time slots (TS) as shown in the
figure:
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!"#
Time slot = $" = 3.9 𝜇 𝑠𝑒𝑐,
Number of bits per slot = 8
$.&
Time per bit = ' = 488 𝑛 𝑠𝑒𝑐.
In TSO (Synchronizing code) it is transmitted every second frame (odd frames)
occupying digits 2 to 8 as follows: 0011011
For even frames (frames without synch. Word), the second bit of TSO is frozen to (1)
so that the synchronizing cannot be imitated. The remaining bite are then used for the
transmission of supervisory signals.
The most common technique for framing is known as added digit framing, in this
scheme, typically, one control bit is added to each TDM frame. An identifiable pattern of bits,
from frame to frame, is used on this "control channel". A typical example is the alternating
bit pattern, 101010 … This is a pattern unlikely to be sustained on a data channel. Thus, to
synchronize, a receiver compares the incoming bits of one frame position to the expected
pattern. If the pattern dose not match, successive bit positions are searched until the pattern
persists over multiple frames. once framing synchronization is established, the receiver
continues to monitor the framing bit channel.
Synchronization in TDM:
Synchronization is vital in TDM. Tl system solved It by adding 1bit/frame, implies
8000 bps for framing with this high bit rate of Synchronization slips then pattern is
scorched and in few milli-seconds can be restored at later time if the channel bank
remains intact (24 ch. together). But if bank were split up into channels, then it
would be advantageous to have Synchronization bit sequence for each channel. This
implies 1 bit/word rather than 1 bit/frame, results in 8000 bps per ch. which is too
much. Their solution was to share the bit (framing-bit) between signaling and
Synchronization as in the CCITT 1.544 Mbps version I of signaling. For large
switched network the problem of Synchronization can be solved by one of two.
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Table below summarized advantages and disadvantages of both FDM and TDM.
A block diagram of TDM for N channels is shown below. Where sampling, quantizing
and coding are shown before the multiplex. The output bit stream can then be multiplexed
again to produce higher bit streams.
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The long-distance carrier system throughout the world was designed to transmit
voice signals over high-capacity transmission links, such as microwave, coaxial cable and
optical fiber.
The increasing demand for efficient communication, particularly for voice traffic, led to the
development of digital hierarchies in telecommunications. These hierarchies provide a standardized
and structured approach to multiplexing digital signals. The core idea is to combine multiple lower
data rate digital signals into a single, higher data rate signal for efficient transmission over long
distances. These higher-rate signals can then be demultiplexed at the receiving end to recover the
original lower-rate signals. By defining specific data rates and multiplexing levels, digital hierarchies
ensure interoperability between different network components and facilitate the scaling of
telecommunications networks to accommodate growing traffic volumes. This structured framework
became essential as telecommunications transitioned from analog to digital systems, allowing for
better management of bandwidth and improved quality of service.
Part of the evolution of these telecommunication networks has been the adaption of
synchronous TDM transmission, structure.
The widely used techniques as:
1- Plesiochronous digital hierarchy (PDH)
2- Synchronous digital hierarchy (SDH)
3- Synchronous optical network (SONET)
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Figure: Plesiochronous digital hierarchies: (a) 1.544 Mbps derived multiplex hierarchy, the 24-channel PDH
TDM ; (b) 2.048 Mbps derived multiplex hierarchy, the 32-channels PDH TDM.
Here's a breakdown of the CEPT hierarchy levels and their frame structure at the primary
level (E1):
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The CEPT hierarchy defines a series of increasing bit rates achieved by multiplexing lower-
level channels. The main levels are:
E1 Frame Structure:
The E1 line is the fundamental building block of the CEPT PDH (Plesiochronous Digital
Hierarchy). Its frame structure is as follows:
• Frame Duration: 125 microseconds (µs). This corresponds to the 8 kHz sampling
rate used for digitizing voice.
• Number of Time Slots: 32 time slots.
• Bits per Time Slot: 8 bits.
• Total Bits per Frame: 32 time slots * 8 bits/time slot = 256 bits.
• Frame Rate: 8000 frames per second (1 / 125 µs).
• Bit Rate: 256 bits/frame * 8000 frames/second = 2,048,000 bits per second (2.048
Mbps).
• Time Slot 0 (TS0): This time slot is primarily used for framing and
synchronization. It carries a specific fixed pattern that allows the receiver to identify
the beginning of each frame and align the time slots correctly. It can also carry alarm
signals and some bits reserved for national use.
• Time Slots 1-15 and 17-31 (TS1-TS15, TS17-TS31): These 30 time slots are
typically used to carry data traffic, which can be digitized voice channels (E0) or
other digital data. Each time slot can carry a 64 kbps channel.
• Time Slot 16 (TS16): This time slot is often used for signaling. It carries information
related to call setup, supervision, and tear-down. However, it can also be used for data
in some configurations.
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• There are also "clear channel E1" or "unchannelized E1" configurations where the
entire 2.048 Mbps bandwidth is used for a single high-speed data stream, and the
framing information in TS0 is still present.
Understanding the E1 frame structure with its 32 time slots, 8 bits per time slot, and
the specific roles of TS0 and often TS16 is crucial for recognizing and working with systems
based on the CEPT hierarchy. Higher levels in the hierarchy (E2, E3, E4, E5) are formed by
multiplexing four lower-level signals together, along with the addition of overhead bits for
synchronization and management.
