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© Dr. D. Pesch, CIT, 2002 1 Telecommunication Concepts Telecommunication Concepts MSc in Software Development  Dr. Dirk Pesch

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Telecommunication ConceptsTelecommunication Concepts

MSc in Software Development

 Dr. Dirk Pesch

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IntroductionIntroduction

• Telecommunication systems functionality based onlayered network approach – ISO/OSI Model

• From a telecommunications software perspective

layers three to seven are most interesting – network 

layer to application layer

• Main principles associated with telecommunications

networking are switching, routing, management and

control

• Telecommunications software covers areas of protocol design, management, and applications

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TelecTelecommunicationommunication NetworksNetworks

• No generally accepted taxonomy into which allcommunication networks fit

• Networks can be classified according to

– Transmission technology

– Scale

• Transmission technology

– digital v. analogue

– point-to-point v. broadcast

– circuit-switched v. packet-switched

With regard to the physical appearance of networks, there is no general accepted

taxonomy into which all networks fit. Many different opinions exist and many

classifications have been attempted. Here, we follow Andrew Tanenbaum, who

proposes to classify networks according to transmission technology and scale.

Transmission technology refers to whether digital or analogue transmission is used.

Most modern communication networks, in particular computer communicationnetworks, use digital transmission technology. However, there are many

communication networks in operation that use analogue transmission technology.

Those networks provide the plain old telephone service (POTS) as well as allow

computers to interconnect using modem technology which converts the digital data

signal of computers into an analogue signal that can be transmitted across an

analogue telephone network.

A second aspect of transmission technology is whether networks are point-to-point

or broadcast networks. Point-to-point networks connect any two network nodes, such

as computers, telephone apparatus, switches, routers, or hubs with a physical

connection. This physical connection can be based on copper, fibre, or radio links.To go from source to destination, data will be routed along a path that can involve

one or more intermediate machines. Broadcast networks have a single

communication channel that is shared by all network nodes. Communication takes

place by one node sending data and all or a group of nodes receiving the data. In the

first case we talk about broadcasting, in the latter about multicasting.

In order to transmit data from source to destination, point-to-point networks use two

different transmission options. The first option establishes a dedicated route between

source and destination along which the information flows. This route is made up of 

dedicated physical links, which are used solely by the communication service in

question. This transmission option is called circuit switching. On the other hand, a

logical connection can be established along which the information, in form of 

packets of data, is transmitted. The logical connection can either use a physical

connection, which is shared with others, or many different physical connections are

used depending on certain circumstances. This transmission option is called packet

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Scale of NetworksScale of Networks

• Personal Area Networks

• Local Area Networks

• Metropolitan Area Networks

• Wide Area Networks

• Internetworks

A personal area network (PAN) is a network in which a number of devices

attached or in close proximity to the human body are interconnected to form a very

small network. A network consisting of a mobile phone, a personal digital assitant

and a wireless handsfree set is an example of a PAN. PANs are a very recent

invention and are typically wireless networks in which all communicating devices

are connected via short-range wireless links. Currently the wireless networkingtechnology being considered for PANs is Bluetooth but other types of wireless short

range systems may be used in the future. A local area network (LAN) is usually

privately owned and links the devices in a single office, building, or campus.

Depending on the needs of an organisation and the type of technology used, a LAN

can be as simple as two PCs and a printer in a home office environment, or it can

extend throughout the campus of a company and include voice, sound, and video

equipment.

A LAN is usually up to a few kilometres is size. LANs are distinguished by (1) their

size, (2) their transmission technology, and (3) their topology.

Example of a LAN is the well know Ethernet, which is probably the most commonLAN technology for office computer networks.

A metropolitan area network (MAN), is basically a bigger version of a LAN and

normally uses similar technology. It might cover a group of nearby corporate offices

or a city and might be either private or public. A MAN can support both data and

voice, and might even be related to the local television network. A MAN just has

one or two cables and does not contain switching elements, which simplifies design.

The main reason for distinguishing MANs as a special class of networks is because a

standard has been adopted for them. This standard is call DQDB (Distributed

Queue Dual Bus) and specified in IEEE 802.6. This MAN standard is used to

provide Switched Megabit Data Services (SMDS) to metropolitan areas. It iswidely used in North America and also in some European countries such as

Germany, where the service is called Datex-M. However, it is expected that the

Asynchronous Transfer Mode (ATM) technology will replace DQDB in the near

future. ATM will provide corporate backbone networks, which are of the size of 

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A WAN consists of end systems, e.g. a computer (host) or even a mobile terminal

(mobile phone), and communication subnets. The job of the subnet is to carry data

from end system to end system. In most WANs, the subnet consists of transmission

lines and switches. Transmission lines, also called circuits, channel, or trunks, movebits between machines. The switching systems are specialised computers as outlines

above.

Many networks exist in the world, e.g. computer networks, packet data networks,

circuit-switched telephone networks, mobile radio networks, etc., often with

different hardware and software. People connected to one network often want to

communicate with people attached to a different one. For example a person may

want to call a friend, who has a mobile phone, from his/her home telephone.

This desire requires connecting together different, and frequently incompatible

networks, sometimes by using machines called gateways to make a connection and

provide the necessary translation, very much like an interpreter. A collection of 

interconnected networks is called an internetwork or just internet.

NOTE: This should not be confused with the term Internet, which refers to the

global computer network using the TCP/IP protocol. However, the origin of the term

Internet is from internetworks, what the Internet basically is.

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Transmission ModesTransmission Modes

• Simplex• Half-Duplex

• Duplex

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Network TopologiesNetwork Topologies

• Mesh topology

• Star topology

• Tree topology

• Ring topology

• Bus topology

• Hybrid topology

• Irregular topology

The term topology refers to the way a network is laid out, either physically or

logically. Two or more devices connect to a link;

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Mesh TopologyMesh Topology

In a mesh topology, every device has a dedicated point-to-point link to every

other device. The term dedicated means that the link carries information only

between the two devices it connects. A fully connected mesh network 

therefore has n(n-1)/2 physical channels to link n devices. To accommodate

that many links, every device on the network must have n-1 input/output

(I/O) ports.

A mesh topology offers advantages over other topologies. First, the use of 

dedicated links guarantees that each connection can carry its data load, thus

eliminating data traffic problems that can occur when more than two device

share a common communication channel. Secondly, a mesh topology is

robust. If one link fails, it does not incapacitate parts of or the entire

communication network. Another advantage is privacy or security. When a

message travels along a dedicated line only the intended recipient sees it.Finally, point-to-point links make fault identification and isolation easy.

Traffic can be routed to avoid links with suspected problems.

The main disadvantage of a mesh are related to the amount of cabling and the

number of I/O ports required. This has implication on the amount of 

hardware required, the available space for cabling and finally the overall

cost, which can be prohibitive. Therefore, mesh topologies are often only

used in backbone networks or the mesh provides only partial connection

between devices.

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Star TopologyStar Topology

Hub/Switch

In a star topology, each device has a dedicated point-to-point link only to a central

controller, e.g. a hub or switch. The devices are not linked to each other. Unlike a

mesh topology, a star topology does not allow direct traffic between devices. The

controller acts as an exchange: If one device wants to send data to another, it sends

to the controller which, which then relays the data to the other connected devices

(see figure above).

A star topology is less expensive then a mesh topology. In a star, each device needs

only one link and one I/O port. This factor also makes it easy to install and

reconfigure. Far less cabling needs to be housed, and additions, moves, and deletions

involve only one connection. Other advantages include robustness. If one link fails,

only the device connected is affected and no other parts of the network. Fault finding

is also easy.

An example of a star configuration is an Ethernet LAN with a hub as a central

controller.

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Tree TopologyTree Topology

Switch Switch

Switch

Switch

A tree topology is a variation of a star. As in a star, nodes in a tree are linked to a

central controller that controls the data traffic in the network. However, not every

device is directly connected into a central hub. The majority of device are connected

to a secondary controller that in turn is connected to the central controller.

The advantages and disadvantages of a tree topology are generally the same as forthe star. The addition of secondary controllers (switches), however, brings two

further advantages. First, it allows more devices to be attached to a central switch

and can therefore increase the distance a signal can travel between devices. Secondly

it allows to scale the size of the network. When the network grows, a single central

controller may easily be overloaded by the amount of devices connected. In a tree

topology the number of devices attached to an individual controller can be scaled.

However, if a link between a secondary controller and the central controller fails, the

entire subtree will be disconnected from the rest of the network.

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Bus TopologyBus Topology -- Example 1Example 1

Cable end Cable endTap

Drop

line

The preceding topologies are all examples of point-to-point transmission technology.

A bus topology, on the other hand, is an example of a broadcast technology. One

long cable is shared by all devices in the network (see figure above). Nodes are

connected to the bus by drop lines and taps. A drop line is a connection running

between the device and the main cable. A tap is a connector that either splices into

the main cable or punctures the sheathing of a cable to create a contact with themetallic core. Due to the electric resistance of the cable, the distance between two

adjacent taps is limited. Also, if a device does not regenerate a signal the overall

length of the cable is limited and thus the size of the network.

Advantages of a bus topology include ease of installation. Backbone cable can be

laid along the most efficient path, then connected to the devices by drop lines of 

various lengths. In this way a bus uses less cabling than the previous topologies.

Disadvantages include difficult reconfiguration and fault isolation. A bus is usually

designed to optimally efficient at installation. It can therefore be difficult to add new

devices. Signal reflections at taps can degrade signal quality. Also, the mechanismwhich controls the sharing of the single communication channel among a number of 

nodes can have a limiting effect on the number of devices that can be connected to a

bus.

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Bus TopologyBus Topology -- Example 2Example 2

Ether

In the previous example of a bus topology, the bus was the physical communication

medium for a number of devices. In the example above, which shows a wireless

broadcast network such as the CB (citizens band) radio.

The ether  represents a logical bus, which represents the single communication

channel that is shared by all radio terminals.

