Lol
Lol
Lol
The World Wide Web (WWW) is a vast network of information distributed globally. The
architecture of the WWW involves various components:
1. Client-Server Model:
The WWW operates on a client-server model, where clients (web browsers)
request and access web content from servers.
2. Client:
A client is typically a web browser like Chrome, Firefox, or Safari.
It comprises three main components:
Controller: Handles user input from the keyboard and mouse.
Client Protocol: Responsible for interacting with the server.
Interpreter: Renders and displays the web content, with interpreters like
HTML, Java, or JavaScript depending on the document type.
3. Server:
Servers store web pages and respond to client requests.
They often use caching to improve efficiency by storing frequently requested files
in memory.
Servers can employ multithreading or multiprocessing to handle multiple
requests simultaneously.
4. Uniform Resource Locator (URL):
To access web content, clients use URLs, which define the address of the web
page.
A URL includes the protocol (e.g., HTTP), host computer (e.g., www.example.com),
port (optional), and path (file location).
5. Cookies:
Originally, the WWW was stateless, but cookies enable stateful interactions.
Cookies store information on the client side, allowing servers to recognize and
interact with returning clients without revealing the stored data to the user.
They are used for various purposes, including authentication, e-commerce,
portals, and advertising.
Web documents are categorized into three main types based on when their content is
determined:
1. Static Documents:
Static documents are fixed-content web pages that are pre-created and stored on
a server.
When a client accesses a static document, a copy of the document is sent to the
client.
HTML (Hypertext Markup Language) is commonly used for creating static web
pages.
2. Dynamic Documents:
Dynamic documents are generated by the web server in real-time when a client
requests them.
Server-side technologies like Common Gateway Interface (CGI) are used to create
dynamic content.
3. Active Documents:
Active documents involve client-side scripting to create dynamic content.
Java applets can be used to run programs on the client's side.
JavaScript is another scripting language that can be used to create small,
interactive programs in web pages.
HTTP is the protocol that underlies the World Wide Web. It is designed for retrieving and
displaying web content. Here's a detailed breakdown of HTTP:
HTTP Basics:
1. Purpose: HTTP is primarily used to access data on the World Wide Web.
2. Functionality: It combines features of both FTP (File Transfer Protocol) and SMTP (Simple
Mail Transfer Protocol). Like FTP, it transfers files, and it operates over the services of TCP
(Transmission Control Protocol). Unlike FTP, HTTP typically uses only one TCP connection.
3. Stateless Protocol: HTTP operates as a stateless protocol. Each interaction between the
client and server is independent. There is no ongoing state or memory of previous
interactions.
4. Message Format: HTTP messages have a format controlled by MIME-like headers. These
messages are not designed for human consumption but are interpreted by HTTP servers
and clients.
HTTP Transaction:
An HTTP transaction involves a client and a server. The client initiates the transaction by
sending a request message, and the server responds with a response message.
HTTP Messages:
HTTP messages consist of two types: request messages (from the client to the server) and
response messages (from the server to the client). Both types of messages share a similar
format: a request/status line, headers, and a body.
Request Line: The first line in a request message is called the request line.
Status Line: The first line in the response message is called the status line.
Status Codes:
Status codes are included in response messages to indicate the outcome of the request.
These codes are three-digit numbers with specific meanings:
100-199: Informational.
200-299: Success.
300-399: Redirection.
400-499: Client-side errors.
500-599: Server-side errors.
HTTP Headers:
Headers exchange additional information between the client and the server. They can be
divided into different types, including general headers, request headers, response
headers, and entity headers.
General Headers:
General headers provide general information about the message and can be present in
both request and response messages.
Request Headers:
Request headers are found only in request messages and specify the client's
configuration and preferred document format.
Response Headers:
Response headers are present only in response messages and provide information about
the server's configuration and the request.
Entity Headers:
Entity headers give information about the body of the document. They are mostly found
in response messages but can also be in some request messages.
