Computer Network Module 3

COMPUTER NETWORKS
(BCSE 3306)
Lecture Notes
Module III
Ajit K Nayak
[email protected]
Department of Computer Science Engineering &
Application
Out Line of Module III
Network Layer, Network Layer Protocols
Transport Layer, Congestion control
Quality of service
Text: “Data Communications and Networking” Third Edition,

Behrouz A Forcuzan, Tata Mc Graw-Hill.
Chapter 19 - Chapter 23
Computer Networking / Module III/ AKN / 2
Lecture I
Network Layer
• Host-to-Host Delivery
• Addressing
• Routing
•Network Layer Protocols
• IPV4
• ARP
• ICMP

Network Layer
¾Protocol used is IP for Network Layer

¾Responsibility of this layer to deliver the
datagram to the correct destination host. i.e. host-to-
host delivery Computer Networking / Module III/ AKN / 4
Classful IP Addresses
Each host on a TCP/IP internet is assigned a unique
32-bit unicast Internet address that is used in all
communication with that host.
Each unicast IP address is a pair(netid, hostid), where
netid identifies a network and hostid identifies a host
on that network
The total address space is 232=4,294,967,296. But all
addresses are not usable
It is represented in dotted decimal notation
128.11.3.31
1000000 00001011 00000011 00011111

Type of communication
Unicast: one-to-one communication. i.e. One
source sends to exactly one destination host
Multicast: one-to-a group. i.e. one sources
sends to a predefined group of destination
hosts simultaneously
Broadcast: one-to-all. i.e. one source sends to
all other hosts available in that network.
Broadcast in Internet is not allowed.
Others: anycast, geocast, etc. read yourself!
Classes of IP addresses
Class A 0.0.0.0 – 127.255.255.255
0 netid hostid
Class B 128.0.0.0 – 191.255.255.255

1 0 netid hostid
Class C 192.0.0.0 – 223.255.255.255

1 1 0 netid hostid
Class D 224.0.0.0 – 239.255.255.255

1 1 1 0 multicast address
Class E 240.0.0.0 – 255.255.255.255

1 1 1 1 reserved for future use
IP Addresses
Class A
First octet defines the netid and first bit is fixed
Max. no of network possible: 27-2=126
All zero and all one values can not be used
24 bits are used for hostid
Max no of hosts 224-2=16,777,214 per network can
be connected to a class A network
Class B
First two octet define the netid and two left bits are
fixed : 214-2=16,382 networks and
216-2=65,534 hosts/network
IP Addresses
Class C: First three octet defines netid and three bits
fixed
221-2=2,097,151 networks
28-2=254 hosts/network
Class D: No net and host ids
First four bits are fixed, remaining 24 bits define multicast
addresses?
Class E: No use

Special Addresses
Network Addresses
Addresses having all zero hostids are used to
identify a network and is not assigned to any host
Specific All 0s
123.50.16.90 123.65.7.34 123.90.123.4

...
123.0.0.0
Class A
Network Address
Find Network addresses of the following IP addresses

24.32.3.29
190.234.211.21
200.23.31.6
Special Addresses contd.
Direct Broadcast Addresses
Used by a router to broadcast a message to all
hosts of a network Specific All 1s
It can only be used as a destination address by
specifying hostid as all 1s
221.45.71.20 221.45.71.64 221.45.71.99
...
221.45.71.255
221.45.71.0 R
Class C networkComputer Networking / Module III/ AKN / 12

Limited Broadcast Addresses
Used by a host to send a message to every other host in
that network All 1s All 1s
It can only be used as a destination address by specifying
netid and hostid as all 1s
Router blocks the packet and discards it.
221.45.71.20 221.45.71.64 221.45.71.99
255
.25
5 .25
5 .25 ...
5 Blocked here
221.45.71.0 R
Class C network
This Host Addresses All 0s All 0s
Used by a DHCP client at bootstrap as a source address to
get a valid IP address from the DHCP server

It is specified by all 0s. The destination is a limited broadcast
address
It is always a Class A address regardless of the network
?.?.?.? 221.45.71.64 221.45.71.99
255
.25
5.2
55. ...
0.0 255
.0.0 Bootstrap server
221.45.71.0 B
221.45.71.1
Class C network
Loop Back Addresses
Used by a host to communicate with itself without
a special network interface
This is the address with first byte as 127 and the
packet never goes out of the machine
P1 Host P2
127 Any
127.0.0.1

Private Network Addresses
These IPs should not be used in internet but one
can use for hosts that do not require direct access
to the Internet
These addresses are filtered by Internet routers and
therefore do not have to be globally unique
10.0.0.0 – 10.255.255.255
172.16.0.0 – 172.31.255.255
192.168.0.0 – 192.168.255.255
Automatic Private IP Addressing
Used by windows machine, if there is no DHCP available
169.254.0.0 – 169.254.255.255
Rfcs: 1466, 1918, 1597, 3927 etc.
Masking
To reach at a host we have two level of hierarchy
1. Reach at destination network 2. Reach at host
Masking is a process that extracts the address of
physical network from an IP address
Mask is an IP having netid all ones and hostid all zeros
141.14.2.21 255.255.0.0 141.14.0.0

Mask
A bit wise and operation is performed

10001101 00001110 00000010 00010101
11111111 11111111 00000000 00000000
141 14 0 0
Problems with classful
There are three main problems with “classful” addressing,
1. Lack of Internal Address Flexibility: Big organizations are
assigned large, “monolithic” blocks of addresses that don't
match well the structure of their underlying internal networks.
2. Inefficient Use of Address Space: The existence of only three
block sizes (classes A, B and C) leads to waste of limited IP
address space.
3. Proliferation of Router Table Entries: As the Internet grows,
more and more entries are required for routers to handle the
routing of IP datagrams, which causes performance problems
for routers. Attempting to reduce inefficient address space
allocation leads to even more router table entries.

Subnetting
This technique helps to divide one physical network
into some smaller subnets (i.e.to create hierarchies)
Advantage:
Increasing popularity of LAN may exhaust the netids
When many hosts connected to a single network the
messages are overcrowded due to the broadcast nature of
LANs
The scheme allows multiple physical networks to
share a same prefix (1980s)
A second extension is also available to divide suffix
and prefix at an arbitrary point called classless
addressing and supernetting (1990s)
Subnetting an Example
.2.20 .7.96 .22.90
...
R 141.14.0.0
.2.20
Without subnet
.7.96
141.14.0.0
.2 141.14.2.0
R .7 141.14.0.0
.22
141.14.7.0
With subnet
141.14.0.0
.22.90
141.14.22.0
Subnetting
Rest of the Internet still fills as if one
network. i.e packet destinated at 141.14.2.21
still reach at router R and it is aware of three
subnets.
Last two octets define two things
1. subnetid 2. hostid
Delivery of packets now involve three steps
1. Delivery to the network
2. Delivery to the subnet
3. Delivery to the host
Example 1
Q. Design 8 subnets from 211.77.20.0
Ans. Taking 3 bits for subnet in last byte, remaining 5 bits are
used for hostid

Example 1 contd.
According to classic IP routing rules, it was not possible to use the

subnets with all zero or all one values. i.e. subnet #0 and subnet #7
However, most modern machines have no troubles using uppermost or
lowermost subnets
Example 2
The network address is x.y.z.0, subnet mask is
255.255.255.248 then design the subnets
From mask it is clear that first five bits of last byte is
used as subnetid and last three bits are used as
hostids
i.e. 25=32 subnets and 23-2=6 hosts/subnets
Subnet #0: x.y.z.0, x.y.z.1, . . ., x.y.z.6, x.y.z.7
......................................
First column is used as subnet id, last column is used as
broadcast address. Computer Networking / Module III/ AKN / 24
IP addresses are used not only to
uniquely identify IP addresses
but also to facilitate the routing
of IP datagrams over networks

Problems with IP Addressing
If a host computer moves from one network to another, its IP
address must change (manually)
Because routing uses the network portion of the IP address, the
path taken by packets traveling to a host with multiple IP address
depends on the address used.
Network 1
I2 I2 I3
R A B
I4 I5
If link I3 fails than A cannot send to B Network 2
Addressing Authorities
IANA: Internet Assigned Number Authority upto 1998
ICANN: Internet Corporation for Assigned Names and Numbers
Dynamic Address configuration
Each computer that is connected to Internet must have
following information
Its IP address
Subnet mask
Router/gateway’s IP address
Name server’s IP address
These information are maintained in operating system and
stored in disk
These information may be acquired by assigning static values
or can also be obtained dynamically when needed
DHCP is designed to assign these information dynamically (on
demand)
It is a client/server program, when client sends a request to
server, server selects an IP address from the pool of unused IP
address for a negotiable period of time (lease time)
Dynamic Host Configuration Protocol
TRANSITION STATES
Initializing state
Client broadcasts a
DHCPDISCOVER message
Selecting state
All the DHCP servers
replies with a
DHCPOFFER message,
which contains IP
address, lease time etc.
client chooses on of the
offers. Client now sends a
DHCPREQUEST message
Requesting state
Remains in this state till it gets the DHCPACK, which creates a binding of
physical and logical address
DHCP contd.
Bound state
After using 50% of the time, client requests for renewal by
sending another DHCPREQUEST, or client can cancel the
lease and go back to the initializing state
Renewing state
If it receives the DHCPACK then the timer is reset or client
goes again for rebinding. If not received till 87.5% of lease
time then goes to rebinding state
Rebinding state
It remains in this state till it receives a DHCPNAK or lease
expires, client goes to initializing state for a fresh process or
goes to bound state if DHCPACK is received
Network Address Translation
Home users and small business can be connected to Internet
via an ADSL or cable modem and every body needs one or
more IP addresses
Due to shortage of IP addresses, the demand may be full filled
by using the private network address through Network address
translation method (NAT)
NAT enables a user to have large set of addresses (private)
internally and one or a small set of addresses externally
(global)
Address translation
NAT contd.
Address translation
All out going packets go through
the NAT router, which replaces
destination address in the packet
with global NAT address.
Similarly all incoming packets
also pass through the NAT
router, which replaces the
destination address with
appropriate private address
using Translation table
Private Private External External Transport
Address Port Address Port Protocol
172.18.3.1 1400 25.8.3.2 80 TCP
172.18.3.2 1401 25.8.3.2 80 TCP
... ... ... ... ...
Routing techniques
Usually routing uses an Internet routing table on each
machine that stores information about possible
destinations and how to reach them
Next Hop Routing
20.0.0.5 30.0.0.6
network network network
10.0.0.0 Q 20.0.0.0 R 30.0.0.0
10.0.0.5 20.0.0.6
Dest Next hop 30.0.0.7
S
10.0.0.0 20.0.0.5 40.0.0.7
20.0.0.0 Direct network
30.0.0.0 Direct 40.0.0.0
40.0.0.0 30.0.0.7
Network-Specific Routing
Instead of one entry for each destination host,
we maintain one entry for total network

