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1.0 INTRODUCTION
A long distance or local telephone conversation between two persons could be
provided by using a pair of open wire lines or underground cable as early as early as
mid of 19th century. However, due to fast industrial development and increased
telephone awareness, demand for trunk and local traffic went on increasing at a rapid rate.
To cater to the increased demand of traffic between two stations or between two
subscribers at the same station we resorted to the use of an increased number of pairs on
either the open wire alignment, or in underground cable. This could solve the problem for
some time only as there is a limit to the number of open wire pairs that can be installed on
one alignment due to headway consideration and maintenance problems. Similarly
increasing the number of open wire pairs that can be installed on one alignment due to
headway consideration and maintenance problems. Similarly increasing the number of
pairs to the underground cable is uneconomical and leads to maintenance
problems.
It, therefore, became imperative to think of new technical innovations which
could exploit the available bandwidth of transmission media such as open wire lines or
underground cables to provide more number of circuits on one pair. The technique used
to provide a number of circuits using a single transmission link is called Multiplexing.
2.0 MULTIPLEXING TECHNIQUES
There are basically two types of multiplexing techniques
i. Frequency Division Multiplexing (FDM)
ii Time Division Multiplexing (TDM)
2.1 Frequency Division Multiplexing Techniques (FDM)
The FDM techniques is the process of translating individual speech circuits (300-
3400 Hz) into pre-assigned frequency slots within the bandwidth of the transmission
medium. The frequency translation is done by amplitude modulation of the audio
frequency with an appropriate carrier frequency. At the output of the modulator a filter
network is connected to select either a lower or an upper side band. Since the intelligence
is carried in either side band, single side band suppressed carrier mode of AM is used.
This results in substantial saving of bandwidth mid also permits the use of low power
amplifiers. Please refer Fig. 1.
FDM techniques usually find their application in analogue transmission systems. An
analogue transmission system is one which is used for transmitting continuously varying
signals.
3.1 FILTERING
Filters are used to limit the speech signal to the frequency band 300-3400 Hz.
3.2 SAMPLING
It is the most basic requirement for TDM. Suppose we have an analogue
signal Fig. 3 (b), which is applied across a resistor R through a switch S as shown in Fig. 3
(a) . Whenever switch S is closed, an output appears across R. The rate at which S is closed
is called the sampling frequency because during the make periods of S, the samples of
the analogue modulating signal appear across R. Fig. 3(d) is a stream of samples of the
input signal which appear across R. The amplitude of the sample is depend upon the
amplitude of the input signal at the instant of sampling. The duration of these sampled
pulses is equal to the duration for which the switch S is closed. Minimum number of
samples are to be sent for any band limited signal to get a good approximation of the
original analogue signal and the same is defined by the sampling Theorem.
0-10 mv 5 mv 0 1000
20-30 mv 25 mv 2 1010
40-50 mv 45 mv 4 1100
Numbers B1 B2 B3 B4 B5 B6 B7 B8
FO X 0 0 1 1 0 1 1 FAW
F1 X 1 Y Y Y 1 1 1 ALARM
F2 X 0 0 1 1 0 1 1 FAW
F3 etc X 1 Y Y Y 1 1 1 ALARM
In frames 1, 3, 5, etc, the bits B3, B4, B5 denote various types of alarms. For
example, in B3 position, if Y = 1, it indicate Frame synchronization alarm. If Y = 1 in
B4, it indicates high error density alarm. When there is no alarm condition, bits B3
B4 B5 are set 0. An urgent alarm is indicated by transmitting "all ones". The code
word for an urgent alarm would be of the form.
X 111 1111
7.0 SIGNALLING IN PCM SYSTEMS
In a telephone network,-the signaling information is used for proper routing
of a call between two subscribers, for providing certain status information like dial
tone, busy tone, ring back. NU tone, metering pulses, trunk offering signal etc. All
these functions are grouped under the general terms "signaling" in PCM
systems. The signaling information can be transmitted in the form of DC pulses (as
in step by step exchange) or multi-frequency pulses (as in cross bar systems) etc.
The signaling pulses retain their amplitude for a much longer period than the
pulses carrying speech information. It means that the signaling information is a
slow varying signal in time compared to the speech signal which is fast changing in
the time domain. Therefore, a signaling channel can be digitized with less number of
bits than a voice channel. In a 30 chl PCM system, time slot Ts 16 in each frame is
allocated for carrying signaling information.
The time slot 16 of each frame carries the signaling data
corresponding to two VF channels only. Therefore, to cater for 30 channels, we
must transmit 15 frames, each having 125 microseconds duration. For carrying
synchronization data for all frames, one additional frame is used. Thus a
group of 16 frames (each of 125 microseconds) is formed to make a "multi-
frame". The duration of a multi-frame is 2 milliseconds. The multi-frame has 16
major time slots of 125 microseconds duration. Each of these (slots) frames has 32
time slots carrying, the encoded samples of all channels plus the signaling and
synchronization data. Each sample has eight bits of duration 0.400 microseconds (3.9/8
= 0.488) each. The relationship between the bit duration frame and multi-frame is
illustrated in Fig. 11 (a) & 11 (b).
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Fig. 2
Encoded TDM (Japanese)
139.264
Fig. 3
Encoded TDM (European)
The term signaling, when used in telephony, refers to the exchange of control
information associated with the establishment of a telephone call on a
telecommunications circuit. An example of this control information is the digits dialed by
the caller, the caller's billing number, and other call-related information.
