Network goals
1.
2.
3.
4.
5.
The main goal of networking is "Resource sharing", and it is to make all programs, data
and equipment available to anyone on the network with out the regard to the physical
location of the resource and the user.
A second goal is to provide high reliability by having alternative sources of supply. For
example, all files could be replicated on two or three machines, so if one of them is
unavailable, the other copies could be available.
Another goal is saving money. Small computers have a much better price/performance
ratio than larger ones. Mainframes are roughly a factor of ten times faster than the fastest
single chip microprocessors, but they cost thousand times more. This imbalance has
caused many system designers to build systems consisting of powerful personal
computers, one per user, with data kept on one or more shared file server machines. This
goal leads to networks with many computers located in the same building. Such a
network is called a LAN (local area network).
Another closely related goal is to increase the systems performance as the work load
increases by just adding more processors. With central mainframes, when the system is
full, it must be replaced by a larger one, usually at great expense and with even greater
disruption to the users.
Computer networks provide a powerful communication medium. A file that was
updated/modified on a network can be seen by the other users on the network
immediately.
Topologies
Bus network
Mesh network
Ring network
Star network
Tree network
A computer network is made of computers which are linked to one another with communication
lines (network cables, etc.) and hardware elements (network adapters, as well as other equipment
for ensuring that data travels correctly). The physical arrangement — that is, the spatial
configuration of the network — is called the physical topology. The different kinds of topology
are:
1.
2.
3.
4.
5.
Bus topology
Star topology
Ring topology
Tree topology
Mesh topology
The logical topology, as opposed to the physical topology, refers to way that data travels along
communication lines. The most common logical topologies are Ethernet, Token Ring and FDDI.
1.
Bus topology
Bus topology is the simplest way a network can be organized. In bus topology, all computers are
linked to the same transmission line by using a cable, usually coaxial. The word "bus" refers to
the physical line that joins all the machines on the network.
The advantages of this topology are that it is easy to implement and functions easily; on the other
hand, it is highly vulnerable, since if one of the connections is defective, the whole network is
affected.
Advantages:
Any one computer or device being down does not affect the others.
Disadvantages: Can't connect a large number of computers this way. It's physically difficult to
run the one communications line over a whole building, for example.
Star topology
In star topology, the network computers are linked to a piece of hardware called a hub. This is a
box which contains a certain number of sockets into which cables coming out of the computers
can be plugged. Its role is to ensure communications between those sockets.
Unlike networks built with bus topology, networks which use star topology are much less
vulnerable, as one of the connections can easily be removed by disconnecting it from the hub,
without paralyzing the rest of the network. The critical point in this network is the hub, as
without it, communication between the computers on the network is no longer possible.
However, a star topology network is bulkier than a bus network, as additional hardware is
required (the hub).
2.
Advantages:
Gives close control of data.
Each PC sees all the data.
User sees up-to-date data always.
If a computer other than the host fails, no other computer is affected.
Disadvantages: If host computer or its software goes down, the whole network is down.
(A backup computer system would be necessary to keep going while
repairs are made.)
3.
Ring topology
In a ring-topology network, computers each take turns communicating, creating a loop of
computers in which they each "have their turn to speak" one after another.
In reality, ring topology networks are not linked together in loops. They are actually
linked to a distributor (called a MAU, Multistation Access Unit) which manages
communication between the computers linked to it, by giving each of them time to
"speak."
The two main logical topologies which use this physical topology are Token ring and
FDDI.
Advantages:
Requires less cabling and so is less expensive.
Disadvantages: If one node goes down, it takes down the whole network.
4.
Mesh Topology
Mesh topologies involve the concept of routes. Unlike each of the previous topologies,
messages sent on a mesh network can take any of several possible paths from source to
destination. (Recall that even in a ring, although two cable paths exist, messages can only
travel in one direction.) Some WANs, most notably the Internet, employ mesh routing.
A mesh network in which every device connects to every other is called a full mesh. As
shown in the illustration below, partial mesh networks also exist in which some devices
connect only indirectly to others.
5.
Tree Topology
Tree topologies integrate multiple star topologies together onto a bus. In its simplest
form, only hub devices connect directly to the tree bus, and each hub functions as the
"root" of a tree of devices. This bus/star hybrid approach supports future expandability of
the network much better than a bus (limited in the number of devices due to the broadcast
traffic it generates) or a star (limited by the number of hub connection points) alone.
Connecting Networks
Networks can be connected to each other, too. There are difficulties in doing so, however. A
combination of software and hardware must be used to do the job.
A gateway connects networks of different kinds, like connecting a network
of PCs to a main frame network. This can be complex!
A bridge connects networks of the same type. This job is simple.
A router connects several networks. A router is smart enough to pick the
right path for communications traffic. If there is a partial failure of the
network, a router looks for an alternate route.
Suppose the accounting, advertising, and shipping departments of a company each have
networks of PCs. These departments need to communicate with each other, but only
sometimes. It would be easier and cheaper to connect them to each other than to put
them all on the same larger network. The best arrangement would be for the
departmental networks to be of the same kind so that a bridge could be used.
Concept of Internet
The Internet is not a network at all, but a vast collection of different networks
that use certain common protocols and provide certain common services.
Architecture of the Internet
A brief overview of the Internet today.
Assume client calls his or her ISP over a dial-up telephone line, as shown in Fig. 1-29. The modem is
a card within the PC that converts the digital signals the computer produces toanalog signals that can pass
unhindered over the telephone system. These signals are transferred to the ISP’s POP (Point of
Presence), where they are removed from the telephone system and injected into the ISP’s regional
network. From this
point on, the system is fully digital and packet switched. If the ISP is the local telco, the POP will
probably be located in the telephone switching office where the telephone wire from the client terminates.
If the ISP is not the local telco, the POP may be a few switching offices down the road. The ISP’s
regional network consists of interconnected routers in the various cities the ISP serves. If the packet is
destined for a host served directly by the ISP, the packet is delivered to the host. Otherwise, it is handed
over to the ISP’s
backbone operator. At the top of the chain are the major backbone operators, companies like AT&T and
Sprint. They operate large international backbone networks, with thousands of routers connected by highbandwidth fiber optics. Large corporations and hosting services that run server farms (machines that can
serve thousands of Web pages per second) often connect directly to the backbone. Backbone operators
encourage this direct connection by renting space in what are called carrier hotels, basically equipment
racks in the same room as the router to allow short, fast connections between server farms and the
backbone. If a packet given to the backbone is destined for an ISP or company served by the backbone, it
is sent to the closest router and handed off there. However, many backbones, of varying sizes, exist in the
world, so a packet may have to go to a competing backbone. To allow packets to hop between backbones,
all the major backbones connect at the NAPs discussed earlier. Basically, a NAP is a room full of routers,
at least one per backbone. A LAN in the room connects all the routers, so packets can be forwarded from
any backbone to any other backbone. In addition to being interconnected at NAPs, the larger backbones
have numerous direct connections between their routers, a technique known as private peering.
The 7 Layers of the ISO-OSI Network Model
1.
Physical Layer
Concerned with the transmission of bits.
How many volts for 0, how many for 1?
Number of bits of second to be transmitted.
Two way or one-way transmission
Standardized protocol dealing with electrical, mechanical and signaling interfaces.
Many standards have been developed, e.g. RS-232 (for serial communication lines).
Example : X.21
2.
Data Link Layer
Handles errors in the physical layer.
Groups bits into frames and ensures their correct delivery.
Adds some bits at the beginning and end of each frame plus the checksum.
Receiver verifies the checksum.
If the checksum is not correct, it asks for retransmission. (send a control message).
Consists of two sublayers:
Logical Link Control (LLC) defines how data is transferred over the cable and provides data
link service to the higher layers.
Medium Access Control (MAC) defines who can use the network when multiple computers are
trying to access it simultaneously (i.e. Token passing, Ethernet [CSMA/CD]).
3.
Network Layer
Concerned with the transmission of packets.
Choose the best path to send a packet ( routing ).
It may be complex in a large network (e.g. Internet).
Shortest (distance) route vs. route with least delay.
Static (long term average) vs. dynamic (current load) routing.
Two protocols are most widely used.
X.25
Connection Oriented
Public networks, telephone, European PTT
Send a call request at the outset to the destination
If destination accepts the connection, it sends an connection identifier
IP (Internet Protocol)
Connectionless
Part of Internet protocol suite.
An IP packet can be sent without a connection being established.
Each packet is routed to its destination independently.
4.
Transport Layer
Network layer does not deal with lost messages.
Transport layer ensures reliable service.
Breaks the message (from sessions layer) into smaller packets, assigns sequence number and
sends them.
Reliable transport connections are built on top of X.25 or IP.
In case IP, lost packets arriving out of order must be reordered.
TCP : (Transport Control Protocol) Internet transport protocol.
TCP/IP Widely used for network/transport layer (UNIX).
UDP (Universal Datagram Protocol) : Internet connectionless transport layer protocol.
Application programs that do not need connection-oriented protocol generally use UDP.
5.
Sessions Layer
Just theory! Very few applications use it.
Enhanced version of transport layer.
Dialog control, synchronization facilities.
Rarely supported (Internet suite does not).
6.
Presentation Layer
Just theory! Very few applications use it.
Concerned with the semantics of the bits.
Define records and fields in them.
Sender can tell the receiver of the format.
Makes machines with different internal representations to communicate.
If implemented, the best layer for cryptography.
7.
Application Layer
Collection of miscellaneous protocols for high level applications
Electronic mail, file transfer, connecting remote terminals, etc.
E.g. SMTP, FTP, Telnet, HTTP, etc
TCP/IP reference model
Network Access Layer – The Network Access Layer is
fairly self explanatory- it interfaces with the physical
network. It formats data and addresses data for subnets,
based on physical hardware addresses. More
importantly, it provides error control for data delivered
on the physical network.



Internet Layer – The Internet Layer provides logical addressing. More specifically, the
internet layer relates physical addresses from the network access layer to logical
addresses. This can be an IP address, for instance. This is vital for passing along
information to subnets that aren’t on the same network as other parts of the network. This
layer also provides routing that may reduce traffic, and supports delivery across an
internetwork. (An internetwork is simply a greater network of LANs, perhaps a large
company or organization.)
Transport Layer – The Transport Layer provides flow control, error control, and serves
as an interface for network applications. An example of the transport layer would be
TCP- a protocol suite that is connection-oriented. We may also use UDP- a
connectionless means of transporting data.
Application Layer – Lastly, we have the Application Layer. We use this layer for
troubleshooting, file transfer, internet activities, and a slew of other activities. This layer
interacts with many types of applications, such as a database manager, email program, or
Telnet.
TCP/IP Model vs OSI Model
Sr.
No.
TCP/IP Reference Model
OSI Reference Model
1
Defined after the advent of Internet.
Defined before advent of internet.
2
Service interface and protocols were not
clearly distinguished before
Service interface and protocols are
clearly distinguished
3
TCP/IP supports Internet working
Internet working not supported
4
Loosely layered
Strict layering
5
Protocol Dependant standard
Protocol independent standard
6
More Credible
Less Credible
7
TCP reliably delivers packets, IP does not
reliably deliver packets
All packets are reliably delivered
TRANSMISSION MEDIA
The means through which data is transformed from one place to another is called transmission or
communication media. There are two categories of transmission media used in computer
communications.
BOUNDED/GUIDED MEDIA
2. UNBOUNDED/UNGUIDED MEDIA
1.
1. BOUNDED MEDIA:
Bounded media are the physical links through which signals are confined to narrow path. These
are also called guide media. Bounded media are made up o a external conductor (Usually
Copper) bounded by jacket material. Bounded media are great for LABS because they offer high
speed, good security and low cast. However, some time they cannot be used due distance
communication. Three common types of bounded media are used of the data transmission. These
are
Coaxial Cable
b) Twisted Pairs Cable
c) Fiber Optics Cable
a)
a) COAXIAL CABLE:
Coaxial cable Coaxial cable is very common & widely used commutation media. For example
TV wire is usually coaxial.
Coaxial cable gets its name because it contains two conductors that are parallel to each other.
The center conductor in the cable is usually copper. The copper can be either a solid wire or
stranded martial.
Outside this central Conductor is a non-conductive material. It is usually white, plastic material
used to separate the inner Conductor form the outer Conductor. The other Conductor is a fine
mesh made from Copper. It is used to help shield the cable form EMI.
Outside the copper mesh is the final protective cover. (as shown in Fig)
The actual data travels through the center conductor in the cable. EMI interference is caught by
outer copper mesh. There are different types of coaxial cable vary by gauge & impedance.
Gauge is the measure of the cable thickness. It is measured by the Radio grade measurement, or
RG number. The high the RG number, the thinner the central conductor core, the lower the
number the thicker the core.
Here the most common coaxial standards.