The Digital Signal (DS) hierarchy is a standardized system used in North America and Japan
to define the data rates and formats for digital communication channels in Time Division
Multiplexing (TDM) systems. It was originally developed by Bell Labs to efficiently transmit
multiple voice signals over digital lines.
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• Frame Structure: A DS-1 frame consists of 193 bits (24 channels * 8 bits/channel +
1 framing bit). These frames are transmitted at a rate of 8,000 frames per second (to
match the sampling rate of DS-0).
• Calculation: (24 channels * 64 kbps/channel) + 8 kbps (framing overhead) = 1536
kbps + 8 kbps = 1544 kbps.
• Significance: DS-1 (often referred to as T1 lines) became a primary standard for
digital transmission of voice and data. It can carry 24 simultaneous voice calls or a
high-speed data connection.
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• Carrier System: T4
• Multiplexing: 168 DS-1 signals (or 28 DS-3 signals) are multiplexed.
• Overhead: Even more overhead is required for management and synchronization at
this high rate.
• Significance: DS-4 represented a very high capacity for its time and was used in
major telecommunications infrastructure.
The DS hierarchy originally extended to even higher levels, such as DS-4A (400.352 Mbps)
and DS-5 (565.148 Mbps), but these saw limited commercial deployment, especially with
the advent of newer technologies like SONET/SDH.
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The "DS" designation refers to the digital signal rate and format, while the "T" designation
(e.g., T1, T3) refers to the physical transmission system or line that carries the DS signal. In
practice, the terms are often used interchangeably (e.g., a DS1 is often called a T1 line).
Figure below shows the frame structure for the 24-channel DS1 system.
DS-1 system which consists of 24 channels and transmit at a rate of 1.544 Mbps
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• All the 24 timeslots within the 125 micro sec timeframe are normally used for
speech channels.
• However, a single bit is contained at the front of the frame before TS1 which is used
to carry the frame alignment pattern. The pattern is sent one-bit at a time over odd
frames – that is dispersed over eight odd frames, rather than bunched into one
frame,
• The frame comprises of 24 eight-bit time slots plus one bit, totalling 193 bits. The
line rate is thus 193 bits per frame, with 8,000 frames/s, giving 1,544 kbit/s – which
is usually written as ‘1.5 Mbit/s’.
• There are two ways that signalling can be conveyed over the 24-ch. system.
o The earlier systems used a method known as ‘bit stealing’ in which the last bit
in each timeslot of every sixth frame was used to carry signalling related to that
channel. This periodic reduction in the size of the PCM word from 8 to 7 bits
introduced a slight, but acceptable, degradation to the quantisation noise.
However, it also prevented the timeslots from being used to carry data at the
full 64 kbit/s (i.e. 8 bits at 8,000 frames/s) and the reduced rate of 56 kbit/s
(i.e. 7 bits at 8,000 frames/s) was the maximum rate that could be supported.
o The recently introduced alternative of common-channel signalling avoids the
need for bit stealing, since the signalling messages are carried in a data stream
carried over the bit 1 at the start of even frames – that is outside of the speech
channels. However, this 4 kbit/s (i.e. 1 bit every other frame) of signalling
capacity was considered too slow and now the DS1 system has been upgraded
to carry common-channel signalling at 64 kbit/s in one of the time slots,
leaving 23 channels for speech.
The following figure shows a DS1/T1 carrier system time division multiplexes PCM encoded
samples from 24 voice band channels for transmission over a single metallic wire pair or a
fiber optic cable.
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The multiplexer has 24 independent inputs and one time-division multiplexed output.
The 24 PCM output signals are sequentially selected and connected through the multiplexer
to the transmission line. To become a T1 carrier, the system has to be line encoded and
placed on special conditioned cables called T1 lines.
𝑏𝑖𝑡𝑠
24 𝑐ℎ𝑎𝑛𝑛𝑒𝑙𝑠 8 𝑏𝑖𝑡𝑠 𝑏𝑖𝑡𝑠 192
𝑓𝑟𝑎𝑚𝑒 8000 𝑓𝑟𝑎𝑚𝑒𝑠
× = 192 => × = 1.536𝑀𝑏𝑝𝑠
𝑓𝑟𝑎𝑚𝑒 𝑐ℎ𝑎𝑛𝑛𝑒𝑙 𝑓𝑟𝑎𝑚𝑒 𝑓𝑟𝑎𝑚𝑒 𝑠𝑒𝑐
Later, an additional bit called the framing bit is added to each frame. The framing bit occurs
once per frame and is recovered at the receiver and its main purpose is to maintain frame
and sample synchronization between TDM transmitter and receiver.
As a result of this extra bit, each frame now contains 193 bits and the line speed for a T1
digital carrier system is 1.544 Mbps. { 193 bits × 8000 frames = 1.544 Mbps}.
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data rate and the number of channels accommodated. However, as one ascends the hierarchy
to higher levels, the Japanese system incorporates certain deviations, reflecting regional
adaptations and potentially distinct capacity planning strategies.3 This initial alignment with
the T-carrier system suggests a possible influence or a parallel evolution in response to
comparable demands in the telecommunications landscape.