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Ring TopologyRing Topology

In a ring topology, each device has a dedicated point-to-point line configuration only

with the two devices on either side of it. A signal, e.g. a message, data, or a packet,

is passed along the ring in one direction, from device to device, until it reaches its

destination. Each device in a ring incorporates a repeater. When a device receives a

signal intended for another device, its repeater regenerates the bits and passes them

along (see figure above).

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Hybrid TopologyHybrid Topology

Switch

HubBus

Ring

Star

Star

Communication networks often combine several of the basic topologies as

subnetworks linked together in a larger topology. One department in a college has

decided to have a ring topology using Token Ring LAN technology, whereas another

department uses a bus topology with an Ethernet LAN. The two subnets can beconnected to each other by a central controller, which may be a hub or a switch.The

so created higher topology is a start topology (see figure above).

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Layered Network ArchitectureLayered Network Architecture

Physical transmission medium

Layer 1

Layer 2

Layer 3

Layer 4

Layer 1

Layer 2

Layer 3

Layer 4

Layer 1 protocol

Layer 2 protocol

Layer 3 protocol

Layer 4 protocol

Layer 1/2 interface

Layer 2/3 interface

Layer 3/4 interface

Host 1 Host 2

Layer 5 Layer 5Layer 5 protocol

Layer 4/5 interface

In order to reduce the design complexity of networks, they are organised as a series

of layers or levels, each one built upon one below it. The number of layers, the name

of each layer, contents of each layer, and the function of each layer differ from

network to network. However, in all networks, the purpose of each layer to offer

certain services to higher layers, shielding those layers from the details of how the

offered services are actually implemented.Layer N on one machine carries on a conversation with layer N on another machine.

The rules and conventions used in this conversation are collectively known as the

layer N protocol. Basically, a protocol is an agreement between the communicating

parties on how communication is to proceed. The key elements of a protocol are

• Syntax - includes such things as the data format, coding and signal levels.

• Semantics - includes control information for co-ordination and error handling.

• Timing - includes speed matching and sequencing.

A five layer network is illustrated in the slide above. The entities comprising the

corresponding layers on different machines are called peers. In other words, it ispeers that communicate using protocols.

In reality, no data are directly transferred from layer N on one machine to layer N on

another machine. Instead, each layer passes data and control information to the layer

immediately below it, until the lowest layer is reached. Below layer 1 is the physical

transmission medium through which actual communication occurs.

Between two pairs of adjacent layers there is an interface. The interface defines

which primitive operations and services the lower layer offers to the upper layer. It is

important in the design of a layer to define clean interfaces so that it is possible to

replace the implementation of one layer by a completely different implementation.

A set of layers and protocols is called a network architecture. The specification of 

an architecture must contain enough information to allow unambiguous

implementation of the functionality of each layer in either software or hardware. The

details of the implementation and the specification of the interfaces are not part of 

the architecture as the are hidden awa inside the machines and are not visible to

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Information Flow and Protocol HierarchyInformation Flow and Protocol Hierarchy

M

MH4

H3 H4 H3M1 M2

H3 H4 M1H2 H2 H3 M2

T2 T2

M

MH4

H3 H4 H3M1 M2

H3 H4 M1H2 H2 H3 M2

T2 T2

Layer 2

protocol

Layer 3 protocol

Layer 4 protocol

Layer 5 protocol

Layer

5

4

3

2

1Layer 1 protocol

Source machine Destination machine

The slide above demonstrates how a message is sent from the top (fifth) layer of one

machine to the top layer of the other. A message,  M , is produced by the protocol

entity in layer 5. This entity may be an application process or an entity providing

service to an even higher layer. The message is passed on to layer 4, where a header

is put in front of the message to identify the message. The header includes control

information, such as sequence numbers, to allow layer 4 on the destination machineto deliver messages in the right order if the lower layers do not maintain sequence.

In some layers headers also contain sizes, times, and other control information. The

resulting unit of header and message is passed on to layer 3. In many networks there

is no real limit to the size of messages transmitted in the layer 4 protocol, but there is

nearly always a limit imposed by the layer 3 protocol. Consequently, layer 3 must

break up the incoming message into smaller units, packets, pre-pending a layer 3

header to each packet. In the example above, the data passed from layer 4 to layer 3

is split into two parts. This divides message M into two parts, M 1 and M 2.

Layer 3 decides which of the outgoing lines to use and passes packets to layer 2.

Layer 2 adds not only a header to each piece, but also a trailer, and gives theresulting unit to layer 1 for physical transmission. At the destination machines the

received data moves upward, from layer to layer, with headers being stripped off and

the original message M being recreated as the data progresses. None of the headers

or trailers of layer N are passed up to layer N+1.

The important aspect to understand about the example in the slide above is the

relation between the virtual and actual communication and the difference between

protocols and interfaces. The peer processes in layer 4 think of their communication

as being horizontal using the layer 4 protocol. Each one is likely to have a procedure

called SendToOtherSide, even though this procedure actually communicates with the

lower layer across the layer 3/4 interface and not with the other side.

Even though the reader might have the impression that protocols are implemented in

software, the lower layers are frequently implemented in hardware. The functionality

of layer 1 is almost always implemented in hardware, often in a specially designed

ASICs.

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Design Issues for LayersDesign Issues for Layers

• Addressing

• Segmentation and re-assembly

• Transmission modes

• Error control

• Flow control

• Routing

• Multiplexing

• Connection and other management

The concept of addressing in a communication architecture is a complex one and

covers a number of issues. At least four separate issues need to be discussed:

• Addressing level

• Addressing scope

• Connection identifiers

• Addressing mode

  Addressing level refers to the level of communications architecture at which an

entity is named, e.g. end system or intermediate system. Such an address is in

general a network level address as for example an IP address in the case of TCP/IP

or a network service access point (NSAP). In general an address identifies a service

access point (SAP) in the protocol hierarchy of the network architecture. A second

issue of addressing is the addressing scope. An IP address is a globally unique

address. In an Ethernet LAN for example, each Ethernet card is identified by an

address which is valid in the sub-network where the card is used.

The concept of connection identifiers comes into play when the connection-orienteddata transfer is considered, e.g. virtual circuits. A connection between the two ends

of a sub-network is identified by a connection identifier or the connection between

two end-systems. The addressing mode is used when uni-cast , multi-cast , or

broadcast  communication is used, that is in point-to-point or point-to-multipoint

connections.

Segmentation and re-assembly takes place when a higher layer passes data packets

to a lower layer, which has restrictions on size for the data segments it can send to its

peer entity or to the layer below. An example of this is ATM (asynchronous transfer

mode) networks. The ATM layer accepts only chunks of 48 bytes from the layer

above, because it process data in form of cells of 53 bytes each, with a 5 byte header,which the layer adds itself, and a 48 byte payload with data from the higher layer. In

order to make sure that the data packets, which have been segmented, arrive in the

right order to the receiving entity, a sequencing function is often used. Each

segment is assigned a sequence number. The receiving side can then re-assemble the

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Error control is used to guard against loss or damage of data and control

information. The level of error control varies depending on the type of data that is

being transmitted. Control data, which is essential for the proper operation of the

communication system, must not experience any damage or loss duringtransmission. Therefore, error control mechanisms make sure that the probability of 

error is very small. On the other hand, if voice is being transmitted, error control

needs not be as stringent as voice communication can sustain damage or loss of 

information. The human brain is very good at correcting or replacing loss of voice

information. Two types of error control can be distinguished, forward error control

(FEC) and automatic repeat request (ARQ) error control. The first type adds

redundancy to the data that is being send. This adding of redundancy, also called

channel coding, is used to detect and also to correct errors in digital data. However,

if more errors were introduced than can be corrected, the received data will remain

erroneous. This type of error control is frequently used in voice communication. The

second type, ARQ mechanisms, are used for error control of data and control

information. Some redundancy is added that allows the receiving side to determine

whether errors were introduced. If the receiving side detects that data is not error

free, it requests the sending side to repeat the transmission. In this case errors in

sequencing of segmented data are also covered. A combination of FEC and ARQ

mechanisms are used in systems where the physical transmission medium is

regarded as highly unreliable. This would be the case in all mobile radio systems.

Flow control is a function performed mainly by the receiving end in order to limit

the amount or rate of data that is send by the transmitting entity. Flow control is used

to manage and also shape the data traffic in the communication system and to avoid

congestion. The simplest form of flow control is a stop-and-wait procedure, in whicheach data packet must be acknowledged before the next can be sent. More efficient

protocols use a sliding window mechanisms, such as HDLC based protocols.

Routing is a function that is used to determine the transmission path between two

end systems across a number of subnets. The transmission route that is being

established depends on a number of factors, such as traffic intensity and congestion,

availability of transmission medium, cost of transmission, transmission delay, and

reliability of transmission among others. Routing functions usually reside in layer 3

of the protocol hierarchy. Routing can be static or dynamic. Static routing is used

mainly in connection-oriented data transmission, where a physical or virtual

connection is established between two end-systems.   Dynamic routing is used inconnectionless data transmission where each data packet carries the destination

address and can be routed independently of other data packets between the two end

systems.

The concept of multiplexing is related to addressing. One form of multiplexing is

supported by means of multiple connections into a single system. For example a

number of virtual connections can terminate in one end system. These virtual

connections are transmitted over a single physical channel, they are multiplexed into

the physical channel. Beside multiplexing of virtual connections into one physical

connection, there can also be logical multiplexing of many logical connections into

another logical connection. There are several ways in which multiplexing of multiplevirtual connections into a physical connection can take place. The most common

forms are based on frequency, time or code multiplexing. The concept of 

multiplexing will be addressed in detail later.

  -

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Interfaces and ServicesInterfaces and Services

Layer N

Layer N - 1

(N) - PDU

(N-1) - SDU

(N-1) - PCI

(N-1) - PDU

(N-1) - SAP

Relationship between layers and interfaces

Interface

The function of each layer is to provide a service for the layer above. The

active elements in each layer are called entities. An entity can be a software

entity (such as a process) or a hardware entity (such as an I/O chip). Entities

in the same layer in different systems are call peer entities. The entities in

layer N implement a service used by layer N+1. In this case layer N is called

the service provider and layer N+1 the service user.