50-Ohm RG-7 or RG-11 : used with thick Ethernet.
50-Ohm RG-58 : used with thin Ethernet
75-Ohm RG-59 : used with cable television
93-Ohm RG-62 : used with ARCNET.
CHARACTERISTICS OF COAXIAL CABLE





Low cost
Easy to install
Up to 10Mbps capacity
Medium immunity form EMI
Medium of attenuation
ADVANTAGES COAXIAL CABLE




Inexpensive
Easy to wire
Easy to expand
Moderate level of EMI immunity
DISADVANTAGE COAXIAL CABLE

Single cable failure can take down an entire network
STP
b) Twisted Pair Cable
UTP
The most popular network cabling is Twisted pair. It is light weight, easy to install, inexpensive
and support many different types of network. It also supports the speed of 100 mps. Twisted pair
cabling is made of pairs of solid or stranded copper twisted along each other. The twists are done
to reduce vulnerably to EMI and cross talk. The number of pairs in the cable depends on the
type. The copper core is usually 22-AWG or 24-AWG, as measured on the American wire
gauge standard. There are two types of twisted pairs cabling
i.
ii.
Unshielded twisted pair (UTP)
Shielded twisted pair (STP)
i.
Unshielded twisted pair (UTP)
UTP is more common. It can be either voice grade or data grade depending on the condition.
UTP cable normally has an impedance of 100 ohm. UTP cost less than STP and easily available
due to its many use. There are five levels of data cabling
Category 1 These are used in telephone lines and low speed data cable.
Category 2 These cables can support up to 4 mps implementation.
Category 3 These cable supports up to 16 mps and are mostly used in 10 mps.
Category 4 These are used for large distance and high speed. It can support 20mps.
Category 5 This is the highest rating for UTP cable and can support up to 100mps.
UTP cables consist of 2 or 4 pairs of twisted cable. Cable with 2 pair use RJ-11 connector and 4
pair cable use RJ-45 connector.
Characteristics of UTP






low cost
easy to install
High speed capacity
High attenuation
Effective to EMI
100 meter limit
Advantages of UTP

Easy installation


Capable of high speed for LAN
Low cost
Disadvantages of UTP

Short distance due to attenuation
ii.
Shielded twisted pair (STP)
It is similar to UTP but has a mesh shielding that’s protects it from EMI which allows for higher
transmission rate. IBM has defined category for STP cable.
Type 1 STP features two pairs of 22-AWG
Type 2 This type include type 1 with 4 telephone pairs
Type 6 This type feature two pairs of standard shielded 26-AWG
Type 7 This type of STP consist of 1 pair of standard shielded 26-AWG
Type 9 This type consist of shielded 26-AWG wire
Characteristics of STP






Medium cost
Easy to install
Higher capacity than UTP
Higher attenuation, but same as UTP
Medium immunity from EMI
100 meter limit
Advantages of STP:


Shielded
Faster than UTP and coaxial
Disadvantages of STP:


More expensive than UTP and coaxial
More difficult installation

High attenuation rate
c) Fiber Optics
Fiber optic cable uses electrical signals to transmit data. It uses light. In fiber optic cable light
only moves in one direction for two way communication to take place a second connection must
be made between the two devices. It is actually two stands of cable. Each stand is responsible for
one direction of communication. A laser at one device sends pulse of light through this cable to
other device. These pulses translated into “1’s” and “0’s” at the other end.
In the center of fiber cable is a glass stand or core. The light from the laser moves through this
glass to the other device around the internal core is a reflective material known as CLADDING.
No light escapes the glass core because of this reflective cladding.
Fiber optic cable has bandwidth more than 2 gbps (Gigabytes per Second)
Characteristics of Fiber Optic Cable:





Expensive
Very hard to install
Capable of extremely high speed
Extremely low attenuation
No EMI interference
Advantages Of Fiber Optic Cable:



Fast
Low attenuation
No EMI interference
Disadvantages Fiber Optics:


Very costly
Hard to install
Baseband and Broadband Transmission
The two ways to allocate the capacity of transmission media are with baseband and broadband
transmissions.
Baseband devotes the entire capacity of the medium to one communication channel. Baseband is
the most common mode of operation. Most LANs function in baseband mode. Baseband
signaling can be accomplished with both analog and digital signals.
Broadband enables two or more communication channels to share the bandwidth of the
communications medium. For example, that the TV cable coming into your house from an
antenna or a cable provider is a broadband medium. Many television signals can share the
bandwidth of the cable because each signal is modulated using a separately assigned frequency.
Television tuner may be used to choose the channel to watch by selecting its frequency. This
technique of dividing bandwidth into frequency bands is called 'Frequency-division
Multiplexing' (FDM) and works only with analog signals. Another technique, called 'Timedivision
Multiplexing'
(TDM),
supports
digital
signals.
Multiplexing is a technique that enables broadband media to support multiple data channels.
Baseband Vs Broadband Transmission
In a baseband transmission, the entire bandwidth of the cable is consumed by a single signal. In
broadband transmission, signals are sent on multiple frequencies, allowing multiple signals to be
sent simultaneously. Also
Baseband Signaling
Broadband Signaling
1
2
Uses digital signaling
No frequency-division multiplexing
Uses analog signaling
Frequency-division multiplexing is possible
3
Bi-directional transmission
Unidirectional transmission
4
Signal travels over short distances
Signal can travel over long distances before being
attenuated
Simplex, Duplex & Multiplex
Simplex transmission allows data to travel only in a single, pre specified direction. For example
in doorbell, the signal can go only from the button to the chime. Two other examples are
television and radio broadcasting. Simplex transmission are not often used because it is not
possible to send back error or control signals to the transmit end.
Half duplex transmission messages can move in either direction , but only one way at a time.
Only one end transmits at a time, the other end receives .The press to talk radio phones used in
police cars employs the half-duplex standard; only one person can talk at a time. Ethernet
networks are often said to be “half-duplex” .
Full duplex transmission works like traffic on a busy two way street the flow moves in two
directions at the same time. Full-duplexing is ideal for hardware units that need to pass large
amounts of data between each other as in mainframe-to-mainframe communications.
Concepts of WAP Technology
WAP stands for Wireless Application Protocol. Per the dictionary definition for each of these
words, we have:



Wireless: Lacking or not requiring a wire or wires: pertaining to radio transmission.
Application: A computer program or piece of computer software that is designed to do a
specific task.
Protocol: A set of technical rules about how information should be transmitted and
received using computers.
WAP is the set of rules governing the transmission and reception of data by computer
applications on, or via, wireless devices like mobile phones. WAP allows wireless devices to
view specifically designed pages from the Internet, using only plain text and very simple blackand-white pictures.
WAP is a standardized technology for cross-platform, distributed computing, very similar to the
Internet's combination of Hypertext Markup Language (HTML) and Hypertext Transfer Protocol
(HTTP), except that it is optimized for:



low-display capability
low-memory
low-bandwidth devices, such as personal digital assistants (PDAs), wireless phones, and
pagers.
WAP is designed to scale across a broad range of wireless networks, like GSM, IS-95, IS-136
and PDC.
The Internet Model
The Internet model makes it possible for a client to reach services on a large number of origin
servers; each addressed by a unique Uniform Resource Locator (URL).
The content stored on the servers is of various formats, but HTML is the predominant. HTML
provides the content developer with a means to describe the appearance of a service in a flat
document structure. If more advanced features like procedural logic are needed, then scripting
languages such as JavaScript or VB Script may be utilized.
The figure below shows how a WWW client request a resource stored on a web server. On the
Internet, standard communication protocols, like HTTP and Transmission Control
Protocol/Internet Protocol (TCP/IP) are used.
The content available at
the web server may be
static or dynamic. Static
content is produced
once and not changed or
updated very often, for
example a company
presentation. Dynamic
content is needed when
the
information
provided by the service
changes more often, for
example
timetables,
news, stock quotes and
account
information.
Technologies such as
Active Server Pages (ASP), Common Gateway Interface (CGI), and Servlets allow content to be
generated dynamically.
The WAP Model
The figure below shows the WAP programming model. Note the similarities with the Internet
model. Without the WAP
Gateway/Proxy the two
models would have been
practically identical.
WAP Gateway/Proxy is the
entity that connects the
wireless domain with the
Internet. You should make
a note that the request that is sent from the wireless client to the WAP Gateway/Proxy uses the
Wireless Session Protocol (WSP). In its essence, WSP is a binary version of HTTP.
A markup language - the Wireless Markup Language (WML) has been adapted to develop
optimized WAP applications. In order to save valuable bandwidth in the wireless network, WML
can be encoded into a compact binary format. Encoding WML is one of the tasks performed by
the WAP Gateway/Proxy.
How WAP Model Works?
When it comes to actual use, WAP works like this:
1. The user selects an option on their mobile device that has a URL with Wireless Markup
language (WML) content assigned to it.
2. The phone sends the URL request via the phone network to a WAP gateway, using the
binary encoded WAP protocol.
3. The gateway translates this WAP request into a conventional HTTP request for the
specified URL, and sends it on to the Internet.
4. The appropriate Web server picks up the HTTP request.
5. The server processes the request, just as it would any other request. If the URL refers to a
static WML file, the server delivers it. If a CGI script is requested, it is processed and the
content returned as usual.
6. The Web server adds the HTTP header to the WML content and returns it to the gateway.
7. The WAP gateway compiles the WML into binary form.
8. The gateway then sends the WML response back to the phone.
9. The phone receives the WML via the WAP protocol.
10. The micro-browser processes the WML and displays the content on the screen.
WAP is a global standard developed by the WAP Forum for wireless devices to access the
Internet and telephony services. WAP can also be used to access data from corporate intranets
through public or private IP networks.
WAP - Key Features
A programming model similar to the Internet's
Though WAP is a new technology but it reuse the concepts found on the Internet. This reuse
enables a quick introduction of WAP-based services, since both service developers and
manufacturers are familiar with these concepts today.
Wireless Markup Language (WML)
HTML language is used to develop web based application. Same way WML is a markup
language used for authoring WAP services, fulfilling the same purpose as HTML does on the
Web. In contrast to HTML, WML is designed to fit small handheld devices.
WML Script
Java Script or VB script is used to enhance the functionality of web applications. Same way
WML Script can be used to enhance the functionality of a service, just as, Java script can be
utilized in HTML. It makes it possible to add procedural logic and computational functions to
WAP based services.
Wireless Telephony Application Interface (WTAI)
The WTAI is an application framework for telephony services. WTAI user agents are able to
make calls and edit the phone book by calling special WML Script functions or by accessing
special URLs. If one writes WML decks containing names of people and their phone numbers,
you may add them to your phone book or call them right away just by clicking the appropriate
hyperlink on the screen.
Optimized protocol stack
The protocols used in WAP are based on well-known Internet protocols, such as HTTP and
Transmission Control Protocol (TCP), but they have been optimized to address the constraints of
a wireless environment, such as low bandwidth and high latency.
WAP is designed in a layered fashion so that it can be extensible, flexible, and scalable. As a
result, the WAP protocol stack is divided into five layers:





Application Layer
Wireless Application Environment (WAE). This layer is of most interest to content
developers because it contains, among other things, device specifications and the content
development programming languages, WML and WMLScript.
Session Layer
Wireless Session Protocol (WSP). Unlike HTTP, WSP has been designed by the WAP
Forum to provide fast connection suspension and reconnection.
Transaction Layer
Wireless Transaction Protocol (WTP). The WTP runs on top of a datagram service such
as User Datagram Protocol (UDP) and is part of the standard suite of TCP/IP protocols
used to provide a simplified protocol suitable for low bandwidth wireless stations.
Security Layer
Wireless Transport Layer Security (WTLS). WTLS incorporates security features that are
based upon the established Transport Layer Security (TLS) protocol standard. It includes
data integrity checks, privacy, service denial, and authentication services.
Transport Layer
Wireless Datagram Protocol (WDP). The WDP allows WAP to be bearer-independent by
adapting the transport layer of the underlying bearer. The WDP presents a consistent data
format to the higher layers of the WAP protocol stack, thereby offering the advantage of
bearer independence to application developers.
Each of these layers provides a well-defined interface to the layer above it. This means that the
internal workings of any layer are transparent or invisible to the layers above it. The layered
architecture allows other applications and services to utilize the features provided by the WAPstack as well. This makes it possible to use the WAP-stack for services and applications that
currently are not specified by WAP.
The WAP protocol
architecture is shown
below alongside a
typical
Internet
Protocol stack.
Note that the mobile
network bearers in the
lower part of the figure
above are not part of
the WAP protocol
stack.
The uppermost layer in
the WAP stack, the
Wireless Application Environment (WAE) provides an environment that enables a wide range of
applications to be used on wireless devices. In the chapter "WAP - the wireless service enabler"
the WAP WAE programming model was introduced. This chapter will focus on the various
components of WAE:



Addressing model
A syntax suitable for naming resources stored on servers. WAP use the same addressing
model as the one used on the Internet, that is, Uniform Resource Locators (URL).
Wireless Markup Language (WML)
A lightweight markup language designed to meet the constraints of a wireless
environment with low bandwidth and small handheld devices. The Wireless Markup
Language is WAP.s analogy to HTML used on the WWW. WML is based on the
Extensible Markup Language (XML).
WMLScript
A lightweight scripting language. WMLScript is based on ECMAScript, the same
scripting language that JavaScript is based on. It can be used for enhancing services

written in WML in the way that it to some extent adds intelligence to the services, for
example procedural logic, loops, conditional expressions, and computational functions.
Wireless Telephony Application (WTA, WTAI)
A framework and programming interface for telephony services. The Wireless Telephony
Application (WTA) environment provides a means to create telephony services using
WAP.
Hardware and Software Requirement
At minimum, developing WAP applications requires a web server and a WAP simulator. Using
simulator software while developing a WAP application is convenient as all the required
software can be installed on the development PC.
Although software simulators are good in their own right, no WAP application should go into
production without testing it with actual hardware. The following list gives a quick overview of
the necessary hardware and software to test and develop WAP applications:





a Web server with connection to the Internet
a WML to develop WAP application
a WAP simulator to test WAP application
a WAP gateway
a WAP phone for final testing
Microsoft IIS or Apache on Windows or Linux can be used as the web server and Nokia WAP
Toolkit version 2.0 as the WAP simulator.
MODEMS
The term modem is a composite word that refers to the two functional entities that make up the
device; a signal modulator and a signal demodulator. A modulator creates a band-pass analog
signal from binary data. A demodulator recovers the binary data from the modulated
signal.Modem stands for modulator and demodulator.
TELEPHONE MODEMS
Traditional telephone lines can carry frequencies between 300 and 3300 HZ, giving them BW of
3000 Hz; All this range is used for transmitting voice, where a great deal of interference and
distortion can be accepted without loss of intelligibility.
The effective BW of a telephone line being used for data Transmission is 2400 Hz, covering the
range from 600 to 3000 Hz.
MODULATION /DEMODULATION
Telco
Telephone
network
Modem
A
Telco
Modem
B
Figure shows the relationship of modems to a communication link. The computer on the left
sends binary data to the modulator portion of the modem; the data is sent as an analog signal on
the telephone lines. The modem on the right receives the analog signal, demodulates it through
its demodulator, and delivers data to the computer on the right.
The communication can be bidirectional, which means the computer on the right can also send
data to the computer on the left using the same modulation and demodulation processes.
Modem standards
V-series standards published by the ITU-T.





V.32
V.32bis
V.34bis
V.90
V.92
V.32
This modem uses a combined modulation and demodulation encoding technique called
trellis-coded modulation. Trellis is essentially QAM plus a redundant bit. The Data stream is
divided into 4-bit sections. Instead of a quad bit, however, a pentabit is transmitted. The value of
the extra bit is calculated from the values of the data bits.
In any QAM system, the receiver compares each received signal point to all valid points
in the constellation and selects the closest point as the intended value.. A signal distorted by
transmission noise can arrive closer in value to an adjacent point than to the intended point,
resulting in a misidentification of the point and an error in the received data.
By adding a redundant bit to each quad bit, trellis-coded modulation increases the amount
of information used to identify each bit pattern thereby reduces the number of possible matches.
The V.32 calls for 32-QAM with a baud rate of 2400. Because only 4 bits of each pentabit
represents data, the resulting speed is 4*2400=9600.
FDX 2400 baud 9600 bps 2-wire
600
1800
3000
Bandwidth diagram
V.32 bis : The V.32 bis modem support 14,400-bps transmission. The V.32 uses 128QAM transmission.
V.34 bis : The V.34 bis modem support 28,800-bps transmission with a 960-point
constellation to a bit rate of 33,600 with a 1664-point constellation.
V.90: Traditional modems have a limitations on the data rate.V.90 modems with a bit
rate of 56,000 bps, called 56Kmodems, are available. Downloading rate is 56K, while the
uploading rate is a maximum of 33.6 kbps.
Traditional modems
In traditional modems data exchange is between two computers, A and B, Through
digital telephone network.
Sampling & noise
PCM
Modem
A
Telephone
network
Inverse
PCM
Modem
A to B
Quantization
noise happens in
the Telco office
near A
B
Traditional modems
Sampling & noise
Inverse
PCM
Modem
A
Telephone
network
B to A
Quantization
noise happens in
the telco office
near B
PCM
Modem
B
After modulation by the modem, an analog signal reaches the telephone company Switching
station. Where it is sampled and digitized to be passed through the digital network. The
quantization noise introduced in the signal at the sampling point limits the data rate according to
the capacity. This limit is 33.6 Kbps.
56K Modems
Communication today is via the Internet. In Uploading, The analog signal must still be sampled
at the switching station, which means the data rate in the uploading is limited to 33.6 Kbps.
There is no sampling in downloading. Data rate in downloading is 56Kbps.
56K MODEMS
Sampling & noise
PCM
Telephone
network
ISP
server
Modem
Uploading,
quantization
noise
A
Inverse
PCM
Telephone
network
ISP
server
Modem
downloading, no
quantization noise
A
V.92: The standard above V.92 is called V.92. These modems can adjust their speed, and if the
noise allows, they can upload data at the rate of 48 Kbps. The modem has additional features.
For example, the modem can interrupt the internet connection when there is an incoming call if
the lines has call-waiting service.
RS 232 INTERFACERS 232 is a standard interface by EIA and RS232C is the latest version of
this interface.
INTERFACING WITH RS232



It expects a modem to be connected to both receiving and transmitting end.
The modem is termed as DCE(Data Communication Equipment) And the computer with
which modem is interfaced is called DTE (Data Terminal Equipment).
The DCE and DTE are linked via a cable whose length does not exceed 50 feet. The DTE
has 35 pins male connector and DCE has 25 pins Female connector.
FEATURES OF RS232 INTERFACE
1. RS232 Signal LEVEL
 RS232 standard follows –ve logic, Logic1 is represented by negative voltage.,
logic0 is represented by +ve voltage.
 Level 1 varies from -3 to -15v and level 0 varies from 3 to 15v
2. RS232 SIGNALS
SL NO PIN NUMBER
SIGNAL
SIGNAL NAME
1
1
---
Frame ground
2
2
TXD
Transmit data
3
3
RXD
Receive data
4
4
RTS
Request to send
5
5
CTS
Clear to send
6
6
DSR
Data Set Ready
7
7
SG
Signal Ground
8
8
RLSD or CD
Received line signal detect or carrier detect
9
20
DTR
Data Terminal Ready
10
22
RI
Ring Indicator
COMMUNICATION BETWEEN DCE AND DTE



Before sending data to the other end the DTE requests the permission from the modem by
issuing RTS signal.
The modem has a method to find out if any telephone line is free and if the other end of
modem is ready.
When the modem finds the communication path is ready for communication it issues
CTS signal to DTE as an acknowledgement.




