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Wednesday, September 28, 2016

SYNCHRONOUS TIME DIVISION MULTIPLEXING



SYNCHRONOUS TIME DIVISION MULTIPLEXING
Characteristics
Synchronous time-division multiplexing is possible when the achievable data rate
(sometimes, unfortunately, called bandwidth) of the medium exceeds the data rate
of digital signals to be transmitted. Multiple digital signals (or analog signals carrying
digital data) can be carried on a single transmission path by interleaving portions
of each signal in time. The interleaving can be at the bit level or in blocks of
bytes or larger quantities. For example, the multiplexer in Figure 7.2b has six inputs
which might each be, say, 9.6 kbps. A single line with a capacity of at least 57.6 kbps
(plus overhead capacity) could accommodate all six sources.
A generic depiction of a synchronous TDM system is provided in Figure 7.6.
A number of signals [mt(t), i = 1, N] are to be multiplexed onto the same transmission
medium. The signals carry digital data and are generally digital signals. The
incoming data from each source are briefly buffered. Each buffer is typically one bit
or one character in length. The buffers are scanned sequentially to form a composite
digital data stream mc(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 the data rates of the mi(t). The digital signal mc(t) may be
transmitted directly or passed through a modem so that an analog signal is transmitted.
In either case, transmission is typically synchronous.
The transmitted data may have a format something like Figure 7.6b. The data
are organized into frames. Each frame contains a cycle of time slots. In each frame,
one or more slots is dedicated to each data source. The sequence of slots dedicated
to one source, from frame to frame, is called a channel. The slot length equals the
transmitter buffer length, typically a bit or a character.
The character-interleaving technique is used with asynchronous 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 receiver, thus
improving efficiency. The bit-interleaving technique is used with synchronous
sources and may also be used with asynchronous sources. Each time slot contains
just one bit.
At the receiver, the interleaved data are demultiplexed and routed to the appropriate
destination buffer. For each input source mi(t), there is an identical
output source which will receive the input data at the same rate at which it was
generated.
Synchronous TDM is called synchronous not because synchronous transmission
is used, but because the time slots are preassigned to sources and fixed. The
time slots for each source are transmitted whether or not the source has data to
send; this is, of course, also the case with FDM. In both cases, capacity is wasted to
achieve simplicity of implementation. Even when fixed assignment is used, however,
it is possible for a synchronous TDM device to handle sources of different data
rates. For example, the slowest input device could be assigned one slot per cycle,
while faster devices are assigned multiple slots per cycle.
TDM Link Control
The reader will note that the transmitted data stream depicted in Figure 7.6 does not
contain the headers and trailers that we have come to associate with synchronous
transmission. The reason is that the control mechanisms provided by a data link protocol
are not needed. It is instructive to ponder this point, and we do so by considering
two key data link control mechanisms: flow control and error control. It should
be clear that, as far as the multiplexer and demultiplexer (Figure 7.1) are concerned,
flow control is not needed. The data rate on the multiplexed line is fixed, and the
multiplexer and demultiplexer are designed to operate at that rate. But suppose that
one of the individual output lines attaches to a device that is temporarily unable to
accept data? Should the transmission of TDM frames cease? Clearly not, as the
remaining output lines are expecting to receive data at predetermined times. The
solution is for the saturated output device to cause the flow of data from the corresponding
input device to cease. Thus, for a while, the channel in question will carry
empty slots, but the frames as a whole will maintain the same transmission rate.
The reasoning for error control is the same. It would not do to request retransmission
of an entire TDM frame because an error occurs on one channel. The
devices using the other channels do not want a retransmission nor would they know
that a retransmission has been requested by some other device on another channel.
Again, the solution is to apply error control on a per-channel basis.
How are flow control, error control, and other good things to be provided on
a per-channel basis? The answer is simple: Use a data link control protocol such as
HDLC on a per-channel basis. A simplified example is shown in Figure 7.7. We
assume two data sources, each using HDLC. One is transmitting a stream of HDLC
frames containing three octets of data; the other is transmitting HDLC frames containing
four octets of data. For clarity, we assume that character-interleaved multiplexing
is used, although bit interleaving is more typical. Notice what is happening.