Services are available at Service Access Points (SAPs). The layer  N SAPs

are the places where layer  N+1 can access the services offered. Each SAP

has an address that uniquely identifies it. As an example, the SAPs in the

telephone system are the sockets into which the telephone apparatus are

plugged, and the SAPs addresses are the telephone numbers of these sockets.

To call someone, one must know the callee’s SAP address.

In order for two layers to exchange information, there has to be an agreed

upon set of rules about the interface. The standard convention in the layeredmodel is that the layer N+1 entity passes a Protocol Data Unit (PDU) to the

layer N entity through the layer N SAP. The PDU consists of a Service Data

Unit (SDU) and Protocol Control Information (PCI), which is added by

the layer entity in order to perform the operation of the layer protocol. The

SDU may also contain Interface Control Information (ICI), which may be

needed by the layer N entity.

In order to transfer the SDU, the layer  N entity may fragment it into several

pieces, each of which is given a header and sent as a separate PDU, such as a

packet.

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ConnectionConnection--Oriented and Connectionless ServicesOriented and Connectionless Services

• Connection-Oriented Service

– modelled after telephone network 

– connection acts like a tube

• Connectionless Service

– modelled after postal system

– Each message (packet, cell) carries full dest. address

• Quality of Service

Layers can offer two types of service to the layers above: connection-oriented and

connectionless service.

To use a connection-oriented service, the service user first requests the

establishment of a connection, uses the connection for information exchange, and

then releases the connection. The essential aspect of the connection is that it acts like

a tube: the sender pushes objects (bits) in one end, and the receiver takes them out inthe same order at the other end.

In contrast, a connectionless service does not first establish a connection. Each

message carries the full destination address, and is routed through the system

independent of other messages. Normally, the message sent first will arrive first.

However, it is possible for messages to ‘overtake’ each other. With a connection-

oriented service this is impossible.

Each service can be characterised by a quality of service. Some services are reliable

in the sense that they never loose data. Usually, a reliable service is implemented by

having the receiver acknowledge the receipt of each message, so that the sender is

sure it has arrived. The acknowledgement process introduces overhead and delays,which are often worth the effort but undesirable. An application where delays are

unacceptable is digitised voice or video traffic (in general any real-time traffic). It is

preferable for telephone users to hear some noise in the background than to wait for

acknowledgements of delivered voice frames.

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Not all application require connections. For example, electronic junk mail

delivery may become common for advertising purposes on the Internet some

day. The junk mail sender may not want to go through the trouble of settingup and later tearing down a connection to send just one item to hundreds of 

users. Furthermore, 100 percent reliability may not be required for this

service. All that is need is a high probability that the junk mail will reach its

destination. Unreliable connectionless service is often called datagram

service, in analogy with telegram service, which does also not provide an

acknowledgement back to the sender.

Still another service is the request-reply service. In this service the sender

transmits a single datagram containing a request; the reply contains the

answer. For example, a query to the local library asking whether Andrew

Tanenbaum’s book “Computer Networks” is available falls into thiscategory. The request-reply service is commonly used to implement

communication in the client-server model: the client issues a request and the

server responds to it.

The table below summarises the most common types of services.

Service Example

Connection- Reliable message stream Sequence of pages

oriented Reliable byte stream Remote login, file transfer

Unreliable connection Digitised voice/video

Connection- Unreliable datagram Electronic junk mail

less Acknowledged datagram Registered mail

Request-reply Database query

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Service PrimitivesService Primitives

• Service is formally specified by primitives (operations)

• Four classes of primitives

– Request

– Indication

– Response

– Confirm

A service is specified by primitives available to a user or other entity to access the

service. These primitives tell the service to perform some action or report on an

action taken by a peer entity. One way to classify the service primitives is to divide

them into four classes as shown in the table below.

Primitive Meaning

Request An entity wants the service to do some work 

Indication An entity is to be informed about an event

Response An entity wants to respond to an event

Confirm The response to an earlier request has come back 

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Service PrimitivesService Primitives -- ExampleExample

CONNECT.request CONNECT.confirm CONNECT.indication CONNECT.response

System A System B

Layer N Layer N

Layer N - 1 Layer N - 1

Connection Establishment

To illustrate the use of primitives, consider how a connection between layers

in two different systems is established. The initiating entity in layer N of 

System A, requests the underlying layer N - 1 to establish a connection by

requesting its service CONNET by issuing a CONNET.request primitive. This

results in a message being send by the layer N - 1 entity in System A to layerN - 1 in System B. The CONNECT service in layer N - 1 of System B notifies

layer N of the establishment request by issuing a CONNECT.indication.

Layer N uses the CONNECT.response primitive to tell layer N - 1 whether it

wants to accept or reject the proposed connection. The layer N - 1 entity in

System B sends a message to the layer N - 1 entity in System A with the

response of the layer N entity in System B. The entity in layer N - 1 of 

System A informs the requesting Layer N entity in a CONNET.confirm

primitive of the outcome of the connection establishment.

Most primitives can have parameters, which specify addresses, service types,

maximum message sizes, caller identity, and a reject or accept field. Thevalue of the parameters varies the connection establishment. A form of 

negotiation takes place and the details are part of the protocol.

Services can either be confirmed or unconfirmed. In a confirmed service

there is a request , indication, response, and confirm. In an unconfirmed

service, there is just a request and an indication. An example of a confirmed

service is the above connection establishment. An example for an

unconfirmed service is data exchange on an established connection , which

typically uses the primitives DATA.request and DATA.indication.

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Relationship of Services to Protocols

Services and protocols are distinct concepts, although they are frequentlyconfused. A service is a set of primitives (operations) that a layer provides to

the layer above. The service defines what operations the layer is prepared to

perform on behalf of the its users, but it says nothing at all about how these

operations are implemented. A service relates to an interface between two

layers, the Service Access Point (SAP), with the lower layer being the service

provider and the upper layer the service user.

A protocol, in contrast, is a set of rules governing the format and meaning of 

messages, frames, or packets that are exchanged by peer entities within a

layer of two different systems. Entities use protocols in order to implement

their service definitions. They are free to change their protocols, provided

they do not change the service that is visible to the user. In this way the

service and the protocol are completely decoupled.

There is a strong analogy with programming languages, in particular object-

oriented languages. A service relates to an object. It defines operations that

can be performed on the data of an object but does not specify how these

operations are implemented. A protocol relates to the implementation of an

object’s operations and as such are hidden from the user.

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The ISO/OSI 7 Layer RMThe ISO/OSI 7 Layer RM

• International Standards Organisation (ISO) Open SystemsInterconnection (OSI) Reference Model

Physical transmission medium

Physical layer

Data Link Layer

Network Layer

Transport layer

Physical layer protocol

Data Link layer protocol

Network layer protocol

Transport layer protocol

Session layerSession layer protocol

Physical layer

Data Link Layer

Network Layer

Transport layer

Session layer

Presentation layerPresentation layer protocol

Application layerApplication layer protocol

Presentation layer

Application layer

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The Internet (TCP/IP) RMThe Internet (TCP/IP) RM

• 5 Layer Reference Model– Host-to-network layer (layers 1 and 2)

• Physical layer

• Multiple Access sublayer

• Link layer

– Subnet (Internet) layer

– Transport layer

– Application layer

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The Physical LayerThe Physical Layer

Transmission of raw bits over a communication channel

• DAC/ADC

• Modulation

• Voltage levels

• Electrical interfaces

• Mechanical connections

• Properties of the physical transmission medium

The physical layer is concerned with transmitting raw bits over a communication

channel. The design issues are basically to make sure that when one side sends a 1

bit, it is received as a 1 bit and not as a 0 bit. Typical characteristics of physical

layers are how many volts should be used to represent a 1 and how many for a 0,

how many microseconds a bit lasts, whether transmission may proceed

simultaneously in both directions, how the initial connection is established and howit is torn down when both sides are finished, and how many pins the network 

connector has and what each pin is used for. The design issues in the physical layer

deal largely with mechanical, electrical, and procedural interfaces, and the physical

transmission medium, which lies below the physical layer.

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The Data Link LayerThe Data Link Layer

Transform a raw data transmission facility into a reliable (error

free) link for the network layer

• Data framing

• Addressing

• Flow control

• Error detection and correction (recovery)

• Synchronisation

• Multiple access control (for broadcast/multipoint channels)

The main task of the data link layer is to tale a raw transmission facility provided

by the physical layer and transform it into a communication line that appears free of 

undetected transmission errors to the network layer. It accomplishes this task by

having the sender break the input data up into data frames (typically a few hundred

or a few thousand bytes), transmit the frames sequentially, and process the

acknowledgement frames sent back by the receiver. Data link layer frames includecontrol information for synchronisation, link management and error detection and

correction.

Another issue that arises in the data link layer (and most of the higher layers as well)

is flow control. Flow control stops a slow receiver from being drowned in data. This

requires some form of traffic regulation mechanism. In the data link layer flow

control and error handling are often integrated.

Broadcast networks have an additional issue in the data link layer: how to control

access to the shared communication channel. A special sublayer of the data link 

layer, the medium access sublayer, deals with this problem.

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The Network LayerThe Network Layer

The network layer controls the operation of the subnet

• Routing

• Congestion control

• Logical addressing

• Address transformation

• Interfacing between heterogeneous networks

The network layer is concerned with controlling the operation of the subnet. A key

design issue is determining how information is routed from source to destination.

Routes can be based on static tables that are “wired into” the network and rarely

change. They can also be determined at the start of each conversation or can be

highly dynamic and change with every packet in order to reflect the network load.

If to many users are using the network it can lead to congestion. The control of congestion is also part of the network layer tasks.

When information travels from one network to another to get to its destination,

addressing needs to be taken into account. This requires translation of local

addresses between two networks. This also requires some form of interfacing

between two networks. The function of accounting comes into the picture at network 

boundaries since all involved operators would like to get a share of the bill.

