telecomms concepts
TRANSCRIPT
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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