standard notes tsn unit iii part 2

8/2/2019 Standard Notes Tsn Unit III Part 2

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SIGNALLINGIn a telecommunication network, signalling systems are as essential as switching

systems and transmission systems. They must be compatible with the switching systems asthey must be able to transmit all the signals required to operate the switches. They must

also be compatible with the transmission system in order to reach the exchange that theycontrol. Thus, design of signalling systems is directly influenced by both switching and

transmission requirements.Exchanges usually send signals over the same circuits in the network as the

connections which they control. This is known as channel associated signaling. In SPC, the

need for more signals to be transmitted between exchanges arise. These signals aretransmitted between two processors of two different exchanges over a separate data

channel. This is known as common channel signalling (CCS). Signaling can be classified as

follows:

Inchannel versus common channel signalling:

INCHANNEL COMMON CHANNEL1.

2.

3.

4.

5.

6.

7.

8.

9.

10.

Trunks are held up during signaling

Signal repertoire is limited

Interference between voice and control signals

may occur

Separate signaling equipment is required for

each trunk and hence is expensive.

Signaling is relatively slow

Speech circuit reliability is assured.

It is difficult to change or add signals.

It is difficult to handle signaling during speechperiod.

Reliability of signaling path is not critical.

Possibility of misuse by customers.

Trunks are not required for signalling.

Extensive signalling repertoire is possible

No interference as the two channels are

physically separated.

Only one set of signalling equipment is required

for a whole group of trunk circuits, therefore is

inexpensive.Signalling is significantly fast.

There is no automatic test of the speech circuit.

There is flexibility to change or add signals.

Signals can be handled anytime.

Reliability of signalling path is critical.

Control channel is generally inaccessible to

customers.

Signalling

Inchannel Common channel

DC Low frequency Voice frequency PCM

Inband Outband

Associated Non-associated

Fig. Signalling techniques


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Customer line signalling:

In a local telephone network, loop/disconnect signaling is used for sending

customers call and clear signals to the exchange. Due to maximum permissible lineresistance (because of minimum line current), there is a limit on the maximum length of the

line and area served by the exchange.

When dial telephones are used, customers send address information by decadicpulsing which is received by a relay circuit. However push-button telephones use DTMF

revolutionized customer line signaling.

FDM carrier systems:

Outband signalling:

In Frequency Division Multiplex (FDM) systems, the carriers are placed at intervalsof 4kHz and the baseband is from 300Hz to 3.4kHz. by using channel filters with a sharp

cut-off, it is possible to insert a narrow-band signalling channel above the speech band

(3.4kHz to 4kHz). This is known as outband signalling .

A DC signal on the input lead M at one terminal causes the signal frequency to besent over the transmission channel. This is detected on the other terminal to give a

corresponding DC signal on the output lead E. If the repeater station containing the FDM

channeling equipment is adjacent to the switching equipment, it is simpler for the latter tosend and receive signals over separate E and M wires than to extract them from and re-

insert them into the speech circuit. The E lead always carries signal from the signaling

apparatus to the switching equipment and the M lead carries signals from the switchingequipment to the signalling apparatus. To use outband signalling successfully in a network,

all routes must use FDM systems with built-in outband signalling.


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Inband (VF) signalling:

Signals that are placed in the outband region need all routes to be equipped with

proper outband signalling FDM systems. This problem is solved if the signals transmittedare placed in the baseband of FDM systems. This is known as inband signallingand this

will function over any circuit which provides satisfactory speech transmission. A voice

frequency signaling system is shown in the figure below.

The line is split when the signal tone is transmitted in order to confine it to the link

concerned. Consequently, the tone spills over before the receiver has operated but this

spill-over is ignored because its duration is less than the length of the signals used. Theunity gain buffer amplifier at the receiving end prevents transients produced by electro-

mechanical switching equipment from reaching the VF receiver.

Since the voice frequency signals are used, there occurs a possibility of signalimitation which is undesirable. The following measures are taken to avoid this:

A signal frequency is chosen at which the energy in speech is low (i.e. above 2khz).

The durations of signals are made longer than the period for which the speechfrequency is likely to persist in speech.

Use is made of the fact that the signal frequency is unlikely to be produced in

speech without other frequencies also being present.