In broadcast networks, the routing problem is simple, so the network layer is often

thin or non-existent.

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The Transport LayerThe Transport Layer

Source-to-destination (end-to-end) delivery of the entireinformation (data stream)

• End-to-end message delivery across one or more subnets

• Service-point (port) addressing

• Segmentation and reassembly

• Multiplexing

• Connection control

The basic function of the transport layer is to accept data from the session layer,

split it up into smaller units if need be, pass these to the network layer, and ensure

that the pieces all arrive correctly at the other end. In this way, the transport layer

provides a true end-to-end connection. The lower layers establish connections only

to their immediate neighbours, whereas a transport layer connection can span several

networks and network layers.Under normal conditions, the transport layer creates a distinct network connection

for each transport connection required by the session layer. If the transport

connection requires high throughput, however, the transport layer might create

multiple connections, dividing the data among the network connections to improve

throughput. On the other hand, network connections can be expensive and the

transport layer might multiplex several connections onto the same network 

connection to reduce cost. In all cases the transport layer is required to make

multiplexing transparent.

The transport layer also determines what kind of service to provide to the session

layer. This can be connection-oriented or connectionless.Many hosts allow multiple connections to enter and leave the host. There needs to be

some form of service point addressing in order to tell which information belongs to

which connection.

In order to maintain end-to-end connectivity the transport layer requires

functionality to establish, maintain and release connections across the network. This

requires some form of naming or addressing. There is also an element of flow

control in the transport layer in order to control the data flow across a network with

possibly links of higher and lower speed.

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The Session LayerThe Session Layer

(Only exists in the OSI RM)

Establish sessions between users on different machines

• Session management

• Dialogue control

• Token management

• Synchronisation

The session layer allows users on different machines to establish sessions between

them. A session allows ordinary data transport, as does the transport layer, but it also

provides enhanced services useful in some applications. A session might be used to

allow a user to log into a remote timesharing system or to transfer a file between two

machines. One of the services of the session layer is to manage dialogue control.

Sessions can allow traffic to go in both directions at the same time, or only onedirection at a time. If half duplex transmission is used, the session layer keeps track 

of whose turn it is.

A related session service is token management. For some protocols, it is essential

that both sides do not attempt the same operation at the same time. To manage these

activities, the session layer provides token exchange.

Another session service is synchronisation. Consider a two hour file transfer between

two machines with a one hour mean time between crashes. In order to avoid to start

the whole transmission over and over again, the session layer inserts check points at

which data transmission can resume after a crash.

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The Presentation LayerThe Presentation Layer

(Only exists in the OSI RM)

Layer ensures interoperability from a syntactical and semantics

point of view

• Translation

• Encryption

• Compression

• Security

The presentation layer, unlike all lower layers, which are just interested in moving

bits around networks reliably, is concerned with syntax and semantics of the

information transmitted.

The functions provided by the presentation layer include translation of characters

between two code systems, for example between ASCII and Unicode, encryption of 

sensitive data for security purposes, and compression of data in order to reducebandwidth requirements.

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The Application LayerThe Application Layer

Enables the user, whether human or software, to access and usethe communication network 

• Network virtual terminal

• File access, transfer, and management

• Mail services, Directory services

• Hypertext transfer (world wide web)

• Control signalling applications in telecommunication networks

– call/session establishment, maintenance, release

– call related and independent supplementary services

The application layer contains a variety of protocols that are commonly needed. In

computer networks the typical application layer protocols are Telnet, FT, X.400

messaging, X.500 directory service.

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A Critique of OSI RMA Critique of OSI RM

• Pro

– The layered concept simplifies design and implementation and thegeneral concept is used in most data and computer communication

networks

• Cons

– The OSI reference model is not a generally suitable model for

communication networks

– The architecture of many real networks cannot easily be mapped

onto the OSI RM

– The layer protocols recommended for the OSI RM are too generic

and complex for many implementations

– The functionality of many layers is not needed in real networks

– The OSI model does not deal well with the concept of planes,which is used in many modern data communication networks

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A Critique of Internet RMA Critique of Internet RM

• Pro– Internet protocols are well thought out and can be efficiently

implemented

– Internet protocols and networks have proven to be extremly usefuland telecommunications is in fact moving towards a unifyingadoption of the Internet protocols

• Cons– Internet RM is not a general definition of a layered network 

architecture and as such not suitable to describe any other network 

– Internet RM is not well defined in terms of service, interface, andprotocol and therefore not suitable as a guide to designing newnetworks

– Some layers within the Internet RM do not distinguish between aninterface and a layer well enough

– Internet RM does not define the functionality of the physical anddata link layers well enough for network design

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Communication ProtocolsCommunication Protocols

• Layers in layered network architecture contain peerprocesses

• Peer processes

– have a common objective, which is achieved through

processing and information exchange

– communicate through lower layers

– consist of an algorithm, which is implemented as a

distributed algorithm or protocol

• Communication Protocols are distributed algorithms

implemented by two or more peer processes toprovide a communication facility to higher layers

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Problems of Distributed AlgorithmsProblems of Distributed Algorithms

Blue

army 1

Blue

army 2

Red

army

Messenger

As indicated above, a communication protocol is an implementation of a distributed

algorithm. In order to gain some insight into the problems associated with distributed

algorithms, we examine the above example involving unreliable communication,

which has in fact no solution.

There are three armies, two coloured blue and one red. The red army separates the

two blue armies. If the two blue armies attack at the same time, they win over the redarmy, but due to the red army’s strength, they lose if they attack independently. The

only communication between the two blue armies is by sending a messenger through

the red army lines. There is a possibility that the messenger will be captured, causing

the message to go undelivered. The blue armies would like to synchronise their

attack at some given time but are unwilling to attack unless assured with certainty

that the other will also attack. Thus, the first blue army might send a message saying

“Let’s attack on Monday noon; please acknowledge if you agree”. The second blue

army, receiving such as message, might send a return message saying “We agree;

please send an acknowledgement if you receive our message”. It is not hard to see

that this strategy leads to an infinite sequence of messages, with the last army tosend a message being unwilling to attack until obtaining a commitment form the

other side.

It is in fact more surprising, that no strategy exist for the two armies to synchronise.

One may try to convince oneself that this is in fact the case by going through the

situation presented above. What you are likely to encounter in this simple mind

experiment is that it is difficult to convince oneself that there is no solution to the

problem. This is so, because we are generally not used to dealing with distributed

decision making problems based on distributed information. If the above conditions

are relaxed as to require only a high probability of simultaneous attack, the problem

can be solved. How?

Fortunately, most problems in real communication networks do not require

simultaneous agreement. Typically, what is required is for one peer process to enter

a given state with the assurance that the other peer process will eventually enter a

corresponding state. Some acknowledgement may berequired for this but a deadlock 

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Error and Flow ControlError and Flow Control

• Communication links are fundamentally unreliable to

more or less extend• In order to provide a reliable communication facility

mechanisms to detect and correct transmission

impairments have to be introduced

• Provision of a reliable communication facility will also

cause some overhead on top of the actual data that is to

be transmitted

• The two communicating parties require to adhere to

common rules of communication• Typically the Data Link Control Layer provides the

means for reliable communication

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Data Link Control LayerData Link Control Layer

Network layer

Data Link Control

layer

Physicallayer/interface

Network layer

Data Link Control

layer

Physicallayer/interface

Virtual synchronous unreliable bit pipe

Communication link 

Packets

Frames

Data

H Data T

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Objectives for Data Link ControlObjectives for Data Link Control

• Frame Synchronisation

• Flow control

• Error detection and correction (error control)

• Addressing

• Framing

– Control information and user data transmission on the same

link 

• Link Management

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Error ControlError Control

• In order to provide reliable communication we mustbe able to

– detect and

– correct any transmission errors

• How can this be achieved?

– Error detection

• Add information to data that will allow to detect bit errors

– Error correction

• Add information to data that allows to correct bit errors

• repeat sending data until received error-free

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Error DetectionError Detection

• Error detection techniques are based on addingredundancy to data messages

• Strategy

– partition data into blocks of n bits

– depending on n bit sequence add additonal k bits according

to some algorithm

– Apply algorithm at receiver to detect whether n bits were

received without bit-error

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Error Detection StrategiesError Detection Strategies

• Parity Check codes• Cyclic Redundancy Check (CRC) codes

• Block codes

– BCH codes

– other block codes

• Convolutional codes

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Forward Error ControlForward Error Control

• Error detection  and correction strategy• Used in cases where re-transmission is not an option

due to real-time constraints

• Error correction by means of adding redundancy

• Two main types of FEC

– Block codes

– Convolutional codes

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Standard CRC PolynomialsStandard CRC Polynomials

• 16 bit– CRC-16 P(X)=X16+X15+X2+1

– CRC-CCITT P(X)=X16+X12+X5+1

• 32 bit

– CRC-32 P(X)=X32+X26+X23+X22+X16+

X12+X11+X10+X8+X7+X5+

X4+X2+X+1

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Implementation of CRCImplementation of CRC

• Typically CRC checks are implemented in digitallogic on integrated circuits together with other DLC

and physical layer functions

• Implementation based on XOR gates + shift register

– register contains n bits, equal to the length of FCS

– up to n XOR gates

– presence or absence of gate corresponds to 1 or 0 in P

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Example CRC Shift RegisterExample CRC Shift Register

C4 C3 C2 C1 C0

INOUT

M = 1010001101 M(X)=X9+X7+X3+X2+1

P = 110101 P(X)=X5+X4+X2+1

FCS=

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Flow ControlFlow Control

• Mechanism to control the speed of transmission of data by sender according to the reception capacity

(buffer space) of receiver

• Flow control based on sequential transmission of 

frames

• Two main types of flow control used

– stop-and-wait

– sliding window

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StopStop--andand--wait Flow Controlwait Flow Control