Host-Specific Routing
Host-specific routes
Although all routing is based on networks and not on specific
hosts, most software allows per-host routes as a special case.
This is helpful for administration purposes like testing,
controlling access and debugging etc.
Net1 Destination Next hop

A
B R
Q R
Net2 Q
Net3 R
Net2 P Net3 Table for host A
B
Default Routing
Default Routes
In this type of routing , it looks in the routing table for the
destination network. If no route appears in the table, the
routing routines send the datagram to a default router
It is useful when the network has a small set of local
addresses and only one connection to the rest of internet
network network
Q
10.0.0.0 20.0.0.0
• Routing table Destination Next hop
S for a host on
20.0.0.0 Q
network 10.0.0.0
Rest of Default S
Internet
Static versus Dynamic Routing Tables
Routing tables may be constructed statically or
dynamically. The success of routing depends on the
consistency of routing table information
Static Routing table
Information entered manually, can be used for small intranet
that does not change very often. It is not a good choice in
Internet where information changes very often
Dynamic Routing table
Updated periodically using the dynamic routing protocols like
RIP, OSPF, or BGP etc.
Dynamic routing is preferred over static routing as the
updation of routing table is done dynamically thus providing
a consistent routing mechanism.
Hierarchical Routing
It is not possible to keep information about each host and or
each network in the routing table of each Internet router
To solve this problem we maintain hierarchical routing.
According to this technique the we maintain partial information
in routers
e.g. if the block assigned to one ISP is a.b.c.d/n and it may
create many subnets of e.f.g.h/m for each of its customers, the
rest of the Internet does not have to be aware of this division.
i.e. all customer of that ISP are defined as a.b.c.d/n to the rest
of Internet
There is only one entry needed for this ISP
The router inside ISP recognizes the sub-blocks and routes the
packets to the destination
To reduce the size of table further the hierarchical routing may
be included. i.e. The routers of ISPs outside Europe will have
only one entry for packets to Europe in their routing tables.
Internet Protocol (IPV4:RFC-791)
Connection less delivery system
Internet service consists of an unreliable, best-effort,
connection less packet delivery system.
Unreliable because delivery is not guaranteed.
i.e.The packet may be lost, duplicated, delayed or delivered
out of order but the service will not detect such conditions,
nor will it inform the sender or receiver.
A sequence of sent from one computer to another may
travel over different paths, or some may be lost while others
are delivered.
It is best-effort delivery because the internet software
makes an earnest attempt to delivery packets
i.e. the internet does not discard packets always.
Unreliability arises only when resources are exhausted or
underlying networks fail.
Internet Protocol (contd.)
The Internet protocol defines unreliable, connection less
delivery mechanism ( IP )
It defines the basic unit of data transfer used throughout the
internet by specifying the exact format of data

It performs routing function, choosing the path over which
the data will be sent

It also includes a set of rules that embody the idea of
unreliable packet delivery.

i.e. It tells how to process the packets, how and when error
message should be generated, and the conditions under

which the packets can be discarded.

Internet Protocol Datagram Format
0 4 8 16 19 24 31
Ver HLen Service Type Total length
Identification Flag Fragment offset
TTL Protocol Header checksum
Source IP
Destination IP
IP Options if any Padding
Data
...

IP Header
Ver: version of IP (4 or 6)
HLen: total length of datagram header (20-60 bytes)
Type of Service: how the datagram should be handled
by the router 0 4 7
Precedence D T R C
Precedence: (3 bits) defines priorities in cases like congestion

TOS bits: low delay, high throughput, high reliability, less cost.
A hint to router as a decision making factor for routing
algorithms. Internet does not guarantee to provide any
particular type of service 0 6 7
IETF redefined the meaning CODEPOINT unused
If last three bits are zero than first three bits define
precedence (backward compatibility)i.e. xxx000
IP Header (contd.)
The 64 code point values maps to an underlying
service definition and is divided into three groups
Pool Codepoint Assigned by
1 xxxxx0 Standards Organization(IETF)
2 xxxx11 Local or Experimental
3 xxxx01 Local or experimental for now
If the standards bodies exhaust all values in pool 1,
they may also choose to assign values in pool 3
Total Length: defines total length of the datagram in
bytes.
i.e. 216-1=65,535 bytes max. including header
IP Header (contd.)
Fragmentation
Each datagram is encapsulated in a datalink frame
before transmission.
It has to travel through different networks and the
frame size differs for different networks and is

defined by MTU of that network
Identification: IP software keeps a global counter
and increments each time a new datagram created.
if the datagram is fragmented then the identification
is copied to each fragment of same datagram
Flags: U D M
3 bit field, D:do not fragment M: more fragment
IP Header (contd.)
D=1: datagram must not be fragmented
D=0: datagram can be fragmented
M=1: It is not the last fragment
M=0: It is the last or only fragment
Fragmentation offset: It shows the relative position of the
fragment, w.r.t. whole datagram
Offset measured in bytes
0/8 = 0
0 1399
1400/8 = 175
0 3999 1400 2799
2800/8 = 350
2800 3999
IP Header (contd.)
Time to Live:It specifies how long in seconds, the
datagram is allowed to remain in the internet system
When a datagram arrives at a router, it records the
time and before sending forward it decrements the
time to live field.
When it becomes zero, the datagram is discarded and
an error message is sent to the source
But to estimate exact time is difficult because routers
do not usually know the transit time for physical
networks.
Thus in practice the time to live acts as a hop limit
rather than an estimate of delay. Each router only
decrements the value by one till it becomes zero.
IP Header (contd.)
Protocol: It defines the higher level protocol
that uses the IP layer service
ICMP- 1, IGMP-2, TCP-6, UDP-17 etc.
Header Checksum: Ensures the integrity of
header values
Divide the packet in to k section of 16 bits each
All sections are added using ones complement
method
The final result is complemented to make checksum
Follow the same method at receiver. If the result is
zero accept else discard the datagram
IP Header Options
IP header is made of two parts: the fixed part and the
variable part. Fixed part is 20 byte long; the variable
part comprises the option which can be a max. of 40
bytes.
These are included primarily for network testing and
debugging
Format Code(8) Length(8) Data (variable length)
Copy Class Number

Code:
It contains copy(1), class(2), and number(5)
Copy = 1: options should be copied to all fragment
Copy = 0: options must be only copied to first fragment
Options field of IP Datagram
Class
00 : used for datagram control, 01: reserved
10: Debugging and management, 11: reserved
Number
Defines the type of options
Length
It defines the total length of the option including the code
field and the length field itself
Data
Contains the data that specific options require

Types of Options
0 : End of option, used if options do not end at end of header
1: no operation, used to align octets
Options 7-byte opt 1

0
Data 8-byte opt
7: Record Route, It is used to record the routers that handles the

datagrams. It can list up to nine router addresses?
The source creates empty fields for the IP addresses in the data field of
the option
Code Length Pointer

First IP Address (empty)
Second IP Address (empty)
Third IP Address (empty)

Types of Options
Whenever a router handles the datagram, it compares
the pointer and length field. If the pointer field is
greater than length field, the list is full.
Else router inserts its IP address at the position
specified by pointer and increments the pointer by
four.
This option requires that two machines must
cooperate. i.e. source must enable record route and
destination must agree to process the resultant list.
9: Strict source route, used by the source to
predetermine a route for the datagram as it travels
through internet
i.e. a source may choose a safer route to the destination
Types of Options
If a datagram specifies a strict source route, all of the
routers defined in the option must be visited in order
by the datagram.
If a datagram reaches at a router not in the list then it
is discarded and error message is sent to the source.
If a datagram reaches at the destination and some
entries were not visited, it will also be discarded and
error message is issued.
i.e. The path between two successive addresses in the
list must consists of a single physical network
It is only useful when the network topology is known
Types of Options
3: Loose source route, It is similar to strict source but allows
multiple network hops between successive address in the list
Both source route options requires routers along the path to
overwrite the list with their local network address.
4: Timestamp, is used to record the time of datagram
processing by the router.
Code Length Pointer OFlow Flags

First IP Address
First Timestamp
...