In order to make use of the last measure, the receiver contains a signal circuit with aband pass filter to accept the signal frequency and a guard circuit with a band stop filter to

accept all other frequencies and reject the signal frequency. The outputs of both circuits are

rectified and compared. If the output from the signal circuit exceeds that from the guard

circuit, the receiver operates and gives an output signal, and vice-versa.

Switchingequipment

Switching

equipment

Receive line

split

VF receiver

VF receiver

Buffer amplifier

Buffer amplifier

Transmit line

split

Forward

Backward

Outgoing signalterminal

Incoming signalterminal

Transmit line

split Receive

line split

Fourwire

circuit

fs

~

~

fs

Fig. Voice frequency(VF) signalling system


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PCM signalling:In this, the DC signals associated with the audio frequency baseband circuits in each

direction are sampled and the signal samples are transmitted within the frame of PCMchannels. It is therefore unnecessary to use VF signalling.

The 2Mbits system has 32 8bit time slots, but it provides only 30 channels. Time slot

0 is used for frame alignment and time slot 16 is used for signaling, as shown above. The

8bits of channel 16 are shared between the 30 channels by a process of multi-framing. 16successive appearances of channel 16 form a multi-frame of 8bit time slots. The first

contains a multi-frame alignment signal and each of the subsequent 15 time slots contain 4bits for each of the two channels. This enables a large number of signals to be exchangedthan is possible with the DC signaling methods. When PCM signaling is used for common

channel signaling, then multi-framing is not needed.

Inter-register signalling:

For register-controlled exchanges, a register in the originating exchange receives

address information from the calling customer and sends out routing digits. This goes on


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till the terminating exchange is reached. This introduces post-dialing delay which is

minimized using inband multi-frequency signaling systems. This enables an operator to

send address information over a junction to an automatic exchange more rapidly than bydialing.

In inter-register signaling systems, the signal initiates a connection to a register. The

register is released after it has set up a connection through its exchange and sent out routingdigits, therefore it cannot receive answer and clear signals. Consequently line signalingis

required in addition to inter-register signaling.

Eitheren-bloc oroverlap signallingmay be used. In en-bloc signaling the completeaddress information is transferred from one register to the next as a single string of digits.

Thus no signal is sent until the complete address information has been received. In overlap

signaling, digits are sent out as soon as possible enabling signaling to take place

simultaneously on two links.Also link by link or end to end signaling may be employed. In link by link signaling,

information is exchanged only between adjacent registers in a multi-link connection. In end

to end signaling, the originating register controls the setting up of a connection until it

reaches its final destination.

Fig. Link-by-link and End-to-end signalling between registers

Common channel signalling:

In common channel signaling, there is a separate data link between the two

processors in two different exchanges. All signals between these two exchanges are

transmitted via this data link. It gives the following advantages:

Information can be exchanged between the processors much more rapidly than

when channel-associated signaling is used. As a result, a much wider repertoire of signals can be used and this enables more

services to be provided to the customers.

Signals can be added or changed by software modification to provide new services.

There is no longer any need for line signaling equipment on every junction whichresults in a considerable cost saving.


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Since there is no line-signalling, the junctions can be used for calls from B to A in

addition to calls from A to B.

Signals relating to a call could be sent while the call is in progress.

Signals between two processors can be exchanged for functions other than call

processing, for example for maintenance or network management purposes.

For a common channel signal, the reliability needs to be much greater than channel-associated signalling because failure of data link could prevent any calls to be made

between the two exchanges. CCS does not provide an inherently checking facility.

Therefore a separate means of checking the functioning of speech circuits must beemployed.

In multi-exchange network there will be many CCS links between exchanges and

they form a signalling network. In principle, CCS networks can pass through different

routes from the connections which they control and they can pass through severalintermediate nodes in the signaling network. This is called non-associated signalling. here

the messages must include labels containing their destinations.

Switching

network

Switching

network

Processor

Processor

Processor

Processor

Switching

network

Switching

network

Backward

signals

Forward

signals

Exchange AJunction

Exchange B

Fig. Channel associated signalling between central processors

Signalling

link

Exchange AJunction

Exchange B

Fig. Common channel signalling between central processors


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In practice, CCS messages are usually only routed through one intermediate node.