T R

T R

T R

T R

T R

t0

t0 + 1

t0 + a

t0 + 1 + a

t0 + 1 + 2a

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SlidingSliding--window Flow Controlwindow Flow Control

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0

0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0

Window expands as

acknowledgements are received

Window of frames that may be transmittedFrames already received

Frame

sequence

number

Last frame

transmitted Window shrinks as

frames are sent

Frames already received Window of frames that may be accepted

Last frame

acknowledged Window shrinks as

frames are received

Window expands as

acknowledgements are sent

Transmitter view

Transmitter view

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SlidingSliding--window Flow Controlwindow Flow Control

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Utilisation of Sliding Window Flow ControlUtilisation of Sliding Window Flow Control

as Function of Window Sizeas Function of Window Size

a

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Automatic Repeat RequestAutomatic Repeat Request

• Error correction through retransmission – backward

error correction

• Three main types of backward error correction

strategies – Automatic repeat request (ARQ)

– Stop-and-wait ARQ

– Go-back-N ARQ

– Selective-repeat ARQ• Retransmission based on flow-control mechanism to

avoid overloading of receive buffer

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StopStop--andand--Wait ARQWait ARQ

• Based on stop-and-wait flow control

• Two types of errors are considered

– frame arrives damaged→ no ACK is sent, timer at

transmitter expires and frame is resent

– ACK from receiver is damaged and transmitter resends

same frame; in order to avoid confusion, frames and ACK

are alternatively marked 0 and 1, respectively

• ARQ scheme is simple but not very efficient

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StopStop--andand--Wait ARQWait ARQ -- ExampleExample

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GoGo--back back --N ARQN ARQ

• Improves efficiency by adopting sliding-window flow

control mechanism

• N denotes length of sliding window

• RR denotes ACK, REJ denotes NACK

• Principle

– When a frame in error is received, destination sends a REJ

and discards erroneous frame and all future frames until theone a frame is correctly received

– Upon receipt of REJ, transmitter must retransmit erroneous

frame and all frames that where sent in the meantime

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GoGo--back back --N ARQ OperationN ARQ Operation

• Damaged Frame received

• A transmits frame i. B detects error but has received (i-1)

correctly. B sends REJ i, A retransmits i and all subsequent

frames

• Frame i was lost in transit. A sends (i+1), B receives out of 

order frame and sends REJ i.

• Frame i is lost. A does not send more frames and B receives

nothing and does not send RR or REJ. A timer at A expiresand A sends RR frame with poll bit P = 1. B sends RR with

next frame it expects and A resends frame i

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GoGo--back back --N ARQ OperationN ARQ Operation

• Damaged RR

– B receives i and sends RR (i+1), which is lost. A may

receive an RR to a subsequent frame before timer expires →

no error

– A’s timer expires and transmits an RR as in the case before.

If RR response from B fails, A will try again for a number

of times and than initiates link reset

– A receives a damaged REJ. A acts like in the case of damaged RR.

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GoGo--back back --N ARQN ARQ

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SelectiveSelective--Reject ARQ OperationReject ARQ Operation

• Based on sliding window flow control mechanism in a

similar fashion as go-back-N

• Only damaged frames are retransmitted by sending

SREJ

– this is more efficient, but receiver has to maintain a large

enough buffer to save post SREJ frames

– transmitter must be able to send out of sequence frames

– receiver must be able to order out-of-sequence frames

• A problem occurs with selective-repeat if the window

size is too large

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SelectiveSelective--Reject and Small Window SizeReject and Small Window Size

•Window size max. half the sequence number max.

– station A sends frames 0 to 6 to station B

– station B receives all 7 frames and cummulatively

acknowledges with RR 7

– Because of noise RR 7 is lost

– A times out and retransmits frame 0

– B has already advanced its receive window to accept frames

7, 0, 1, 2, 3, 4, 5. Thus it assumes that frame 7 has been lostand that this is frame 0, which it accepts

The problem with this scenario is that there is an overlap between

the sending and receiving windows. To overcome the problem the

window size should be no more than half the sequence numbers.

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SelectiveSelective--Reject ARQReject ARQ

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Utilisation for various ARQ SchemesUtilisation for various ARQ Schemes

((PPbb=10=10--33))

a

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FramingFraming

• Information in data and computer communication

links is typically send in chunks of finite size called

packets or frames

• The task of framing is to flag start and end of a frame

so that the receiving end can identify where

successive frames start and end

• Three protocols are in use for framing

– character oriented framing

– bit oriented framing

– length oriented frame

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CharacterCharacter--based Framingbased Framing

• Character codes such as ASCII provide binary

representation of communication control characters

• SYN (synchronous idle) is such character that is used

when DLC has nothing to send

• STX (start of text) and ETX (end of text) used to

indicate start and end of a frame

• Practical character-oriented framing protocols suchas IBM binary synchronous communication system

(BSC) are more complex

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CharacterCharacter--based Framebased Frame -- ExampleExample

SYN SYN STX Header Packet ETX CRC SYN

Frame

SYN = Synchronous Idle

STX = Start of Text

ETX = End of Text

CRC = Cyclic Redundancy Check 

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Transparent ModeTransparent Mode

• The DLE (data link escape) character is inserted to

indicate start of transparent mode

• DLE is inserted before STX to indicate start of a

frame

• DLE not inserted if STX or ETX are part of the data

field

• DLE also inserted to indicate appearacne of DLE indata field

• DLE STX is start of frame

• DLE DLE STX is appereance of DLE STX in data

field

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BitBit--oriented Framingoriented Framing

• In bit oriented framing a flag, that is a known sequence of bits,

marks the start and end of a frame

• Typically, the flag is encoded as 01111110

• In order to avoid having the sequence 01111110 within the data

field, bit stuffing is used. Bit stuffing inserts a 0 after each

sequence of five 1s. The receiver deletes the zero after a string

of five 1s. If a 1 follows a sequence of five 1s, the frame is

declared to be finished

• Some DLC implementations use a sequence of seven 1s as an

abnormal termination of a frame and a sequence of 15 1s

indicates that the link is idle.

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Frame SizesFrame Sizes

• Two frame size options are possible

– fixed size frames

– variable length frames

• Fixed size frames

– Since not all packet sizes are constant, the frame’s data field

needs some additional bits, called fill, to bring it up to

required length at all times

– Problem here is to determine where data ends and fill starts

• Variable length frames

– require length field that indicates length of packet based on

multiples of octets

– Overhead similar to overhead due to bit stuffing

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DLC Protocols for link initialisationDLC Protocols for link initialisation

• Two typical protocols for DLC link initialisation

– Master-Slave protocol for link initialisation

– Balanced protocol for link initialisation

• Master-Slave Protocol

– One node is master and the other slave during initialisation

– Inititialise and disconnect frames and their

acknowlegements are sent accoring to the stop-and-wait

ARQ protocol

• Balanced Protocol

– both nodes can be master and slave at the same time

– the balanced protocol consist of two consecutively,

synchronised running master-slave protocols

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MasterMaster--Slave ProtocolSlave Protocol

INIT

ACKI

DISC

ACKD

INIT

Initiating Up Disconnecting Down Initiating

Up Down

Data B → A link free of data

Data A → B link free of data

Node A

Node B

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Balanced ProtocolBalanced Protocol

INIT

ACKD

INIT

ACKI

DISC

ACKI

DISC

ACKD

ACKI

Node A

Node B

ACKI ACKD

INIT

ACKD

ACKI

INIT

ACKD

Up

Up

Down

Down

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Switching in Telecommunication NetworksSwitching in Telecommunication Networks

• Switching was created for the first telephone networks• Office switch with a telephone operator (telephonist)

• Automatic switching introduced by Strowger

• Strowger switch (step-by-step switching) first circuit

switching

• Telephone networks use circuit switching

• Data and computer networks use packet switching

(sometimes called cell switching)

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Switching NetworksSwitching Networks

B

A

C

D

F

E

1

4

2

3

5

6

7

End station

Communicating network node

For transmission of data beyond a local area, communication is typically achieved by

transmitting data from source to destination through a network of intermediate

switching nodes. This switched-network design is sometimes used to implement LANs

and MANs as well. The switching nodes are not concerned with the content of the data

but rather their purpose is to provide a switching facility that will move the data from

node to node until they reach their destination.The figure in the slide above illustrates a simple switching network. The end devices

that wish to communicate may be referred to as stations. The stations may be

computers, terminals, telephones, or other communicating devices. We will refer to the

switching devices whose purpose it to provide communications as nodes, which are

connected to each other in some topology by transmission links. Each station attaches

to a node, and the collection of nodes is referred to as a communications network . The

type of network, a wide area network , discussed here, is also referred to as switched 

communication network .

Data entering the network from a station are routed to the destination by being

switched from node to node. For example, data from station A intended for station Fare sent to node 4. Data may then be routed via nodes 5 and 7 or nodes 6 and 7 to the

destination.

Two quite different technologies are used in wide-are switched networks:

• circuit switching and

• packet switching.

These technologies differ in the way the nodes switch data from one link to another on

the route from source to destination.

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Circuit Switching NetworksCircuit Switching Networks

• Dedicated connection path between two stations• One logical connection on each physical connection

• Three phases

– Circuit establishment

– Data transfer

– Circuit disconnection

Circuit establishment: Before any signals can be transmitted, an end-to-end (station-to-

station) circuit must be established. For example, station A sends a request to node 4

requesting a connection to station E. Typically, the link from A to 4 is a dedicated line, so

that part of the connection already exists. Node 4 must find the next leg in a route leading

to node 7. Based on routing information and measures of availability and perhaps cost,

node 4 selects the link to node 5, allocates a free channel, using FDM or TDM, on thatlink and sends a message requesting connection to E. So far, a dedicated path has been

established from A through 4 to 5. Because a number of stations may be attached to 4, it

must be able to establish internal paths from multiple stations to multiple nodes. The

remainder of the process proceeds similarly. Node 5 dedicates a channel to node 7 and

internally ties that channel to the channel from node 4. Node 7 completes the connection

to station E. In completing the connection, a test is made to determine if E is busy or is

prepared to accept the connection.