Types of Options
Length and pointer fields are used to specify the
length of the space reserved for the option and the
location of the next unused slot.
Oflow(4) contains an integer count of routers that
couldnot supply timestamp because the option was
too small
Flag(4), controls the exact format of the option and
tells how routers should supply timestamps.
0: Record timestamps only, omit IP addresses
1: Precede each timestamp by an IP address
3: IP addresses are specified by sender; a router only
records a timestamp if the next IP address in the list
matches the router’s IP address
Routing IP Datagrams
Routing is the process of choosing a path over which
to send packets, and router refers to a computer
making the choice
The goal of IP is to provide a virtual network that
encompasses multiple physical network and offers a
connection less datagram delivery service
Routing is divided into two forms
1. Direct delivery: Transmission of a datagram from one
computer across a single physical network directly to another
2. Indirect delivery: Transmission of datagram to a
destination not attached directly to the senders network, thus
forcing the sender to pass the datagram to a router for delivery
Datagram delivery over a single Network
In this case the final destination of the datagram is a
host connected to the same physical network
• Extraction of network address
takes a few machine instructions
R making the process extremely
efficient
• The sender extracts the network address of destination IP and
compares it to the network portion of its own IP .
• If a match is found then the delivery is direct and it does not
involve routers
• Now the destination IP address is used to find its physical
address for actual datalink layer delivery?
Indirect Delivery
It is more difficult because the sender must identify a
router to which the datagram can be sent
• The datagram goes from router to
router until it reaches the destination
R network
• At the destination network it
R performs direct delivery to reach
at the host
• How can a host know which router to use for a given

destination?
• How can a router know where to send datagrams?
Mapping Internet Address to Physical Address
Delivery of a packet requires two levels of addressing.
Hosts and routers are recognized at the network
level by their logical addresses, which is universal and
implemented in software
But at physical level devices are recognized by their
physical addresses
Therefore, the packet to be sent from A to B should
be mapped to the physical address of B
Address mapping must be performed at each step
along a path from original source to ultimate
destination
i.e 1. Last hop addressing 2. Intermediate addressing
Mapping Internet Address Physical Address
Last hop addressing
Packet’s internet address is mapped to the final
destinations physical address
Intermediate addressing
At any point along the path packet is mapped to
intermediate routers physical address (as destination)
Address resolution problem
The problem of mapping logical to physical address is
called the ‘address resolution problem’.
There are two technologies followed by TCP/IP to resolve
the problem.
1. Resolution through direct mapping
2. Resolution through Dynamic binding
Resolution through Direct Mapping
In proNET token ring network, the administrator chooses
small integers for physical addresses while installing an
interface.
Now to have a efficient address resolution one can find a
function PA = f (IA) to calculate the numbers.
i.e. if f is simple then the mapping will be simple
Another way is to keep a table containing address pairs
(logical, physical) and a hash function may be used to
search that table
Another advantage in this method is, if one interface of a
computer is changed then also the same physical address
can be used for the new interface
Also new computers can be added to the network without
changing the existing assignments.
Resolution through dynamic binding
In Ethernet technology the 48 bit physical address is
assigned when manufactured
Thus the physical address of a computer changes each time
an interface is changed.
Because the physical address is 48 bit long and not assigned
by the user thus it is impossible to devise a function for
mapping as in previous case
To avoid maintaining a mapping table (not possible !) the
designers developed a protocol to bind addresses
dynamically known as ‘Address Resolution Protocol’
ARP provides a mechanism that is both reasonably efficient
and easy to maintain

Resolution through dynamic Binding
Idea
Sender broadcasts a special
packet that asks the A B C D
destination about its
physical address
Destination recognizes the
packet and sends a reply
containing its physical A B C D
address
Now the sender uses
physical address to send
packets directly to
destination A B C D
ARP Packet Format (RFC-826)
Hardware Type Protocol Type
H/W length Protocol length Operation
Sender Hardware Address
Sender Protocol Address
Target Hardware Address
Target Protocol Address
H/W Type: 16 bit field defines type of LAN e.g. Ethernet=1
Protocol Type: 16 bit field defining IP version e.g.
IPV4=0080016
Hlen: 8 bit, length of hardware address e.g. Ethernet = 6
Plen : 16 bit, length of logical address
Operation : 8 bit, request=1, reply 2
Address Resolution Protocol
Encapsulation
ARP packet is encapsulated directly in to a
datalink frame
SFD Dest Add Source Add Type Data CRC
ARP Packet
Refinements
If the target machine is down or too busy to
accept the request? i.e sender may not receive a
reply (1) or it is delayed(2)
Retransmit the request for (1) or it restores the
original outgoing packet till it resolves the address
ARP Implementation
ARP Cache
After receiving an ARP reply, it saves the IP address and
corresponding hardware address in its cache for successive
lookups
But problem occurs if receiver crashes in between and
source gets no information but keep on sending
To resolve above problem a timer is used, when it expires
the information in the cache is erased and normal procedure
starts again
Another refinement possible is, senders IP-Physical address
binding can also be updated in receivers cache before
processing the ARP request

Four cases using ARP

Limitations with IP
A datagram travels from router to router till it reaches
one that can deliver directly to its final destination
If a router cannot route a datagram?
If the router detects an unusual condition that affects
its ability to forward the datagram?
In an connectionless system, each router operates
autonomously, i.e without coordination of sender.
and
IP fails to deliver the datagram if
The destination is temporarily or permanently disconnected
The TTL expires
The intermediate routers become so congested that they
cannot process the incomingComputer
traffic Networking / Module III/ AKN / 66
The Internet Control Message Protocol
To allow routers in an internet to report errors or provide
information about unexpected circumstances, one mechanism
is attached with IP is called
“The Internet Control Message Protocol”, ICMP
ICMP allows routers to send error or control messages to other
router or hosts; It provides communication between the IP
software on one machine and the IP software on another
i.e. The ultimate destination of an ICMP message is not an
application program or user on destination but the IP software
of that machine
ICMP is not restricted only to routers but is allowed to be used
by any arbitrary machine to get some information.
ICMP messages travel across internet in the data portion of IP
datagrams
Error Reporting / Error Correction
When a datagram causes an error, ICMP can only report
the error condition back to the original source of the
datagram.
The source must take some action to correct the error
It cannot be used to inform intermediate routers about
the problem
An Example
If a datagram follows a path R1, R2, . . ., Rk and Rk has the
incorrect information and mistakenly routes the datagram to
Re
Now Re cannot use ICMP to report the error back to Rk but it
can send a report back to the original source
And the original source has no control over the misbehaving
router. In fact it is not possible for the source to know which
router (Rk) causes the problem Computer Networking / Module III/ AKN / 68
ICMP Message
Message Delivery
It requires two levels of encapsulation
Header ICMP Data
Header Datagram Data
Header Frame Data

– Even though ICMP messages are encapsulated and sent
using IP datagrams, it is not considered a higher level
protocol, but a required part of IP
– It is Because, it needs to travel across several physical
networks to reach their final destination
ICMP Message Format
Type (8 bit) Code (8 bit) Checksum (16 bit)
Rest of Header
Data . . .
(Variable size)
Type : identifies the message type
Code : provides further information about the
message type
Checksum : error detection
ICMP messages that report errors always include the
header and first 64 bit data bits of the datagram
causing the problem
ICMP Message Format (contd.)
Type Message Ping: One of the most
0 Echo Reply frequently used
3 Destination unreachable debugging tool that
4 Source Quench invokes ICMP echo
5 Redirect (change route) request and echo reply
8 Echo Request
messages
9 Router Advertisement - Any machine that
10 Router solicitation receives an echo request
11 Time Exceeded for a datagram formulates an echo reply
12 Parameter problem on a datagram and return it to the
original sender
The total table is available in page 133 of D.E. Comer
Echo Request and Reply Message
Type(8 / 0) Code (0) Checksum
Identifier Sequence no
Data . . .
(optional)
Optional Data is a variable length field that
contains data to be returned to sender
Identifier and Sequence number are used by
the sender to match replies to request.
The Type field specifies whether the message
is a request (8) or reply (0)
Reports of Unreachable Destinations
Type-3 Code (0-15) Checksum
Unused - all zeros
Part of the received IP datagram including IP header +
first 8 byte of datagram data
When a router cannot forward or deliver an IP datagram, it sends
a ‘destination unreachable’ message back to the original
source
The code field contains an integer that further describes the
problem
Code Meaning Cause
0: Network unreachable (h/w failure)
1: host unreachable (do)
2: Protocol unreachable (receiving protocol not running)
3: Port unreachable (receiving appl. Prg not running)
4: fragmentation required (D bit set) etc.
Congestion and Datagram flow control
Type-4 Code -0 Checksum
Unused - all zeros
IP header + first 8 byte of datagram data
IP doesn't have a flow control (rate of sending and receiving)
mechanism, which may lead to congestion. i.e
The router eventually exhausts memory and discards additional
datagrams arrived
‘Source quench’ message has been designed to add a kind
flow control to IP.
When a datagram is discarded, it sends a source quench
message to the sender, which helps in
Reporting source that datagram is discarded
Make the source aware of congestion and to slow down

Route change requests
Router Internet Address
Routers are assumed to know correct routes; hosts begin with
minimal routing information and learn new routes from routers
If a host sends a datagram to an incorrect router, then the
router forwards the datagram in correct destination and sends
a ‘redirect message’ to the host.
Now host updates its table accordingly
Code
0: redirection for the network
1 : redirection for the host