This is known as quasi-associated signalling. The intermediate node is called signal

transfer point (STP). Since CCS signals may be routed via an STP, each message contains

a destination point code and also an originating point code. The transmission bearers usedfor a CCS network are channels in the main transmission bearer network.

CCITT signalling system no.7

This was the first CCS system to be standardized internationally. This was used in

analog networks and it used bit rates of 2.4kbits/s and 4.8kbits/s. it used modems totransmit over analog telephone channels. It used fixed size signal units of 28bits. A later

version for use in digital networks added four padding bits to be compatible with 8bit PCM

time slots. However this has now been replaced by the CCITT signalling system no.7.


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High-level data-link control protocol (HDLC):

Flag Address Control Information Check Flag

1 octet 1 or 2 octets 1 octet variable 2 octets 1 octet

fig. Frame structure for high-level data-link control (HDLC) protocol

The level 2 protocol used in the CCITT no.7 signalling uses the international standard

known as high level data-link control (HDLC). Messages are sent by packets contained

within frames having the format shown above.

The beginning and end of each HDLC message is indicated by a unique combinationof digits(01111110) known as a flag. These sequence of digits can occur in the message

also and must not be interpreted as a flag. This is done by 0 bit insertion and deletion which

is also called stuffing and un-stuffing respectively. When sending digits of a messagebetween two flags, the sending terminal inserts a 0 after every sequence of five consecutive

1s. the receiving terminal deletes this 0.

The opening flag is followed by bit fields for address and control informationfollowed by the data field containing the message information. Between the data and the

closing fields, there is an error-check field, which enables the receiving system to detect if

the frame is erroneous and request re-transmission.

Signal messages are passed from the central processor of the sending exchange to theCCS system. This consists of 3 micro-processor based sub-systems:

1. signalling control subsystem

2. signalling transmission subsystem3. error control subsystem

The signalling control subsystem structures the messages in the appropriate format

and queues them for transmission. Messages are then passed to the signalling termination

subsystem, where complete signal units (SU) are assembled using sequence numbers andcheck bits generated by the error control subsystem. At the receiving terminal, the reverse

sequence is carried out.The system can be modeled as a stock of protocols:

1. Level 1: The physical level

It is the means of sending bit-streams over a physical path. It uses time slot

16 of a 2Mbit/s PCM system or time slot 24 of a 1.5Mbit/s system.2. Level 2: Data-link level:

It performs the function of error control, link initialization, error-rate

monitoring, flow control and de-lineation of messages.3. Level 3: Signalling network level:

It provides functions required for a signaling network. Each node in thenetwork has a signal point code which is a 14 bit address. Every message containsthe point code of the originating and terminating nodes for that message.

4. Level 4: User level:

This must be fully compatible with the level 3 of the model.


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Signal units:

Information that has to be sent in structures into a signal unit (SU) by the signalling

control unit. The SU is based on the HDLC protocol. SUs are of 3 types:1) The message signal unit (MSU): This transfers information supplied by a user port

(level 4) via the signaling network level (level 3).

2) The link-status signal unit(LSSO): This is used for link initialization and errorcontrol

3) The fill-in signal unit(FISU): This is sent to maintain alignment when there is no

signal traffic.

The format of MSU is shown below:

Flag BSN BIB FSN FIB LI Spare SIO SIF Check Flag

8 7 1 7 1 6 2 8 8n 16 8

fig. Message signalling unit

Flag BSN BIB FSN FIB LI Spare SF Check Flag

8 7 1 7 1 6 2 8 or16 16 8

fig. Link status signalling unit

Flag BSN BIB FSN FIB LI Spare Check Flag

8 7 1 7 1 6 2 16 8

fig. Fill-in unit

SF: status field BIB: backward indicator bit

SIF: signaling information field BSN: backward sequence numberSIO: service information octet FIB: forward indicator bit

LI: length indicator FSN: forward sequence number

Fig. Format of signal units in CCITT no.7 signalling system

Messages are of variable length and are sent in 8-bytes as follows:

1) Opening and closing flags are used to delimit signals. They have the code pattern01111110.

2) The forward indicator bit (FIB), backward indicator bit (BIB), forward sequence

number (FSN) and backward sequence number (BSN) are used for error correction.3) The length indicator (LI) gives the length of the SU. Value of LI greater than 2

indicates that the SU is a message signal unit.