Data transfer: Information can now be transmitted from A through the network to E. The

data may be analog or digital, depending on the nature of the network. As networks

evolve to fully integrated digital networks, the use of digital (binary) transmission forboth voice and data is becoming the dominant method. Generally, the connection is full-

duplex.

Circuit disconnection: After some period of data transfer, the connection is terminated,

usually by the action of one of the two stations. Signals must be propagated to nodes 4, 5,

and 7 to de-allocate the dedicated channel resources.

Circuit switching can be rather inefficient. Channel capacity is dedicated for the duration

of a connection, even if no data is being transferred. For a voice connection, utilisation

may be rather high, but still is well below 100%. For terminal-to-computer connection,

the capacity may be idle during most of the time. However, after circuit establishment,

the network is virtually transparent to the user and delay is at a minimum with only signalpropagation delays.

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Public Circuit Switched Network Public Circuit Switched Network 

• A public telecommunication network can be described

by

– Subscribers

– Local Loop

– Exchanges (switches)

– Trunks

• Example networks are

– Public switched telephone network (PSTN)

– Private (automatic) branch exchange (PABX)

The best known example of a circuit-switched network is the public telephone network.

This is actually a collection of one or more national networks interconnected to form a

global service. Although originally designed and implemented to service analog

telephone subscribers, the network handles an ever increasing amount of data traffic

via modem and is gradually being converted to a fully digital network. Another well

known application of circuit-switching is the private (automatic) branch exchange(PABX), used to interconnect telephones within a building of offices.

A public telecommunications network consists of four generic architectural

components:

Subscribers: The devices that attach to the network. It is still the case that most

subscriber devices to public telecommunications networks are telephones, but the

percentage of data traffic is exponentially increasing.

Local loop: The link between the subscriber and the network, also referred to as the

subscriber loop. Almost all local loop connections use twisted pair wire. The length of 

a local loop is typically in the range from a few kilometres to a few tens of kilometres.

Often multiplexing points are used in order to bundle individual links.

Exchanges (switches): The switching centres in the network. A switching centre that

directly supports subscribers is know as end office or local exchange (LE). Typically, a

local exchange will support up to a few thousand subscribers in a localised area. There

are many hundreds of local exchanges across Ireland, so that it is impractical for each

LE to have a direct link to each of the other LEs across the country. Rather

intermediate switching nodes, called trunk exchange, are used. Switches that represent

nodes that connect only trunk exchanges are often called tandem switch.

Trunks: The branches between exchanges. Trunks carry multiple voice-

frequency circuits using either FDM or synchronous TDM.

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Public Circuit Switched Network Public Circuit Switched Network 

Exchanges

(Switches)

Local Loop

Subscriber

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Switching ConceptsSwitching Concepts

• Elements of a modernswitching node

– Digital switch

– Network interface

– Control unit

• Switching techniques

– Space division switching

– Time division switching

Control unit

Digital switch

Network 

interface

   F  u   l   l  -   d  u  p   l  e  x   l   i  n  e  s   t  o  a   t   t  a  c   h  e   d   d  e  v   i  c  e  s

At the heard of a modern switching node is a digital switch. The function of the digital

switch is to provide a transparent signal path between any pair of attached devices. The

path is transparent in that it appears to the attached pair of devices that there is a direct

connection between them. Typically, the connection must allow full-duplex

transmission.

The network interface element represents the functions and hardware needed toconnect digital devices, such as data processing devices and digital telephones, to the

network. Analog telephones can also be attached if the network interface contains

analog to digital conversion logic. Trunks to other digital switches carry TDM signals

and provide the links for constructing multiple node networks.

The control unit performs three general tasks. First it establishes connections, secondly,

the control unit must maintain the connection. Because the digital switch uses time-

division principles this may require ongoing manipulation of the switching elements.

Third the control unit must tear down the connection, either in response to a request

from one of the parties or for its own reasons.

An important characteristic of a circuit-switching device is whether it is blocking or

non-blocking. Blocking occurs when the network is unable to connect two stations

because all possible paths between them are already in use. A blocking network is one

in which such blocking is possible. Hence, a non-blocking network permits all stations

to be connected (in pairs) at once and grants all possible connection requests as long as

the called party is free.

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Space Division SwitchingSpace Division Switching

Space division switch

(Crossbar Switch)

First stage Second stage Third stage

2x2 switch

5x2 switch

2x5 switch

Three stage space division switch

Space division switching was originally developed for the analog environment and has been

carried over into the digital domain. The fundamental principles are the same, whether the

switch is used to carry analog or digital signals. As its name implies, a space division switch

is one where the signals paths are physically separated from one another. Each connection

requires the establishment of a physical path through the switch solely used to the transfer of 

signals between the two endpoints.The left figure in the slide above shows a simple crossbar matrix with 5 full-duplex I/O

lines. The matrix has 5 inputs and 5 outputs. Interconnection is possible between any two

lines by enabling the appropriate crosspoint. Therefore, N I/O lines require N2 crosspoints.

This indicates the limitations of the crossbar switch:

• number of crosspoints grows with the square of the number of I/O lines

• the loss of a crosspoint prevents connection between two particular devices

• the crosspoints are inefficiently utilised. Even if all connections are active, only a fraction

of all crosspoints is used.

To overcome these limitations, multiple-stage switches are employed. The right figure inthe above slides shows a three stage switch. This arrangement has several advantages

• the number of crosspoints is reduced, increasing crossbar utilisation.

• There is more than one path through the switch to connect two endpoints, increasing

reliability. Of course, a multiple stage crossbar switch requires a more complex control unit.

A consideration with a multistage space division switch is that it may be blocking. It should

be clear from the figure above that a single-stage crossbar switch is non-blocking.

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Time Division SwitchingTime Division Switching

Control unit

Network 

interface

   F  u   l   l  -   d  u  p   l  e  x   l   i  n  e  s   t  o  a   t   t  a  c   h  e   d   d  e  v   i  c  e  s

TDM bus switch

1

2

3

4

5

Control

logic

2→ 5

4→ 6

3→ 1

1→ 3

5→ 2

6

6→ 4

Control of a TDM bus switch

Virtually all modern switches use digital time division techniques for establishing and

maintaining “circuits”. Time-division switching uses input lines based on synchronous

time division multiplexing. The slots on the TDM line are manipulated by the control

logic to route data from an input line to the dedicated output line. There are a number

of variations on this basic concept. Here only the concept of TDM bus switching is

examined.The left figure in the above slide shows how TDM can be extended to provide a

switching functionality. Each device attaches to the switch through a full-duplex line.

These lines are connected through controlled gates to a high-speed digital bus. Each

line is assigned a time slot for providing input. For the duration of the slot, that line’s

gate is enabled, allowing s small burst of data onto the bus. For that same time slot one

of the other gates is enabled for output. During successive time slots, different

input/output pairings are enabled, allowing a number of connections to be carried over

a shared bus.

The right figure in the above slide is an example that suggests how the control for a

TDM bus switch can be implemented. Lets assume that the propagation time on the busis 0.01µsec. Time on the bus is organised into 30.06µsec frames of six 5.01µsec slots

each. A control memory indicates which gates are to enabled during each time slot. In

this example, size words of memory are needed. A controller cycles through the

memory at a rate of one cycle every 30.06µsec.

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Control SignallingControl Signalling

• Means by which network is managed and connections are established,

maintained and terminated

• Control signalling functions

– Audible communication with subscriber - ringing one, dialling tone, busy

signal, etc.

– Transmission of number dialled to switches for attempt to complete connection

– Transmission of information between switches indicating that call cannot be

completed

– Transmission of information between switches indicating that call has ended

– A signal to make telephone ring

– Transmission of information for billing purposes

– Transmission of information indicating status or equipment and trunks

– Transmission of information for diagnosing and isolating system faults

– Control of special equipment such as satellite channel equipment

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Signalling in CircuitSignalling in Circuit--Switched NetworksSwitched Networks

Description Comment

In-channel

Inband Transmit control sign als The simp lest technique. It is

in the same band of frequencies necessary for call info signals and

used by the voice signals may be used for other control

signals. Inband can be used over

any type of subscriber interface

Out-of-band Transmit control signals over Unlike inband, out-of-band provides

the same facilities as voice signals continuous supervision for the

but in a different part of the duration of the connection

frequency band

Common Channel Transmit control signals over Reduces call setup time compared

signall ing channels that are with in-channel methods. It is also

dedicated to control signals more adaptable to evolvingand are common to a number funct ional needs.

of voice channels.

Control signalling needs to be considered in two contexts: signalling between a

subscriber and the network and signalling within the network. Typically, signalling

operates differently within these two contexts. The signalling between a telephone or

other subscriber device and the local exchange to which it attaches is, to a large extend,

determined by the characteristics of the subscriber device and the needs of the human

user. Signals within the networks are entirely computer to computer. The internalsignalling is concerned not only with the management of the subscriber calls but with

the management of the network itself. Therefore, mapping between the less complex

subscriber signalling techniques and the more complex network signalling techniques

must be possible.

Traditional control signalling in circuit-switched networks has been on a per-trunk or

in-channel basis. With in-channel signalling, the same channel is used to carry control

signals and the call the control signals relate to. Such signalling originates at the

subscriber and follows the same path as the call itself. Two forms of in-channel

signalling are in use: inband and out-of-band.

 Inband signalling uses the same frequency band as the call and the signals have thesame electromagnetic properties. Due to this method the information that can be

carried is very limited. However, an advantage is that it is impossible to setup a call on

a faulty speech path. Out-of-band  signalling uses a narrow band within the 4kHz

speech band that is not used by speech. Signalling is possible whether voice signals are

on the line or not and thus continuous supervision of a call is possible. However, an

out-of-band scheme needs extra electronics to handle the signalling band.