Body:
The body of an HTTP message can contain the document to be sent or received. It can be
present in both request and response messages.
HTTP versions prior to 1.1 used non-persistent connections, where a new TCP connection
was established for each request/response. HTTP 1.1, by default, uses persistent
connections, where the server can leave the connection open for multiple requests.
Proxy Server:
HTTP supports proxy servers, which are intermediary servers that store copies of recent
responses. When a client sends a request, the proxy server can serve the response from
its cache if available, reducing load on the original server and improving latency.
HTTP is a fundamental protocol for the web, enabling the exchange of data and content between
clients and servers, and its principles are key to understanding how information is retrieved from
the World Wide Web.
The provided text explains TELNET, a client/server program that allows a user to access
applications on a remote computer. Here's a detailed breakdown of the text, including the
discussion of TELNET modes:
In local login, a terminal driver accepts keystrokes from the user, interprets them, and
invokes the desired application program.
In remote login, users access application programs or utilities on a remote machine.
Keystrokes are sent to the local operating system, which forwards them to the TELNET
client.
The TELNET client translates these keystrokes into a universal character set called
Network Virtual Terminal (NVT) characters, which are sent over the Internet.
The TELNET server on the remote machine changes these characters into a format
understandable by the remote computer.
NVT uses two sets of characters: one for data and the other for control.
Both sets consist of 8-bit bytes. Control characters are identified by the highest-order bit
being set to 1.
Embedding:
TELNET uses a single TCP connection with the server using port 23 and the client using an
ephemeral port.
Control characters are embedded within the data stream to distinguish between data and
control characters. This is accomplished using a special control character called "Interpret
as Control" (IAC).
Options:
TELNET allows clients and servers to negotiate options, which are extra features available
for more sophisticated terminals.
Option Negotiation:
Option negotiation is necessary to use any of the options, and it involves a process of
negotiation through commands like WILL, DO, WONT, and DONT.
Mode of Operation:
TELNET implementations operate in one of three modes: default mode, character mode,
or line mode.
Default Mode:
In the default mode, echoing is done by the client, and the client sends characters to the
server only when a complete line is typed.
Character Mode:
In character mode, each character is sent immediately from the client to the server, and
the server echoes the character back to be displayed on the client's screen.
Line Mode:
Line mode compensates for deficiencies in default and character modes. In this mode,
line editing (including echoing, character erasing, and line erasing) is done by the client,
which then sends the entire line to the server.
In summary, TELNET is a versatile protocol for remote access and allows users to interact with
remote computers, and its modes cater to various needs, including reduced network traffic and
efficient line editing.
The process of establishing and releasing TCP connections is a fundamental part of the
Transmission Control Protocol (TCP). Below is an explanation of the TCP connection
establishment and release processes as described in the provided text:
TCP connections are released independently for each direction, meaning that each end
can initiate the release process.
1. Either party can send a TCP segment with the FIN (Finish) bit set to indicate that it
has no more data to transmit in one direction.
2. Upon receiving the FIN segment, the receiving end acknowledges the FIN.
3. The direction that sent the FIN is now closed for new data, although data may still
continue to flow in the other direction.
1. Both ends can independently send FIN segments to signal the end of data
transmission from their respective directions.
2. When each side receives the FIN from the other, they acknowledge it.
3. Once both directions have acknowledged the FIN segments, the connection is
considered released.
Timers are used to handle situations where an acknowledgment for a FIN segment is not
received within a reasonable time.
If no acknowledgment arrives within twice the maximum packet lifetime, the sender of
the FIN segment assumes that the other side is no longer listening and releases the
connection.
The text briefly mentions a SYN flood attack, which can be used to tie up resources on a
host by sending a stream of SYN segments.
To defend against SYN floods, SYN cookies can be used. These are cryptographic
sequence numbers that allow a host to verify the validity of an acknowledgment without
having to remember it.