Detecting Circular or long routes
Unused
This message is generated in two cases
Code 0: TTL exceeded
¾ If there are errors in one or more routing table a datagram
may travel in a loop. After some time when TTL becomes zero
the datagram is discarded and a ‘Time exceeded’ message is
sent to source
Code 1: Fragment reassembly time exceeded
¾ If all fragments that belong to one datagram don’t arrive at
the destination within a time limit then the fragments are
discarded and a Time exceeded message is sent to the source
Reporting Other Problems
Pointer Unused
If a router or destination discovers an
ambiguous or missing value in any field of the
datagram header then it sends a ‘Parameter
problem’ message back to source
Code 0: Error in header fields
Pointer field points to the byte with problem
Code 1: Required part of option is missing
Pointer field not used in this case
Clock Synchronization and Transit Time Estimation
Type(13-14) Code -0 Checksum
Identifier Sequence number
Source: Originate time stamp
Destination: Receive time stamp
Destination: Transmit time stamp (departure)
‘Time Stamp message’ is used by two machines to determine the round
trip time needed for an IP datagram to travel between them
Each time the fields hold a no representing time measured in
milliseconds from midnight in GMT
Calculation:
Sending time = receive TS - Originate TS
Receiving time = datagram return time - Trnsmit TS
Round trip time = sending time + receiving time
Obtaining a subnet mask
Type(17-18) Code -0 Checksum
Address Mask
‘Address mask request/reply’ message are used
by a host to obtain its mask from a router
Router Discovery
Type(9) Code -0 Checksum
Nun addr Addr size Life time
Router Address 1
Preference level 1
Router Address 2
Preference level 2...
Router Solicitation/Advertisement
Type(10) Code -0 Checksum
ICMP supports a router discovery scheme that allows hosts to

discover router address.
A host can broadcast a ‘router solicitation’ message. The
routers that receive the message broad cast their routing
information using ‘router advertisement’ message
ICMP router discovery scheme helps in two ways
1. Instead of providing a statically configured router address
via a boot strap protocol, the scheme allows a host to obtain
information from router itself
2. The mechanism uses a soft state technique with timers to
prevent hosts from retaining a route after a router crashes
Routers advertise their information periodically, and a host discards a
route if the timer for a route expires (30min, 10min)
Lecture II
Network Layer Protocols
• IPV6
• ICMPR6
• Unicast Routing protocols
• RIP
• OSPF

IPv6: Need for an alternative
IPv4 has two level address structure (?) and
categorized into 5 classes. The use of address space
is inefficient
The internet must accommodate realtime audio and
video transmission, which requires min delay and
reservation of resources
The Internet must accommodate encryption and
authentication of data for some application
Not only the computers but various devices
including house hold devices, hand held devices,
telephones etc. needs IP address
Characteristics of IPv6
Larger Address Space: 128 bit long
Huge increase in address space
Better header format
options are separated from base header
New options
To add new functionalities
Allowance for extension
To support new technologies
Support for resource allocation
To support traffic such as real-time audio and video
Support for more security
Encryption and authentication mechanism
RFCs
1365, 1550, 1678, . . .

IPv6 address
• 128 bits are divided into eight sections of hexadecimal nos, each 2
byte long sections separated by colons
• The address may be abbreviated, i.e the leading zeros can be
omitted (not trailing zeros)
• consecutive sections consisting of zeros

can be replaced with double semicolons
• if there are two runs of zero section than
only one of them can be abbreviated
Unicast Addresses
Defines two types of unicast addresses
Geographically based unicast address (left for future
definition)
Provider based unicast address (discussed below)
Type identifier: 3 bit field defines the address as a provider-

based address
Unicast Addresses contd.
Registry identifier: 5bit field indicates the agency that has
registered the address.currently three registry has been
defined.
INTERNIC: center for North America
RIPNIC: center for European registration
APNIC: for Asian and Pacific countries
Provider indentifier: variable-length field identifies the provider
for Internet access (like ISP). A 16 bit length is recommended
for this field
Subscriber identifier: a 24 bit is assigned to an organization
subscribing to the Internet via provider
Subnet identifier: a 32 bit is assigned to define a subnet under
the territory of a subscriber
Node identifier: a 48 bit is assigned for the identity of the node
connected to subnet
Multicast addresses
First 8 bits all 1s
Flag: 4bit field that defines
the group address as either
permanent or transient
Scope: 4 bit field defines
scope of the group address
Group ID: 112 bits identifies
group
Anycast addresses
A packet destinated for anycast address is delivered to only
one member of the anycast group. i.e. member having
shortest route
No block is assigned to for this anycast address
Reserved addresses
Start with eight zeros
Unspecified address is used when a host does not know its own
address
Loopback address is used by a host to test itself
Compatible address is used during the transition from IPv4 to
IPv6. i.e. when passing from IPv6 to IPv6 via IPv4 network
Mapped address is also used during transition when sending
from Ipv6 to IPv4 computer
Local addresses
Used when an organization wants to use IPv6
without being connected to Internet
Nobody outside the organization can send a message

to the nodes using these addresses
A link local address is used in an isolated subnet
A site local address is used in an isolated site with
several subnets
Format of an IPv6 datagram
• Each packet is
composed of a
mandatory base
header (40 bytes)
followed by a
payload.
• Payload consists of
two parts (65535
bytes)
• Optional
extension header
• Data from an
upper layer

Base Header
Version(4): version of IP
Priority(4): priority of the packet w.r.t. congestion
Flow level(3byte): special handling for a particular flow
of data
Payload length(2 byte): total length of datagram
excluding base header
Next header(8): either one of the optional extension
headers used by IP or the header for an upper layer
protocol like UDP, TCP
Hop Limit(8): same as TTL
Source Address(16byte): IP of source
Source Address(16byte): IP of destination
Comparison between IPv4 and IPv6 packet headers

Extension header
The base header can be followed by six extension headers
Hop-by-hop Option
Is used when the source needs to pass information to all
routers visited by the datagram. Three options are defined
Pad1: 1 byte, designed for alignment purposes
PadN: used when 2 or more bytes needed for alignment
Jumbo payload: is used to define a payload longer than 65535
bytes
Fragmentation
Only original source can fragment after using a path MTU
discovery to get the smallest MTU supported by any network
on the path
If it will not use the technique then it must fragment a
datagram to a size <= 576 bytes
Extension header contd.
Authentication
It validates sender, and ensures integrity of data
Encrypted Security Payload
It provides confidentiality and guards against
eavesdropping
Source Routing
Uses the concept of strict/loose source routing
Destination Option
Is used when the source needs to pass information to
the destination only. Intermediate routers are not
permitted access too this information
Comparison between IPv4 options and IPv6 extension headers

Transition from IPv4 to IPv6
Because of huge systems using IPV4 that’s
why three strategies were proposed for
smooth transition
Dual stack
A station should run both IPv4 and
IPv6 simultaneously until all the
Internet uses IPv6
If DNS returns IPV4 address then
source sends IPV4 packet else IPV6
packet
Tunneling
When two computers using IPV6 want
to communicate with each other and
the the packet has to pass through a
region that uses IPV4
Therefore IPV6 packet is encapsulated
in an IPV4 datagram when it enters
that IPv4 region Computer Networking / Module III/ AKN / 96
Transition from IPv4 to IPv6
Header Translation
It is necessary when the majority of the Internet
has moved to IPv6
i.e. If sender uses IPv6 but receiver uses IPv4
Header must be completely translated
It uses mapped address of IPv6

ICMPv6
Comparison of error-reporting messages in ICMPv4 and ICMPv6
Comparison of query messages in ICMPv4 and ICMPv6

Unicast Routing Protocols
A routing protocol allows routers share their knowledge
(routing information) about the network with other routers.
They maintain a table to keep routing information. This table
gets updated periodically after receiving information from
neighbouring routers
Routers use routing table to decide about the best route based
on a cost metric
Cost metric
Hop count: cost of passing through any network is same. i.e. passing
through one network costs 1 hop
Max throughput: throughput is more in passing through an fiber than in
radio link
Min delay: delay is less in fiber than satellite link
Reliability: some networks may be more reliable than others, it is
decided based on a policy.
Various routing protocols available are RIP, OSPF etc.
Routing Information Protocol
It is based on Distance Vector routing, which uses Bellman-
Ford algorithm for calculating the routing table
Distance Vector Routing
In this scheme, each router periodically (30 s) shares (broadcasts) its
own routing information with its neighbours
Every router keeps a routing table that has three columns in its simplest
form for each entry about a network
• A, B,C, D are
(routers)
• To: destination
network
• Cost: hop count
• Next: next hop
RIP Updating
Receive: a response RIP message
1. Add one hop to the hop count for each advertised
destination.
2. Repeat the following steps for each advertised destination:
1. If (destination not in the routing table)
1. Add the advertised information to the table.
2. Else
1. If (next-hop field is the same)
1. Replace entry in the table with the advertised one.
2. Else
1. If (advertised hop count smaller than one in the table)
1. Replace entry in the routing table.
3. Return.