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implementation and parameters within Japan. Consulting these specific TTC documents is
essential for a thorough technical understanding of the J-carrier hierarchy.
As the table illustrates, the primary rate (Level 1) is identical in both hierarchies, with J1 and
DS1 operating at 1.544 Mbps and carrying 24 voice channels. The second level also shows
convergence, with J2 and DS2 both at approximately 6.3 Mbps and accommodating 96 voice
channels. However, a significant difference emerges at the third level, where J3 operates at
32.064 Mbps with 480 channels, while DS3 has a higher data rate of 44.736 Mbps and
supports 672 channels. The fourth and fifth levels also show variations in data rates and
channel capacities. Notably, the North American system includes an intermediate level, DS1C
(3.152 Mbps, 48 channels), which is not present in the J-carrier hierarchy.
Historically, both hierarchies evolved within the PDH framework, with the J-carrier system
likely influenced by the earlier development of the T-carrier system in North America by
AT&T's Bell Labs.
The primary rate for the E-carrier system (E1) is 2.048 Mbps, which is higher than J1's 1.544
Mbps, and it accommodates 32 voice channels compared to J1's 24.This difference at the
foundational level propagates through the higher levels of the hierarchy, with the E-carrier
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system generally exhibiting higher data rates and channel capacities at each corresponding
level. The E-carrier system was standardized by CEPT as a refinement and improvement of
the earlier T-carrier technology.
Frame Synchronization
With TDM systems, it is important not only that a frame has to be identified, but also
individual timeslots within the frame be identified. There are several methods used to
establish frame synchronization, including added digit, robbed digit, added channel,
statistical and unique coding. Considerable amount of overhead is added to transmission to
achieve frame synchronization.
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• 2 Mbit/s service signals are multiplexed to 140 Mbit/s for transmission over optical fiber
or radio
• In PDH, different frame is used for transmission and in data layer. Hence multiplexing and
de-multiplexing is very complex.
• The maximum capacity for PDH is 566 Mbps, which is limited in bandwidth.
• Tolerance is allowed in bit rates.
• PDH allows only Point-to-Point configuration.
Every manufacturer has its own standards; PDH also has different multiplexing hierarchies
making it difficult to integrate interconnecting networks together.
CCITT Standards:
a) An international equivalent to Bell system level-1 of bit rate 1.544 Mbps and 24
channels, but not identical to the Bell’s T1-carrier.
b) 30 voice channel level 1 CEPT at 2.084 Mbps.
In its first standard (1.544 Mbps, 24 channels). It differs from Bell system T1-carrier
in the way framing and signaling are arranged for the block of channels (24 channel).
• 193 bit/frame with 8 bits per time slot for speech, and the alignment
bit (framing bit) is the first not the 193th.
Version 1:
With a common signaling channels associated with block of 24 channels. This is
arranged by having odd frames alignment bit in the form 101010… in successive
frames (i.e., frame 1, bit is 1, frame 3 bit is 0, frame 5, bit is 1 and so on).
In even frames of this multi frame the bit is used for signaling as a common signaling
channel at a bit rate of 4000 bps.
Version 2:
Signaling is associated with each channel and here there are 2-signaling bit streams
that are associated with each channel.
• Bit 8 of each channel in frame 6 is reserved for signaling channel A.
• Bit 8 of each channel in frame 12 is reserved for signaling channel B.
Figure below shows, version 2 of signaling in Mutli frame CCITT standard of 24 Ch.
at 1.544 Mbps.
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In 1st scheme CCITT of 1.544 Mbps has a word length of 8 bits at 64,000 bps/ch.
whereas in the second scheme (12 multi- frame) 8 bit but every 6th word in a channel
has only 7 usable bits. For the CCITT 2.048 Mbps, 30 V.C. standard, signaling carrier
is accomplished by having one time slot in frame assigned to signaling. In multi-frame
of 16 frames with 256 bits per frame if results in 500 bps for each channel for
signaling, the overall signaling channel (TS 16) is sub multiplexed to give 4 each 500
bps per Ch.. the 4 each 500 bps came from assigning 4 bits of each word to a channel.
SDH Specifications:
§ It can be interfaced with PDH network via interfacing operating at, 1.5Mb/s, 2Mb/s,
6 Mb/s, 34 Mb/s, 45 Mb/s, and 140 Mb/s.
§ ATM (Asynchronous Transfer Mode).
§ It is based on byte interleaving, i.e. all bits in a byte belong to the same (Ts) channel.
§ Channels are added or dropped from each network at all rates down to 2 Mb/s as
shown in the figure below.
§ To drop 2 Mb/s no need to drop all other levels but it can be dropped directly from
the STM-1.
§ Reconfiguration can be made from a remote network management with no manual
changes.
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The hierarchy levels are designed by the following abreviation STM-N (Synchronous
Transport Module level-N thus:
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The fame can be logically viewed as a matrix of 9 rows of 270XN octates, each with
transmission being one row at a time from left to right and top to bottom.
The first 9XN columns of the frame are devoted to Section Overhead (SOH). The N is the level
of STM (N = 1, 4, 16 ,64).
Information payload, also known as Virtual Container level 4 (VC-4), used to transport low
speed tributary signals. Contains low rate signals and Path Overhead (POH), Location: rows
#1 ~ #9, columns #10 ~ #270.