The provided text also includes a finite state machine diagram illustrating the different states a
connection can be in, the events that trigger state transitions, and the actions taken upon those
events. These state transitions are fundamental for understanding the behavior of TCP
connections.
The TCP header is an essential part of the Transmission Control Protocol (TCP) segment, and it
contains various fields that facilitate communication between devices over a network. Let's break
down the fields in the TCP header as described in the provided text, and visualize them in a neat
diagram:
Here is a summary of the fields in the TCP header, along with their descriptions:
The TCP header structure is followed by an optional variable-length Options field, which can
extend up to 40 bytes to accommodate different options. These options allow hosts to negotiate
and communicate additional capabilities for the TCP connection.
Let's analyze the provided partial dump of a TCP header in hexadecimal format:
(i) What is the source port number? The source port number is the first 16 bits of the TCP header,
which corresponds to the first two hexadecimal digits: 05 32 (in hexadecimal) or 1314 (in
decimal). So, the source port number is 1314.
(ii) What is the application being used? Determining the specific application based solely on the
source port number can be challenging, as it varies. However, certain well-known port numbers
are associated with specific applications. You can refer to the IANA (Internet Assigned Numbers
Authority) service name and port number registry to find common applications associated with
specific port numbers. Keep in mind that applications can use dynamic or non-standard port
numbers, so it's not always possible to determine the application definitively from the port
number alone.
(iii) What is the sequence number? The sequence number is the next 32 bits in the TCP header,
which correspond to the following 8 hexadecimal digits: 00000001 00000000 (in hexadecimal). To
get the decimal equivalent, we'll convert the hexadecimal values to decimal:
(iv) What is the acknowledgment number? The acknowledgment number follows the sequence
number in the TCP header. In the provided dump, it corresponds to the 32-bit value: 500207FF (in
hexadecimal). Converting this hexadecimal value to decimal:
500207FF (in hexadecimal) is 134493439 (in decimal). So, the acknowledgment number is
134493439.
Keep in mind that the interpretation of the source port and acknowledgment numbers is
relatively straightforward, while identifying the application based on the source port may require
additional context and knowledge of commonly associated port numbers for specific
applications.
The TCP (Transmission Control Protocol) service model is a fundamental aspect of the TCP
protocol suite used for communication over the Internet. Here's a note on various aspects of the
TCP service model:
In summary, the TCP service model is a fundamental framework for establishing reliable
connections between sockets. It provides the necessary mechanisms for identifying services using
port numbers, ensuring reliable and full-duplex communication, and handling data as a byte
stream. While TCP has evolved and adapted over time, it remains one of the most widely used
transport layer protocols for Internet communication.
Here's a table summarizing the key differences between UDP (User Datagram Protocol) and TCP
(Transmission Control Protocol):
Connection
Type Connectionless Connection-oriented
Order of
Delivery No guarantee of order Guaranteed order of delivery
Congestion
Control No congestion control mechanism Congestion control mechanisms
Overhead Lower overhead due to smaller header Higher overhead due to larger header
Suitable for real-time applications and Suitable for reliable data transfer, web
scenarios where some packet loss is browsing, email, and applications where data
Use Cases acceptable integrity is crucial
Please note that UDP and TCP serve different purposes. UDP is preferred for real-time
applications where low overhead and speed are critical, even if some packet loss is acceptable. In
contrast, TCP is used when data integrity and reliability are of paramount importance, even if it
comes at the cost of increased overhead and a slight delay due to the connection establishment
process.
The Open Shortest Path First (OSPF) protocol is a widely used interior gateway routing protocol
for determining the best paths for routing data packets within an Autonomous System (AS).
Here's an explanation of the OSPF protocol:
In summary, OSPF is a robust and efficient routing protocol used for determining the best paths
for routing data packets within an Autonomous System. It provides scalability through
hierarchical area design, maintains routing consistency, and ensures reliable communication
among routers through a variety of message types.
ARP (Address Resolution Protocol) is a crucial protocol used to map IP addresses onto data link
layer addresses, such as Ethernet addresses. It plays a vital role in local network communications.