Example of updating a routing table

Initial and Final routing tables in an example network

Problems with RIP: Count-to-infinity
Count to infinite A B C D E F
Suppose there is a Initially 1, - 2, B 3, C 4, D 5, E
network as shown
Each router keeps the After 1 exchange 3, C 2, B 3, C 4, D 5, E
information about A After 2 exchanges 3, C 4, B 3, C 4, D 5, E
initially as shown
Now A goes down or link After 3 exchanges 5, C 4, B 5, C 4, D 5, E
between A and B Brakes
After 4 exchanges 5, C 6, B 5, C 6, D 5, E
At the first packet
exchange B will not After … exchanges . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
receive any message ∞ ∞ ∞ ∞ ∞
from A
•The number of exchanges required depends
But C tells B that it has a
path to A of length 2 on the numerical value used for infinity.
B now updates its own
information about A •In RIP the value is kept 16, that’s why it
according updation algo can’t be used in large systems
and make it 3
Open shortest path First (OSPF)
It is based on link state routing that uses dijkstra’s
algorithm
Link state routing
In this scheme, each router shares the knowledge about its
own neighbours to all other routers using flooding
Each router maintains a database about its neighbours and
sends it when there is a change or after a large period.
The idea is that all routers should have a complete topology
of the network. From this topology the router can calculate
the shortest path between itself and the destination network
using dijkstra’s graph algorithm
The topology is represented as a graph, where vertices are
networks or routers and edges are links.
A cost is associated with each link
Link state Routing
Learning about neighbours
When router is booted, it sends a hello packet on each
point-to-point line
The router at the other end sends back a reply
Measuring Link cost
One echo packet is sent and its time is recorded, other side
sends the packet back immediately and the time of receiving
is recorded again
The test is conducted several times and the average RTT is
calculated for better result
Building the Link state packets
Identity of sender, sequence #, age, a list of neighbours
with their link costs
Link state Knowledge
Whole topology can be compiled from the partial
knowledge of each node

Formation of shortest path tree
The dijkstra’s algorithm creates a single source shortest path tree
given a graph(topology), each node is assigned a cumulative cost
from root to that node (called weight or total cost)

Lecture III
Transport Layer
• User Datagram Protocol

• Transmission Control protocol
• Congestion Control and Quality of
services

Transport Layer
¾Protocols used for Transport Layer are UDP or TCP

¾The responsibility of transport layer is to deliver the
message to the receiving process/Application. i.e.
process to process delivery
Review
Internet layer provides a host-to-host packet delivery
The next problem is turn this service to process-to-
process delivery
The Transport layer supports communication between
the end application programs, thus called end-to-end
protocol
The underlying networks upon which the transport
protocol operates has certain limitations like, it may
Drop messages
Reorder Messages
Deliver duplicate copies of messages
Limit messages to some finite size
Delivery messages after a long delay
Review
The operating system supports multiprogramming
But specifying that a particular process on a particular
machine is the ultimate destination for a datagram is
misleading, because
Processes are created and destroyed dynamically(pid),
senders seldom know enough to identify a process on
another machine
Processes may be replaced without informing to the senders
We need to identify destinations from the functions they
implement without knowing the process
Instead of thinking a process as the ultimate
destination, we will imagine that the machine contains
a set of abstract points called protocol ports (integer
nos.) Computer Networking / Module III/ AKN / 112
Review
Operating system provides two types of access to
ports
1. Synchronous access
computation stops during a port access operation.
i.e. if a process attempts to extract data from a port, then
the operating system temporarily blocks the process till data
is passed to the process and then restarts it
2. Asynchronous access
Ports are buffered, so that data arrives before a process is
ready to access will not be lost
To achieve buffering the protocol software places the
packets that arrive for a particular protocol port in a (finite)
queue
Each message must carry the destination port on
source Computer Networking / Module III/ AKN / 113
Types of data deliveries

Port Addressing
At transport layer, port number is used to deliver a message to
the correct process out of several processes running on
destination host
Port numbers are 16 bit integers between 0-65535.
The client program defines itself with a port number, chosen
randomly by transport layer called ephemeral port numbers
The server program uses well known port number.
i.e. client gets a new port number each time it runs, but the
port number for server is fixed
IANA defines some ranges
Well-know ports: 0-1023 are assigned and controlled by IANA for some
well-know server processes
Registered ports: 1024-49151 are not assigned or controlled by IANA,
but can be used by processes
Dynamic ports: 49151-65535 are neither controlled nor registered, called
ephemeral ports Computer Networking / Module III/ AKN / 115
Other features
Socket Address
The IP address and port number pair defines the socket
address
The client and server’s socket addresses define client and
server processes uniquely
A pair of socket address (client and server’s) uniquely defines a
connection.
Multiplexing and demultiplexing
At the sender side, there may be several processes need to
send packets, but there is one transport layer protocol.
Therefore the protocol accepts messages from different
processes differentiated by their port numbers and interleaves
them
At the receiver side, the transport layer receives interleaved
packets from network layer and passes to appropriate
application after processing Computer Networking / Module III/ AKN / 116
Other features contd.
Connection-less vs connection-oriented service
In a connection less service, packets are sent from
one party to another, without establishing the
connection
In case of connection oriented, a connection is
established, data transferred, then connection is
released
Reliable vs unreliable
Reliability is achieved by providing error and flow
control at transport layer (data transmission)
It becomes a slower and more complex service
Where as unreliable services are faster and simple to
implement (real-time application)
The User Datagram Protocol (UDP)
It is the simplest possible transport protocol that
extends the host-to-host delivery into a process-to-
process communication service.
It only adds a level of demultiplexing, s.t. multiple
application process on each host are allowed to share
the network.
Aside from this requirement, UDP adds no other
functionality to the best effort service.
UDP provides an unreliable connection less delivery
service.
It uses IP to carry messages, but adds the ability to
distinguish among multiple destinations within a given
host computer. Computer Networking / Module III/ AKN / 118
The UDP message format
UDP Source Port UDP Destination Port
UDP message length UDP Checksum
Data . . .
Port nos may vary from 0-65535, and source port is

optional. These are used to demultiplex datagrams
The Length field contains a count of datagram in octets.
Minimum length is 8
Checksum is optional and zero is kept if not computed
The UDP checksum provides the only way to guarantee
that data has arrived intact and should be used
Checksum Calculation
UDP uses the same checksum algorithm as IP
But UDP covers more information than is present in
UDP datagram
It prepends a pseudo-header to the UDP datagram
Appends an octets of zeros to pad the datagram to an exact
multiple of 16 bits
And computes checksum over entire object
UDP pseudo-Header
Source IP
Destination IP
Zero Protocol UDP Length
Checksum Calculation (contd.)
Checksum calculation at the Sender end.
Add pseudo-header to the user datagram
Fill the checksum field with zeros
Divide the total bits in to 16 bit words
If total bytes are not even, add one byte of all zeros
Add all 16-bit sections using one’s complement arithmetic
Complement the result and insert the result in checksum
field
Drop the pseudo header and any padding used
Deliver the datagram
Checksum calculation at the Receiver end.
Perform the operation same as above
If complement is zero drop pseudo-header and padding and
accept the datagram. Otherwise discard the datagram
Checksum Calculation (contd.)
153.18.8.105
171.2.14.10
Zero 17 15
1027 13
15 0
U D P T
E S T padding
Assignment
Calculate the checksum of the user datagram at sender side
and also test it for the receiver side
Checksum Calculation an example

Problem with Checksum Calculation
Pseudo-header contains source and destination IP
addresses
i.e. IP addresses must be known at UDP layer
Destination IP address is supplied by the user.
But what about source IP, which is yet to be computed in
IP layer?
Solution 1: UDP software asks the IP layer to compute addresses
Solution 2: UDP software computes addresses and after checksum
calculation sends it to IP layer.
IP layer need to fill remaining IP header fields
But any of the solution violates the abstraction of layers
i.e. It is clearly a compromise of pure separation needed
for practical reasons
UDP Operation
Connection less service
Each datagram sent by UDP is an independent datagram.
Data grams are not numbered, also there is no connection
establishment thus different datagrams may follow different
path
It cannot send a stream of data, i.e. each request must be
small enough to fit into one user datagram
Flow and error control
No flow control hence no window mechanism. Receiver may
overflow
No error control hence sender does not know if a message is
lost or duplicated
Multiplexing and Demultiplexing
In a host running a TCP/IP software, there is only one UDP but
possibly several processes, that need to use services of UDP
Port1 Port2 Port3 Port1 Port2 Port3
UDP Multiplexer UDP DeMultiplexer
IP IP
• At sending side UDP accepts messages from different processes,
differentiated by their port nos.Then it is passed to IP layer
• At receiving side UDP receives datagrams from IP. After error
checking drops the header and delivers to the appropriate processes
Well known ports used for UDP

Use of UDP
It is suitable for process that requires simple
and fast request-response communication like
DNS
Suitable for process with internal flow and error
control mechanism like tftp
Suitable for multicasting
Used for management process such as SNMP
Used for route update protocols like RIP