The octets of the HDLC frames from the two sources are shuffled together for
transmission over the multiplexed line. The reader may initially be uncomfortable
with this diagram, as the HDLC frames have lost their integrity in some sense. For
example, each frame check sequence (FCS) on the line applies to a disjointed set of
bits. Even the FCS is not in one piece! However, the pieces are reassembled correctly
before they are seen by the device on the other end of the HDLC protocol.
In this sense, the multiplexing/demultiplexing operation is transparent to the
attached stations; to each communicating pair of stations, it appears that they have
a dedicated link.
One refinement is needed in Figure 7.7. Both ends of the line need to be a
combination multiplexer/demultiplexer with a full-duplex line in between. Then
each channel consists of two sets of slots, one traveling in each direction. The individual
devices attached at each end can, in pairs, use HDLC to control their own
channel. The multiplexer/demultiplexers need not be concerned with these matters.
Framing
So we have seen that a link control protocol is not needed to manage the overall
TDM link. There is, however, a basic requirement for framing. Because we are not
providing flag or SYNC characters to bracket TDM frames, some means is needed
to assure frame synchronization. It is clearly important to maintain framing synchronization
because, if the source and destination are out of step, data on all channels
are lost.
Perhaps the most common mechanism 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
does 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. If the pattern breaks down, the receiver
must again enter a framing search mode.
Pulse Stuffing
Perhaps the most difficult problem in the design of a synchronous time-division
multiplexer is that of synchronizing the various data sources. If each source has a
separate clock, any variation among clocks could cause loss of synchronization.
Also, in some cases, the data rates of the input data streams are not related by a simple
rational number. For both these problems, a technique known as pulse stuffing
is an effective remedy. With pulse stuffing, the outgoing data rate of the multiplexer,
excluding framing bits, is higher than the sum of the maximum instantaneous
incoming rates. The extra capacity is used by stuffing 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 fixed locations in the multiplexer
frame format so that they may be identified and removed at the demultiplexer.
Example
An example, from [COUC95], illustrates the use of synchronous TDM to multiplex
digital and analog sources. Consider that there are 11 sources to be multiplexed on
a single link:
Source 1: Analog, 2-kHz bandwidth.
Source 2: Analog, 4-kHz bandwidth.
Source 3: Analog, 2-kHz bandwidth.
Sources 4-11: Digital, 7200 bps synchronous.
As a first step, the analog sources are converted to digital using PCM. Recall
from Lesson 4 that PCM is based on the sampling theorem, which dictates that a
signal be sampled at a rate equal to twice its bandwidth. Thus, the required sampling
rate is 4000 samples per second for sources 1 and 3, and 8000 samples per second
for source 2. These samples, which are analog (PAM), must then be quantized
or digitized. Let us assume that 4 bits are used for each analog sample. For convenience,
these three sources will be multiplexed first, as a unit. At a scan rate of
4 kHz, one PAM sample each is taken from sources 1 and 3, and two PAM samples
are taken from source 2 per scan. These four samples are interleaved and converted
to 4-bit PCM samples. Thus, a total of 16 bits is generated at a rate of 4000 times
per second, for a composite bit rate of 64 kbps.
For the digital sources, pulse stuffing is used to raise each source to a rate of
8 kbps, for an aggregate data rate of 64 kbps. A frame can consist of multiple cycles
of 32 bits, each containing 16 PCM bits and two bits from each of the eight digital
sources. Figure 7.8 depicts the result.
Digital Carrier Systems
The long-distance carrier system provided in the United States and throughout the
world was designed to transmit voice signals over high-capacity transmission links,
such as optical fiber, coaxial cable, and microwave. Part of the evolution of these
telecommunications networks toward digital technology has been the adoption of
synchronous TDM transmission structures. In the United States, AT&T developed
a hierarchy of TDM structures of various capacities; this structure is used in Canada
and Japan as well as in the United States. A similar, but unfortunately not identical,
hierarchy has been adopted internationally under the auspices of ITU-T
(Table 7.3).