4) The service information octet (SIO) indicates the user port appropriate to themessage.

5) The signalling information field (SIF) may consist of upto 272 octets and contains

the information to be transmitted.

6) The error-check field is immediately before the closing flag. It contains 16 bitsgenerated as a cyclic redundancy check code.

Traffic.


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In telecommunication system, traffic is defined as the occupancyof the server in the network. There are two types of traffic viz. voicetraffic and data traffic. For voice traffic, the calling rate is defined as thenumber of calls per traffic path during the busy hour. In a day, the 60minutes interval in which the traffic is highest is called busy hour (BH).

Grade of Service. In telephone field, the so called busy hour traffic are used forplanning purposes. Once the statistical properties of the traffic areknown, the objective for the performance of a switching system shouldbe stated. This is done by specifying a grade of service (GOS). GOS is ameasure of congestion expressed as the probability that a call will beblocked or delayed. Thus when dealing with GOS in traffic engineering,the clear understanding of blocking criteria, delay criteria andcongestion are essential.

Blocking criteria.

If the design of a system is based on the fraction of calls blocked(the blocking probaility), then the system is said to be engineered on ablocking basis or call loss basis. Blocking can occur if all devices areoccupied when a demand of service is initiated. Blocking criteria areoften used for the dimensioning of switching networks and interofficetrunk groups. For a system designed on a loss basis, a suitable GOS isthe percentage of calls which are lost because no equipment is availableat the instant of call request.

Delay criteria.If the design of a system is based on the fraction of calls delayedlonger than a specified length of time (the delay probability), the systemis said to be a waiting system or engineered on a delay basis. Delaycriteria are used in telephone systems for the dimensioning of registers.In waiting system, a GOS objective could be either the percentage ofcalls which are delayed or the percentage which are delayed more thana certain length of time.

Congestion.It is the condition in a switching center when a subscriber can not obtain

a connection to the wanted subscriber immediately. In a circuitswitching system, there will be a period of congestion during which nonew calls can be accepted. There are two ways of specifying congestion.

1. Time congestion.It is the probability that all servers are busy. It is also called the

probability ofblocking.


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2. Call congestion.It is the proportion of calls arising that do not find a free server.

Call congestion is a loss system and also known as the probability of losswhile in a delay system it is referred to as the probability of waiting. If

the number of sources is equal to the number of servers, the timecongestion is finite, but the call congestion is zero. When the number ofsources is large in comparison with servers, the probability of a new callarising is independent of the number already in progress and thereforethe call congestion is equal to the time congestion. In general, time andcall congestions are different but in most practial cases, thediscrepancies are small.

3.Measure of GOS.GOS is expressed as a probability. The GOS of 2% (0.02) mean

that 98% of the calls will reach a called instrument if it is free. Generally,

GOS is quoted as P.02 or simply P02 to represent a network busyprobability of 0.02. GOS is applied to a terminal-toterminal connection.For the system connection many switching centers, the system isgenerally broken into following components.

(i) an internal call (calling subscriber to switching office)(ii) an outgoing call to the trunk network (switching office to trunk)(iii) The trunk network (trunk to trunk)(iv) A terminating call (switching office to called subscriber)

The GOS of each component is called component GOS.

The GOS for internal calls is 3 to 5%, for trunk calls 1-3%, foroutgoing calls 2% and for terminating calls 2%. The overall GOS of asystem is approximately the sum of the component grade of service. Inpractice, in order to ensure that the GOS does not deterioratedisastrously if the actual busy hour traffic exceeds the mean, GOS arespecified 10% or 20% more of the mean.

TELECOMMUNICATIONS TRAFFIC

In case of telecommunication systems, it is required to design the

system in accordance with the number of calls that are in progress at any

point of time and the total number of subscribers that are connected to the

network. Teletraffic engineering involves the design of the number of


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switching equipment required and the number of transmission lines required

for carrying telephone calls.

In teletraffic engineering the term trunkis used to describe any entity

that will carry one call. The arrangement of trunks and switches within an

exchange is called its trunking.