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Common Channel SignallingCommon Channel Signalling

Associated signalling

Non-associated signalling

Speech

Signalling

Switching points

Signalling transfer points

As public network become more complex and provide a richer set of services, the

drawbacks of in-channel signalling become more apparent. The information transfer

rate is quite limited and with inband signalling only available if there are no voice

signals on the circuit. Out-of-band signalling provides only a very limited bandwidth.

With these limitations it is difficult to provide more complex control messages in order

to manage the increasing complexity of evolving network technology. A morepowerful approach is required. This approach is based on common channel signalling.

In this approach the signalling path is physically distinct from the path for voice and

other subscriber signals. The common channel can be configured with the bandwidth

required to carry control signals for a rich variety of functions. With dropping costs for

hardware this concept has become so attractive that it is being introduced in all public

telecommunication networks. The control signals are messages passed between

switches as wells as between a switch and the network management centre. Thus, the

control-signalling portion of the network is a distributed computer network carrying

short messages.

Two modes of operation are used, the associated mode and the non-associated mode.In the associate mode (shown above) the common channel closely tracks along the

entire length of the inter-switch trunks. The non-associated mode is more complex, but

more powerful; with this the network is augmented by additional nodes, known as

signal transfer points. There is now no close or simple assignment of control channels

to trunk groups. In effect, there are now two separate networks, with links between

them so that the control portion of the network can exercise control over the switching

nodes that carry the subscriber calls. This mode is used in modern telecommunication

networks and the control signalling architecture is called Common Channel Signalling

System No. 7 (SS#7).

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Packet SwitchingPacket Switching

• Data are transmitted in short packets (≤ 1000bytes)• Packet consists of control and data part

– Control part contains address

• Two approaches to switching

– datagrams

– virtual circuits

A key characteristic of circuit-switched networks is that resources within the network are

dedicated to a particular call. For voice connections, the resulting circuit will enjoy a high

percentage of utilisation because most of the time one party or the other is talking.

However, as the circuit-switching network is more and more utilised for data

transmission, two shortcomings become apparent

• In a typical user/host data connection much of the time the line is idle. Therefore, theresource usage is inefficient.

• In a circuit-switching network the connections provide for transmission at constant data

rate. Thus, each of the two devices must transmit and receive at the same data rate as the

other; this limits the utility of the network in interconnecting a variety of different

computing devices and terminals.

Packet switching addresses these problems by transmitting data in short packets of 

usually no more than about 1000 bytes. If a data stream is longer it will be broken up into

a series of packets, each packet containing a portion of the overall data stream. A packet

also contains some control information. The control information, at a minimum, includes

the information that the network requires in order to be able to route a packet through the

network and deliver it to the destination. At each node en route, the packet is is received,

stored briefly, and passed on to the next node.

This approach has a number of advantages over circuit switching

• Line efficiency is greater as a single node-to-node link can be dynamically shared by

many packets over time. Packets are queued up and transmitted as rapidly as possible

over the link. By contrast, with circuit switching, time on a node is pre-allocated using

synchronous TDM.

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• A packet switching network can perform data-rate conversion. Two stations of 

different data rates can exchange packets because each connects to its node at its

proper data rate.

• When traffic become heavy on a circuit-switching network, some calls are blocked;that is the network refuses to accept additional connection requests until the load

decreases. On a packet-switching network, packets are still accepted but delivery delay

increases

• Packets can be transmitted with different priorities attached. Thus, if a node has a

number of packets queued for transmission, it can transmit higher-priority packets first.

These packets will therefore experience less delay than lower-priority packets.

The key question is now how a packet-switching network attempts to transmit a

sequence of packets from source to destination. Two approaches are used in

contemporary networks: datagram and virtual circuit.

In the datagram approach each packet is treated independently, with no reference to

packets that have already been transmitted. The implications of this technique are that a

routing decision has to be made for each packet. Therefore, each packet must contain

the full address of its detstination. It is possible that packets get lost somewhere along

the path and that packets arrive out of sequence if routed on a path where transmission

takes longer than on others. The advantage of this approach is that no connection

establishment is required and in cases where only a short message needs to be sent this

approach achieves fast transmission. Each packet is referred to as a datagram. This

technique is commonly used in the Internet.

In the virtual circuit approach, a pre-planned route is established before any packets are

sent. Because the route is fixed for the duration of the transmission, it is somewhat

similar to a circuit in a circuit-switching network, and is referred to as a virtual circuit.

Each packet now contains a virtual-circuit identifier as well as data. Each node on the

pre-established route knows where to direct such packets; no routing decisions are

required. The pre-establishment of the route does not mean that this is a dedicated path.

A packet is still buffered at each node, and queued for output over a line. The

difference from the datagram approach is that the node need not make a routing

decision for each packet; it is made only once for all packets using the virtual circuit.

The advantages of the virtual circuit approach are that services such as sequencing and

error control can be associated with a virtual circuit. The virtual circuit approach alsoallows to control the load in the network better as the number of virtual circuits per line

can be limited and thus the potential load per line.

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Packet SizePacket Size

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Comparison of Switching TechniquesComparison of Switching Techniques

Circuit switching Datagram packet switching Virtual-circuit packet switching

Dedicated transmission path No dedicated path No dedicated path

Continuous transmission of data Transmission of packet Transmission of packets

Fast enough for interactive Fast enough for interactive Fast enough for interactive

Message are not stored Packets may be stored unti l de livered Packets stored until delivered

Path established for entire call Route established for each packet Route established for entire call

Call setup delay; n egligib le Packet transmission delay Call setup delay; p acket transmission

transmission delay delay

Busy signal if called party busy Sender may be notified if packet Sender notified of connection denial

not delivered

Overload may block call setup; no Overload increases packet delay Overload may block call setup;

delays for established calls increases packet delay

Electromechanical or computerised Small switching nodes (computers) Small switching nodes (computers)

switching

User responsible for message loss Network may be responsible for Network may be responsible for packet

protection individual packets sequencesUsually no speed or code Speed and code conversion Speed and code conversion

conversion

Fixed bandwidth transmission Dynamic use of bandwidth Dynamic use of bandwidth

No overhead bits after call setup Overhead bits in each message Overhead bit in each packet

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Routing in Wide Area NetworksRouting in Wide Area Networks

• Determining a path from source to destination withrespect to certain criteria

• Typically:routing along the best path in terms of 

– smallest number of hops

– smallest delay (desirable in most cases)

• Routing based on smallest delay influenced by

– Packet transmission time

– Queuing and processing delay

The goal of all routing procedures is to determine the best path in terms of the smallest

number of hops and/or the smallest delay. The smallest delay is typically desirable in

most networks, particularly in packet switched networks, as the queuing delay takes up

the bulk of the delay. However, it is often impossible to determine the delay, in

particular the queuing delay as it depends on the load of the nodes and the network.

Most routing algorithms attempt to route the packet over the best guess path. This isachieved by assigning fixed or varialbe cost to a path. Approaches to determining link 

cost are

• unit cost, which delivers minimum hop path

• cost inverse to link data rate, which yields the minimum transmission time and

achieves load balancing

• cost equal to average delay experienced, which is estimated over some interval

• two costs, low cost when queue is below a threshold, high cost, when queue length

grows beyond a certain bound.

For virtual circuit routing, a path is chosen at setup time. Although the path chosenmay provide minimum delay at setup time, there is no guarantee that this state will

prevail throughout the duration of the connection.

For datagram routing, routing decisions may be made for each packet individually, and

thus the ideal of minimum delay is closer to fulfillment.

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Routing in Wide Area NetworksRouting in Wide Area Networks

• Requirements for routing strategies– Correctness

– Simplicity

– Robustness

– Stability

– Fairness

– Optimality

– Efficiency

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Elements of Routing TechniquesElements of Routing Techniques

• Performance Criteria

– Number of Hops

– Cost

– Delay

– Throughput

• Decision time

– Packet (Datagram)

– Session (virtual circuit)

• Decision place

– Each node (distributed)

– Central node (centralised)

– Originating node (source)

• Network information

source

– None

– Local

– Adjacent node

– Nodes along route

– All nodes

• Network information

update timing

– Continuous

– Periodic

– Major load change– Topology change

The selection of a route is generally based on some performance criterion. The simplest

criterion is to choose the minimum-hop route through the network. This is an easily measured

criterion and should minimise the consumption of network resources. A generalisation of the

minimum-hop criterion is the least-cost routing. In this case a cost is associated with each link 

and the optimal route is the one that has the least cost associated with it.

Routing decisions are made on the basis of some performance criterion. Two key characteristicsof the decision are the time and the place that the decision is made. Decision time is based on

whether the decision is made on a per packet basis (datagram approach) or on a virtual circuit

basis. The term decision place refers to which node or nodes in the network are responsible for

the routing decision. Most common is distributed routing, in which each node has the

responsibility of selecting an output link for routing packets as they arrive. For centralised

routing, the decision is made by some designated node, such as a network control centre. The

last approach is source routing. This allows the user to decide upon a route that meets criteria

local to that user. The decision time and place are independent design variables for packet

switching networks.

Most routing strategies require that decisions be based on knowledge of the topology of thenetwork, traffic load, and link cost. Surprisingly, some strategies use no such information and

yet manage to get packets through.; flooding and some random strategies are in this category.

With distributed routing the individual node may make use of local information such as the cost

of each outgoing link. Each node might also collect information from adjacent nodes such as

the state of congestion they are currently experiencing.

Information update timing is a function of both the information source and the routing strategy.

If no information is used there is no requirement for updating. If only local updating is used the

update timing is essentially continuous. For a fixed routing strategy information is never

updated and for an adaptive strategy it is updated from time to time.