The basis of the TDM hierarchy (in North America and Japan) is the DS-1
transmission format (Figure 7.9), which multiplexes 24 channels. Each frame contains
8 bits per channel plus a framing bit for 24 X 8 + 1 = 193 bits. For voice transmission,
the following rules apply. Each channel contains one word of digitized
voice data. The original analog voice signal is digitized using pulse code modulation
(PCM) at a rate of 8000 samples per second. Therefore, each channel slot and,
hence, each frame must repeat 8000 times per second. With a frame length of
193 bits, we have a data rate of 8000 X 193 = 1.544 Mbps. For five of every six
frames, 8-bit PCM samples are used. For every sixth frame, each channel contains
a 7-bit PCM word plus a signaling bit. The signaling bits form a stream for each
voice channel that contains network control and routing information. For example,
control signals are used to establish a connection or to terminate a call.
The same DS-1 format is used to provide digital data service. For compatibility
with voice, the same 1.544-Mbps data rate is used. In this case, 23 channels of
data
are provided. The twenty-fourth channel position is reserved for a special sync
byte, which allows faster and more reliable reframing following a framing error.
Within each channel, seven bits per frame are used for data, with the eighth bit used
to indicate whether the channel, for that frame, contains user data or system control
data. With seven bits per channel, and because each frame is repeated
8000 times per second, a data rate of 56 kbps can be provided per channel. Lower
data rates are provided using a technique known as subrate multiplexing. For this
technique, an additional bit is robbed from each channel to indicate which subrate
multiplexing rate is being provided; this leaves a total capacity per channel of
6 X 8000 = 48 kbps. This capacity is used to multiplex five 9.6-kbps channels, ten
4.8-kbps channels, or twenty 2.4-kbps channels. For example, if channel 2 is used to
provide 9.6-kbps service, then up to five data subchannels share this channel. The
data for each subchannel appear as six bits in channel 2 every fifth frame.
Finally, the DS-1 format can be used to carry a mixture of voice and data
channels. In this case, all 24 channels are utilized; no sync byte is provided.
Above this basic data rate of 1.544 Mbps, higher-level multiplexing is achieved
by interleaving bits from DS-1 inputs. For example, the DS-2 transmission system
combines four DS-1 inputs into a 6.312-Mbps stream. Data from the four sources
are interleaved 12 bits at a time. Note that 1.544 X 4 = 6.176 Mbps. The remaining
capacity is used for framing and control bits.

ISDN User-Network Interface
ISDN enables the user to multiplex traffic from a number of devices on the user's
premises over a single line into an ISDN (Integrated Services Digital Network).
Two interfaces are defined: a basic interface and a primary interface.
Basic ISDN Interface
At the interface between the subscriber and the network terminating equipment,
digital data are exchanged using full-duplex transmission. A separate physical line
is used for the transmission in each direction. The line coding specification for the
interface dictates the use of a pseudoternary coding scheme.' Binary one is represented
by the absence of voltage; binary zero is represented by a positive or negative
pulse of 750 mV +lo%. The data rate is 192 kbps.
The basic access structure consists of two 64-kbps B channels and one 16-kbps
D channel. These channels, which produce a load of 144 kbps, are multiplexed over
a 192-kbps interface at the S or T reference point. The remaining capacity is used
for various framing and synchronization purposes.
The B channel is the basic user channel. It can be used to carry digital data
(e.g., a personal computer connection), PCM-encoded digital voice (e.g., a telephone
connection), or any other traffic that can fit into a 64-kbps channel. At any
given time, a logical connection can be set up separately for each B channel to separate
ISDN destinations. The D channel can be used for a data-transmission connection
at a lower data rate. It is also used to carry control information needed to
set up and terminate the B-channel connections. Transmission on the D channel
consists of a sequence of LAPD frames.
As with any synchronous time-division multiplexed (TDM) scheme, basic
access transmission is structured into repetitive, fixed-length frames. In this case,
each frame is 48 bits long; at 192 kbps, frames must repeat at a rate of one frame
every 250 psec. Figure 7.10 shows the frame structure; the upper frame is transmitted
by the subscriber's terminal equipment (TE) to the network (NT); the lower
frame is transmitted from the NT to the TE.