We can check the number of calls in progress at different intervals of

time for a whole day and then tabulate the results. If a graph is plotted taking

the number of calls in progress on y-axis and the time of the day on x-axis

the graph would look like this:-

fig 1.

The maximum number of calls occurs between 8:00 and 10:00 am for

this particular exchange. This hour which corresponds to the peak traffic of

the exchange is called the busy hour.

The busy hour varies for different exchanges and the teletraffic curve

also varies for different exchanges from what is shown in the fig 1.

Exchanges in which offices and business establishments predominate

usually have a busy hour between 10:00 and 11:00 am. Residential

exchanges have a busy hour normally between 4:00 and 5:00 pm.

The limit of traffic:The teletraffic intensity or simply the traffic is defined as the average

no. of calls in progress. The unit of traffic is erlang (named after the Danish

pioneer in teletraffic A.K.Erlang).It is a dimensionless quantity.

On a group of trunks, the average number of calls in progress

depends on both the no. of calls which arrive and their duration. The


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duration of a call is called its holding time because it holds the trunk for that

time.

Consider a holding time T for a group of 3 trunks:

Fig. 2Example of 1 erlang of traffic carried on 3 trunks

Figure 2(a) shows 1 erlang of traffic resulting from one truck being busy for

the holding time T. Figure 2(b) shows 1 erlang of traffic resulting from two

trunks with each trunk being busy for 50% of the time T. Figure 2(c) shows

1 erlang of traffic being carried by three trunks with each of the trunks being

busy for 33.33% of the time T.

Sometimes the traffic is also expressed in terms of hundreds of call

seconds per hour ( CCS).

1 erlang = 36 CCS

Mathematically traffic can be represented by the following equation:

A=Ch/T (1) where A=traffic in erlangs

C=average number of callsarriving during time T

h=average holding time

From eqn 1, if T=h, A=C. Thus traffic in erlangs can be defined as

mean number of calls arriving during a period equal to the mean duration of

the calls (average holding time).

1

2

3

1

2

3

1

2

3a)>..

b)

c)


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A single trunk cannot carry more than one call, therefore A


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It is essential to keep a record of the traffic that is offered to a

telephone exchange in order to upgrade the system capacity as and when

required.

Initially the number of calls used to be measured manually. Later

automatic traffic recorders were installed in automatic exchanges. In modern

SPC systems, a separate sub-program keeps count of the traffic generated.

Mathematical model:

A mathematical model needs to be developed in order to study

telecommunications traffic. Such a model is based on two assumptions:

a) pure chance traffic

b) statistical equilibrium

a) The assumption of pure chance traffic means that call arrivals and call

terminations are independent random events. It also implies that thenumber of sources generating calls is very large. Since call arrivals are

independent random events, the occurrence of calls is not affected by

previous calls, therefore traffic is sometimes called memoryless

traffic.

The number of call arrivals in a given time T has a poissonian

distribution given by,

P(n) = x/x!.e- (3) where x is the number arrivals in time T

is the mean number of call arrivals in time T

i. The intervals between calls arrivals are intervals between

independent random events and these intervals have a negative

exponential distribution,

P(x>=t) = e -t/ (4)

where is the mean interval between call arrivals

ii. The call durations, T are intervals between independent random

events (call termination). Therefore the call durations also have

a negative exponential distribution.

P(T>=t) = e-t/h (5) where h is the average holding time

b) Statistical equilibrium means that the generation of traffic is a

stationary random process i.e. the probabilities do not change for the


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period being considered. Consequently the mean number of calls in

progress remains constant.

The state transition diagram is shown for a group of N trunks. The

total number of states that N trunks can have is N+1. The number of calls in

progress varies randomly and lies between 0 and N. the state transition

diagram shown is called a simple Markov chain. The probabilities P(0),P(1), . are called the state probabilities and the P j,k , P k,i are called

transition probabilities of the Markov chain. In case of statistical equilibrium

these probabilities will have a fixed value and they will not change.

Using Markov chains we can proceed to prove that if call arrivals has a

Poissonian distribution, then the calls in progress will also have a Poissonian

distribution:IntroductionThe specification of SS7 started during 1970s when it started to become clear that the future

switching systems would be program controlled, that their switching fabrics would also be

implemented in silicon and closer to the end of the decade that even subscriber access in thoseswitching systems would be digital.