When the signaling is performed on the same circuit that will ultimately carry the
conversation of the call, it is termed Channel Associated Signaling (CAS). This is the
case for earlier analogue trunks, MF and R2 digital trunks, and DSS1/DASS PBX trunks.
In contrast, SS7 signaling is termed Common Channel Signaling (CCS) in that the
path and facility used by the signaling is separate and distinct from the
telecommunications channels that will ultimately carry the telephone conversation. With
CCS, it becomes possible to exchange signaling without first seizing a facility, leading to
significant savings and performance increases in both signaling and facility usage.
CAS potentially results in lower available bandwidth for the payload. For example, in the
PSTN the use of out-of-band signalling within a fixed bandwidth reduces a 64 kbit/s DS0
to 56 kbit/s. Because of this, and the inherent security benefits of separating the control
lines from the payload, most current telephone systems rely more on Common Channel
Signaling (CCS).
For example, in the public switched telephone network (PSTN) one channel of a
communications link is typically used for the sole purpose of carrying signaling for
establishment and Tear down of telephone calls. The remaining channels are used
entirely for the transmission of voice data. In most cases, a single 64kbit/s channel is
sufficient to handle the call setup and call clear-down traffic for numerous voice and data
channels.
CCS offers the following advantages over CAS, in the context of the PSTN:
The most common CCS signaling methods in use today are Integrated Services Digital
Network (ISDN) and Signaling System 7 (SS7).
ISDN signaling is used primarily on trunks connecting end-user private branch exchange
(PBX) systems to a central office. SS7 is primarily used within the PSTN. The two
signaling methods are very similar since they share a common heritage and in some
cases, the same signaling messages are transmitted in both ISDN and SS7.
CCS is distinct from in-band or out-of-band signaling, which are to the data band what
CCS and CAS are to the channel.
SS7 is a set of telephony signaling protocols which are used to set up most of the world's
public switched telephone network telephone calls. The main purpose is to set up and tear
down telephone calls. Other uses include number translation, prepaid billing mechanisms,
short message service (SMS), and a variety of other mass market services.
It is usually abbreviated as Signaling System No. 7, Signaling System #7, or just SS7. In
North America it is often referred to as CCSS7, an acronym for Common Channel
Signaling System 7. In some European countries, specifically the United Kingdom, it is
sometimes called C7 (CCITT number 7) and is also known as number 7 and CCIS7.
There is only one international SS7 protocol defined by ITU-T in its Q.700-series
recommendations. There are however, many national variants of the SS7 protocols. Most
national variants are based on two widely deployed national variants as standardized by
ANSI and ETSI, which are in turn based on the international protocol defined by ITU-T.
Each national variant has its own unique characteristics. Some national variants with
rather striking characteristics are the China (PRC) and Japan (TTC) national variants.
SS7 is designed to operate in two modes: Associated Mode and Quasi-Associated Mode.
When operating in the Quasi-Associated Mode, SS7 signaling progresses from the
originating switch to the terminating switch, following a path through a separate SS7
signaling network composed of STPs. This mode is more economical for large networks
with lightly loaded signaling links. The Quasi-Associated Mode of signaling is the
predominant choice of modes in North America.
SS7 clearly splits the signaling planes and voice circuits. An SS7 network has to be made
up of SS7-capable equipment from end to end in order to provide its full functionality.
The network is made up of several link types (A, B, C, D, E, and F) and three signaling
nodes - Service switching point (SSPs), signal transfer point (STPs), and Service Control
Point (SCPs). Each node is identified on the network by a number, a point code.
Extended services are provided by a database interface at the SCP level using the SS7
network.
The links between nodes are full-duplex 56, 64, 1,536, or 1,984 kbit/s graded
communications channels. In Europe they are usually one (64 kbit/s) or all (1,984 kbit/s)
timeslots (DS0s) within an E1 facility; in North America one (56 or 64 kbit/s) or all
(1,536 kbit/s) timeslots (DS0As or DS0s) within a T1 facility. One or more signaling
links can be connected to the same two endpoints that together form a signaling link set.
Signaling links are added to link sets to increase the signaling capacity of the link set.
In Europe, SS7 links normally are directly connected between switching exchanges using
F-links. This direct connection is called associated signaling. In North America, SS7
links are normally indirectly connected between switching exchanges using an
intervening network of STPs. This indirect connection is called quasi-associated
signaling. Quasi-associated signaling reduces the number of SS7 links necessary to
interconnect all switching exchanges and SCPs in an SS7 signaling network.
SS7 links at higher signaling capacity (1.536 and 1.984 Mbit/s, simply referred to as the
1.5 Mbit/s and 2.0 Mbit/s rates) are called High Speed Links (HSL) in contrast to the low
speed (56 and 64 kbit/s) links. High Speed Links (HSL) are specified in ITU-T
Recommendation Q.703 for the 1.5 Mbit/s and 2.0 Mbit/s rates, and ANSI Standard
T1.111.3 for the 1.536 Mbit/s rate. There are differences between the specifications for
the 1.5 Mbit/s rate. High Speed Links utilize the entire bandwidth of a T1 (1.536 Mbit/s)
or E1 (1.984 Mbit/s) transmission facility for the transport of SS7 signaling messages.
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