The DTE issues DTR signal when it is powered on, error free and ready for logical
connection through the modem.
The modem issues a DSR signal to indicate that it is powered on and it is error free.
The data is transferred by TXD signal from DTE to DCE and RXD signal receives data
from DCE to DTE.
The RI and RLSD signals are used with the dialed modem, when the telephone link is
shared.
Communication
25 pin female connector
25 pin male connector
MODEM
TERMINAL
DCEStandards
Serial Communication
DTE
RS232 is the most commonly used, because almost every computer is equiped with one or more
serial ports. Because RS232 signals share a common ground, they are sensitive for interference,
therefore the distance is limited to 15 meters. RS232 can only be used for point to point
connections (one sender, one recipient).
3
RS422 is a high speed digital interface - unlike RS232 which uses signals with reference to
3
ground, RS449 receivers look for the difference between two wires. By twisting the two wires
and making a "twisted pair" any noise or interference picked up on one wire will also be picked
22
up on the other, because both wires pick up the same interference the RS449 differential
22
interface just shifts in voltage level with reference to ground, but does not change with respect to
each other. The receivers are only looking at the difference in voltage level of each wire to the
8
other not to ground.
8
RS423 is a not commonly used enhanced version of the RS232 standard. The only difference is
that it can be used for greater distances. The interface looks like it uses
7 differential signals, but in
fact, all B pins are connected7 to ground. RS423 can be used for point to point or multidrop (one
sender, up to 10 recipients) communications.
RS449 is a not commonly used enhanced version of the RS232 standard. Another name for
RS449 is V.11 Like RS 485 and RS 422 it uses differential signals over twisted pair cables which
reduces errors caused by noise or interference. RS449 can be used for point to point or multidrop
(one sender, up to 10 recipients) communications.
RS485 is the most enhanced RS232 based serial interface available. It can handle high speeds
and long cable distances up to 4000 feet. Like RS 449 and RS 422 it uses differential signals
over twisted pair cables which reduces errors caused by noise or interference. RS485 uses
multipoint technology, it is possible to connect 32 devices on a single RS485 bus. Connection
speeds can be as high as 35 Mbs
Specifications for RS232, RS423, RS422, and RS485
X-21 Digital Interface
CCITT X21 is a physical and electrical interface that uses two types of circuits:
balanced (X.27N.1 1) and and unbalanced (X.26N.10). CCITT X.21 calls out the DA-15 (also
know by DB-15) connector.
The physical interface between the DTE and the local PTT-supplied DCE is defined in ITU-T
recommendation X.21. The DCE provides a full-duplex, bit-serial, synchronous transmission
path between the DTE and the local PSE. It can operate at data rates from 600bps to 64Kbps. A
second standard, X.21bis has been defined for use on existing (analog) networks. An X.21bis is a
subset of EIA-232D/V.24 therefore allowing existing user equipment to be readily interfaced
using this standard. It should perhaps be emphasized here that V24 defines the data terminal
equipment interface to the modem and is not concerned with the interface between the modem
and the line itself. The modems themselves therefore form part of the conceptual physical
connection. The V24 interface is thus independent of both modulation technique and data
throughput rate.
The X.21 interface protocol is concerned only with the set-up and clearing operations between
DTE and DCE associated with each call. The control of the ensuing data transfer is the
responsibility of the link layer.
X21 Overview
X.21 is a state-driven protocol running full duplex at 9600 bps to 64 Kbps with subscriber
networks. It is a circuit-switching protocol using Synchronous ASCII with odd parity to connect
and disconnect a subscriber to the public-switching network.
The data-transfer phase is transparent to the network. Any data can be transferred through the
network after Call Establishment is made successfully via the X.21 protocol. The call-control
phases which are used were defined in the CCITT (now ITU) 1988 "Blue Book"
Recommendations X.1 - X.32.
Signals Provided
The signals of the X21 interface are presented on a 15-pin connector defined by ISO Document
4903. The electrical characteristics are defined in CCITT Recommendations X.26 and X.27,
which refer to CCITT Recommendations V.10 and V.11.
X.21 provides eight signals:
Signal Ground (G) -
This provides reference for the logic states against the other
circuits. This signal may be connected to the protective ground
(earth).
DTE Common Return
(Ga) -
Used only in unbalanced-type configurations (X.26), this signal
provides reference ground for receivers in the DCE interface.
Transmit (T) -
This carries the binary signals which carry data from the DTE to
the DCE. This circuit can be used in data-transfer phases or in
call-control phases from the DTE to DCE (during Call Connect
or Call Disconnect).
Receive (R) -
Controlled by the DTE to indicate to the DCE the meaning of the
data sent on the transmit circuit. This circuit must be ON during
data-transfer phase and can be ON or OFF during call-control
phases, as defined by the protocol.
Indication (I) -
The DCE controls this circuit to indicate to the DTE the type of
data sent on the Receive line. During data phase, this circuit must
be ON and it can be ON or OFF during call control, as defined
by the protocol.
Signal Element Timing
(S) -
This provides the DTE or DCE with timing information for
sampling the Receive line or Transmit line. The DTE samples at
the correct instant to determine if a binary 1 or 0 is being sent by
the DCE. The DCE samples to accurately recover signals at the
correct instant. This signal is always ON.
Byte Timing (B) -
This circuit is normally ON and provides the DTE with 8-bit byte
element timing. The circuit transitions to OFF when the Signal
Element Timing circuit samples the last bit of an 8-bit byte. Callcontrol characters must align with the B lead during call-control
phases. During data- transfer phase, the communicating devices
bilaterally agree to use the B lead to define the end of each
transmitted or received byte. The C and I leads then only monitor
and record changes in this condition when the B lead changes
from OFF to ON, although the C and I leads may be altered by
the transitions on the S lead. This lead is frequently not used.
X.21 Protocol Operation
As stated previously, X.21 is a state protocol. Both the DTE and DCE can be in a Ready or NotReady state.
The Ready state for the DTE is indicated by a continuous transmission of binary 1's on the T
lead. The Ready state for the DCE is continuous transmission of binary 1's on the R lead. During
this continuous transmission of Ready state, the control leads are OFF.
During the Not-Ready state, the DCE transmits binary 0's on the R lead with the I lead in the
OFF state.
The DTE Uncontrolled Not-Ready is indicated by transmission of binary 0's with the C lead in
the OFF state. The DTE Uncontrolled Not-Ready state signifies that the DTE is unable to accept
calls due to an abnormal condition.
The DTE Controlled Not-Ready state sends a pattern of alternating 1's and 0's on the T lead with
the C lead OFF. This state indicates that the DTE is operational, but unable to accept incoming
calls.
The characters sent between the DTE and DCE during call-control phases are International
Alphabet 5 (IA5), defined by CCITT Recommendation V.3. At least two Sync characters must
precede all sequences of characters sent between the DTE and DCE to establish 8-bit byte
synchronization between the transmitter and the receiver. If the Byte Timing (B) lead is used,
these Sync characters must align with the B lead timing signals.
Need for data link layer
The data link layer transforms the physical layer, a raw transmission facility, to a link
responsible for node-to-node (hop-to-hop) communication. Specific responsibilities of the data
link layer include
1. framing
2. addressing
3. flow control
4. error control
5. media
6. access control
The data link layer divides the stream of bits received from the network layer into manageable
data units called frames. The data link layer adds a header to the frame to define the addresses of
the sender and receiver of the frame. If the rate at which the data are absorbed by the receiver is
less than the rate at which data are produced in the sender, the data link layer imposes a flow
control mechanism to avoid overwhelming the receiver. The data link layer also adds reliability
to the physical layer by adding mechanisms to detect and retransmit damaged, duplicate, or lost
frames. When two or more devices are connected to the same link, data link layer protocols are
necessary to determine which device has control over the link at any given time.
Reliable Transmission
Frames are sometimes corrupted while in transit, with an error code like CRC used to detect such
errors. While some error codes are strong enough also to correct errors, in practice the overhead
is typically too large to handle the range of bit and burst errors that can be introduced on a
network link. Even when error-correcting codes are used (e.g., on wireless links), some errors
will be too severe to be corrected. As a result, some corrupt frames must be discarded. A linklevel protocol that wants to deliver frames reliably must somehow recover from these discarded
(lost) frames. This is usually accomplished using a combination of two fundamental
mechanisms—acknowledgments and timeouts. An acknowledgment (ACK for short) is a small
control frame that a protocol sends back to its peer saying that it has received an earlier frame.
By control frame we mean a header without any data, although a protocol can piggyback an
ACK on a data frame it just happens to be sending in the opposite direction. The receipt of an
acknowledgment indicates to the sender of the original frame that its frame was successfully
delivered. If the sender does not receive an acknowledgment after a reasonable amount of time,
then it retransmits the original frame. This action of waiting a reasonable amount of time is
called a timeout. The general strategy of using acknowledgments and timeouts to implement
reliable delivery is sometimes called automatic repeat request (normally abbreviated ARQ).
Stop-and-Wait Protocol
Figure 2.20 Timeline for stopand-wait with 1-bit sequence
number.
The simplest ARQ scheme is the stop-and-wait algorithm. The idea of stop-and-wait is
straightforward: After transmitting one frame, the sender waits for an acknowledgment before
transmitting the next frame. If the acknowledgment does not arrive after a certain period of time,
the sender time out and retransmits the original frame. Figure 2.19 illustrates four different
scenarios that result from this basic algorithm.
Next figure is a timeline, a common way to depict a protocol’s behavior. The sending side is
represented on the left, the receiving side is depicted on the right, and time flows from top to
bottom. Figure 2.19(a) shows the situation in which the ACK is received before the timer
expires, (b) and (c) show the situation in which the use of error-correcting codes in networking is
sometimes referred to as forward error cor- rection (FEC) because the correction of errors is
handled “in advance” by sending extra information, rather than waiting for errors to happen and
dealing with them later by retransmission. The original frame and the ACK, respectively, are
lost, and (d) shows the situation in which the timeout fires too soon. By “lost” we mean that the
frame was corrupted while in transit, that this corruption was detected by an error code on the
receiver, and that the frame was subsequently discarded. There is one important subtlety in the
stop-and-wait algorithm. Suppose the sender sends a frame and the receiver acknowledges it, but
the acknowledgment is either lost or delayed in arriving. This situation is illustrated in timelines
(c) and (d) of Figure 2.19. In both cases, the sender times out and retransmit the original frame,
but the receiver will think that it is the next frame, since it correctly received and acknowledged
the first frame. This has the potential to cause duplicate copies of a frame to be delivered. To
address this problem, the header for a stop-and-wait protocol usually includes a 1-bit sequence
number—that is, the sequence number can take on the values 0 and 1—and the sequence
numbers used for each frame alternate, as illustrated in Figure 2.20. Thus, when the sender
retransmits frame 0, the receiver can determine that it is seeing a second copy of frame 0 rather
than the first copy of frame 1 and therefore can ignore it (the receiver still acknowledges it, in
case the first ACK was lost).
The main shortcoming of the stop-and-wait algorithm is that it allows the sender to have only
one outstanding frame on the link at a time, and this may be far below the link’s capacity.
Consider, for example, a 1.5-Mbps link with a 45-ms round-trip time. This link has a delay ×
bandwidth product of 67.5 Kb, or approximately 8 KB. Since the sender can send only one frame
per RTT, and assuming a frame size of 1 KB, this implies a maximum sending rate of
BitsPerFrame ÷ TimePerFrame
= 1024 × 8 ÷ 0.045
= 182 Kbps
or about one-eighth of the link’s capacity. To use the link fully, then, we’d like the sender to be
able to transmit up to eight frames before having to wait for an acknowledgment.
Sliding Window
Consider again the scenario in which the link has a delay × bandwidth product of 8 KB and
frames are of 1-KB size.We would like the sender to be ready to transmit the ninth frame at
pretty much the same moment that the ACK for the first frame arrives. The algorithm that allows
us to do this is called sliding window, and an illustrative timeline is given in Figure 2.21.
The Sliding Window Algorithm
The sliding window algorithm works as follows. First, the sender assigns a sequence number,
denoted SeqNum, to each frame. For now, let’s ignore the fact that SeqNum is implemented by a
finite-size header field and instead assume that it can grow infinitely large. The sender maintains
three variables: The send window size, denoted SWS, gives the upper bound on the number of
outstanding (unacknowledged) frames that the sender can transmit; LAR denotes the sequence
number of the last acknowledgment received; and LFS denotes the sequence number of the last
frame sent. The sender also maintains the following invariant:
LFS − LAR ≤ SWS
When an acknowledgment arrives, the sender moves LAR to the right, thereby allowing the
sender to transmit another frame. Also, the sender associates a timer with each frame it transmits,
and it retransmits the frame should the timer expire before an ACK is received. Notice that the
sender has to be willing to buffer up to SWS frames since it must be prepared to retransmit them
until they are acknowledged.
The receiver maintains the following three variables:
1. RWS denotes the receive window size, gives the upper bound on the number of out-oforder frames that the receiver is willing to accept.
2. LAF denotes the sequence number of the largest acceptable frame and
3. LFR denotes the sequence number of the last frame received.
4.
The receiver also maintains the following invariant: LAF − LFR ≤ RWS
This situation is illustrated in Figure 2.23. When a frame with sequence number SeqNum arrives,
the receiver takes the following action. If SeqNum ≤ LFR or SeqNum > LAF, then the frame is
outside the receiver’s window and it is discarded. If LFR < SeqNum ≤ LAF, then the frame is
within the receiver’s window and it is accepted. Now the receiver needs to decide whether or not
to send an ACK. Let SeqNumToAck denote the largest sequence number not yet acknowledged,
such that all frames with sequence numbers less than or equal to SeqNumToAck have been
received. The receiver acknowledges the receipt of SeqNumToAck, even if higher-numbered
packets have been received. This acknowledgment is said to be cumulative. It then sets LFR =
SeqNumToAck and adjusts LAF = LFR + RWS.
For example, suppose LFR = 5 (i.e., the last ACK the receiver sent was for sequence number 5),
and RWS = 4. This implies that LAF = 9. Should frames 7 and 8 arrive, they will be buffered
because they are within the receiver’s window. However, no ACK needs to be sent since frame 6
is yet to arrive. Frames 7 and 8 are said to have arrived out of order. (Technically, the receiver
could resend an ACK for frame 5 when frames 7 and 8 arrive.) Should frame 6 then arrive—
perhaps it is late because it was lost the first time and had to be retransmitted, or perhaps it was
simply delayed—the receiver acknowledges frame 8, bumps LFR to 8, and sets LAF to 12. If
frame 6 was in fact lost, then a timeout will have occurred at the sender, causing it to retransmit
frame 6. We observe that when a timeout occurs, the amount of data in transit decreases, since
the sender is unable to advance its window until frame 6 is acknowledged. This means that when
packet losses occur, this scheme is no longer keeping the pipe full. The longer it takes to notice
that a packet loss has occurred, the more severe this problem becomes. Notice that in this
example, the receiver could have sent a negative acknowledgement (NAK) for frame 6 as soon as
frame 7 arrived. However, this is unnecessary since the sender’s timeout mechanism is sufficient
to catch this situation, and sending NAKs adds additional complexity to the receiver. Also, as we
mentioned, it would have been legitimate to send additional acknowledgments of frame 5 when
frames 7 and 8 arrived; in some cases, a sender can use duplicate ACKs as a clue that a frame
was lost. Both approaches help to improve performance by allowing early detection of packet
losses. Yet another variation on this scheme would be to use selective acknowledgments. That is,
the receiver could acknowledge exactly those frames it has received, rather than just the highestnumbered frame received in order. So, in the above example, the receiver could acknowledge the
receipt of frames 7 and 8. Giving more information to the sender makes it potentially easier for
the sender to keep the pipe full, but adds complexity to the implementation. The sending window
size is selected according to how many frames we want to have outstanding on the link at a given
time; SWS is easy to compute for a given delay×bandwidth product. On the other hand, the
receiver can set RWS to whatever it wants.
Two common settings are RWS = 1, which implies that the receiver will not buffer any frames
that arrive out of order, and RWS = SWS, which implies that the receiver can buffer any of the
frames the sender transmits. It makes no sense to set RWS > SWS since it’s impossible for more
than SWS frames to arrive out of order.
Frame Order and Flow Control
The sliding window protocol is perhaps the best-known algorithm in computer networking. It can
be used to serve three different roles. The first role is to reliably deliver frames across an
unreliable link. In general, the algorithm can be used to reliably deliver messages across an
unreliable network. This is the core function of the algorithm.
The second role that the sliding window algorithm can serve is to preserve the order in
which frames are transmitted. Since each frame has a sequence number, the receiver makes
sure that it does not pass a frame up to the next-higher-level protocol until it has already passed
up all frames with a smaller sequence number. That is, the receiver buffers (i.e., does not pass
along) out-of-order frames.
The third role that the sliding window algorithm sometimes plays is to support flow
control—a feedback mechanism by which the receiver is able to throttle the sender. Such a
mechanism is used to keep the sender from overrunning the receiver, that is, from transmitting
more data than the receiver is able to process. This is usually accomplished by augmenting the
sliding window protocol so that the receiver not only acknowledges frames it has received, but
also informs the sender of how many frames it has room to receive. The number of frames that
the receiver is capable of receiving corresponds to how much free buffer space it has.
HDLC- Bit-Oriented Protocols
Framing: A sequence of bits is transmitted over a point-to-point link from adaptor to adaptor. In
packet-switched networks, blocks of data (called frames at this level), not bit streams, are
exchanged between nodes. It is the network adaptor that enables the nodes to exchange frames.
When node A wishes to transmit a frame to node B, it tells its adaptor to transmit a frame from
the node’s memory. This results in a sequence of bits being sent over the link. The adaptor on
node B then collects together the sequence of bits arriving on the link and deposits the
corresponding frame in B’s memory. Recognizing exactly what set of bits constitutes a frame,
that is, determining where the frame begins and ends is the central challenge faced by the
adaptor- framing problem.
There are several ways to address the framing problem.
1. Byte-Oriented Protocols (BISYNC, PPP, DDCMP)
2. Bit-Oriented Protocols (HDLC)
HDLC is a Bit-Oriented Protocol. Unlike byte-oriented protocols, a bit oriented protocol is not
concerned with byte boundaries—it simply views the frame as a collection of bits. These bits
might come from some character set, such as ASCII, they might be pixel values in an image, or
they could be instructions and operands from an executable file. The Synchronous Data Link
Control (SDLC) protocol developed by IBM is an example of a bit-oriented protocol; SDLC was
later standardized by the ISO as the High-Level Data Link Control (HDLC) protocol.
HDLC denotes both the beginning and the end of a frame with the distinguished bit sequence
01111110. This sequence is also transmitted during any times that the link is idle so that the
sender and receiver can keep their clocks synchronized. Because this sequence might appear
anywhere in the body of the frame—in fact, the bits 01111110 might cross byte boundaries—bitoriented protocols use the analog of the DLE (data-link-escape) character, a technique known as
bit stuffing. Bit stuffing in the HDLC protocol works as follows. On the sending side, any time
five consecutive 1s have been transmitted from the body of the message (i.e., excluding when the
sender is trying to transmit the distinguished 01111110 sequence), the sender inserts a 0 before
transmitting the next bit. On the receiving side, should five consecutive 1s arrive, the receiver
makes its decision based on the next bit it sees (i.e., the bit following the five 1s). If the next bit
is a 0, it must have been stuffed, and so the receiver removes it. If the next bit is a 1, then one of
two things is true: Either this is the end-of-frame marker or an error has been introduced into the
bit stream. By looking at the next bit, the receiver can distinguish between these two cases: If it
sees a 0 (i.e., the last eight bits it has looked at are 01111110), then it is the end-of frame marker;
if it sees a 1 (i.e., the last eight bits it has looked at are 01111111), then there must have been an
error and the whole frame is discarded. In the latter case, the receiver has to wait for the next
01111110 before it can start receiving again, and as a consequence, there is the potential that the
receiver will fail to receive two consecutive frames. Obviously, there are still ways that framing
errors can go undetected, such as when an entire spurious end-of-frame pattern is generated by
errors, but these failures are relatively unlikely. An interesting characteristic of bit stuffing, as
well as character stuffing, is that the size of a frame is dependent on the data that is being sent in
the payload of the frame. It is in fact not possible to make all frames exactly the same size, given
that the data that might be carried in any frame is arbitrary.
Multiplexing
In telecommunications and computer networks, multiplexing (also known as muxing) is a
process where multiple analog message signals or digital data streams are combined into one
signal over a shared medium. The aim is to share an expensive resource. For example, in
telecommunications, several phone
calls may be transferred using one
wire. It originated in telegraphy, and
is
now
widely
applied
in
communications.
The multiplexed signal is transmitted
over a communication channel,
which
may
be
a
physical
transmission
medium.
The
multiplexing divides the capacity of the low-level communication channel into several higherlevel logical channels, one for each message signal or data stream to be transferred. A reverse
process, known as demultiplexing, can extract the original channels on the receiver side. A
device that performs the multiplexing is called a multiplexer (MUX), and a device that performs
the reverse process is called a demultiplexer (DEMUX).
Types of multiplexing
1.
Frequency-division multiplexing (FDM)
The deriving of two or more simultaneous, continuous channels from a
transmission medium by assigning a separate portion of the available frequency
spectrum to each of the individual channels.
FDMA (frequency-division multiple access): The use of frequency division to
provide multiple and simultaneous transmissions
•
•
Transmission is organized in frequency channels, assigned for an exclusive use by a
single user at a time
If the channel is not in use, it remains idle and cannot be used by others
•
•
•
2.
There are channeling frequency plans elaborated to avoid mutual co-channel and
adjacent-channel interference among neighboring stations
The use of a radio channel or a group of radio channels requires authorization (license)
– for each individual station or for group of stations
Frequency division duplexing
– 2 radio frequency channels for each duplex link (1 up-link & 1 down-link or 1
forward link and 1 reverse link)
Time-division multiplexing (TDM)
– Time Division Multiplex: A single carrier frequency channel is shared by a number of
users, one after another. Transmission is organized in repetitive “time-frames”. Each
frame consists of groups of pulses - time slots.
– Each user is assigned a separate time-slot.
– TDD – Time Division Duplex provides the forward and reverse links in the same
frequency channel.
3.