AT&T, RBOCs, BT, other European and Japanese operators played key roles in specifying

SS7. They sought to remove the limitations of analogue signaling systems that were in use inelectromechanical switching systems in their networks. The limitation included: a limited set of

signals, difficulty of signaling after call setup, difficulty of signaling independent of the voicecircuit, high level of use of network capacity because of slow signaling and dedicated signaling

resources


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Specification of SS7 took place in parallel and even earlier than the work on OSI in theInternational Standardization Organization. The same principle of hierarchical layers as in OSI

was used in both. However, the purpose and the design goals of the two protocol sets aredifferent and as a result there are significant differences in the allocation of functions on

different layers in OSI and levels in SS7

There are two variants of SS7. One for ETSI (European Telecommunications Standards

Institute) and the other for ANSI markets. The latter is mainly in use in the Americas, SouthKorea and Japan. This presentation is mainly based on ETSI specifications and makes only

marginal references to ANSI.

Design considerationsThe solution to the limitations of analogue signaling was a fully digital,message based system

that could carry all kinds of information needed for network services. Moreover, to make

signaling independent of voice circuits it was obvious that an out-of-band or common channel

signaling was needed.Common channel refers to the idea that one timeslot allocated forsignaling will carry all signaling needed for many voice slots and even for many PCM

lines. Let us verify this idea.

Let us assume that 1000 bits are needed on average for signaling for a single call2. Let us

further assume that an average call takes 3 minutes. It is easy to calculate that a PCM line can

carry no more than 30 call minutes in a minute and each call minute on average will create 333signaling bits. For a PCM full of calls this would give a total of 10 kbit/minute = 167 bit/s.

It follows for this simple calculus that one 64 kbit/s timeslot can carry call signaling for many

PCM lines. If we take as the dimensioning criteria that signaling channels should be filled notmore than 20% of the time, the number of PCM-lines we can serve with a single signaling

timeslot would be:

Nrof PCM lines = 0,2 * 64 000/167 = 76 2300 timeslots.

On the other hand, let us recall the discussion on the structure and capacity of an exchange. We

gave an example of an exchange with 8000 PCM lines and calculated that about 7000 of themmight be used for connections to other exchanges. What if there are e.g. 1000 PCM lines

connecting two exchanges.

These PCMs would carry the maximum of:1000 30 = 30 000 timeslots or simultaneous telephone calls.

With our earlier assumptions, one signaling channel can support approximately 76 PCM lines

and ca. 14 signaling channels would be required to support the call traffic on 1000 PCM lineswith 30 000 timeslots.

The above calculus is not meant to be exact but rather show the orders of magnitude of in

capacity requirements for signaling and the carriage of voice. Another major consideration inSS7 design was reliability of signaling. During 1970s operators and users were used to noisy

transmission. On the other hand from the business perspective a lost or misrouted call maymean lost revenue for the operator or an unsatisfied subscriber.


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Below fig shows the telephone network service reliability as a serial reliability diagramdepicting that all the boxes must work for the service to work. For such a system the weakest

link determines the reliability of the whole system. By eliminating the impact of signaling witha limited cost it became possible to leverage the expensive long term investment into the voice

path transmission plant and the operators other infrastructure.

It follows from the reliability requirements that it would be unacceptable to loose signalingconnectivity between two exchanges that between them carry for example 1000 simultaneous

calls because of a failure of a single signaling channel (even 100 simultaneous calls mighttrigger to say the same). Also sometimes a connection between two exchanges uses more than

80 PCM lines. From these two considerations, we have a requirement that signaling channelsneed to be able to replicate each other and also share load. Replication means that it should be

possible detect signaling channel failures quickly and to automatically move the load off the

failed signaling channel to another channel that has been preconfigured to handle the load.

Sharing the load means that at least it must be possible to split the PCMs between twoexchanges onto different signaling channels. This split can be a configuration matter because

adding and reducing the number of PCMs between two exchanges is not a frequent event andalways requires management actions from the operator.

SS7 conceptsSS7 defines a signaling network that is made ofsignaling points (network nodes) andsignaling links. In practice most times a signaling link is a PCMtimeslot.Among the signaling

points from a call perspective are the originating point(OP) and the destination point(DP).Obviously, addresses of the OP and DP need to appear in signaling messages. These addresses

are calledsignaling point codes. OPC stands for Originating Point Code and DPCfor Destination Point Code.