Let's explain the ARP protocol and how it works with a simple sketch:
ARP is a protocol used for dynamically mapping a 32-bit IP address (Layer 3) to a 48-bit Ethernet
address (Layer 2) on a local network. It enables devices within the same network segment to
discover each other's hardware addresses, which are essential for delivering data packets. ARP
operates at the data link layer and is used primarily within the same broadcast domain or LAN.
1. Initialization:
Each device on a local network has a unique 48-bit Ethernet address (MAC
address) and at least one IP address.
When a device needs to communicate with another device on the same network,
it needs to know the target device's Ethernet address.
2. ARP Request:
Suppose Host 1 wants to send a packet to Host 2, and it knows Host 2's IP
address but not its Ethernet address.
Host 1 sends an ARP request as a broadcast message to the entire local network.
The ARP request contains:
Sender's IP address (Host 1's IP)
Sender's MAC address (Host 1's MAC)
Target IP address (Host 2's IP)
A placeholder for the target's MAC address (initially set to
00:00:00:00:00:00).
3. ARP Response:
When Host 2 receives the ARP request and recognizes its IP address in the
request, it replies to Host 1 with an ARP response.
The ARP response contains:
Sender's IP address (Host 2's IP)
Sender's MAC address (Host 2's MAC)
Target IP address (Host 1's IP)
Target MAC address (Host 1's MAC).
4. Caching ARP Information:
Host 1 receives the ARP response, updates its ARP table with the mapping of
Host 2's IP to Host 2's MAC address.
This ARP information is cached to avoid future ARP requests for Host 2 on the
local network.
5. Sending Data Packet:
Now that Host 1 knows the MAC address of Host 2, it can encapsulate the IP
packet within an Ethernet frame with the correct destination MAC address and
send it to Host 2.
Host 2, upon receiving the Ethernet frame, extracts the IP packet and processes it.
This process ensures that devices within the same local network can discover each other's
Ethernet addresses when needed, enabling the successful transmission of data packets. ARP plays
a crucial role in local network communications.
ICMP (Internet Control Message Protocol) is a network protocol used for various diagnostic and
error-reporting purposes within IP networks. ICMP messages help maintain the proper
functioning of the network and provide valuable feedback to network administrators. Here are
some common types of ICMP messages:
1. Echo Request and Echo Reply (Ping) (Type 8 and Type 0):
Echo Request (Type 8): Sent by a host to request an "echo" from another host,
often referred to as "pinging."
Echo Reply (Type 0): Sent by the target host in response to an Echo Request,
indicating its availability and responsiveness.
2. Destination Unreachable (Type 3):
Used to indicate that a destination host or network is unreachable for various
reasons, such as network congestion, unreachable host, or protocol unreachable.
3. Time Exceeded (Type 11):
Used to indicate that a packet has exceeded its time-to-live (TTL) value while
traversing through routers. It helps detect routing loops or network issues.
4. Redirect Message (Type 5):
Sent by a router to inform a host that a better route is available for a specific
destination.
5. Router Advertisement and Router Solicitation (Type 9 and Type 10):
These messages are used in the context of IPv6 to facilitate the autoconfiguration
of network interfaces and to discover routers on the local network.
6. Parameter Problem (Type 12):
Used to indicate that a problem has been detected with the IP header, such as an
unrecognized option or an incorrect length.
7. Timestamp Request and Timestamp Reply (Type 13 and Type 14):
Timestamp Request (Type 13): Sent to request timestamps for diagnostic and
timing purposes.
Timestamp Reply (Type 14): Sent in response to a Timestamp Request,
providing timing information.
8. Address Mask Request and Address Mask Reply (Type 17 and Type 18):
These messages are used to determine the subnet mask of a network, especially
in older versions of ICMP.
9. Source Quench (Type 4):
Sent to indicate to a sender that its traffic is causing congestion within the
network, and it should slow down the rate of transmission.