Code-division multiplexing (CDM)
Code Division Multiple Access or Spread Spectrum communication techniques
FH: frequency hoping (frequency synthesizer controlled by pseudo-random
sequence of numbers)
DS: direct sequence (pseudo-random sequence of pulses used for spreading)
TH: time hoping (spreading achieved by randomly spacing transmitted pulses)
Other techniques


Hybrid combination of the above techniques (radar and other applications)
Random noise as carrier
Transmission is organized in time-frequency “slots”. Each link is assigned a sequence of the
slots, according to a specific code. Used e.g. in Bluetooth system
Switching
SWITCHING is a methodology to establish connection between two end points in any network.
switched network consists of a series of interlinked nodes called switches. These are devices
capable of creating temporary connections between two or more devices. Switching is categories
in following



CIRCUIT SWITCHING
MESSAGE SWITCHING
PACKET SWITCHING
Message Switching
In this switching method, a different strategy is used, where instead of establishing a dedicated
physical line between the sender and the receiver, the message is sent to the nearest directly
connected switching node. This node stores the message, checks for errors, selects the best
available route and forwards the message to
the next intermediate node.
The line becomes free again for other
messages, while the process is being
continued in some other nodes. Due to the
mode of action, this method is also known as
store-and-forward technology where the
message hops from node to node to its final
destination. Each node stores the full
message, checks for errors and forwards it.
In this switching technique, more devices
can share the network bandwidth, as compared with circuit switching technique. Temporary
storage of message reduces traffic congestion to some extent. Higher priority can be given to
urgent messages, so that the low priority messages are delayed while the urgent ones are
forwarded faster. Through broadcast addresses one message can be sent to several users. Last of
all, since the destination host need not be active when the message is sent, message switching
techniques improve global communications.
However, since the message blocks may be quite large in size, considerable amount of storage
space is required at each node to buffer the messages. A message might occupy the buffers for
minutes, thus blocking the internodal traffic.
Basic idea



Each network node receives and stores the message
Determines the next leg of the route, and
Queues the message to go out on that link.
Advantages




Line efficiency is greater (sharing of links).
Data rate conversion is possible.
Even under heavy traffic, packets are accepted, possibly with a greater delay in delivery.
Message priorities can be used, to satisfy the requirements, if any.
Disadvantages: Message of large size monopolizes the link and storage
Packet Switching
The basic approach is not much different from message switching. It is also based on the same
‘store-and-forward’ approach. However, to overcome the limitations of message switching,
messages are divided into subsets of equal length called packets. This approach was developed
for long-distance data communication (1970) and it has evolved over time. In packet switching
approach, data are transmitted in short packets (few Kbytes). A long message is broken up into a
series of packets as shown in Fig. 4.2.2. Every packet contains some control information in its
header, which is required for routing and
other purposes.
Main difference between Packet switching
and Circuit Switching is that the
communication lines are not dedicated to
passing messages from the source to the
destination. In Packet Switching, different
messages (and even different packets) can
pass through different routes, and when
there is a "dead time" in the communication between the source and the destination, the lines can
be used by other sources.
There are two basic approaches commonly used to packet Switching: virtual-circuit packet
switching and datagram packet switching. In virtual-circuit packet switching a virtual circuit is
made before actual data is transmitted, but it is different from circuit switching in a sense that in
circuit switching the call accept signal comes only from the final destination to the source while
in case of virtual-packet switching this call accept signal is transmitted between each adjacent
intermediate node as shown in Fig. 4.2.3. Other features of virtual circuit packet switching are
discussed in the following subsection
Virtual Circuit Packet Switching Networks
An initial setup phase is used to set up a route between the intermediate nodes for all the packets
passed during the session between the two end nodes. In each intermediate node, an entry is
registered in a table to indicate the route for the connection that has been set up. Thus, packets
passed through this route, can have short headers, containing only a virtual circuit identifier
(VCI), and not their destination. Each intermediate node passes the packets according to the
information that was stored in it, in the setup phase. In this way, packets arrive at the destination
in the correct sequence, and it is guaranteed that essentially there will not be errors. This
approach is slower than Circuit Switching, since different virtual circuits may compete over the
same resources, and an initial setup phase is needed to initiate the circuit. As in Circuit
Switching, if an intermediate node fails, all virtual circuits that pass through it are lost. The most
common forms of Virtual Circuit
networks are X.25 and Frame
Relay, which are commonly
used for public data networks
(PDN).
Datagram Packet Switching
Networks
This approach uses a different,
more dynamic scheme, to
determine the route through the
network links. Each packet is
treated as an independent entity, and its header contains full information about the destination of
the packet. The intermediate nodes examine the header of the packet, and decide to which node
to send the packet so that it will reach its destination. In the decision two factors are taken into
account:
• The shortest ways to pass the packet to its destination - protocols such as RIP/OSPF are used to
determine the shortest path to the destination.
• Finding a free node to pass the packet to - in this way, bottlenecks are eliminated, since packets
can reach the destination in alternate routes.
Thus, in this method, the packets don't follow a pre-established route, and the intermediate nodes
(the routers) don't have pre-defined knowledge of the routes that the packets should be passed
through.
Packets can follow different routes to the
destination, and delivery is not guaranteed
(although packets usually do follow the same
route, and are reliably sent). Due to the nature
of this method, the packets can reach the
destination in a different order than they were
sent, thus they must be sorted at the
destination to form the original message. This
approach is time consuming since every router has to decide where to send each packet. The
main implementation of Datagram Switching network is the Internet, which uses the IP network
protocol.
Advantages:
• Call setup phase is avoided (for transmission of a few packets, datagram will be faster).
• Because it is more primitive, it is more flexible.
• Congestion/failed link can be avoided (more reliable).
Problems:
• Packets may be delivered out of order.
• If a node crashes momentarily, all of its
queued packets are lost.
Reduction of transmission time because of
parallelism in transmission in packet switching
technique
Virtual Circuit Versus Datagram Packet Switching
Key features of the virtual circuit packet switching approach are as follows:
•Node need not decide route
•More difficult to adopt to congestion
•Maintains sequence order
•All packets are sent through the same predetermined route
On the other hand, the key features of the datagram packet switching are as follows:
•Each packet is treated independently
•Call set up phase is avoided
•Inherently more flexible and reliable
Circuit Switching
In this networking method, a connection called a circuit is set up between two devices, which is
used for the whole communication. Information about the nature of the circuit is maintained by
the network. The circuit may either be a fixed one that is always present, or it may be a circuit
that is created on an as-needed basis. Even if many potential paths through intermediate devices
may exist between the two devices communicating,
only one will be used for any given dialog. In a
circuit-switched network, before communication can
occur between two devices, a circuit is
established between them. This is shown as a thick
blue line for the conduit of data from Device A to
Device B, and a matching purple line from B back to A.
Once set up, all communication between these devices
takes place over this circuit, even though there Fig. circuit switching
are other
possible ways that data could conceivably be passed over the network of devices between them.
The classic example of a circuit-switched network is the telephone system. When you call
someone and they answer, you establish a circuit connection and can pass data between you, in a
steady stream if desired. That circuit functions the same way regardless of how many
intermediate devices are used to carry your voice. You use it for as long as you need it, and then
terminate the circuit. The next time you call, you get a new circuit, which may (probably will)
use different hardware than the first circuit did, depending on what's available at that time in the
network.
Packet Switching
In this network type, no specific path is used for data transfer. Instead, the data is chopped up
into small pieces called packets and sent over the network. The packets can be routed, combined
or fragmented, as required to get them to their eventual destination. On the receiving end, the
process is reversed—the data is read from the packets and re-assembled into the form of the
original data. A packet-switched network is more analogous to the postal system than it is to the
telephone system (though the comparison isn't
perfect.) In a packet-switched network, no circuit
is set up prior to sending data between devices.
Blocks of data, even from the same file or
communication, may take any number of paths as
it journeys from one device to another.
Datagram Networks
Example for Datagram Network
Example for Virtual Circuit
Network
Two basic approaches to packet switching are common: The most common is datagram
switching (also known as a "best-effort network" or a network supporting the connection-less
network service).This is what is used in the network layer of the Internet.
Datagram Packet Networks
Datagram transmission uses a different scheme to determine the route through the network of
links. Using datagram transmission, each packet is treated as a separate entity and contains a
header with the full information about the intended recipient. The intermediate nodes examine
the header of a packet and select an appropriate link to an intermediate node which is nearer the
destination. In this system, the packets do not follow a pre-established route, and the
intermediate nodes (usually known as "routers") do not require prior knowledge of the routes that
will be used.
A datagram network is analogous to sending a message as a series of postcards through the
postal system. Each card is independently sent to the final destination (using the postal system).
To receive the whole message, the receiver must collect all the postcards and sort them into the
original order. Not all postcards need be delivered by the postal system, and not all take the same
length of time to arrive.
In a datagram network delivery is not guaranteed (although they are usually reliably sent).
Enhancements, if required, to the basic service (e.g. reliable delivery) must be provided by the
end systems (i.e. user's computers) using additional software. The most common datagram
network is the Internet which uses the IP network protocol. Applications which do not require
more than a best effort service can be supported by direct use of packets in a datagram network
(using the User Datagram Protocol (UDP) transport protocol). Such applications include Internet
Video, Voice Communication, messages notifying a user that she/he has received new email, etc.
Most Internet applications need additional functions to provide reliable communication (such as
end-to-end error and sequence control). Examples include sending email, browsing a web site, or
sending a file using the file transfer protocol (ftp). This reliability ensures all the data is received
in the correct order with no duplication or omissions. It is provided by additional layers of
software algorithms implemented in the End Systems (A,D). Two examples of this are the
Transmission Control Protocol (TCP), and the Trivial File Transfer Protocol (TFTP) which uses
UDP.
One merit of the datagram approach is that not all packets need to follow the same path (route)
through the network (although frequently packets do follow the same route). This removes the
need to set-up and tear-down the path, reducing the processing overhead, and a need for
Intermediate Systems to execute an additional protocol.
Packets may also be routed around busy parts
of the network when alternate paths exist.
This is useful when a particular intermediate
system becomes busy or overloaded with
excessive volumes of packets to send. It can
also provide a high degree of fault tolerance,
when an individual intermediate system or
communication circuit fails. As long as a
route exists through the network between
two end systems, they are able to
communicate. Only if there is no possible way to send the packets, will the packets be discarded
and not delivered. The fate (success/failure) of an application therefore depends only on
existence of an actual path between the two End Systems (ESs). This is known as "fate sharing" since the application shares the "fate" of the network. There is another type of network known as
a virtual circuit network
Congestion Control
When one part of the subnet (e.g. one or more routers in an area) becomes overloaded,
congestion results. Because routers are receiving packets faster than they can forward them, one
of two things must happen:
1. The subnet must prevent additional packets from entering the congested region until those
already present can be processed.
2. The congested routers can discard queued packets to make room for those that are arriving.
We now consider the problem of congestion and some possible solutions. Three general
approaches:
1. prevent it altogether
2. congestion avoidance
3. deal with it if it occurs
Preallocation of Resources
Preallocation schemes aim to prevent congestion from happening in the first place. For example,
we can require that resources be preallocated before any packets can be sent, guaranteeing that
resources will be available to process each packet. In virtual circuit networks, for example, the
sender opens a connection before sending data. The circuit setup operation selects a path through
the subnet, and each router on the path dedicates buffer space and bandwidth to the new circuit.
What happens when a user attempts to open a virtual circuit and the subnet is congested? The
subnet can refuse to open the connection, forcing the user to wait until sufficient resources
become available. Note: The ability of the subnet to reject requests to open connections is an
important
property of connection oriented networks.
Traffic Shaping
Control the rate at which packets are sent (not just how many). A congestion avoidance method.
Relates to Quality of Service (QoS) as means to provide more reliable service despite variable
transmission patterns. At set up, the sender and carrier negotiate a traffic pattern (shape).Leaky
Bucket Algorithm used to control rate in a datagram network. A single-server queue with
constant service time. If bucket (buffer) overflows then packets are discarded. Enforces a
constant output rate regardless of burstiness of input. Does nothing when input is idle. In contrast
the Token Bucket Algorithm causes a token to be generated periodically, which during idle
periods can be saved up. Related to traffic shaping is how specification, where a particular
quality of service is agreed upon between sender, receiver and carrier.
Isarithmic Congestion Control
Another approach to congestion avoidance is to limit the total number flow of packets in the
subnet at any one time. The idea is similar to the token ring:
1. When an router accepts a packet from a host, it must obtain a permit before sending the packet
into the subnet.
2. Obtaining a permit is analogous to \seizing the token", but there can be many permits in the
subnet. When an router obtains a permit, it destroys it.
3. The destination router regenerates the permit when it passes the packet to the destination host.
Issues:
1. Although we have limited the total number of packets in the subnet, we have no control over
where in the subnet those packets will be. Thus, an router in one part of the subnet might be
congested, while an router in another part remains idle, but unable to process packets for lack of
permits.
2. Regenerating lost permits is di_cult, because no single node knows how many permits are
currently in the subnet.
3. How to distribute permits? Distribute among the routers or centralize for a known access
point.
Virtual Circuits Admission Control
Refuse to set up new connections if congestion is present. Also helps with congestion avoidance.
Flow Control
Flow control is aimed at preventing a fast sender from overwhelming a slow receiver. Flow
control can be helpful at reducing congestion, but it can't really solve the congestion problem.For
example, suppose we connect a fast sender and fast receiver using a 9.6 kbps line:
1. If the two machines use a sliding window protocol, and the window is large, the link will
become congested in a hurry.
2. If the window size is small (e.g., 2 packets), the link won't become congested. Note how the
window size limits the total number of packets that can be in transmission at one time.
Flow control can take place at many levels:
 User process to user process (end-to-end). Later, we'll see how TCP uses flow control at
the end-to-end level.
 Host to host. For example, if multiple application connections share a single virtual
circuit between two hosts.