SS7 also specifies that intermediate signaling points that do not process the call itself may

appear on the way from the OP to the DP. These intermediate points are called Signaling

Transfer Points or STPs and work either on MTP or SCCP levels. These are useful e.g in alarge country for the purpose of concentrating or aggregating signaling from a large number of

local or transit exchanges to a small set of Intelligent Network (IN) Nodes. We will come

back to IN later on this course.In ETSI networks signaling point codes are 14 bits long. In ANSI networks the point codes are

24 bits long. In an ETSI signaling network there may be some 16 000 unique point codes.

Since the codes are allocated to Exchanges and not to users, this is quite enough. However, thismeans that an SS7 signaling network is not global in itself. Instead, the global signaling

network is formed from independent islands that each are an SS7 signaling network. Theislands are bridged by higher level functions that process telephony signaling. Asignaling link

is a connection between two signaling points. A signaling link set is a set of signaling links


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such that all links have the same signaling end points. A sequence of signaling link setsbetween two end points forms a signaling route. A signaling route differs from a link set

because of the possible intermediate STPs. Finally, the set of all signaling routes connectingtwo signaling points is asignaling route set. These concepts are depicted in below fig

set and signaling route set help to meet the requirement of high signaling capacity between twolarge exchanges through load sharing. Because the dimensioning rule is that each signaling link

is supposed to carry less than 0.2 Erl during a busy hour, we can always take one signalingchannel out of use inredundancy.

In Channel Associated Signaling timeslot 16 was dedicated to signaling. The same agreement

could be used also for SS7. However, all timeslots in a PCM system except timeslot zero are

the same. In a digital telephone network it is up to the operators to configure the use of thosetimeslots as they please. In Finland, the agreement was that timeslot 1 was reserved for

signaling purposes in SS7 networks.

SS7 Protocol ArchitectureFigure 5.3 depicts the levels and the user and application parts in the SS7 architecture.


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Figure 5.3: SS7 protocol architecture.

On the left the Figure shows comparison to OSI layers. MTP stands for Message Transfer Part,SCCP for Signaling Connection Control Part, TC for Transaction Capabilities, INAP for IN

Application Part, CAP for CAMEL Application Part that is used in GSM for a similar purposeas INAP in wire line networks. MAP stands for Mobile Application Part, BSSAP for Base

Station Subsystem Application Part, ISUP for ISDN User Part, TUP for Telephony User Part

and MUP and HUP are legacy systems that we do not need to worry about. There is alsoOMAP that is Operations and Maintenance Application Part. MTP covers two and a half OSIlevels. Compared to OSI what is missing is global reachability for messaging. This capability is

added with the help of SCCP. MTP + SCCP then together correspond to three lower layers inOSI.The primitive interfaces between levels are left for vendors to design.Therefore, instead of

layers we talk about levels. One reason that justifies looser layering in SS7 than in OSI is that

performance requirements were seen very important for SS7. Strict layering was sacrificed for

the sake of high performance and high reliability. Lets recall that this design decision wasmade sometime late 1970s or early 1980s and with the level of computing power available at

that time was probably reasonable. The task of MTP is to carry signaling messages betweenexchanges. These MTP messages will carry in their payload Application Part or User Part

messages.

MTP service is connectionless, connections ( i.e. signaling links, link sets etc) between

signaling points are pre-configured. This is natural, because business relationships are rather

stable between operators and telephone traffic is rather predictable. MTP uses DPCs and OPCsfor addressing. SCCP uses a richer set of information items for addressing than MTP. Global

addressability among telephone exchanges and other network nodes is reached by usingroutable telephone numbers in SCCP message addressing. SCCP supports both connectionless

and connection oriented messaging services.


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User Parts contain the telephone call signaling functions. Application Parts such as MAP andINAP add non-call related signaling capabilities.Application Parts use the TC services and TC

uses SCCP. TUP never uses SCCP but rather sits directly on top of MTP. ISUP may, inprinciple use SCCP but in practice it does not need it for regular telephony services and thus

also in practice relies only on MTP for messaging.

standard notes tsn unit iii part 2

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