These ICMP messages play a critical role in network troubleshooting, error detection, and
communication between network devices. Network administrators use ICMP messages to identify
and address issues, ensuring the efficient operation of IP networks.
Packet fragmentation is the process of breaking down a large data packet into smaller fragments
to fit within the Maximum Transmission Unit (MTU) size of the network medium. It is an essential
technique when dealing with networks that have limitations on the size of data packets that can
be transmitted. There are two main types of fragmentation:
1. Nontransparent Fragmentation:
Responsibility: The sending device or host is responsible for fragmentation.
Process: When a data packet is larger than the MTU of the outgoing network link,
the sending device breaks down the packet into smaller fragments that fit within
the MTU size.
Headers: Each fragment created by the sending device includes its own headers,
including network layer (e.g., IP) and transport layer (e.g., TCP or UDP) headers.
Reassembly: When the fragments reach their destination, the receiving device or
final destination host must reassemble the fragments back into the original
packet.
Intermediate Devices: In nontransparent fragmentation, intermediate network
devices, such as routers, are not aware of the fragmentation process. They treat
each fragment as an independent packet and forward them to their destination
without knowledge of the original packet's structure.
2. Transparent Fragmentation:
Responsibility: Network devices and routers along the path of the packet are
responsible for fragmentation.
Process: When a data packet is larger than the MTU of an outgoing network link,
the intermediate network devices along the path will detect the oversize packet
and fragment it into smaller pieces that fit within the MTU size of the outgoing
link.
Headers: In transparent fragmentation, the intermediate network devices modify
the packet headers to account for fragmentation, and the process is entirely
transparent to the sender and receiver.
Reassembly: The sender sends the original packet, and the receiver receives the
reassembled packet as if no fragmentation occurred.
Intermediate Devices: In transparent fragmentation, intermediate network
devices are actively involved in the process of breaking down and reassembling
the packet. This ensures that packets can traverse the network without issues
related to packet size limitations.
The choice between nontransparent and transparent fragmentation depends on the network
configuration and the capabilities of the devices involved. Transparent fragmentation can be
particularly useful when dealing with devices that do not support or cannot handle fragmentation
and helps ensure that packets can smoothly traverse networks with varying MTU sizes. However,
it's essential to be aware that fragmentation can introduce overhead, increase network latency,
and potentially impact network performance, so efficient network design and the use of
techniques like Path MTU Discovery (PMTUD) are encouraged to reduce the reliance on
fragmentation.
Tunneling is a networking technique that enables the transmission of data securely across an
untrusted network, such as the internet, by encapsulating data packets from one network
protocol within the packets of another protocol. This creates a "tunnel" through which the data
can traverse, protecting it from unauthorized access, interception, and tampering. Tunneling is
commonly used to ensure data privacy and security when transmitting sensitive information over
unsecured networks or when connecting remote networks.
1. Encapsulation: Tunneling involves encapsulating data packets of one network protocol inside
the packets of another network protocol. This means that the original data is placed within a new
packet structure used by the second protocol. The inner data packet is often referred to as the
"payload."
2. Creating a Secure Path: The process of encapsulation allows data to travel through the
untrusted or public network as if it were within a secure, private network. The encapsulated data
remains hidden from potential eavesdroppers on the public network.
3. Security and Privacy: Tunneling typically involves encryption to secure the data within the
tunnel. Encryption ensures that even if data packets are intercepted, they cannot be easily
deciphered without the encryption key.
Point-to-Point Tunneling Protocol (PPTP): Used for creating VPNs and secure
connections over the internet. Widely used but considered less secure due to
vulnerabilities.
Layer 2 Tunneling Protocol (L2TP): Often used in combination with IPSec for enhanced
security. Suitable for creating secure point-to-point or site-to-site connections.
Internet Protocol Security (IPSec): Provides security services, including encryption and
authentication, for IP packets. It can be used for creating secure connections and VPNs.