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Routing in CircuitRouting in Circuit--Switched NetworksSwitched Networks

• Routing finds a path through more than one switchingnode depending on a certain set of criteria

• Alternate routing

– Used in SS7 networks

– Fixed alternate routing

– dynamic alternate routing

• multi-alternate routing

• dynamic non-hierarchical routing

• Adaptive routing

– Dynamic traffic management

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Routing Strategies for PacketRouting Strategies for Packet--SwitchedSwitched

NetworksNetworks

• Fixed routing

• Flooding

• Random routing

• Adaptive routing

– failure

– congestion

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Example Routing AlgorithmsExample Routing Algorithms

• Least Cost Algorithms– Most routing algorithms are based on two basic least-cost

algorithms

– Dijkstra’s Algorithm

– Bellman-Ford Algorithm

• Principle

– Given a network of nodes connected by bidirectional links,

where each link has a cost associated with it in each

direction, define the cost of a path between two nodes as the

sum of the costs of the links traversed. For each pair of nodes, find the path with the least cost.

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Example PacketExample Packet--Switched Network Switched Network 

1

2 3

4 5

6

8

5

2

3

1

7

6

35

8

22

2

1

1

11

33

4

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Dijkstra’sDijkstra’s AlgorithmAlgorithm

• Find the shortest paths from a given source node to all other

nodes, by developing the paths in order of increasing path

length.• The algorithm has proceeds in stages and by the kth stage the

shortest paths to the k nodes closest to the source node have

been determined

• Algorithm is defined formally as

N = set of nodes in the network 

s = source node

T = set of nodes so far incorporated by the algorithm

w(i, j) = link cost from node i to node j; w(i, i) = 0; w(i, j) = ∞ if 

nodes are not directly connected

L(n) = cost of the least-cost path from node s to n that is currently

known to algorithm

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Dijkstra’sDijkstra’s AlgorithmAlgorithm

• Initialisation

T = {s}, i.e. set of nodes so far incorporated consists of only the source node

L(n) = w(i, j) for n ≠s, i.e. the initial path cost are simply the link costs

• Get Next Node

Find the neighbouring node not in T that has the least-cost path from node s

and incorporate that node into T; Also incorporate the edge that is incident

on that node and a node in T the contributes to the path

Find

Add x to T; add to T the edge that is incident on x to L(x)

• Update Least-Cost Paths

If the latter term is the minimum, the path from s to n is now the path from s to

x concatenated with the edge from x to n

The final edges are called the spanning tree of the network 

(((( )))) (((( )))) j L x LT  xT  j∉∉∉∉

====∉∉∉∉ minsuch that

(((( )))) (((( )))) (((( )))) (((( ))))[[[[ ]]]] T nn xw x Ln Ln L ∈∈∈∈++++====  allfor,,,min

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Dijkstra’sDijkstra’s AlgorithmsAlgorithms

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BellmanBellman--Ford AlgorithmFord Algorithm

• Find the shortest paths from a given source node subject to theconstraint that the paths contain at most one link; then find the

shortest paths with a constraint of paths of at most two links

and so on.

• The algorithm is defined formally as

s = source node

w(i, j) = link cost from node i to j; w(i, i) = 0; w(i, j) = ∞ if nodes

are not directly connected

h = maximum number of links in a path at the current stage of the

algorithm

Lh(n) = cost of the least-cost path from node s to node n under the

constraint of no more than h links

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BellmanBellman--Ford AlgorithmFord Algorithm

• Initialisation

L0

(n) = ∞, for all n ≠ s

Lh(s) = 0; for all h

• Update

For each successive h ≥ 0:

For each n ≠ s, compute

Connect n with the predecessor node j that acheves the minimum, and

eliminate any connection of n with a different predecessor node formed

during an earlier iteration. The path from s to n terminates with the link 

from j to n.

(((( )))) (((( )))) (((( ))))[[[[ ]]]]n jw j Ln L h j

h ,min1 ++++====++++

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BellmanBellman--Ford AlgorithmFord Algorithm

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Cost Function forCost Function for DijkstraDijkstra and BFand BF

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Traffic and Congestion ControlTraffic and Congestion Control

• Communication networks are designed for a particular

traffic load

• If traffic load increases beyond a certain point,

congestion occurs

• Traffic management is used to avoid congestion

• Congestion control is used to resolve a state of 

congestion

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Congestion ControlCongestion Control

• Packet-switching network is network of queues that

may become highly loaded or overloaded

• High load or overload situation needs to be controlled

– choke source by dedicated control packet

– use routing information to influence packet generation rate

– Use probe packet to find least congested route

– add congestion information to packets

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Traffic Management/ControlTraffic Management/Control

• Traffic control is also often called flow control• Traffic management is mainly used in packet-

switched networks

• Traffic management in packet-switched networks tries

to avoid congestion

• Traffic control in circuit-switched networks is based

on call blocking

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Objectives of Traffic ControlObjectives of Traffic Control

• Limiting delay

• Limiting buffer overflow

• Fairness

For real-time applications, such as voice and video transmission, excessively

delayed packets are useless, as they would reduce the quality of the

applications significantly. For such applications, a limited delay is essential

and should be the chief concern of traffic management algorithms; for

example such applications may be given a high priority for transmission.

For other applications, a small average delay per packet is desirable but itmay not be crucial. For these applications, the network layer traffic

management does not necessarily reduce delay; it simply shifts the delay

from the network layer to the higher layers. That is, by restricting entrance

into the subnet, traffic management keeps packets waiting outside the subnet

rather than in the queues of the subnet. In this way, traffic management

avoids wasting subnet resources in packet retransmissions and helps prevent

a disastrous traffic jam inside the subnet. Retransmission in this scenario can

occur in two ways:

• build-up of queues causes buffer overflow and packets are discarded,

• slow acknowledgements, due to excessive delays, can cause the source to

retransmit packets because it mistakenly thinks that packets were lost.

In certain cases sessions generating packets at high rate can capture almost

buffer space and exclude slow rate source from transmission. In order to

prevent this from happening, a buffer management scheme needs to be

implemented. In such a scheme packets are divided into different classes. At

each node separate buffer space is reserved for different classes, while some

buffer space is shared by all classes.

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When offered traffic must be cut back in order to avoid congestion, it must be

done fairly. The notion of fairness is complicated, however, by the presence

of different session (connection) priorities and service requirements. Forexample, some sessions need a minimum guaranteed rate and a strict upper

bound on network delay. Thus, while it is appropriate to consider simple

notions of fairness within classes of similar sessions, the notion of fairness

between classes is complex and involves the requirements of those classes. In

general, real-time sessions would be favoured with respect to delay but may

have to suffer some loss of packets whereas data sessions would have to

suffer more delay at the source but the network would make sure that no loss

occurs. However, it can easily be anticipated that fairness is a complex issue

which can result in a multi-dimensional optimisation problem in order to

achieve optimum fairness.

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Functions of Traffic ControlFunctions of Traffic Control

• Call or packet blocking (admission control)

• Packet scheduling - Window flow control

• Source rate control (traffic shaping)

• Network resource allocation

Call or packet blocking is regulated by a traffic management function that is called

admission control. Admission control allows or denies admission to the network based

on whether the parameters that the connection requires can be fulfilled. Parameters in

this case are average and peak data rate, packet delay variation, packet loss rate, etc.

Packet scheduling is facilitated by a window based flow control mechanisms in much

the same way as is used in data link layer protocols such as HDLC. However, as can beseen in HDLC, this kind of flow control is not well suited to high-speed transmission

since it would require large window sizes to make use of a high data rate. It is also not

well suited to wide area networks were propagation delays are large and waiting for

acknowledgement packets reduces the throughput. A further problem is that window

based mechanisms do not regulate the end-to-end delay well and do not guarantee a

minimum data rate, which is important for the transmission of real-time services such

as voice and video.

Another form of traffic management more suited to high-speed transmission lines is

rate control. This form of traffic management gives each session or connection a

guaranteed data rate, which is commensurate to its needs. This rate should lie withincertain limits that depend on the session type.

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The main considerations in setting source rates are:

• Delay-throughput trade-off - increasing throughput by setting the rates too

high runs the risk of buffer overflow and excessive delay

• Fairness - if session rates must be reduced to accommodate some new

sessions, the rate reduction must be done fairly, while obeying the minimum

rate requirement for each session.

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Leaky Bucket SchemeLeaky Bucket Scheme

Arriving permits at a rate of one

per 1/r sec (turned away if there is

no space in the permit queue)

Permit queue (limited space W)

Arriving

packets

Queue with packets

without a permit

Queue of packets

with a permit

In order to implement a session rate of  r packets/sec one could admit only

one packet every 1/r  seconds. This, however, amounts to a form of time

division multiplexing and amounts to large delays when the traffic load is

bursty. A more appropriate implementation is to admit as many as W packets

(W > 1) every W/r seconds. This allows a burst of as many as W packets into

the network without delay, and is better suited for a dynamically changingload. This approach achieves some sort of traffic smoothing and reduces the

burstiness for which TDM causes long delays.

An implementation of this kind of traffic management mechanisms is the so

called leaky bucket scheme. An allocation of  W  packets is given to each

session, and a count  x of the unused portion of this allocation is kept at the

source. Packets from the session are admitted to the network as long as x > 0.

In the leaky bucket scheme the count is incremented periodically, every 1/r 

seconds, up to a maximum of W packets. Another way to view this scheme is

to imagine that for each session, there is a queue of packets without a permit

and a bucket or permits at the session’s source. The packet at the head of thepacket queue obtains a permit once one is available in the permit bucket and

then joins the set of packets with permits waiting to be transmitted (see figure

in slide above). Permits are generated at the desired input rate r of the session

(one permit every 1/r seconds) as long as the number is the permit bucket

does not exceed a certain threshold W . The leaky bucket scheme is used in

ATM networks to shape the source data rate such that it maintains the

parameters of the agreed traffic contract.

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Congestion ControlCongestion Control

No congestion

Mild

congestion

Severe

congestion

No congestion

Mild

congestion

Severe

congestion

Offered load

Offered load

   T   h  r  o  u  g   h  p  u   t

   D  e   l  a  y

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Functions of Congestion ControlFunctions of Congestion Control

• When congestion occurs, one or more of the followingfunctions are used to resolve congestion

– Discard packets

– Send control packet

– Use routing information

– Use end-to-end probe packet

– Add congestion information to packets