The frame format of STM-1 : 155.52 Mbps is shown in the figure below.
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Figure: STM-1 frame chopped up into 9 segments, stacked on top of each other as shown
And the total No. of bits in 9 rows of the frame = 270 X8X9 = 19440 bits.
The advantage of SDH multiplexer is that it can accept inserting or droping 2 Mbps, or 1.5
Mbps, or 6 Mbps, or 34 Mbps, or 45 Mbps, or 140 Mbps. On the STM-1 as shown.
To find the payload rate we find the bit rate of the data sent with the over head bits.
• No. of bits per frame= 261x8x9 = 18792 bits in 125 𝜇 𝑠𝑒𝑐 of the frame
Figure (a) below illustrates the relationship between various multiplexing elements of SDH
and shows generic multiplexing structures. Figure (b) illustrates one multiplexing example
for SDH, where there is direct multiplexing from container-1 using AU-3.
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Figure (a): Basic generalized SDH multiplexing structure. (From Figure 1/G.709, ITU-T Rec. G.709.)
Figure (b): SDH multiplexing method directly from container-1 using AU-3. (From, Figure 2-3/ G.708, ITU-T Rec.
G.708.)
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Terms of SDH:
Index of the container (n), n= (11, 12, 2, 3, 4). The rated of the container depends on the
signal which is being transported.
The mapping of a signal in the corresponding container is specified in ITU-T Rec. G.707.
Virtual Container-n (VC-n): A virtual container is the information structure used to support
path layer connection in the SDH. It consists of information payload and POH information
fields organized in a block frame that repeats every 125 μsec or 500 μsec. Alignment
information to identify VC-n frame start is provided by the server network layer. Two types
of virtual container have been identified:
1. Lower-Order Virtual Container-n, VC-n (n = 1, 2). This element comprises a single C-n (n =
1, 2), plus the basic virtual container POH appropriate to that level.
2. Higher-Order Virtual Container-n, to VC-n (n = 3, 4). This element comprises a single C-n (n
= 3, 4), an assembly of tributary unit groups (TUG-2s), or an assembly of TU-3s, together with
virtual container POH appropriate to that level.
So, VC-n: it is made up of C-n and POH. POH: LO Path OverHead. The POH is an additional
transport-capacity designed for the container (for management, supervision and
monitoring). It carries details on e.g. the payload contents.
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TU-n= VC-n + PTR, n= 11, 12, 2, 3, (PTR: indicates the beginning of VC-12)
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From the figure the entry bits for each box are equivalent e.g. 1x34 Mbps = 7x6 Mbps = 21x2
Mbps = 28x1.5 Mbps, and any three of this are equivalent to 140 Mbps, which from STM-1.
Thus one ADM (Add-Drop Multiplexer) can be used to handle 155 Mbps with possibility to
add or drop any of the bit streams without adding any other lower level mixes, as shown
from the figure below.
Where, as for PDH, a drop of 2 Mbps stream requires additional lower level mixers as
shown below.
Thus, the figures below show the line transmission systems for SDH and PDH.
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3. SONET
Signal Hierarchy
The SONET specification defines a hierarchy standardized digital data rates as shown
in the Table below. the lowest level, referred to as STS-1 (Synchronous Transport Signal level
1) or OC-1 (optical carrier level 1), is 51.84 Mb/s. This rate can be used to carry a single T3
or DS-3 signals (i.e 44.736 Mb/s) or 34 Mb/s or a group of lower rate signals such as T1, T2,
E1, 4×E1. Multiple STS-1 signals can be combined to form an STS-N (or OC-N) signal. The
signal is created by interleaving by to from N. STS-1 signals that are mutually synchronized.
from the following table we see the STM-1 corresponds to STS-3.
SONET designation ITU-T designation Data Rate Mbps Payload rate Mbps
STS-1 /oc-1 - 51.84 50.112
STS-3 /oc-3 STM-1 155.52 150.336
STS-9 /oc-9 STM-3 466.56 451.008
STS-12 /oc-12 STM-4 622.08 601.344
STS-18 /oc-18 STM-6 933.12 902.016
STS-24 /oc-24 STM-8 1244.16 1202.688
STS-36 /oc-36 STM-12 1866.24 1804.032
STS-48 /oc-48 STM-16 2488.32 2405.376
NOTE: OC: Optical Carier and STS: Synchronous Transport Signal
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• 4 T1 gives T2 and
• 6 T2 gives T3 and so on.
• Similarly 30 DS0 –this is a European system – gives E1 line.
• So E1, is about 2 Mbps:
• 4 E1 gives E2;
• E3 is a 34 Mbps line.
• And then I suddenly jump right up to this thing called OC3 (Optical Container -3).
This 155 Mbps is 3 of the basic STS-1 rates that we mentioned earlier.
Compatibilty
It is worth noting that the internet working between SDH and SONET systems is
possible at matched bit rates; for example STM-4 and OC-12; so they interoperate. A slight
modification to the overhead is required as they are structured little differently so there will
always be a little something; but anyway that is not very serious. So they do interoperate.
SONET Terms
We will talk about some SONET terms now; for example,
• Envelope: This envelope is the payload. Basically, after all encapsulation, etc., you
remember that finally near the bottom we have this layer-2 and this layer-2 protocol
will encapsulate it and then hand it over to SONET at the lower level, maybe at the
physical level. So whatever this layer-2 hands over to SONET is the payload; the rest
of it are kind of system overheads –
payload + some end-system overhead also goes into this payload.
