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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