Each frame of 48 bits includes 16 bits from each of the two B channels and
4 bits from the D channel. The remaining bits have the following interpretation. Let
us first consider the frame structure in the TE-to-NT direction. Each frame begins
with a framing bit (F) that is always transmitted as a positive pulse. This is followed
by a dc balancing bit (L) that is set to a negative pulse to balance the voltage. The
F-L pattern thus acts to synchronize the receiver on the beginning of the frame. The
specification dictates that, following these first two bit positions, the first occurrence
of a zero bit will be encoded as a negative pulse. After that, the pseudoternary rules
are observed. The next eight bits (Bl) are from the first B channel; this is followed
by another dc balancing bit (L). Next comes a bit from the D channel, followed by
its balancing bit. This is followed by the auxiliary framing bit (FA), which is set to
zero unless it is to be used in a multiframe structure. There follows another balancing
bit (L), eight bits (B2) from the second B channel, and another balancing bit (L);
this is followed by bits from the D channel, first B channel, D channel again, second
B channel, and the D channel yet again, with each group of channel bits followed
by a balancing bit.
The frame structure in the NT-to-TE direction is similar to the frame structure
for transmission in the TE-to-NT direction. The following new bits replace some of
the dc balancing bits. The D-channel echo bit (E) is a retransmission by the NT of
the most recently received D bit from the TE; the purpose of this echo is explained
below. The activation bit (A) is used to activate or deactivate a TE, allowing the
device to come on line or, when there is no activity, to be placed in low-powerconsumption
mode. The N bit is normally set to binary one. The N and M bits may
be used for multiframing. The S bit is reserved for other future standardization
requirements.
The E bit in the TE-to-NT direction comes into play to support a contention
resolution function, which is required when multiple TE1 terminals share a single
physical line (i.e., a multipoint line). There are three types of traffic to consider:
B-channel traffic. No additional functionality is needed to control access to
the two B channels, as each channel is dedicated to a particular TE at any
given time.
D-channel traffic. The D channel is available for use by all the subscriber
devices for both control signaling and packet transmission, so the potential for
contention exists. There are two subcases:
a Incoming traffic: The LAPD addressing scheme is sufficient to sort out the
proper destination for each data unit.
0 Outgoing traffic: Access must be regulated so that only one device at a time
transmits. This is the purpose of the contention-resolution algorithm.
The D-channel contention-resolution algorithm has the following elements:
1. When a subscriber device has no LAPD frames to transmit, it transmits a
series of binary ones on the D channel; using the pseudoternary encoding
scheme, this corresponds to the absence of line signal.
2. The NT, on receipt of a D-channel bit, reflects back the binary value as a
D-channel echo bit.
3. When a terminal is ready to transmit an LAPD frame, it listens to the stream
of incoming D-channel echo bits. If it detects a string of 1-bits equal in length
to a threshold value Xi, it may transmit; otherwise, the terminal must assume
that some other terminal is transmitting, and wait.
4. It may happen that several terminals are monitoring the echo stream and
begin to transmit at the same time, causing a collision. To overcome this condition,
a transmitting TE monitors the E bits and compares them to its transmitted
D bits. If a discrepancy is detected, the terminal ceases to transmit and
returns to a listen state.
The electrical characteristics of the interface (i.e., 1-bit = absence of signal)
are such that any user equipment transmitting a 0-bit will override user equipment
transmitting a 1-bit at the same instant. This arrangement ensures that one device
will be guaranteed successful completion of its transmission.