So these together form what is known as the envelope; this is a SONET term.
• Overhead: Other bits and bytes which are used for management, that means OAM&P
portion, goes as the overhead of SONET.
• Concatenation: Then there is the concept of concatenation; that means, unchannelized
envelope can carry super rate data payloads, for example, ATM, e.g. OC-3c, the method
of concatenation is different from that of T carrier hierarchy.
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STS-1 Frame:
• Each STS frame lasts 125 microseconds. As mentioned, this 125 microseconds time
period, time epoch, is sort of sacred in this whole domain because 125 microseconds
is what is required for a DS0 channel. Remember this is a time division multiplexing,
which means that if you have a 125 microsecond kind of slot, then some of the DS0
bytes can take these bytes. So if you have a 125 microsecond slot, if 1 byte travels in
this frame, then that is enough for 1 DS0 channel. In SONET we have very
sophisticated and very fast equipment; that means this is a time division multiplexing
system; within this 125 microseconds, not only 1 byte can go but a lot of other bytes
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can go. That means a lot of channels can travel together in this 125 microseconds
frame. So each STH frame lasts 125 microseconds; how many bytes are going in there
depends on whether it is STS -1 or STS -2 or STS –N, etc. So 125 microseconds as I
mentioned is 8000 frames/s.
• STS -1 frame has 6480 bits = 810 octetes (bytes). That means in 125 microsecond slot
or frame, we are putting in 810 bytes. Theoretically, that means it can carry 810 DS0
or voice signals; actually it is not 810, it is lesser than that because a number of these
bytes are used for different types of overheads. We have these 810 bytes; the octets
are understood in terms of a table of 9 rows and 90 columns; so let us look at this
figure below. We have a SONET frame or an SDH frame, which has 9 rows; you can
see the 9 rows on this side and then 90 columns, total 90 columns. Out of these 90
columns, 3 columns, these are sort of used for overhead and these 87 columns are
used for payload or for envelope. The envelope contains the payload as well as little
bit of overhead, which we will come to later on. This is how after every 90 bytes, we
come back to again another 3 bytes of this overhead. This is how it is to be understood.
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similarly path overhead would contain something else. But these are required for
these OAM capabilities that we have in SONET.
Overhead Streams
Let us look at these overheads separately;
• Section Overhead (SOH), which defines and identifies frames and monitors section
errors and communication between sections terminating equipment. So these are its
functions: it identifies frames; monitors section errors – if there are errors, it monitors
section errors; and communication between section terminating equipment, maybe
two repeaters or a repeater and multiplexer, and so on.
• Line Overhead (LOH): locates first octet of SPE and monitors line errors and
communication between terminating equipment. We will come back to this locating
of the first octet of synchronous payload envelope (SPE). This is a very interesting
feature and we will talk about this separately. Previously we talking about section
errors; so line errors and communication between terminating equipment, etc., is
taken care of by the line overhead. Apart from that, line overhead contains this
pointer, which points to the first byte of the SPE. And then there is a path overhead;
and as mentioned earlier path overhead is really inside the envelope. Path overhead
verifies connection path; you remember path means from end to end; that means
from the end to end multiplexer is a path. Whether the connection has been
established or not, it monitors path errors, receivers’ status, communication between
path termination equipment, and so on. This is the POH .
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Add/Drop Multiplexers
Just as some of the equipment that we use in SONET, one of the most important of
these is the add/drop multiplexer. They are important because at certain point in the
network, what might happen is that there are some sources which want to send into the
network. They will sort of go so there is this SONET equipment, which is ADM let us say, and
SONET stream is flowing let us say like this. There may be something that wants to upload
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and travel along with this thing. At the same time, this may be the destination location for
some of the other signals which originated elsewhere; they have to be dropped here. So some
signals have to be dropped, some signals have to be added. So this multiplexer can handle
that and that is very important. That is why they are called add drop multiplexers. This
stream is itself of course flowing at a tremendous rate, whatever that rate is.
• SONET/SDH is a synchronous system with the master clock accuracy of 1 in 109,
which you will see is highly accurate. It shows when you come in some kind of CCM
clock somewhere and then there is a protocol for distributing and maintaining this
clock over the entire network.
• Frames are sent byte by byte.
• ADMs can add drop smaller tributaries into the main SONET/SDH stream and it was
mensioned earlier how that is done. Within that frame you can send a lot of bytes; you
can take out some of the bytes and add some of the bytes. That is how you take out
some of the smaller tributaries and add some of the smaller tributaries.
Grooming
Grooming means, we group the traffic in some format. So you want to keep this
group in one particular way; it could be that there is a one group of streams for whom you
want to give higher priority or you want to give higher quality of service Qos. So you have
to group them together. Similarly there may be multiple groups;
• so it enables grouping traffic with similar destination, Qos, etc., which is a part of
grooming.
• It enables multiplexing or extracting streams also – that is also part of grooming.
• Narrow /wider, broad-band and optical crossconnects may be used for grooming.