router to router: For example, in virtual circuits.
Load Shedding/Discarding Packets (No Preallocation)
At the other end of the spectrum, we could preallocate no resources in advance, and take our
chances that resources will be available when we need them. When insufficient resources are
present to process existing packets, discard queued packets to make room for newly arriving
ones.
Who retransmits the discarded packets?
Two cases: connection oriented and connectionless.
In datagram (connectionless) networks, the sending host (transport layer) retransmits discarded
packets (if appropriate). In virtual circuit networks, the previous-hop router retransmits the
packet when it fails to receive an acknowledgment.
Failure to preallocate resources leads to two problems: potential deadlock and unfairness.
First, let us consider deadlock. Suppose that all of an router's buffers hold packets. Because the
router has no free buffers, it cannot accept additional frames. Unfortunately, it also ignores
frames containing ACKs that would free up some of those buffers! Suppose further, that two
adjacent routers, A and B, are sending packets to each other. Since both are waiting for the other
to accept a packet, neither can proceed. This condition is known as a deadlock.
Solution: Reserve at least one buffer for each input line and use it to hold incoming
packets. Note that we can extract the ACK field and still discard the packet, if we don't have
buffers to hold it.
Advantage of discarding packets when congested: Easy to implement.
Disadvantages:
1. Wastes resources. The network may have expended considerable resources processing a
packet that is eventually discarded.
2. Non-deterministic. There is fewer guarantees than with virtual circuits that packets will ever
reach their destination.
3. Requires that sending hosts pay attention to congestion. If the network can't prevent a host
from sending data, a host can overload the network. In particular, a” broken” host may cause the
network to become overly congested.
4. In the extreme case, congestion collapse occurs. The network becomes so overloaded, that few
packets reach their destination. Meanwhile, the sending hosts continue to generate more data
(both retransmissions and new packets). This condition occurred several times back in 1987, and
the Internet/Arpanet became unusable for a period of hours to days.
Random Early Detection (RED)
Start dropping packets before a router runs out of buffer space. Basic idea:
Choke Packets
ECN (Explicit Congestion Notification) is an
example where instead of dropping packets,
routers mark packets indicating that router is
congested. Routers can monitor the level of
congestion around them, and when congestion is
present, they can send choke packets to the
sender that say “slow down".
How can an router measure congestion? An
router might estimate the level of congestion by
measuring the percentage of buffers in use, line utilization, or average queue lengths.
Advantage: Dynamic. Host sends as much data as it wants, the network informs it when it is
sending too much.
Disadvantage:
1. Difficult to tune. By how much should a host slow down? The answer depends on how much
traffic the host is sending, how much of the congestion it is responsible for,and the total capacity
of the congested region. Such information is not readily available in practice.
2. After receiving a choke packet, the sending host should ignore additional choke packets for a
short while because packets currently in transmission may generate additional choke packets.
How long? Depends on such dynamic network conditions as delay.
Variations exist.
LAN: Baseband versus Broadband
There are two LAN transmission options, Baseband and Broadband. Baseband LANs which are
the most prevalent are single channel systems that support a single transmission at any given
time. Broadband LANs which are most unusual support multiple transmissions via frequency
channels.
Broadband LANs
Broadband LANs are multichannel typically based on coaxial cable as the transmission media,
although fiber optic cable is also used. Individual channels offer bandwidth of 1 to 5 Mbps, with
20 to 30 channels typically Aggregate bandwidth is as much 500MHz. The characteristics may
be given as follows:




Digital Signal onto RF carrier (Analog)
Channel allocation based on FDM.
Head-End for bi-directional transmission.
Stations connected via RF modems radio modems accomplish the digital-to-analog
conversion process providing the transmitting device access to an analog channel.
Advantages of Broadband



Data, voice and video can be accomplished on broadband channel.
Greater distances
Greater bandwidth
Disadvantages of Broadband



Cable Design
Alignment and maintenance
High cost, requires modems

Lack of well developed standards
Baseband LANs
Baseband LAN is single channel, supporting a single communication at a time. They are digital
in nature. The total bandwidth of 1 to 100 Mbps is provided over coaxial cable, UTP,STP or
fiber optic cable. Distance limitations depend on the medium employed and the specifics of the
LAN protocol.
Baseband LAN is the most popular and the most highly standardized. Ethernet, Token passing,
Token Ring and FDDI LANs are all baseband. They are intended only for data, as data
communication is, after all, the primary reason for the existence of LANs. The characteristics of
this system may be summarized as follows:




Unmodulated digital signal.
Single channel.
Bi-directional propagation via T connectors.
No need of modems-low cost installation
Advantages of Baseband




Simplicity
Low cost
Ease of installation and maintenance
High rates
Disadvantages of Baseband


Limited distances
Data and voice only
Carrier Sense Networks-CSMA/CD
CSMA is a network access method used on shared network topologies such as Ethernet to
control access to the network. Devices attached to the network cable listen (carrier sense) before
transmitting. If the channel is in use, devices wait before transmitting. MA (multiple access)
indicates that many devices can connect to and share the same network. All devices have equal
access to use the network when it is clear. Even though devices attempt to sense whether the
network is in use, there is a good chance that two stations will attempt to access it at the same
time. On large networks, the transmission time between one end of the cable and another is
enough that one station may access the cable even though another has already just accessed it.
There are two methods for avoiding these so-called collisions, listed here:

CSMA/CD (carrier sense multiple access/collision detection)
CD (collision
detection) defines what happens when two devices sense a clear channel, then attempt to

transmit at the same time. A collision occurs, and both devices stop transmission, wait for
a random amount of time, then retransmit. This is the technique used to access the 802.3
Ethernet network channel. This method handles collisions as they occur, but if the bus is
constantly busy, collisions can occur so often that performance drops drastically. It is
estimated that network traffic must be less than 40 percent of the bus capacity for the
network to operate efficiently. If distances are long, time lags occur that may result in
inappropriate carrier sensing, and hence collisions.
CSMA/CA (carrier sense multiple access/collision avoidance) In CA (collision
avoidance), collisions are avoided because each node signals its intent to transmit before
actually doing so. This method is not popular because it requires excessive overhead that
reduces performance.
Ring Network




A ring network is a network topology in which each node connects to exactly two other
nodes, forming a single continuous pathway for signals through each node - a ring. Data
travels from node to node, with each node along the way handling every packet.
Because a ring topology provides only one pathway between any two nodes, ring
networks may be disrupted by the failure of a single link. A node failure or cable break
might isolate every node attached to the ring.
FDDI networks overcome this vulnerability by sending data on a clockwise and a
counterclockwise ring: in the event of a break data is wrapped back onto the
complementary ring before it reaches the end of the cable, maintaining a path to every
node along the resulting "C-Ring".
Many ring networks add a "counter-rotating ring" to form a redundant topology. Such
"dual ring" networks include Spatial Reuse Protocol, Fiber Distributed Data Interface
(FDDI), and Resilient Packet Ring.
Ring network - Advantages