The algorithm includes a primitive priority mechanism based on the threshold
value Xi. Control information is given priority over user data. Within each of these
two priority classes, a station begins at normal priority and then is reduced to lower
priority after a transmission. It remains at the lower priority until all other terminals
have had an opportunity to transmit. The values of Xi are as follows:
Control Information
Normal priority XI = 8
Lower priority XI = 9
User Data
Normal priority X2 = 10
Lower priority X2 = 11
Primary ISDN Interface
The primary interface, like the basic interface, multiplexes multiple channels across
a single transmission medium. In the case of the primary interface, only a point-topoint
configuration is allowed. Typically, the interface supports a digital PBX or
other concentration device controlling multiple TEs and providing a synchronous
TDM facility for access to ISDN. Two data rates are defined for the primary interface:
1.544 Mbps and 2.048 Mbps.
The ISDN interface at 1.544 Mbps is based on the North American DS-1
transmission structure, which is used on the T1 transmission service. Figure 7.11a
illustrates the frame format for this data rate. The bit stream is structured into
repetitive 193-bit frames. Each frame consists of 24 8-bit time slots and a framing
bit, which is used for synchronization and other management purposes. The same
time slot repeated over multiple frames constitutes a channel. At a data rate of
1.544 Mbps, frames repeat at a rate of one every 125 psec, or 8000 frames per second.
Thus, each channel supports 64 kbps. Typically, the transmission structure is
used to support 23 B channels and 1 64-kbps D channel.
The line coding for the 1.544-Mbps interface is AM1 (Alternate Mark Inversion)
using B8ZS.
The ISDN interface at 2.048 Mbps is based on the European transmission
structure of the same data rate. Figure 7.11b illustrates the frame format for this
data rate. The bit stream is structured into repetitive 256-bit frames. Each frame
consists of 32 &bit time slots. The first time slot is used for framing and synchronization
purposes; the remaining 31 time slots support user channels. At a data rate
of 2.048 Mbps, frames repeat at a rate of one every 125 psec, or 8000 frames per second.
Thus, each channel supports 64 kbps. Typically, the transmission structure is
used to support 30 B channels and 1 D channel.
The line coding for the 2.048-Mbps interface is AM1 using HDB3.

SONET/SDH
SONET (Synchronous Optical Network) is an optical transmission interface originally
proposed by BellCore and standardized by ANSI. A compatible version,
referred to as Synchronous Digital Hierarchy (SDH), has been published by ITU-T
in Recommendations G.707, G.708, and G.709.~S ONET is intended to provide a
specification for taking advantage of the high-speed digital transmission capability
of optical fiber.
Signal Hierarchy
The SONET specification defines a hierarchy of standardized digital data rates
(Table 7.4). The lowest level, referred to as STS-1 (Synchronous Transport Signal,
level 1) or OC-1 (Optical Carrier level I ) , ~is 51.84 Mbps. This rate can be used to
carry a single DS-3 signal or a group of lower-rate signals, such as DS1, DSlC, DS2,
plus ITU-T rates (e.g., 2.048 Mbps).
Multiple STS-1 signals can be combined to form an STS-N signal. The signal
is created by interleaving bytes from N STS-1 signals that are mutually synchronized.
For the ITU-T Synchronous Digital Hierarchy, the lowest rate is 155.52 Mbps,
which is designated STM-1. This corresponds to SONET STS-3. The reason for the
discrepancy is that STM-1 is the lowest-rate signal that can accommodate an
ITU-T level 4 signal (139.264 Mbps).
Frame Format
The basic SONET building block is the STS-1 frame, which consists of 810 octets
and is transmitted once every 125 ps, for an overall data rate of 51.84 Mbps (Figure
7.12a). The frame can logically be viewed as a matrix of 9 rows of 90 octets each,
with transmission being one row at a time, from left to right and top to bottom.
The first three columns (3 octets X 9 rows = 27 octets) of the frame are
devoted to overhead octets. Nine octets are devoted to section-related overhead
and 18 octets are devoted to line overhead. Figure 7.13a shows the arrangement of
overhead octets, and Table 7.5 defines the various fields.
The remainder of the frame is payload, which is provided by the path layer.
The payload includes a column of path overhead, which is not necessarily in the first
available column position; the line overhead contains a pointer that indicates where
the path overhead starts. Figure 7.13b shows the arrangement of path overhead
octets, and Table 7.5 defines these.
Figure 7.12b shows the general format for higher-rate frames, using the
ITU-T designation.

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