If you look at figure, you have this narrowband, this SONET layer and optical layer. In the
narrow band, we have this DS0 grooming and then in the DS1 grooming, there is a whide
band and then the broadband DS3 grooming – so the rates are going up, starting from the 64
kbps, it is going up. When you are going up for the STS 48, you are in optical domain; that
means STS-48 is STM-16, so that is a high rate. The point is that, at that rate, most probably,
you are well in the optical domain. Then, finally, you can go to all optical domain; that means
wavelength, waveband, and fiber grooming – there are different levels of grooming,
depending on what you want to do.
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Figure: Grooming
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So you will go with this public infrastructure or public switched tele PSDN network
or whoever is maintaining this communication equipment. Usually telecom people
maintain it in most of the places. Anyway, they have a sort of fiber going from one
place to another, which contains very high-speed links. What you want is your T1 line
should join them, sort of get transported over the distance and then go and feed into
another T1 line at the destination. That is what you want. You want your T1 line to
sort of have a separate sort of existence – just like in a compartment, we have different
passengers. Passengers have their own individual entity but together they are packed
into one compartment and then they travel. Similarly your T1 line is going to ride onto
to this very fast stream and travel to the destination.
• VT 1.5: Most popular quickley access T1 line within the STS-1 frame.
• VT payload (a.k.a VT SPE) How do you find out about the difference? How do you
separate them in the SPE? The point is,
• you require one more level of pointer used to access it.
o You can access a T1 with just a 2-pointer operation, first from the LOH – you
remember, you go to the SPE, just like that. Similarly, you go to the different
tributaries or different zcontainers using just one more level of pointer. This
flexibility was not there earlier;
• So it was very complex to do the same function in DS-3. For example, accessing DS-0
within DS-3 requires full demultiplexing, stacked multiplexing, etc. So you require full
demultiplexing; that is not required in SONET. The point is that the other streams may
go; where in that frame your bytes really are traveling for the stream or for the
container or for the tributary that you are interested; you just extract it, others keep
on traveling as they are. So you do not demultiplex the whole thing and that gives a
great advantage of add/drop multiplexing.
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companies are connected through that and major service providers use this as a means of
transport.
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Types of TDM:
• The two basic forms of TDM are: Synchronous TDM (STDM) and Asynchronous (or)
Statistical TDM (STATDM)
• In synchronous TDM, time slot ‘x’ is assigned to user m alone and cannot be used by
any other user or other device. T-1 and ISDN telephone lines are common examples
of synchronous time division multiplexing. Asynchronous TDM (or statistical TDM
STATDM) networks assign time slots only when they are to be used and delete them
when they are idle. STATDM is used in high density and high traffic applications.
I. Synchronous TDM:
Interleaving:
The interleaving can be at the bit level or in blocks of bytes or larger quantities.
• TDM can be visualized as two fast-rotating switches, one on the multiplexing side
and the other on the demultiplexing side.
• The switches are synchronized and rotate at the same speed, but in opposite
directions.
• On the multiplexing side, as the switch opens in front of a connection, that
connection has the opportunity to send a unit onto the path. This process is called
interleaving.
• On the demultiplexing side, as the switch opens in front of a connection, that
connection has the opportunity to receive a unit from the path.
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For example, a multiplexer with six inputs each of 9.6 kb/s can result a multiplexed
signal that can be transmitted in a single line with a capacity of at least 57.6 kbps.
• The bit interleaving technique is used with synchronous sources and may be used
with asynchronous source.
• The character interleaving technique is used with synchronous sources. Each Time
slot contains one character of data. Typically, the start and stop bits of each
character are eliminated before transmission and reinserted by the received thus
improving effected.
Figure below shows a number of signals mi (t), i =1, N, are multiplexer into the same
transmission medium. The incoming data from each source are briefly buffered. Each
buffer is typically one bit and one character in length. The buffers are scanned
sequentially to form a composite digital signal stream me (t). The scan operation is
sufficiently rapid so that each buffer is emptied before more data can arrive. Thus, the
data rate of mc (t) must at least equal the sum of data rates of mi (t). mc (t) can be
transmitted directly or passed through modem. In either case transmission is
typically synchronous.
At the receiver, the interleaved data are demultiplexed and routed the appropriate
destination buffer.
• Synchronous TDM is called synchronous because the time slots are reassigned to
source and fixed. The time slots for each source are transmitted whether or not the
source has data to send. Synchronous TDM can handle sources of different data rates.
For example, the slowest input device could be assigned one time slot per frame,
while the faster device is assigned more than one time slot per frame.
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Empty Slots:
Synchronous TDM is not as efficient as it could be. If a source does not have data to
send, the corresponding slot in the output frame is empty. Following Figure shows a case in
which one of the input lines has no data to send and one slot in another input line has
discontinuous data.
One problem with TDM is how to handle a disparity in the input data rates. We
assumed that the data rates of all input lines were the same. However, if data rates are not
the same, three strategies, or a combination of them, can be used. We call these three
strategies multilevel multiplexing, multiple-slot allocation, and pulse stuffing.
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c) Pulse Stuffing: Sometimes the bit rates of sources are not multiple integers of
each other. Therefore, neither of the above two techniques can be applied. One
solution is to make the highest input data rate the dominant data rate and then
add dummy bits to the input lines with lower rates. This will increase their
rates. This technique is called pulse stuffing, bit padding, or bit stuffing. The
idea is shown in Following Figure. The input with a data rate of 46 kbps is
pulse-stuffed to increase the rate to 50 kbps. Now multiplexing can take place.