Reliable Stream Transport Service
Stream Orientation
Data is converted into stream of bits, divided into
octets at source machines
The stream delivery service on the destination
machine passes to the receiver exactly the same
sequence of octets that the sender has passed.
Virtual Circuit Connection
Before data transfer can start, both the
applications interact with their respective OS for a
connection
i.e. one application places a call, which must be accepted
by the other
Properties of Reliable Delivery Service
During transfer, protocol software on the two machines continue
to communicate to verify that data is received correctly otherwise
report the failure to appropriate S/W for necessary action
Therefore, Application programs view the connection as a
dedicated H/W circuit.
The reliability is an illusion provided by the stream delivery service
called virtual circuit
Buffered Transfer
The protocol software is free to divide/combine the stream into
packets independent of pieces the application program transfers.
At the sending side, a PUSH mechanism forces protocol S/W to
transfer all the data that has been generated without waiting to fill
a buffer.
At the other end PUSH causes it to make the data available to
application without delay Computer Networking / Module III/ AKN / 130
Properties of Reliable Delivery Service
Unstructured Stream
TCP/IP stream service doesn’t honour structured
data stream
i.e. There is no way for a payroll application to have
the stream service mark the boundaries between
employee records
Full Duplex Connection
Connections provided by TCP/IP stream service
allow concurrent transfer on both directions
The advantage is control information for one stream
can be send back to the source in datagrams
carrying data in the opposite direction
Transmission Control Protocol
Reliability
+ve acknowledgement with retransmission
Sender Receiver ¾The sender keeps a record of each
Pkt Recv Pkt packet it sends and waits for an ack
Send Ack before sending the next pkt
Recv Ack ¾Sender also starts a timer and
Send Pkt retransmits a packet if the timer
expires before receiving the ack
• Disadvantages
• Duplication of data / Ack due to premature retransmission
• To avoid confusion caused by delayed or duplicated Ack, seq. no. is
sent back with Ack
• Wasting of substantial amount of N/W bandwidth
END-to-END vs Point-to-Point
1. TCP needs an explicit connection establishment s.t.
two parties establish some shared state to enable the
sliding window algorithm to begin
2. Variations in RTT are possible due to various
reasons.(?) Therefore timeout mechanism that triggers
retransmissions must be adaptive.
3. How late a packet can arrive at the destination? IP
throws packets away after their TTL expires, TCP
assumes that each packet has a max. segment life
time(MSL).
TCP has to be prepared for very old packets to suddenly show
up at the receiver, potentially confusing the sliding window
algorithm.
END-to-END issues
4. In case of point-to-point link
delay × bandwidth ≈ window size ≈ buffer space
The amount of resources dedicated to any one TCP
connection highly variable, especially considering that
any one host can potentially support hundreds of TCP
connections at the same time
i.e TCP must include a mechanism that each side ‘learn’ what
resources the other side is able to apply to the connection
5. TCP connection has no idea what links will be
traversed to reach at the destination.
The sending machine might be connected directly to a
relatively fast Ethernet and somewhere in the middle a slower
link has to traversed, which leads to ‘congestion’
TCP Segment
Appl process Appl process ¾TCP is a byte oriented protocol.
Read bytes
i.e. It describes the service
Write bytes
provided to appl. process.
TCP Send TCP Recv ¾The pkts exchanged between
buffer buffer TCP peers are called segments
segment segment
TCP has three mechanisms to trigger the transmission of a
segment
1. TCP maintains a variable, maximum segment Size (MSS), and it sends a
segment as soon as it has collected MSS bytes from sending process
2. Sending process invokes push operation to effectively flush the buffer of
unsent bytes
3. A timer that periodically fires; the resulting segment contains as many
bytes as are currently in buffer
TCP Segment Header Format
0 4 10 16 19 24 31
Src Port Dst Port
Sequence Number
Acknowledgement
HLen unused Flags Advertised window
Checksum Urgent pointer
Options (variable length) Padding
Data
...

TCP Header Format Explanation
SrcPort and DstPort, identify the source and destination
application programs respectively
A TCP connection is identified by a 4-tuple
{SrcPort, SrcIPAddr, DstPort, DstIPAddr}
Because TCP is a byte oriented protocol, each byte of
data has a sequence number
SeqNum field contains the sequence number for the first
octet of data carried in that segment
Ack field defines the octet number that is expected next
AdvertisedWindow contains the buffer space available at
receiver seqNum
Sender Receiver
Ack+advWin
TCP Header Format Explanation
Flags: 6 bits, when set it is understood as follows
5. SYN: Synchronize seq. nos during connection
6. FIN: Terminate the connection
4. RESET: reset the connection
3. PUSH: request for push
1. URG: urgent pointer is valid
2. ACK:
Urgent pointer specifies the position, where the urgent
data ends.
Options: TCP header can have 40 bytes of optional
information
TCP Header Options
Max Seg Size(MSS): 4bytes determined at the
time of connection establishment
Window Scale factor:3bytes
Used to increase the window size
New window size=window size × 2scaleFactor
Largest value possible for scale factor is 16
i.e. 216 × 216 = 232 max size of seq. number
Time Stamp: 10 bytes
Used to calculate round trip time

Connection Establishment
Client Server
SYN, seq
Num=x
Nu m= y The algorithm used is called
CK s e q
SYN+A three-way-handshaking
ACK=
y+1
¾ The client sends a segment to the server stating (flags=SYN,

seqNum=x )
¾ Then server responds with a single segment that both
acknowledges (Flags=ACK, Ack=x+1) and states it own
beginning seqNum (Flags=SYN, seqNum=y)
¾ Finally client responds with a third segment that acknowledges
the server’s sequence number (flags=ACK, Ack= y+1)
Connection Termination, four-way-handshaking
Client Server
FIN, seqN ¾ The client sends a segment to
um=x the server stating (flags=FIN,
x+ 1 seqNum=x )
AC K =
q Nu m=y ¾ Then server responds with a
FIN, se single segment that acknowledges
ACK=
y+1 (Flags=ACK, Ack=x+1)
¾ now the connection is in half
close mode. i.e. server can send
data (remaining) but client can’t
¾ Finally server sends a segment to the client stating (flags=FIN,

seqNum=y )
¾ The client responds with a segment that acknowledges the
server’s sequence number (flags=ACK, Ack= y+1)
Connection Resetting
TCP may request for resetting a connection. i.e.
the current connection is destroyed.
Resetting is done in one of the following three
cases
The TCP of one side has requested a connection to
a non-existent port. TCP of other side sends a
segment with RST bit set
One TCP may want to abort the connection due to
an abnormal situation
The TCP on one side may discover that the TCP on
the other side has been idle for a long time

TCP State Transition
To keep track of all the different events during
connection establishment to connection termination
The TCP of both sides are implemented as a finite
state machine and is represented in a state transition
diagram
Notations
The states are shown using ovals
Transition from one state to another is shown using directed
lines
Each line is contains two strings separated by slash. First
string is input to TCP and second is output
Dotted lines represent server and solid lines represent client
State transition diagram
Client Diagram
Starts in CLOSED state
When receives an Active
open request from client
application, it sends a SYN
segment to server and
goes to SYN-SENT state
Client TCP receives a
SYN+ACK segment from
server TCP. It sends an
ACK to server TCP and
goes to ESTABLISHED
state
This is the data transfer
state. Client remains in
this state till data
transmission continues Computer Networking / Module III/ AKN / 144
State transition diagram contd.
Client Diagram
Client TCP receives a close request from its application program.
It sends a FIN segment to the other TCP and goes to FIN-
WAIT-1 state
When the ACK is received from server TCP, it goes to FIN-
WAIT-2 state. The connection is closed in one direction
Client receives a FIN segment from server TCP and sends an
ACK and goes to TIME-WAIT state
When client TCP is in this state it starts a timer and waits till the
timer goes off.
The value of this timer is set to double the MSL
The client TCP remains in this state to let all duplicate packets,
if any arrive to be discarded. After the time-out the client goes
to CLOSED state again
State transition diagram contd.
Server Diagram
Server TCP starts with CLOSED state
It receives a passive open request from the server application
and goes to LISTEN state
IT now receives a SYN segment from the client TCP and sends a
SYN+ACK segment to client TCP and goes to SYN-Rcvd state
It then receives ACK from client TCP and goes to ESTABLISHED
state. Data transfer occurs between client and server
applications
After data transmission it receives a FIN segment from client
TCP, it now sends an ACK and goes to CLOSE-WAIT state
Server TCP receives a close request from server application
program and sends a FIN segment to client TCP and goes to
LAST-ACK state
When it receives the last ACK from client it goes to CLOSED
state again Computer Networking / Module III/ AKN / 146
TCP’s Sliding Window
1. It guarantees the reliable delivery of data,
2. It ensures data is delivered in order and
3. It enforces flow control between sender and receiver
The algorithm places a small, fixed size virtual window
on the stream sequence and transmits all octets that lie
inside the window without receiving an Ack.
Three pointers are maintained into the send buffer
Sending Application Receiving Application

TCP TCP
LastByteWritten LastByteRead
LastByteAckd LastByteSent NextByteExpected LastByteRecvd

Direction of transmission Computer Networking / Module III/ AKN / 147
Reliable and Ordered Delivery
TCP on sending side maintains a send buffer, this
buffer is used to store data that has been sent but not
yet acknowledged, as well as data that has been written
by the sending application, but not transmitted
On other side, TCP maintains a receive buffer that
holds data that arrives out of order, as well as the data
that is in correct order but that application process has
not yet read it
The relations among send buffer pointers can be as
follows
LastByteAckd ≤ LastByteSent and
LastByteSent ≤ LastByteWritten
bytes to the left of LastByteAcked and bytes to the right
of LastByteWritten need notComputer
be saved
Networking / Module III/ AKN / 148
Reliable and Ordered Delivery
Similarly at the receive buffer
LastByteRead < NextByteExpected is true As a byte
cannot be read by the application until it is received
NextByteExpected ≤ LastByteRecvd + 1
i.e. if data has arrived in order, NextByteExpected points to
the byte after LastByteRecvd
¾ if data has arrived out of order, NextByteExpected points to
the start of the first gap in data

¾ The bytes to the left of LastByteRead need not be buffered
because they have already been read by the local process

¾ bytes to the right of LastByteRecvd need not be buffered
because they have not yet arrived.

TCP Flow Control
Both buffers are of finite size defined by
MaxSendBuffer and MaxRcvBuffer.
Receiver sends a window advertisement that it can
buffer. At receiving side, it maintains as
LastByteRecvd – LastByteRead ≤ MaxRcvBuffer to avoid
overflowing its buffer, it therefore advertises a window size
of
AdvertisedWindow = MaxrecvBuffer- ((NextByteExpected-1) -
LastByteRead) i.e. the free space remaining in receive buffer
NextByteExpected-1 is same as LastByteExpected in case of
inorder receive, it will be different if out of order receive
If the receiving process is reading data just as fast as
it arrives, then the advertised window stays open.
TCP Flow Control
If the receiving process falls behind, then
advertise window shrinks and eventually goes to
zero
On the other hand sender end TCP ensures that
LastByteSent – LastByteAcked ≤ AdvertisedWindow
i.e. it calculates How much data it can send as
EffectiveWindow = AdvertisedWindow –
(LastByteSent – LastByteAcked) i.e. how much extra
bytes it can send
Also sending side should ensure that the local
process doesn’t overflow the send buffer, that is
LastByteAcked ≤ MaxSendBuffer
tries
itten to + y(LastByteWritten
write y bytes and
– LastByteAcked) > Computer
MaxSendBuffer – LastByteAcked)
Networking then TCP
/ Module blocks
III/ AKN +
/ 151
TCP Flow Control
How does the sending side know that the advertised
window is no longer zero?
i.e. once the receiver side has advertised a window size of 0,
the sender is not permitted to send any more data, which
mince it has no way to discover that the advertised window
is no longer zero at some time in the future.
Solution: the sending side persists in sending a
segment with one byte of data every so often. The
data may not be accepted but eventually it gets a
response whenever send buffer becomes free.
The size of MSS is set to MTU of the directly
connected network minus the size of TCP and IP
header s.t. can be sent without fragmentation
Adaptive Retransmission
TCP retransmits each segment if an Ack is not received
in a certain period of time(RTT)
But choosing an appropriate timeout value is very
difficult and TCP uses adaptive retransmission
mechanism
Original Algorithm:
TCP sends a data segment, records the time. When Ack for
that segment arrives, it reads the time again. Difference
between two times gives a SampleRTT.
TCP then computes a weighted average between the previous
estimate and this new sample as
EstimatedRTT = α × EstimatedRTT + (1 - α) × SampleRTT
α between 0.8 and 0.9 used to smooth the EstimatedRTT
Adaptive Retransmission
Then TimeOut = 2 × EstimatedRTT
Problems
Ack does not acknowledges a transmission but receipt of data. i.e. it is
difficult to associate an ACK with an transmission or retransmission
Associating the ACK with original transmission may be an over estimate
and associating with retransmission may be an under estimate as shown
in two figures
Solution?
Receiver Sender Receiver
Sender
Original T
Original Tran r ansmissio
smission n
SampleRTT
Retransmiss SampleRTT
ion AC K
Retran
A CK smissio
n
Original transmission Retransmission

Congestion Control
Congestion is a situation which may occurs when the load on
the network is greater than the capacity of the network
i.e. The number of packets sent to the router is much more then
the Number of packets the router can handle.
Router has so many packets queued that it runs out of buffer
space and has to start dropping packets, which is a worst
condition
Therefore to control the congestion we try to avoid heavy data
traffic that may cause congestion
If the rate of packet arrival rate is higher than
processing rate then input queues becomes longer
If the rate of packet departure rate is higher than

processing rate then output queues becomes longer
Traffic descriptors
Average data rate = amount of
data/total time
Peak datarate= max datarate of
the traffic
Max. burst size= max length of
time the traffic is generated at
the peak rate
Effective bandwidth= is a
function of average datarate,
peak data rate, and max. burst
size

Traffic Profiles
Constant-bit-rate traffic:
Datarate is constant throughout
Variable bit rate:

The rate of data flow changes in time
Bursty:
The datarate changes suddenly in a
very short period of time. This type of traffic
creates congestion in a network.

Network performance
Delay vs Load
When load is much less than the capacity of
the network, the delay is at a minimum
Delay composed of propagation delay and
processing delay, which is negligible!
When load reaches the network capacity, the
delay increases sharply because waiting time
is added to the delay
Throughput vs Load
Throughput is the number of packets passing
through the network in unit time
when the load is below capacity, the
throughput increases proportionally with load
When load reaches the network capacity,
throughput declines sharply due to discarding
of packets followed by retransmissions further
makes things worse
Congestion Control
Two categories of mechanisms for congestion control
Open Loop: congestion prevention
Closed Loop: congestion removal
Open Loop: preventing congestion
Retransmission policy
The retransmission policy and retransmission timers must be designed to
optimize the efficiency and to prevent congestion
Window Policy
The selective repeat is better than Go-Back-N policy for congestion
control?
ACK Policy
If ACK is not received, sender slows down, help prevent congestion
Discarding Policy
Selective discarding of less sensitive packets when likelihood of
congestion increases
Admission Policy
Before admitting for a flow it checks the resources
Congestion Control: closed Loop
Closed Loop: removal of congestion, if occurs
Back Pressure
Router informs previous routers to slow down (recursive)
Choke Point
Router informs source to slow down by sending a special packet
Implicit Signaling
Source predicts about congestion and slows down (like delay in getting
ACK)
Explicit Signaling
Router sends an explicit signal by setting a bit in the packet
Backward signaling:The bit can be set in a packet moving in the opposite
direction. This bit warns the sender to slow down
Forward signaling:The bit can be set in a packet moving in the direction
of congestion. This bit warns the destination to slow down. Receiver
slows down sending ACK
Congestion Control TCP
When congestion occurs in a router and some packets
might be dropped, then sender retransmits those
packets. This may create more congestion and more
dropping of packets.
The condition become so worse that the system can
pass no more data. This situation is called congestion
collapse
i.e. If the cause of the lost segment is congestion,
retransmission of the segment does not remove
the cause—it aggravates it.
To avoid this situation, TCP assumes that the cause of
a lost segment is due to congestion in the network and
takes necessary action to remove congestion.
Congestion Control TCP contd.
The window size is decided not only by the
receiver’s advertisement but also by congestion
in the network
Actual Window = Min(receiver’s window,
Congestion window)
Congestion avoidance
To avoid congestion we have two strategies
Slow start and additive increase till there is no congestion
Multiplicative Decrease, if congestion occurs

Slow start
At the beginning of a connection TCP sets the congestion
window size = 1MSS
For each segment ACK it receives the congestion window size
is increased by 1 MSS till it reaches a threshold value = ½ of
allowable window size i.e.
ACK for 1 seg –> congestion window size = 2 MSS
ACK for 2 segs -> congestion window size = 4 MSS
ACK for 4 segs -> congestion window size = 8 MSS
. . . -> congestion window size = ½ advt. Window
Additive Increase
After the size reaches the threshold, it increases the size by
one for each received ACK.
i.e. ACK may be received for several segments but increase is
only by 1 MSS
This strategy continues till it receives ACK before time-out or
congestion window size = advt. Window size.
Multiplicative Decrease
The only way to guess that a congestion has occurred is
through a lost segment. i.e. if the sender does not receive
ACK before time-out
If congestion occurs than threshold value is set to ½ of
congestion window and congestion window is set to 1MSS
again

Congestion control in frame relay
Frame relay is designed for high throughput
and low delay but congestion decreases
throughput and increases delay
Frame relay does not have flow control, but
allows user to transmit bursty data that can
cause congestion
For congestion avoidance, Frame relay protocol
uses 2 bits the frame to warn the source and
destination about the congestion.
Backward Explicit congestion Notification (BECN) bit
Forward Explicit congestion Notification (FECN) bit
BECN bit
It warns the sender about congestion in the
network using two methods
Method 1: the switch uses response frames from
the receiver
Method 2: the switch can use a predefined
connection, DLCI=1023 to send special frames for
this specific purpose
Sender responds by reducing data rate

FECN bit
Used to warn the receiver about the congestion
If there is an ACK mechanism at the higher level the receiver
can delay the ACK, thus forcing the source to slow down
Four cases of congestion in Frame Relay

Quality of Service (QoS)
Is an assurance from the network for a particular kind
of service
e.g. network uses retransmission strategy to make
sure that data arrives correctly.
This service is ok for non-real time application. But
may not be ok for real-time applications as it does-not
guarantee timeliness
i.e. we need a new service model in which, application
that need higher assurances can ask the network for
that
A network that can provide these different level of
services is said to support QoS.
Flow characteristics
Reliability
Lack of reliability means losing a packet or ACK, which may or may not
needs retransmission
Example: Email, file transfer needs retransmission
Audio and video may not need retransmission
Delay (Source-to-destination delay)
Application can tolerate delay in different degrees
Example: multimedia application need minimum delay, but in case of file
transfer or email it is less important
Jitter
Is a variation in delay for packets belonging to same flow.
Audio and Video cannot tolerate high jitter
No effect for file or mail transfer
Bandwdth
Different application needs different BW
In video transmission we need million of bits to refresh a color screen
While total no of bits in an email may not reach even a million
Techniques to Improve QoS
Common methods are scheduling, traffic shaping,
admission control,and resource reservation
Scheduling (FIFO, priority and weighted fair queuing)
When packets from different flows arrive at a router, It is
needed to treat the different flows in a fair and appropriate
manner. Some techniques are as follows
FIFO Queuing with tail drop
In this queuing, packets wait in a buffer until the node is ready to
process them
If average arrival rate is higher than the average processing rate, the
queue will fill up and new packets will be discarded without regard to
which flow the packet belongs to or how important the packets is?
It is simplest and most widely used in Internet routers

Scheduling Techniques contd.
Priority Queuing
Each packet is marked with a priority class
The router implements multiple FIFO queues, one for each priority class
It processes packets of higher priority first and moves on to the next
priority if the higher priority one is empty
If there is a continuous flow in a high priority queue, then this will create
a starvation problem in others
Therefore this should be optimized to put hard limits on how much high
priority traffic can be inserted in the queue
These scheduling is used in Internet to protect most important packets
like routing updates

Scheduling Techniques
Weighted fair Queuing
The packets are still assigned to different classes before
inserting to the queues
The router than serves queues in around-robin fashion
according to the weight of the queue
i.e. for above example: 3 pkts from first, two from 2nd and
one from 3rd queue