Generic Routing Encapsulation (GRE): A simple, lightweight protocol used for creating
point-to-point connections. It does not provide encryption but is often used in
combination with other encryption protocols.
Secure Socket Tunneling Protocol (SSTP): A Microsoft-developed protocol for creating
secure connections, often used in combination with VPNs.
Tunneling is a fundamental technique for ensuring the security and interoperability of networks,
especially when dealing with different network protocols and the need to protect data while
traversing untrusted networks like the internet.
Inter-domain routing (also known as Exterior Gateway Protocol, EGP) and intra-domain routing
(also known as Interior Gateway Protocol, IGP) are two fundamental components of routing in
computer networks. They serve different purposes and operate at different levels of network
hierarchy. Here's a comparison and contrast of inter-domain and intra-domain routing in tabular
form:
Protocol (BGP) and handles like OSPF, RIP, and EIGRP, which
(AS) routing operate
BGP uses path attributes, IGPs use various metrics (e.g., hop
Metrics including count,
Examples Border Gateway Protocol (BGP) Open Shortest Path First (OSPF),
Aspect Inter-Domain Routing Intra-Domain Routing
Routing
In summary, inter-domain routing and intra-domain routing serve distinct purposes and operate
at different levels of the network hierarchy. Inter-domain routing handles routing between
autonomous systems (ASes) on a global scale and is often subject to policy-based routing
decisions. Intra-domain routing, on the other hand, manages routing within a single AS or
domain, typically using metrics to make routing decisions and maintaining a smaller, localized
routing table. Each type of routing has its own challenges, requirements, and protocols tailored
to its specific use case.
Border Gateway Protocol (BGP) is the primary Exterior Gateway Routing Protocol used to
implement internetworking, particularly in the context of the global internet. BGP plays a crucial
role in routing data between autonomous systems (ASes) and enabling connectivity and data
exchange between networks that are under different administrative controls. Here's a detailed
explanation of how BGP is used for internetworking:
BGP is specifically designed for inter-domain routing, which involves routing between
different ASes, each under separate administrative control. It operates at the network
layer (Layer 3) and is responsible for exchanging routing information between these ASes.
BGP allows network administrators to implement a wide range of routing policies. These
policies can involve political, security, or economic considerations. For example, an AS
may want to:
Restrict the flow of commercial traffic on an educational network.
Avoid sending traffic through specific countries or regions for security reasons.
Choose routes based on cost considerations.
Opt for certain routes due to performance or reliability concerns.
Ensure that traffic to or from specific organizations does not transit through
certain other organizations.
IXPs are crucial for facilitating the exchange of internet traffic between different ISPs and
networks. These points play a vital role in improving the efficiency of internet traffic
exchange by reducing the need for data to travel long distances through external
networks.
BGP is a path vector protocol. It maintains a path or route history along with the next hop
router's information. This path history is used to detect and prevent routing loops,
enhancing routing stability.
BGP advertisements include the AS path, which represents the sequence of ASes the
route has traversed. This path information helps in route selection and loop detection.
To propagate BGP routes inside an ISP, a variant called iBGP (internal BGP) is often used.
iBGP ensures that BGP routes are disseminated within the AS and helps in selecting the
best route among multiple options.
10. Early Exit or Hot-Potato Routing: - The strategy of choosing the quickest route to exit the
AS is known as "early exit" or "hot-potato routing." It may lead to asymmetric routing paths
where incoming and outgoing traffic follow different routes.
11. Tiebreakers: - Tiebreakers are used when multiple routes have the same level of preference.
For example, the shortest AS path is often chosen as a tiebreaker.
In conclusion, BGP is the protocol that enables internetworking on a global scale by allowing
different ASes to exchange routing information and implement complex routing policies. It plays
a critical role in defining how data flows across the internet, ensuring that data is delivered
efficiently and in accordance with various policies and agreements between ASes.
To allocate IP address ranges for organizations A and B, we need to consider the number of IP
addresses requested and ensure that the allocations do not overlap. In this case, Organization A
requests 4,000 IP addresses, and Organization B requests 2,000 IP addresses.