The extra capacity is used by stufQing extra dummy bits or pulses into each
incoming signal until its rate is raised to that of a locally-generated clock
signal.
The stuffed pulses are Inserted at Qixed locations in the multiplexer frame
format so that they may be identiQied and removed at the de multiplexer.
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T= 2Tb
for a given durstion T we want to have n pulses at the output and thus we have (n-a)
pulses at the input where a is No. of stuff bits.
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Then, a = n- T/Tb1
a = n- 1250/139 = n – 8.99
n= T/Tb2 = 1250/125 = 10
a= 10 − 8.99 ≈ 10 − 9 ≈ 1
or we have 1 stuffing pulse after the 9th pulse of the signal ( i.e 1250/139 ≈ 9)
if T = 250 𝜇 𝑠𝑒𝑐
n= T/Tb2 = 250/125 = 2
thus, wither a duration T = 250 𝜇 𝑠𝑒𝑐 there is 0.2 stuff pulse, which means we have 1 stuff
pulse every T/0.2 = 5 T = 5 X 250 = 1250 𝜇 𝑠𝑒𝑐.
Example:
A synchronous TDM is used to multiplex 3 analog sources and 11 digital sources on a single
link. The analog signals are sampled and PCM encoded with 4 bits per sample. Design this
multiplexer. The sources are:
Source 1 analog, 2khz bandwidth
Source 2 analog, 4khz bandwidth
Source 3 analog, 2khz bandwidth
Source 4-11 digital, each 7200bps synchronous.
Sampling frequency of source 1 is 4khz.
Sampling frequency of source 2 is 8khz
Sampling frequency of source 3 is 4khz
These sources can be multiplexed using commentator with rotation rate
4000/sec.
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Other approach is to multiplex the 8 digital signals to obtain a digital data rate of 8X8 kbps
= 64 kbps, then multiplex it with the multiplexed analog signals 0f 64 kbps.
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the former, some slots are empty because the corresponding line does not have data
to send. In the latter, however, no slot is left empty as long as there are data to be
sent by any input line.
• The statistical TDM allocate dynamically time slots on demand and as with
synchronous TDM, the statistical multiplexer has a number of I/O lines on one
side and a higher speed multiplexed line on the other.
• Each I/O line has a buffer associated with it. So there are n I/O lines but only k
(where k < n) time slots are available on the TDM frame for input, the
function of the multiplexer is to scan the Input buffers, collecting data until a
frame is Qilled, as then send the frame.
• On the output, the multiplexer receives a frame and distributes the slots of
data to the appropriate output buffers.
• The data rate on the statistical TDM multiplexed line is less than the sum of the
data rates of the attached devices. This because statistical TDM takes
advantage of the fact that the attached devices are not all transmitting all of the
time.
• With Statistical multiplexing, control bits must be included in the frame. The
following figure shows the overall frame format for a statistical TDM multiplexer.
• The frame includes a beginning flag and ending flag to indicate the start and end of
frame, an address field that indicates the transmitting device, a control field, a
statistical TDM subframe, and a Frame Check Sequence field (FCS), which provides
error detection.
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The above figure shows the frame when only one data source is transmitting. The
transmitting devise is identified in the address field. The data field is variable and this
scheme works well in times of light loads, but inefficient for heavy loads.
The above figure shows a way to improve efficiency by allowing more than one data source
to be included within a single frame.
Addressing:
• Above Figure also shows a major difference between slots in synchronous TDM and
statistical TDM.
• An output slot in synchronous TDM is totally occupied by data; in statistical TDM, a
slot needs to carry data as well as the address of the destination.
• In synchronous TDM, there is no need for addressing; synchronization and
preassigned relationships between the inputs and outputs serve as an address.
• In statistical multiplexing, there is no fixed relationship between the inputs and
outputs because there are no preassigned or reserved slots. We need to include the
address of the receiver inside each slot to show where it is to be delivered. The
addressing in its simplest form can be n bits to define N different output lines with
n=log2N. For example, for eight different output lines, we need a 3-bit address.
Slot Size:
Since a slot carries both data and an address in statistical TDM, the ratio of the data size
to address size must be reasonable to make transmission efficient. For example, it would be
inefficient to send 1 bit per slot as data when the address is 3 bits. This would mean an
overhead of 300 percent. In statistical TDM, a block of data is usually many bytes while the
address is just a few bytes.
No Synchronization Bit:
There is another difference between synchronous and statistical TDM, but this time it is at
the frame level. The frames in statistical TDM need not be synchronized, so we do not need
synchronization bits.
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Bandwidth:
In statistical TDM, the capacity of the link is normally less than the sum of the capacities of
each channel. The designers of statistical TDM define the capacity of the link based on the
statistics of the load for each channel. If on average only x percent of the input slots are
filled, the capacity of the link reflects this. Of course, during peak times, some slots need to
wait.
Performance:
In statistical multiplexers the data rate of its output is less than the sum of data rates
of the inputs. This is anticipated that the average amount of input is less than the capacity of
the multiplexed line. But in general cases may arise where during peak periods the input
exceeds capacity. The solution to this problem is to include buffer in the multiplexer to hold
temporary excess input.
There is a trade-off between the size of the buffer used and the data rate of the line.
We would like to use the smaller possible buffer and smallest possible data rate. But a
reduction in one requires an increase in the other. Note that we are not concerned with the
cost of the buffer, but the more buffering there is the longer the delay. Thus, the trade-off is
really between the system response time and the speed of the multiplexed line.
To examine the mentioned trade off. Consider the following parameters:
N = number of input sources
R = data rate of each source bps
M = effective capacity of multiplexed line (bps) (taking into account the
overhead bits introduced by the multiplexer)
i.e. M represents the maximum rate at which data bits can be transmitted.
𝛼 = mean fraction of time each source is transmitting
0< 𝛼 <1
The compression of the multiplexer is: K=M/NR
which is ratio of multiplexed line capacity to total maximum input.
Thus, for a given data rate M, if K=0.25 means there are 4 times as many devices being
handed as by a synchronous TDM using the same link capacity.
that is:
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N2 = N1/K or K=N1/N2
𝛼<𝐾<1
The value of K=1 corresponds to synchronous TDM, and if 𝐾 < 𝛼 thus the input will exceed
the multiplexer capacity.
The parameter 𝜌 can be defined as the utilization or fraction of total link capacity being
used. Thus,
𝛼𝑁𝑅 𝛼 𝜆
𝜌= = = 𝜆𝑆 =
𝑀 𝐾 𝑀
Where,
𝑏
𝜆 = 𝛼𝑁𝑅 = 𝑎𝑣𝑒𝑟𝑎𝑔𝑒 𝑎𝑟𝑟𝑖𝑣𝑎𝑙 𝑟𝑎𝑡𝑒 𝑖𝑛
𝑠
Let
𝜎( = 𝑠𝑡𝑎𝑛𝑑𝑎𝑟𝑑 𝑑𝑒𝑣𝑖𝑎𝑡𝑖𝑜𝑛 𝑜𝑓 𝑞
𝜌 = 𝜆𝑆
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𝜌"
𝑞= +𝜌
2(1 − 𝜌)
𝑆(2 − 𝜌)
𝑡( =
2(1 − 𝜌)
1 3𝜌" 5𝜌$ 𝜌)
𝜎( = X𝜌 − + −
1−𝜌 2 𝜎 12
𝜆 = 𝛼𝑁𝑅
𝑆 = 1/𝑀
• The average arrival rate 𝜆 b/s is the total input NR times the fraction of time 𝛼 that
each source is transmitting.
• The service time S in seconds is the time it takes to transmit one bit which is 1/M.
• Note that,
𝛼𝑁𝑅 𝛼 𝜆
𝜌 = 𝜆𝑆 = = =
𝑀 𝐾 𝑀
The parameter 𝜌 is the utilization or fraction of total link capacity being used.
• For example, if the capacity M is 50 kb/s and 𝜌 = 0.5, the load on the system is 25
kb/s.
• The parameter q is a measure of the amount of buffer space being used in the
multiplexer.
• 𝑡( is a measure of the average delay encountered by an input source. The figure
below gives mean buffer size versus utilization.
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The above figures assume that data are being transmitted in 1000 bit frames, figure(a),
shows average number of frames that must be buffered as a function of the average
utilization of the multiplexed line. Thus, as utilization of the line increases the buffer
size increases, and consequently the delay increases as shown in Fig. (b). The
utilization is expressed as a percent of the total line capacity.
• If average input load is 5000 b/s and line capacity is 5000 b/s then
utilization is 100%.
• But if the line capacity is 7000 bps, then the utilization is 5000/7000×100.
• From the two Qigures we conclude that utilization above 80% is clearly
undesirable. In this case (i.e., when 𝜌 = 80%) the mean buffer size is 2.4.
• Note that the average buffer size being used depends only on 𝜌 and not
directly on M. For example, consider the following two cases.
Thus proportionally, a smaller amount of buffer space per source is needed for
multiplexers that handle a larger number of sources.
Also Fig (b) shows that the average delay will be small as the link capacity increases
for constant utilization.
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So far we have been considering average queue length, and hence the average
amount of buffer capacity needed. There will be some fixed upper bound on the
buffer size avallable. The varlance of the queue size grows with u@liza@on. Thus
at a higher level of u@liza@on, a larger buffer is needed to hold the backlog. Fig.
3.9.6 shows strong dependence of over flow probability on u@liza@on.
Transmultiplexers:
The interface between FDM and TDM systems can be done by two methods:
A. Use of back analog and digital channel banks with individual voice channels
connected between as shown:
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B. Use of transmultiplexers which directly translates FDM into PCM signals and
vice versa.
The first two configurations have been recognized by the ITU-T Recommendation G793 and
G794 respectively.
The algorithms used in FDM-TDM translation in general are based on the use of digital signal
processing as illustrated below. The FDM signal is digitized and processed such as filtering,
modulator, and amplification is performed on the digital representation of the FDM signal to
produce a PCM signal.
In the reverse direction, the PCM signal is digitally processed to produce a digital version of
a FDM signal. A digital to analog converter is then used to produce the conventional FDM
signal.
Many design approaches have been used, and no single technique is considered standard
practice. The following figure illustrates the block diagram of 60 channel Transmultiplexer.
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