Traffic shaping
Is a mechanism to control the amount and the rate of the
traffic sent to the network.
Two techniques used 1. Leaky Bucket, 2. Token Bucket
Leaky Bucket
The idea is to have a constant bit rate traffic in the
network in spite of bursty data coming from source.
i.e. if a bucket has a

small hole at the bottom,
the water leaks from the
bucket at a constant rate
and is independent of the
rate of input to the bucket
Leaky bucket implementation
When the packets are of same fixed size then one packet can
be pushed to network per clock tick
If packets are of variable size than more packets per tick may
be allowed.
i.e. if rule is 1024 bytes per tick then one 1024 byte packet is
allowed per tick, two 512 bytes per tick and four 256 byte
packets per tick and so on
Algorithm
for each clock tick{
1. Initialize a byte counter to n
2. while n ≥ size of the packet
3. send the packet and decrement the counter by the packet size.
4. Stop the transmission till next tick
}
Where n is max number of bytes allowed per tick
Leaky bucket contd.
A leaky bucket algorithms shapes bursty traffic into fixed-rate
traffic by averaging the data rate. The packets will be dropped
if the buffer is full
This algorithm prevents congestion by avoiding instantaneous
heavy traffic at the output line
The buffer capacity should be carefully designed s.t. it should
be able to store the bursty data for short period of time,
otherwise packets will be dropped
Example:
data comes at a rate 25 Mbps, one 40ms burst every second. Design the
leaky bucket
Solution:
total data per sec = 25Mbps * 40 *10-3 = 1Mb
Thus capacity of buffer can be chosen as 1Mb
Uniform output rate may be chosen as 2Mbps, s.t. it will take 500ms to
drain the complete data
Token Bucket
The leaky bucket is restrictive. i.e. if a host is idle then bucket
becomes empty, if the host has bursty data then bucket allows
only an average rate.
But the token bucket algorithm allows idle hosts to accumulate
credit for the future in form of tokens
Algorithm:
token bucket holds tokens generated by a clock at the rate of one token per
∆T sec or n tokens per sec
It consumes one token per packet sent
i.e. to send a packet
there should be a token
available in the bucket

Token Bucket contd.
Leaky bucket and Token Bucket provides different kind of
traffic shaping
The Leaky bucket algorithm does not allow idle hosts to save up
permission to send large bursts later
But the token bucket algorithm does allow saving, up to the
max size of bucket. i.e. bursts upto the size of bucket can be
sent at once
The token bucket algorithm throws away tokens when the
bucket fills up but never discard packets.
But the Leaky bucket discards packets when bucket fills up
One variation to Token bucket
Each token represent the right to send k bytes in place of one
packet.
A packet can only be sent if enough tokens are available to
cover length in bytes. Fractional tokens are kept for future use
Quality of Service (QoS)
Two models have been proposed to
provide Quality of Service in the Internet
Integrated Services (IntServ)
Is a flow based QoS model designed for IP. i.e. a
user needs to create a flow, a kind of virtual
circuit, from the source to destination and inform
all routers about the resource requirement.
Differentiated Services (DiffServ)
Is a class based QoS model designed for IP. i.e.
the applications, or hosts, define the type of
service they need each time they send a packet.
Integrated services features
Signals
IP is a connection less protocol
To implement a flow based service a signaling protocol is used
to run over IP that provides the signaling mechanism for
making reservation
The protocol is named as Resource Reservation Protocol
Flow Specification
has two parts: Rspec and Tspec
Rspec(resource specification)
Defines the resource that the flow needs to reserve (buffer, bw etc.)
Tspec(Traffic specification)
Traffic characterization of the flow
Admission
After receiving flow specification the router decides to admit or
deny the flow Computer Networking / Module III/ AKN / 179
Integrated services features
Two service classes are defined
Guaranteed Service Class
Designed for real time traffic that needs guaranteed
minimum end-to-end delay. (multimedia)
end-to-end delay = sum of delays in routers + propagation
delay + setup mechanism
Only delay in router can be guaranteed by router
The amount of end-to-end delay and the data rate must be
defined by the application
Controlled-Load Service Class
Designed for applications that can accept some delays, but
are sensitive to an overload network and to the danger of
losing packets
Example application are file transfer, email etc.
Resource ReserVation Protocol (RSVP)
The resource reservation protocol is a signaling
protocol to help IP create a flow and consequently
make a resource reservation
The signaling system of RSVP is designed for
multicasting to enable it to provide resource
reservation for all kinds of traffic including multimedia,
which often uses multicasting
In this case the receivers (not the sender) makes the
reservation
It has several types of messages for above tasks. Two
of them are used for resource reservation, i.e. Path
message and Resv message
RSVP Path message
A Path message travels from the sender and reaches
all the receivers (downstream) in multicast path
On the way path message stores the necessary
information for the receivers.
A new message is created when the path diverges.

RSVP Recv message
Receiver sends a recv message,
which travels towards sender
(upstream) and makes a resource
reservation on the routers that
support RSVP
If a router does not support RSVP
on the path, it routes packet using
traditional delivery methods
Reservation merging
Resources are not reserved for each receiver in a
flow.
Reservation is merged to larger of the two (or
more) requests
As different qualities for multimedia is required by
different receivers, thus different requirements for
the same flow Computer Networking / Module III/ AKN / 183
Reservation Styles
When there are more than one flow, the router needs
to make a reservation to accommodate all of them
RSVP defines three types of reservation styles
Wild card Filter: router creates a single reservation
for all senders based on largest request. This is
used when flow from different receivers do not
occur at the same time
Fixed Filter: router creates a distinct reservation for
each flow. It is used when there is a high
probability that from different receivers occurs at
the same time
Shared Explicit: creates a single reservation which
can be shared by a set of flows
Differtiated services
Problems with integrated services
Scalability
This model requires that each router keep information for
each flow, which is impractical as load on routers will
increase
Service type limitation
It provides two services 1. Guaranteed and control load
Solutions
The routers do not have to store information about flows.
i.e. The applications, or hosts, define the type of service
they need each time they send a packet
The per-flow service is changed to per class service.
The router routes the packet based on the class of service
This is called DifferentiatedComputer
services
Networking / Module III/ AKN / 185
Differentiated service features
Each packet contains a field called DS field.
The value of this field is set by the first router
designated as the boundary router.
contains two sub-fields:
Differentiated services code point: defines per hop behavior
(PHB) and an unused field
DE PHB(default PHB) same as TOS.
EF PHB (expedited forwarding) provides following services
like Low loss, Low latency, Ensured bandwidth.
AF PHB (Assured forwarding) delivers the packet with a high
assurance as long as the class traffic does not exceed the
traffic profile of the node. The users of the network need to
be aware that some packets may be discarded
Traffic conditioner
The DS node uses traffic conditioners like
Meters: checks to see if the incoming flow matches the
negotiated traffic profile
Marker: can remark a packet that is using best-effort delivery
or down-mark a packet based on information received from
the meter.
Shaper: reshapes the traffic if not compliant with negotiated
traffic
Dropper: discards a packet if flow severely violates the
negotiated profile

END of module III
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COMPUTER NETWORKS

Text: “Data Communications and Networking” Third Edition,

Computer Networking / Module III/ AKN / 3

¾Protocol used is IP for Network Layer

1000000 00001011 00000011 00011111

 Class B 128.0.0.0 – 191.255.255.255

 Class C 192.0.0.0 – 223.255.255.255

 Class D 224.0.0.0 – 239.255.255.255

 Class E 240.0.0.0 – 255.255.255.255

Computer Networking / Module III/ AKN / 9

123.50.16.90 123.65.7.34 123.90.123.4

Find Network addresses of the following IP addresses

Class C networkComputer Networking / Module III/ AKN / 12

get a valid IP address from the DHCP server

Computer Networking / Module III/ AKN / 15

141.14.2.21 255.255.0.0 141.14.0.0

 A bit wise and operation is performed

Computer Networking / Module III/ AKN / 18

Computer Networking / Module III/ AKN / 22

 According to classic IP routing rules, it was not possible to use the

Computer Networking / Module III/ AKN / 25

Computer Networking / Module III/ AKN / 33

Net1 Destination Next hop

Net2 P Net3 Table for host A

internet by specifying the exact format of data

the data will be sent

unreliable packet delivery.

message should be generated, and the conditions under

Computer Networking / Module III/ AKN / 39

Computer Networking / Module III/ AKN / 40

 Precedence: (3 bits) defines priorities in cases like congestion

frame size differs for different networks and is

Copy Class Number

Computer Networking / Module III/ AKN / 48

Options 7-byte opt 1

Data 8-byte opt

 7: Record Route, It is used to record the routers that handles the

Code Length Pointer

Computer Networking / Module III/ AKN / 49

Code Length Pointer OFlow Flags

Computer Networking / Module III/ AKN / 52

• How can a host know which router to use for a given

Computer Networking / Module III/ AKN / 60

Computer Networking / Module III/ AKN / 64

Computer Networking / Module III/ AKN / 65

Header Datagram Data

Header Frame Data

Computer Networking / Module III/ AKN / 74

Computer Networking / Module III/ AKN / 75

 ICMP supports a router discovery scheme that allows hosts to

Computer Networking / Module III/ AKN / 81

Computer Networking / Module III/ AKN / 83

• consecutive sections consisting of zeros

 Type identifier: 3 bit field defines the address as a provider-

 Nobody outside the organization can send a message

Computer Networking / Module III/ AKN / 90

Computer Networking / Module III/ AKN / 92

Computer Networking / Module III/ AKN / 95

Computer Networking / Module III/ AKN / 97

Class B 128.0.0.0 – 191.255.255.255

Class C 192.0.0.0 – 223.255.255.255

Class D 224.0.0.0 – 239.255.255.255

Class E 240.0.0.0 – 255.255.255.255

A bit wise and operation is performed

According to classic IP routing rules, it was not possible to use the

Precedence: (3 bits) defines priorities in cases like congestion

7: Record Route, It is used to record the routers that handles the

ICMP supports a router discovery scheme that allows hosts to

Type identifier: 3 bit field defines the address as a provider-

Nobody outside the organization can send a message

Port nos may vary from 0-65535, and source port is

Four cases of congestion in Frame Relay

i.e. if a bucket has a