All stations have equal access
Each node on the ring acts as a repeater, allowing ring networks to span greater distances
than other physical topologies.
Because data travels in one direction high speeds of transmission of data are possible
When using a coaxial cable to create a ring network the service becomes much faster.
Ring network - Disadvantages




Often the most expensive topology
Numerous connections to other computers mean that a computer is less likely to become
isolated from the network due to a failure
If one node fails, the rest of the network could also fail
Damage to the ring will affect the whole network
IEEE 802 Standards
The IEEE (Institute of Electrical and Electronic Engineers) is a technical association of industry
professionals with a common interest in advancing all communications technologies. The
LAN/MAN Standards Committee (LMSC), develops LAN (local area network) and MAN
(metropolitan area network) standards, mainly for the lowest two layers in the OSI reference
model. LMSC is also called the IEEE Project 802, so the standards it develops are referenced as
IEEE 802 standards. IEEE 802 standards define physical network interfaces such as network
interface cards, bridges, routers, connectors, cables, and all the signaling and access methods
associated with physical network connections.
While connecting computers through networks we need to have set of rules/standards for the data
to travel from one computer to other computer. One such set of rules for the networking traffic
to follow is IEEE802 standards developed by IEEE .The standards such as IEEE 802 helps
industry provide advantages such as, interoperability, low product cost, and easy to manage
standards. The IEEE 802 standards are further divided into many parts –
802.1,802.2,802.3(Ethernet CSMA/CD) etc.
X.25
X.25 is an International Telecommunication Union-Telecommunication Standardization Sector
(ITU-T) protocol standard for WAN communications that defines how connections between user
devices and network devices are established and maintained. X.25 is designed to operate
effectively regardless of the type of systems connected to the network. It is typically used in the
packet-switched networks (PSNs) of common carriers, such as the telephone companies.
Subscribers are charged based on their use of the network.
X.25 Devices and Protocol Operation
X.25 network devices fall into three general categories: data terminal equipment (DTE), data
circuit-terminating equipment (DCE), and packet-switching exchange (PSE). Data terminal
equipment devices are end systems that communicate across the X.25 network. They are usually
terminals, personal computers, or network hosts, and are located on the premises of individual
subscribers. DCE devices are communications devices, such as modems and packet switches,
that provide the interface between DTE devices and a PSE, and are generally located in the
carrier's facilities. PSEs are switches that compose the bulk of the carrier's network. They
transfer data from one DTE device to another through the X.25 PSN.
Figure: DTEs, DCEs, and
PSEs Make Up an X.25
Network
illustrates
the
relationships among the three
types of X.25 network
devices:
Figure:
DTEs,
DCEs, and PSEs Make Up an
X.25 Network
Packet Assembler/Disassembler
The packet assembler/disassembler (PAD) is a device found in X.25 networks. PADs are used
when a DTE device, such as a character-mode terminal, is too simple to implement the full X.25
functionality. The PAD is located between a DTE device and a DCE device, and it performs
three primary functions: buffering (storing data until a device is ready to process it), packet
assembly, and packet disassembly. The PAD buffers data sent to or from the DTE device. It also
assembles outgoing data into packets and forwards them to the DCE device and adds an X.25
header. Finally, the PAD disassembles incoming packets before forwarding the data to the DTE
and removes the X.25 header .
Figure: The PAD Buffers,
Assembles, and Disassembles
Data Packets illustrates the basic
operation of the PAD when
receiving packets from the X.25
WAN:
Figure: The PAD Buffers, Assembles, and Disassembles Data Packets
X.25 Session Establishment
X.25 sessions are established when one DTE device contacts another to request a communication
session. The DTE device that receives the request can either accept or refuse the connection. If
the request is accepted, the two systems begin full-duplex information transfer. Either DTE
device can terminate the connection. After the session is terminated, any further communication
requires the establishment of a new session.
X.25 Virtual Circuits
A virtual circuit is a logical connection created to ensure reliable communication between two
network devices. A virtual circuit denotes the existence of a logical, bidirectional path from one
DTE device to another across an X.25 network. Physically, the connection can pass through any
number of intermediate nodes, such as DCE devices and PSEs. Multiple virtual circuits (logical
connections) can be multiplexed onto a single physical circuit (a physical connection). Virtual
circuits are demultiplexed at the remote end, and data is sent to the appropriate destinations.
Figure: The PAD Buffers, Assembles, and Disassembles Data Packets illustrates the basic
operation of the PAD when receiving packets from the X.25 WAN:
Figure: Virtual Circuits Can Be Multiplexed onto
a Single Physical Circuit illustrates four separate
virtual circuits being multiplexed onto a single
physical circuit:
Two types of X.25 virtual circuits exist: switched and permanent. Switched virtual circuits
(SVCs) are temporary connections used for sporadic data transfers. They require that two DTE
devices establish, maintain, and terminate a session each time the devices need to communicate.
Permanent virtual circuits (PVCs) are permanently established connections used for frequent and
consistent data transfers. PVCs do not require that sessions be established and terminated.
Therefore, DTEs can begin transferring data whenever necessary because the session is always
active.
The basic operation of an X.25 virtual circuit begins when the source DTE device specifies the
virtual circuit to be used (in the packet headers) and then sends the packets to a locally connected
DCE device. At this point, the local DCE device examines the packet headers to determine
which virtual circuit to use and then sends the packets to the closest PSE in the path of that
virtual circuit. PSEs (switches) pass the traffic to the next intermediate node in the path, which
may be another switch or the remote DCE device.
When the traffic arrives at the remote DCE device, the packet headers are examined and the
destination address is determined. The packets are then sent to the destination DTE device. If
communication occurs over an SVC and neither device has additional data to transfer, the virtual
circuit is terminated.
The X.25 Protocol Suite
The X.25 protocol suite maps to the lowest three layers of the OSI reference model. The
following protocols are typically used in X.25 implementations: Packet-Layer Protocol (PLP),
Link Access Procedure, Balanced (LAPB), and those among other physical-layer serial
interfaces (such as EIA/TIA-232, EIA/TIA-449, EIA-530, and G.703).
Figure: Key X.25 Protocols Map to the Three Lower Layers of the OSI Reference Model maps
the key X.25 protocols to the layers of the OSI reference model:
Figure: Key X.25 Protocols Map to the Three Lower Layers of the OSI Reference Model
Packet-Layer Protocol
PLP is the X.25 network layer protocol. PLP manages packet exchanges between DTE devices
across virtual circuits. PLPs also can run over Logical Link Control 2 (LLC2) implementations
on LANs and over Integrated Services Digital Network (ISDN) interfaces running Link Access
Procedure on the D channel (LAPD).
The PLP operates in five distinct modes: call setup, data transfer, idle, call clearing, and
restarting.
Call setup mode is used to establish SVCs between DTE devices. A PLP uses the X.121
addressing scheme to set up the virtual circuit. The call setup mode is executed on a per-virtualcircuit basis, which means that one virtual circuit can be in call setup mode while another is in
data transfer mode. This mode is used only with SVCs, not with PVCs.
Data transfer mode is used for transferring data between two DTE devices across a virtual
circuit. In this mode, PLP handles segmentation and reassembly, bit padding, and error and flow
control. This mode is executed on a per-virtual-circuit basis and is used with both PVCs and
SVCs.
Idle mode is used when a virtual circuit is established but data transfer is not occurring. It is
executed on a per-virtual-circuit basis and is used only with SVCs.
Call clearing mode is used to end communication sessions between DTE devices and to
terminate SVCs. This mode is executed on a per-virtual-circuit basis and is used only with SVCs.
Restarting mode is used to synchronize transmission between a DTE device and a locally
connected DCE device. This mode is not executed on a per-virtual-circuit basis. It affects all the
DTE device's established virtual circuits.
Four types of PLP packet fields exist:




General Format Identifier (GFI) - Identifies packet parameters, such as whether the packet
carries user data or control information, what kind of windowing is being used, and whether
delivery confirmation is required.
Logical Channel Identifier (LCI) - Identifies the virtual circuit across the local DTE/DCE
interface.
Packet Type Identifier (PTI) - Identifies the packet as one of 17 different PLP packet types.
User Data - Contains encapsulated upper-layer information. This field is present only in data
packets. Otherwise, additional fields containing control information are added.
Session Layer
The session layer resides above the transport layer, and provides “value added” services to the
underlying transport layer services. The session layer (along with the presentation layer) add
services to the transport layer that are likely to be of use to applications, so that each application
doesn't have to provide its own implementation. It is the thinnest layer in the OSI model. At the
time the model was formulated, it was not clear that a session layer was needed.
The session layer provides the following services:
Dialog management: Deciding whose turn it is to talk. Some applications operate in half-duplex
mode, whereby the two sides alternate between sending and receiving messages, and never send
data simultaneously. In the ISO protocols, dialog management is implemented through the use of
a data token. The token is sent back and forth, and a user may transmit only when it possesses
the token.
Synchronization: Move the two session entities into a known state. The transport layer handles
only communication errors, synchronization deals with upper layer errors. In a file transfer, for
instance, the transport layer might deliver data correctly, but the application layer might be
unable to write the le because the le system is full. Users can split the data stream into pages,
inserting synchronization points between each page. When an error occurs, the receiver can
resynchronize the state of the session to a previous synchronization point. This requires that the
sender hold data as long as may be needed.
Synchronization is achieved through the use of sequence numbers. The ISO protocols provide
both major and minor synchronization points. When resynchronizing, one can only go back as
far as the previous major synchronization point. In addition, major synchronization points are
acknowledged through explicit messages (making their use expensive). In contrast, minor
synchronization points are just markers.
Activity management: Allow the user to delimit data into logical units called activities. Each
activity is independent of activities that come before and after it, and an activity can be processed
on its own. Activities might be used to delimit les of a multiple transfer. Activities are also used
for quarantining, collecting all the messages of a multi-message exchange together before
processing them. The receiving application would begin processing messages only after all the
messages had arrived. This provides a way of helping insure that all or none of a set of
operations are performed.
For example, a bank transaction may consist of locking a record, updating a value, and then
unlocking the record. If an application processed the first operation, but never received the
remaining operations (due to client or network failures), the record would remain locked forever.
Quarantining addresses this problem.
Exception handling: A General purpose mechanism for reporting errors.
Note: The TCP/IP protocols do not include a session layer at all.
Remote Procedure Call (RPC)
Remote Procedure Call (RPC) provides a different paradigm for accessing network services.
Instead of accessing remote services by sending and receiving messages, a client invokes
services by making a local procedure call. The local procedure hides the details of the network
communication.
When making a remote procedure call:
1. The calling environment is suspended, procedure parameters are transferred across the
network to the environment where the procedure is to execute, and the procedure is executed
there.
2.When the procedure finishes and produces its results, its results are transferred back to the
calling environment, where execution resumes as if returning from a regular procedure call. The
main goal of RPC is to hide the existence of the network from a program. As a result, RPC
doesn't quite t into the OSI model:
1. The message-passing nature of network communication is hidden from the user. The user
doesn't first open a connection, read and write data, and then close the connection. Indeed, a
client often doesn’t even know they are using the network!
2. RPC often omits many of the protocol layers to improve performance. Even a small
performance improvement is important because a program may invoke RPCs often. For example,
on (diskless) Sun workstations, every file access is made via an RPC.
RPC is especially well suited for client-server (e.g., query-response) interaction in which the
flow of control alternates between the caller and callee. Conceptually, the client and server do
not both execute at the same time. Instead, the thread of execution jumps from the caller to the
callee and then back again. The following steps take place during an RPC:
1. A client invokes a client stub procedure, passing parameters in the usual way. The client stub
resides within the client's own address space.
2. The client stub marshalls the parameters into a message. Marshalling includes converting the
representation of the parameters into a standard format, and copying each parameter into the
message.
3. The client stub passes the message to the transport layer, which sends it to the remote server
machine.
4. On the server, the transport layer passes the message to a server stub, which demarshalls the
parameters and calls the desired server routine using the regular procedure call mechanism.
5. When the server procedure completes, it returns to the server stub (e.g., via a normal
procedure call return), which marshalls the return values into a message. The server stub then
hands the message to the transport layer.
6. The transport layer sends the result message back to the client transport layer, which hands the
message back to the client stub.
7. The client stub demarshalls the return parameters and execution returns to the caller.
RPC Issues
Issues that must be addressed:
Marshalling: Parameters must be marshalled into a standard representation. Parameters consist of
simple types (e.g., integers) and compound types (e.g., C structures or Pascal records).
Moreover, because each type has its own representation, the types of the various parameters must
be known to the modules that actually do the conversion. For example, 4 bytes of characters
would be uninterpreted, while a 4-byte integer may need to the order of its bytes reversed.
Semantics: Call-by-reference not possible: the client and server don't share an address space.
That is, addresses referenced by the server correspond to data residing in the client's address
space.
One approach is to simulate call-by-reference using copy-restore. In copy-restore, call-byreference parameters are handled by sending a copy of the referenced data structure to the server,
and on return replacing the client's copy with that modified by the server. However, copy-restore
doesn't work in all cases. For instance, if the same argument is passed twice, two copies will be
made, and references through one parameter only changes one of the copies.
Binding: How does the client know who to call, and where the service resides?
The most flexible solution is to use dynamic binding and and the server at run time when the
RPC is first made. The first time the client stub is invoked, it contacts a name server to determine
the transport address at which the server resides.
Transport protocol: What transport protocol should be used?
Exception handling: How are errors handled?
Binding
We'll examine one solution to the above issues by considering the approach taken by Birrell and
Nelson. Binding consists of two parts:
Naming refers to what service the client wants to use. In B&N, remote procedures are named
through interfaces. An interface uniquely identifies a particular service, describing the types and
numbers of its arguments. It is similar in purpose to a type definition in programming languages.
For example, a “phone” service interface might specify a single string argument that returns a
character string phone number. Locating refers to finding the transport address at which the
server actually resides. Once we have the transport address of the service, we can send messages
directly to the server. In B&N's system, a server having a service to offer exports an interface for
it. Exporting an interface registers it with the system so that clients can use it. A client must
import an (exported) interface before communication can begin. The export and import
operations are analogous to those found in object-oriented systems.
Interface names consists of two parts:
1. A unique type specifies the interface (service) provided. Type is a high-level specification,
such as “mail” or “file access".
2. An instance specifies a particular server oering a type (e.g., file access on wpi").
Name Server
B&N's RPC system was developed as part of a distributed system called Grapevine. Grapevine
was developed at Xerox by the same research group the developed the Ethernet. Among other
things, Grapevine provides a distributed, replicated database, implemented by servers residing at
various locations around the internet. Clients can query, add new entries or modify existing
entries in the database. The Grapevine database maps character string keys to entries called
RNames. There are two types of entries:
Individual: A single instance of a service. Each server registers the transport address at which its
service can be accessed and every instance of an interface is registered as an individual entry.
Individual entries map instances to their corresponding transport addresses.
Group: The type of an interface, which consists of a list of individual RNames. Group entries
contain RNames that point to servers providing the service having that group name. Group
entries map a type (interface) to a set of individual entries providing that service. For example, if
wpi and bigboote both offered le access, the group entry le access” would consists of two
individual RNames, one for wpi and bigboote's servers.
When a server wishes to export an interface:
1. It calls its server stub, which then calls Grapevine, passing it the type and instance of the
service it wishes to register. Once the interface has been registered with Grapevine, it can be
imported by the client. Note: Grapevine insures that both an individual and a group entry has
been established for the exported service.
2. The server stub then records information about the instance in an internal export table. In
B&N, there is one export table per machine, containing entries for all currently exported
interfaces. This table is used to map incoming RPC request messages to their corresponding
server procedure.
3. Each entry in the export table contains: a unique identifier that identifies that interface, and a
pointer to the server stub that should be called to invoke the interface service. The unique
identifier is never reused. If the server crashes and restarts, new identiers are used.
The client binds to an exported service as follows:
1.The client stub calls the Grapevine database to nd an instance of the desired type.
2. The database server returns the desired group entry, and the client chooses one of the
individual servers.
3. The client stub sends a message to the selected server stub asking for information about that
instance.
4. The server stub returns the unique identifier and index of the appropriate export table entry.
The client saves the [identier, index] pair as it will need it when actually making an RPC.
5. The client is now ready to actually call the remote procedure:
(a) The client sends the export table index and the unique identifier together with the parameters
of the call to the server.
(b) Upon receipt of a message, the server stub uses the table index contained in the message to
nd the appropriate entry in its export table.
(c) The server stub then compares the provided identifier with the one in the table. If they dier,
reject the procedure call as invalid. Otherwise call the server stub procedure.
Binding Notes
Note: The unique identifiers in the export table change whenever a server crashes and restarts,
allowing the client to detect server restarts between calls. In those cases where a client doesn't
care if the server has restarted, it simply rebinds to another instance of the interface and restarts
the remote call.
Note: Identiers are managed by the server and are not stored in the Grapevine database. Storing
them in the Grapevine database would reduce the number of messages exchanged during the
binding phase. However, the current approach greatly reduces the load on the Grapevine servers.
In most cases, when a server exports an interface to Grapevine, the entry will have been
registered previously, and no updates to the database are required (updates are expensive because
the database is distributed).
Using Grapevine's database provides late binding. Binding callers to specific servers at runtime
makes it possible to move the server to another machine without requiring changes to client
software. Finally, the separate registering of types and instances provides great flexibility. Rather
than binding to a specific instance, a client asks for a specific type. Because all instances of a
type implement the same interface, the RPC support routines would take the list of instances
returned by Grapevine and chose the one that is closest to the client.
How is binding done on other systems?
Semantics of RPC
Unlike normal procedure calls, many things can go wrong with RPC. Normally, a client will
send a request, the server will execute the request and then return a response to the client. What
are appropriate semantics for server or network failures?
Possibilities:
1. Just hang forever waiting for the reply that will never come. This models regular procedure
call. If a normal procedure goes into an infinite loop, the caller never finds out. Of course, few
users will like such semantics.
2. Time out and raise an exception or report failure to the client. Of course, finding an
appropriate timer value is difficult. If the remote procedure takes a long time to execute, a timer
might time-out too quickly.
3. Time out and retransmit the request. While the last possibility seems the most reasonable, it
may lead to problems. Suppose that:
1. The client transmits a request, the server executes it, but then crashes before sending a
response. If we don't get a response, is there any way of knowing whether the server acted on the
request?
2. The server restarts, and the client retransmits the request. What happens? Now, the server will
reject the retransmission because the supplied unique identifier no longer matches that in the
server's export table. At this point, the client can decide to rebind to a new server and retry, or it
can give up.
3. Suppose the client rebinds to the another server, retransmits the request, and gets a response.
How many times will the request have been executed? At least once, and possibly twice. We
have no way of knowing. Operations that can safely be executed twice are called idempotent. For
example, fetching the current time and date, or retrieving a particular page of a file.
Is deducting $10,000 from an account idempotent? No. One can only deduct the money once.
Likewise, deleting a file is not idempotent. If the delete request is executed twice, the
first attempt will be successful, while the second attempt produces a nonexistent file error.
RPC Semantics
While implementing RPC, B&N determined that the semantics of RPCs could be categorized in
various ways:
Exactly once: The most desirable kind of semantics, where every call is carried out exactly once,
no more and no less. Unfortunately, such semantics cannot be achieved at low cost; if the client
transmits a request, and the server crashes, the client has no way of knowing whether the server
had received and processed the request before crashing.
At most once: When control returns to the caller, the operation will have been executed no more
than once. What happens if the server crashes? If the server crashes, the client will be notified of
the error, but will have no way of knowing whether or not the operation was performed.
At least once: The client just keeps retransmitting the request until it gets the desired response.
On return to the caller, the operation will have be performed at least one time, but possibly
multiple times.
Transport Layer
The design of a transport protocol is more challenging than the design of a data link protocol due
to a major difference between the environments in which the two layers operate.
• Two data link protocol entities communicate directly via a physical channel.
• At the transport layer, the protocol entities communicate through a set of networks (not
always known).
Introduction
The transport layer aims at ensuring end to end reliable data transfer. In order to do so, transport
protocol performs is function on top of a network layer. The complexity of the transport layer
depends on the service provided by the network layer.
Virtual Circuit Service:
• The messages are delivered in order from sender to receiver, without error, loss, or
duplication.
• The transport protocol becomes relatively simple.
Datagram Services:
• Packet can be lost, corrupted, etc. It is up to the hosts to make sure that messages are
delivered in order without loss, error, or duplication.
• The transport protocol becomes complex and sophisticated.
OSI defines 5 classes of transport protocols:
TP0 to TP4. TP0 is supposed to work on top
of a perfect network, while TP4 is for the
worst kind of networks.
Simplified view of protocol architecture
(importance of transport layer)
Quality of Service
QoS can be characterized by a number of specific parameters. OSI transport service allows the
applications to specify preferred, acceptable, and unacceptable values for these parameters at the
time a connection is set up.
It is up to the transport layer to examine these parameters, and depending on the kind of network
services available to it, determine whether it can provide the required service.
1. Connection establishment delay
2. Connection establishment failure probability
3. Throughput
4. Transit delay
5. Residual error rate
6. Transfer failure probability
7. Connection release delay
8. Connection release failure probability
9. Protection
10. Priority
Transport Service
Provided to the session layer.
Connection-less Service:
Service Primitives:
T - UNITDATA.request (callee, caller, qos, user_data)
T - UNITDATA.indication (callee, caller, qos, user_data)
Connection Oriented Service:
Three phases:
• Transport Connection (TC) establishment (This is a logical connection)
• Data Transfer
• Transport Connection Release
Terminology
• Callee: Transport address (TSAP) to be called
• Caller: Transport address (NSAP) used by calling transport entity
• Exp_wanted: Boolean flag specifying whether expedited data will be sent
• Qos: Quality of service desired
• User_data: 0 or more bytes of data transmitted but not examined
• Reason: Why did it happen
• Responder: Transport address connected to at the destination
Transport Protocol
Two transport layer entities communicate with each other by exchanging TPDUs.
The different TPDU types necessary to implement the basic set of services introduced earlier are
as follows:
CR: Connection request
CC: Connection Confirm
DR: Disconnect request
DC: Disconnect confirm
DT: Data
AK: Acknowledgement
The use of each TPDU will be described in relation to the various services previously described.
Examples of Transport Protocols
In this section we will look at specific transport protocols from the TCP/IP architecture.
TCP (Transmission Control Protocol) : Connection oriented, reliable end to end.
Header format
TCP connection establishment
• Allow each end to now the other exists and
willing to communicate
• Negotiation of optional parameters for the
communication
• At both ends: Allocation of transport entity
resources done at this time.
First attempt to establish a connection 2-way handshaking
States of TCP entities: Active Open vs. Passive Open
Actually A sends SYN j and B sends SYN j …
2-way handshake -> obsolete SYN -> problems
Check the following scenario:
SYNi has been traveling in the Internet for a while and suddenly shows up at the B side.
There is a need for 3-way handshaking .
Connection Termination
Either or both sides can initiate the termination of the connection.
Termination can be abrupt or graceful (must accept incoming data until FIN received)
One scenario:
On one side:
•
•
•
•
•
•
A Transport Service (TS) user calls Close request primitive
Transport entity (TE) sends FIN segement, requesting termination
Connection on this side is placed in FIN WAIT state
Continue to accept data from the other TE and deliver data to user
User does not send any more data from this side
When FIN received from the other TE, informs user and closes connection
The other side:
• The TE receives FIN segment
• Informs TS user, and goes into CLOSE WAIT state
• Continues to accept data from TS user and transmit it
• When TS user issues CLOSE primitive, TE sends FIN
Now, the connection is closed
TCP Entity state diagram: state and transitions
5.1.5. Flow Control
Fixed size Sliding Window does not work properly, because of unreliable networks, and variable
conditions for the TCP entity …
Credit scheme: make the size of the window variable � receiver can acknowledge and still can
restrain the sender window, because receiver is not ready to receive all what sender can send
after receiving the ACK.
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Credit Scheme: Data transfer as a stream of octets numbered modulo 2
Flow control by credit allocation of number of octets
Congestion Control (Issues)
We will discuss a few issues here:
Retransmission timer management
• Estimate round trip delay by observing pattern of delays
• Set time to value somewhat greater than estimate
• Simple average
• Exponential average
• RTT Variance Estimation (Jacobson’s algorithm)
Window Management
• Slow start
1. awnd = MIN[credit, cwnd]
2. Start connection with cwnd=1
3. Increment cwnd at each ACK, to some max
Problems …
• Dynamic windows sizing on congestion
1. When a timeout occurs, set slow start threshold to half current congestion window:
ssthresh=cwnd/2
2. Set cwnd = 1 and slow start until cwnd = ssthresh Increasing cwnd by 1 for every ACK
3. For cwnd >=ssthresh, increase cwnd by 1 for each RTT
User Datagram Protocol (UDP)
Connectionless service: Segments may arrive in disorder, corrupted, and forwarded to the
application as they arrive. Some applications such as Simple Network Management Protocol
(SNMP) are implemented on top of UDP. Reliability is not an important criterion for these
applications. Fewer overheads than TCP.
Header format
Checksum for header only. Very simple interface for programming.
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