We will allocate these IP address ranges based on their requested counts. To do this, we'll use
CIDR notation to specify the IP address range and subnet mask.
To determine the Total length, Identification, DF (Don't Fragment), MF (More Fragments), and
Fragment Offset fields of the IP header in each packet transmitted over the three links, we need
to perform IP fragmentation to ensure that the packets match the maximum frame sizes for each
link. Let's go through the process:
1. Original IP Packet:
Data size: 900 bytes
TCP header size: 20 bytes
The total size of the original packet, including the IP header, can be calculated as follows: Total
size = Data size + TCP header size + IP header size
We need to consider the IP header size, which is typically 20 bytes for IPv4. The IP header size is
not explicitly mentioned, so we'll assume it's 20 bytes.
Total size = 900 bytes (Data) + 20 bytes (TCP header) + 20 bytes (IP header) = 940 bytes
Now, let's perform fragmentation for each link based on the maximum frame sizes.
2. Link A-R1 (Maximum frame size: 1024 bytes): Since the original packet's total size (940
bytes) is smaller than the maximum frame size, there's no need to fragment it for this link.
IP Header (Link A-R1):
Total length: 940 (original packet size)
Identification: A unique identifier (e.g., 12345)
DF: 0 (Not set to Don't Fragment)
MF: 0 (No more fragments)
Fragment Offset: 0 (No fragmentation)
3. Link R1-R2 (Maximum frame size: 512 bytes): We need to fragment the packet for this
link. The maximum frame size is 512 bytes, so we'll create two fragments.
Fragment 1 (First 512 bytes):
Total length: 512 bytes
Identification: The same unique identifier as the original packet (e.g., 12345)
DF: 0 (Not set to Don't Fragment)
MF: 1 (More fragments)
Fragment Offset: 0 (This is the first fragment)
Fragment 2 (Remaining data):
Total length: 428 bytes (the remaining data)
Identification: The same unique identifier (e.g., 12345)
DF: 0 (Not set to Don't Fragment)
MF: 0 (No more fragments)
Fragment Offset: 64 (Offset of the second fragment in the original packet)
4. Link R2-B (Maximum frame size: 512 bytes): The second fragment (428 bytes) from the
previous step is smaller than the maximum frame size, so there's no need to fragment it
further.
IP Header (Link R2-B):
Total length: 428 (fragment size)
Identification: The same unique identifier (e.g., 12345)
DF: 0 (Not set to Don't Fragment)
MF: 0 (No more fragments)
Fragment Offset: 0 (No fragmentation)
In summary, the Total length, Identification, DF, MF, and Fragment Offset fields of the IP header
in each packet transmitted over the three links are as described above. Please note that the
"Identification" field typically holds a unique identifier for the entire original packet, and the same
identifier is used for all fragments generated during the fragmentation process.
To fragment the 2048-byte TCP message for delivery across two networks (N1 and N2) with
different Maximum Transmission Unit (MTU) sizes and header sizes, we will need to perform IP
fragmentation. The goal is to generate fragments that match the MTU of each network while
keeping track of the offset for each fragment. Let's go through the process:
Total size of the original packet, including the IP header: Total size = Data size + TCP header size
+ IP header size
We need to account for the IP header size, which is not provided but typically 20 bytes for IPv4.
Total size = 2048 bytes (Data) + 20 bytes (TCP header) + 20 bytes (IP header) = 2088 bytes
Network N1:
Network N2:
Now, let's calculate the sizes and offsets of the sequence of fragments delivered to the network
layer at the destination host:
In summary, the sequence of fragments delivered to the network layer at the destination host
consists of two fragments for Network N1 (both 1024 bytes) and three fragments for Network N2
(512 bytes, 512 bytes, and 40 bytes) with their respective sizes and offsets as described above.
Advantages of Tunneling:
Disadvantages of Tunneling: