If a server has a high volume of connections that frequently enter the TIME_WAIT state, it can lead to resource exhaustion. Each connection in this state consumes system resources, such as memory and available ports. If the server is unable to free up these resources due to a large number of connections remaining in TIME_WAIT, it may eventually reach the limit of available ports, leading to connection failures for new incoming requests. This situation can severely impact the server’s performance, especially under high load conditions. To mitigate this issue, network administrators can consider adjusting the TCP settings on the server, such as reducing the TIME_WAIT duration or implementing techniques like connection pooling or reusing sockets. However, these adjustments must be made with caution, as they can introduce risks of packet misordering or data loss if not managed properly. Understanding the implications of the TIME_WAIT state is essential for maintaining optimal network performance and ensuring that the server can handle incoming connections efficiently.

ULAs are structured in the format of FC00::/7, where the first seven bits are reserved for the ULA prefix. This means that any address starting with FC00 or FD00 is considered a ULA. The uniqueness of these addresses is guaranteed by the fact that they are generated using a random 40-bit global ID, which minimizes the risk of address conflicts when merging networks or connecting different segments of an organization. On the other hand, Global Unicast Addresses are routable on the Internet and would not meet the requirement of internal-only communication. Link-Local Addresses, which are used for communication within a single local network segment, are not suitable for communication across different networks, as they are only valid within the local link and are not routable beyond that. Lastly, Multicast Addresses are used for one-to-many communication and do not serve the purpose of unique addressing for devices within a local network. Thus, the most appropriate choice for the administrator’s needs is the Unique Local Address, as it allows for internal communication while ensuring that addresses remain unique within the organization. This understanding of IPv6 address types is crucial for effective network design and management, particularly in environments that require both internal and external communication capabilities.

During this handshake, the server presents its digital certificate, which contains its public key and is signed by a trusted Certificate Authority (CA). This step is vital for authenticating the server’s identity, ensuring that clients are communicating with the legitimate server and not an imposter. The client verifies the certificate against a list of trusted CAs, and if the verification is successful, it proceeds to generate a unique session key. After the handshake, symmetric encryption is employed for the actual data transmission. This is because symmetric encryption is generally faster and more efficient for encrypting large amounts of data compared to asymmetric encryption. The session key, which is derived during the handshake, is used for this symmetric encryption, ensuring confidentiality and integrity of the data being transmitted. In contrast, the other options present misconceptions about the SSL/TLS process. For instance, using a single shared key for both encryption and authentication before data transmission does not align with the SSL/TLS protocol, which relies on a combination of asymmetric and symmetric encryption. Similarly, relying solely on IP address verification undermines the security provided by digital certificates, and stating that SSL/TLS operates only on asymmetric encryption throughout the session ignores the efficiency benefits of symmetric encryption after the handshake. Thus, understanding the nuanced steps and components involved in SSL/TLS is essential for securing network communications effectively.

In contrast, options that suggest UDP guarantees packet delivery or order (such as options b and c) are incorrect. UDP does not provide any guarantees regarding the delivery of packets; they may arrive out of order, be duplicated, or even be lost without notification. This lack of reliability is a trade-off for the speed and efficiency that UDP offers. Furthermore, the notion that UDP establishes a dedicated connection (as stated in option d) is fundamentally incorrect. UDP is inherently connectionless, meaning that each packet is treated independently, and there is no need for a dedicated path between sender and receiver. This allows for a more efficient use of network resources, particularly in scenarios where timely delivery is prioritized over reliability. In summary, the key advantage of UDP in this context is its ability to transmit data quickly without the overhead of establishing and maintaining a connection, making it suitable for applications that can tolerate some level of data loss.

The Session Layer (Layer 5) is responsible for establishing, managing, and terminating sessions between applications. It ensures that the data exchange is synchronized and that the communication remains open for the duration of the session. If there is a failure at this layer, devices may not be able to establish a session, leading to issues such as the inability to share files. On the other hand, the Transport Layer (Layer 4) is responsible for end-to-end communication and error recovery, ensuring that data is delivered reliably. While it plays a crucial role in data transmission, it does not directly manage sessions. The Network Layer (Layer 3) is responsible for routing packets across the network, and the Data Link Layer (Layer 2) deals with node-to-node data transfer and physical addressing. Neither of these layers is responsible for session management. In summary, the inability to establish a session for file sharing indicates a problem at the Session Layer, as this layer is specifically designed to handle the initiation and termination of sessions between applications. Understanding the functions of each layer in the OSI model is essential for effective troubleshooting in networking scenarios.

The primary security features provided by Tunnel mode include confidentiality, integrity, and authentication. Confidentiality is achieved through encryption, which ensures that the data within the original packet cannot be read by unauthorized parties. Integrity is maintained by using hashing algorithms to verify that the data has not been altered during transmission. Additionally, authentication ensures that the packets are coming from a legitimate source, preventing spoofing attacks. In contrast, Transport mode only encrypts the payload of the original packet, leaving the original IP header exposed. This can be a significant security risk, as attackers can see the source and destination IP addresses, potentially allowing them to target specific devices or networks. Furthermore, while Tunnel mode may require additional configuration and resources, such as VPN gateways, it is generally considered more secure than Transport mode for scenarios where the entire packet needs protection. Overall, the choice of Tunnel mode is particularly advantageous in scenarios where secure communication between remote sites is necessary, as it provides a comprehensive security solution that protects both the data and the routing information.

The formula to calculate the number of usable host addresses is given by: $$ \text{Usable Hosts} = 2^n – 2 $$ where \( n \) is the number of bits available for host addresses. In this case, since we have 16 bits for hosts, we can substitute \( n = 16 \): $$ \text{Usable Hosts} = 2^{16} – 2 = 65,536 – 2 = 65,534 $$ The subtraction of 2 accounts for the network address (172.16.0.0) and the broadcast address (172.16.255.255), which cannot be assigned to individual hosts. This large number of usable addresses (65,534) allows for significant scalability within the corporate network. It means that the organization can accommodate a large number of devices, such as computers, printers, and servers, without needing to readdress or redesign the network frequently. Additionally, the use of Class B addresses is particularly advantageous for medium to large organizations that anticipate growth, as it provides flexibility in subnetting. Furthermore, the IT department can implement various subnetting strategies to further divide the network into smaller segments for different departments or functions, enhancing both performance and security. For example, they could create subnets for different departments (e.g., HR, Finance, IT) while still maintaining a manageable number of addresses within each subnet. This approach not only optimizes network traffic but also simplifies management and improves security by isolating departmental traffic. In summary, understanding the implications of using Class B addresses and the calculation of usable host addresses is crucial for effective network design and future scalability.

In this scenario, the neighboring networks are using channels 2, 3, and 4. If the administrator selects channel 1, it will be the least affected by interference from these neighboring networks. Channel 1 operates at a frequency of 2412 MHz, while channel 2 operates at 2417 MHz, channel 3 at 2422 MHz, and channel 4 at 2427 MHz. The overlap of channel 1 with channels 2, 3, and 4 is minimal compared to the overlap that would occur if channel 6 or channel 11 were selected. Choosing channel 6 (2437 MHz) would result in significant interference from channel 4 (2427 MHz) and channel 5 (2432 MHz), while selecting channel 11 (2462 MHz) would also be affected by channels 10 (2457 MHz) and 9 (2452 MHz). Therefore, the optimal choice for the administrator, in order to minimize interference and maximize throughput, is channel 1. This decision is crucial in environments with multiple overlapping networks, as it directly impacts the quality of service and overall user experience. Understanding the implications of channel selection in relation to frequency overlap and interference is essential for effective wireless network management.

In this scenario, the neighboring networks are using channels 2, 3, and 4. If the administrator selects channel 1, it will be the least affected by interference from these neighboring networks. Channel 1 operates at a frequency of 2412 MHz, while channel 2 operates at 2417 MHz, channel 3 at 2422 MHz, and channel 4 at 2427 MHz. The overlap of channel 1 with channels 2, 3, and 4 is minimal compared to the overlap that would occur if channel 6 or channel 11 were selected. Choosing channel 6 (2437 MHz) would result in significant interference from channel 4 (2427 MHz) and channel 5 (2432 MHz), while selecting channel 11 (2462 MHz) would also be affected by channels 10 (2457 MHz) and 9 (2452 MHz). Therefore, the optimal choice for the administrator, in order to minimize interference and maximize throughput, is channel 1. This decision is crucial in environments with multiple overlapping networks, as it directly impacts the quality of service and overall user experience. Understanding the implications of channel selection in relation to frequency overlap and interference is essential for effective wireless network management.

Moreover, OSPF supports hierarchical network design through the use of areas, which helps in managing large networks more effectively. By segmenting the network into areas, OSPF can limit the scope of routing updates, thus enhancing scalability and performance. This hierarchical approach also allows for better load balancing and fault tolerance, as OSPF can quickly reroute traffic in the event of a link failure. In contrast, distance-vector protocols, which rely on metrics such as hop count, can become inefficient in larger networks due to their slower convergence times and susceptibility to routing loops. Additionally, OSPF is designed to handle a larger number of routes and requires more memory and CPU resources than simpler protocols, but this trade-off is justified in environments where performance and scalability are critical. Therefore, the advantages of OSPF in this scenario include its efficient use of bandwidth, faster convergence, and scalability, making it a suitable choice for the corporate network’s future growth. The other options present misconceptions about OSPF’s nature and capabilities, such as incorrectly categorizing it as a distance-vector protocol or suggesting that it is not suitable for large networks. Understanding these nuances is essential for network administrators when selecting the appropriate routing protocol for their specific needs.

In contrast, a switch operates at the data link layer (Layer 2) and is capable of intelligently forwarding data packets only to the specific device for which the data is intended. It does this by maintaining a MAC address table that maps the MAC addresses of connected devices to their respective ports. When a switch receives a data packet, it examines the destination MAC address and forwards the packet only to the port associated with that address. This targeted approach reduces unnecessary traffic on the network, leading to improved performance and efficiency. The other options present misconceptions: while switches can indeed manage VLANs (Virtual Local Area Networks) to segment network traffic, hubs do not have this capability at all. Additionally, switches typically consume more power due to their advanced functionalities, but this does not inherently make them less efficient for small networks; rather, it enhances their capability to manage traffic effectively. Lastly, the claim that hubs can filter traffic based on MAC addresses is incorrect, as hubs do not possess any filtering capabilities; they simply transmit data indiscriminately to all ports. Thus, the transition from a hub to a switch is justified by the need for improved data handling and network efficiency.

In contrast, the principle of redundancy focuses on creating backup systems or components to ensure network reliability and availability, which, while important, does not directly address the need for effective communication between diverse systems. The principle of scalability pertains to the network’s ability to expand and accommodate growth, which is a vital consideration for future-proofing the network but does not specifically relate to the immediate need for interoperability. Lastly, the principle of segmentation involves dividing a network into smaller, manageable sections to enhance performance and security, which is a valid strategy but does not encapsulate the overarching goal of enabling communication across different departments. Thus, the administrator’s focus on ensuring that various devices and systems can communicate effectively aligns with the principle of interoperability, making it the most relevant concept in this scenario. Understanding these principles is essential for network administrators to design efficient, secure, and adaptable networks that meet organizational needs.

In contrast, Infrastructure as a Service (IaaS) offers virtualized computing resources over the internet, which gives users more control over the infrastructure but requires them to manage everything from the operating system up to the application. This model is more suited for organizations that need extensive customization and control over their environment, which may not align with the company’s goal of focusing on application development. Software as a Service (SaaS) delivers fully functional applications over the internet, where users access software hosted on the provider’s servers. While this model is convenient for end-users, it does not provide the flexibility needed for developers to create and manage their applications. Function as a Service (FaaS) is a serverless computing model that allows developers to execute code in response to events without managing servers. While it offers scalability and ease of use, it may not provide the comprehensive development environment that PaaS offers, making it less suitable for a full application development lifecycle. Thus, PaaS stands out as the optimal choice for the company, as it strikes a balance between ease of use and the necessary tools for application development, allowing them to scale and innovate without the overhead of infrastructure management. This understanding of the nuances between different cloud service models is crucial for making informed decisions in cloud strategy and deployment.

In contrast, Infrastructure as a Service (IaaS) offers virtualized computing resources over the internet, which gives users more control over the infrastructure but requires them to manage everything from the operating system up to the application. This model is more suited for organizations that need extensive customization and control over their environment, which may not align with the company’s goal of focusing on application development. Software as a Service (SaaS) delivers fully functional applications over the internet, where users access software hosted on the provider’s servers. While this model is convenient for end-users, it does not provide the flexibility needed for developers to create and manage their applications. Function as a Service (FaaS) is a serverless computing model that allows developers to execute code in response to events without managing servers. While it offers scalability and ease of use, it may not provide the comprehensive development environment that PaaS offers, making it less suitable for a full application development lifecycle. Thus, PaaS stands out as the optimal choice for the company, as it strikes a balance between ease of use and the necessary tools for application development, allowing them to scale and innovate without the overhead of infrastructure management. This understanding of the nuances between different cloud service models is crucial for making informed decisions in cloud strategy and deployment.

The format for a ULA is FC00::/7, which means that any address starting with the prefix FC00 or FD00 is considered a unique local address. This allows devices within the same local network to communicate directly without needing a global address, fulfilling the requirement for internal communication. On the other hand, Global Unicast Addresses are routable on the internet and would not meet the requirement of avoiding public addresses for internal communication. Link-Local Addresses, which are used for communication within a single local network segment, are not suitable for communication across different networks, as they are only valid on the local link and cannot be routed. Lastly, Multicast Addresses are used for one-to-many communication but do not serve the purpose of direct device-to-device communication within a local network. Thus, the most appropriate choice for the administrator’s needs is the Unique Local Address, as it provides the necessary functionality for internal communication without exposing the devices to the global internet. This understanding of IPv6 address types is crucial for effective network design and management, particularly in environments that prioritize security and internal communication efficiency.

In this scenario, the neighboring networks are operating on channels 2, 3, and 4. Choosing a channel that is as far away from these channels as possible will help minimize interference. Channel 1 is the lowest channel and does not overlap with channels 2, 3, or 4, making it an optimal choice. Channel 6, while also non-overlapping with channels 2, 3, and 4, is closer in frequency to these channels and could experience some interference, albeit less than channels 2, 3, or 4. Channel 11, on the other hand, is the highest non-overlapping channel and is further away from the channels in use, but it is not the best choice in this specific scenario since it is the furthest from the interference source. Thus, while channels 6 and 11 are valid options, channel 1 is the most effective choice for minimizing interference from the neighboring networks operating on channels 2, 3, and 4. This decision is supported by the principles of frequency reuse and channel selection in wireless networking, which emphasize the importance of selecting channels that reduce co-channel and adjacent-channel interference to enhance overall network performance.

On the other hand, a packet filtering firewall operates at a more basic level, examining packets in isolation based on predefined rules such as source and destination IP addresses, ports, and protocols. While it can effectively block unwanted traffic, it lacks the ability to understand the context of the traffic flow, making it less secure for complex applications. A proxy firewall acts as an intermediary between the user and the internet, handling requests on behalf of clients. This type of firewall can provide additional security by hiding the internal network structure and filtering traffic at the application layer. However, it does not maintain the state of connections in the same way a stateful firewall does. Lastly, an application-layer firewall focuses on filtering traffic based on specific applications, providing deep packet inspection and the ability to enforce security policies at the application level. While this is beneficial for certain scenarios, it may not provide the comprehensive connection state management that a stateful firewall offers. Given the requirements of maintaining connection states and handling application-level traffic securely, a stateful firewall is the most appropriate choice. It combines the benefits of both packet filtering and connection tracking, ensuring robust security for the corporate network while allowing for efficient management of active connections.

When a TCP connection is established, each byte of data is assigned a unique sequence number. This numbering allows the receiving end to reorder packets that may arrive out of sequence. If a packet is lost during transmission, the receiving TCP stack can detect this by noticing a gap in the sequence numbers. It will then request the sender to retransmit the missing packets, ensuring that all data is received correctly and in the right order. While the three-way handshake is essential for establishing a reliable connection, it does not directly address the integrity of the data once the connection is active. Flow control mechanisms are important for managing the rate of data transmission to prevent overwhelming the receiver, but they do not guarantee that the data arrives in the correct order or that all packets are received. Multiplexing allows multiple connections to share a single port, but it does not contribute to the reliability or integrity of the data being transmitted. Thus, the combination of sequence numbers and acknowledgments is fundamental to TCP’s reliability, making it the most critical feature for maintaining data integrity in scenarios where packet loss or out-of-order delivery may occur. This understanding of TCP’s mechanisms is crucial for anyone working with network protocols, as it highlights the importance of reliable data transmission in network communications.

Moreover, HTTPS provides data integrity, meaning that the information sent and received cannot be altered or corrupted during transmission without detection. This is particularly important in preventing man-in-the-middle attacks, where an attacker could intercept and modify the data being sent between the client and server. While the other options present some misconceptions, they do not accurately reflect the primary benefits of HTTPS. For instance, while HTTPS may have some performance overhead due to encryption, it is not inherently faster than HTTP; in fact, it can be slightly slower due to the additional processing required for encryption and decryption. Additionally, HTTPS does not reduce server resource requirements; in fact, it may require more resources to manage the encryption process. Lastly, HTTPS does not simplify development; rather, it adds complexity by necessitating the management of SSL/TLS certificates and ensuring secure configurations. In summary, the primary advantages of implementing HTTPS over HTTP in this context are the enhanced confidentiality and integrity of sensitive data, which are critical for maintaining customer trust and complying with data protection regulations.

In addition to categorization, the report should include specific remediation steps tailored to each vulnerability. For instance, if outdated software is identified, the report should recommend updating to the latest version and provide links to relevant patches or updates. Assigning responsibilities ensures accountability, as different team members may be responsible for different aspects of remediation. This can be complemented by establishing deadlines for each task, which helps prioritize actions based on urgency and resource availability. Furthermore, the documentation should be clear and accessible, allowing both technical and non-technical stakeholders to understand the findings and the necessary actions. This comprehensive approach not only facilitates immediate remediation efforts but also serves as a reference for future audits and security assessments, ensuring continuous improvement in the organization’s security posture. By contrast, the other options lack the depth and structure needed for effective risk management, as they either provide insufficient detail, omit critical recommendations, or fail to create a lasting record of the findings.

1. Calculate the number of full segments: $$ \text{Number of full segments} = \left\lfloor \frac{\text{Total file size}}{\text{MSS}} \right\rfloor = \left\lfloor \frac{10000}{1460} \right\rfloor = 6 $$ 2. Calculate the remaining bytes: $$ \text{Remaining bytes} = \text{Total file size} – (\text{Number of full segments} \times \text{MSS}) = 10000 – (6 \times 1460) = 10000 – 8760 = 1240 $$ Since there are remaining bytes (1240 bytes), an additional segment is needed to transmit these bytes. Therefore, the total number of segments required is: $$ \text{Total segments} = \text{Number of full segments} + 1 = 6 + 1 = 7 $$ Now, regarding the significance of the TCP sliding window mechanism: this mechanism is crucial for managing the flow of data between the client and server. It allows the sender to send multiple segments before needing an acknowledgment for the first one, thus optimizing the use of the network’s bandwidth. The sliding window adjusts dynamically based on network conditions, such as congestion and the receiver’s buffer capacity, ensuring that data is transmitted efficiently while minimizing the risk of packet loss. This adaptability is essential for maintaining a reliable connection, especially when dealing with large files or varying network speeds. The sliding window also plays a vital role in error recovery, as it allows TCP to retransmit lost segments without requiring the entire file to be resent, thereby enhancing the overall robustness of the transmission process.

In contrast, full-duplex communication allows devices to send and receive data simultaneously, which is not a feature of traditional hubs, as they only support half-duplex communication. Therefore, the configuration of the hub does not contribute to the inefficiency in this scenario. Additionally, while a limited number of ports can create a bottleneck, the primary issue in this case is the inherent design of the hub itself, which leads to excessive broadcasting and collisions. Lastly, using a high-speed connection that exceeds device capabilities could lead to data loss, but this is less common than the fundamental issues caused by the hub’s broadcasting nature. Understanding the limitations of hubs is crucial for network administrators. In modern networking, switches are preferred over hubs because they operate at Layer 2 of the OSI model and can intelligently forward packets only to the intended recipient, significantly reducing collisions and improving overall network efficiency. This nuanced understanding of how hubs function and their impact on network performance is essential for diagnosing and resolving network issues effectively.

In contrast, full-duplex communication allows devices to send and receive data simultaneously, which is not a feature of traditional hubs, as they only support half-duplex communication. Therefore, the configuration of the hub does not contribute to the inefficiency in this scenario. Additionally, while a limited number of ports can create a bottleneck, the primary issue in this case is the inherent design of the hub itself, which leads to excessive broadcasting and collisions. Lastly, using a high-speed connection that exceeds device capabilities could lead to data loss, but this is less common than the fundamental issues caused by the hub’s broadcasting nature. Understanding the limitations of hubs is crucial for network administrators. In modern networking, switches are preferred over hubs because they operate at Layer 2 of the OSI model and can intelligently forward packets only to the intended recipient, significantly reducing collisions and improving overall network efficiency. This nuanced understanding of how hubs function and their impact on network performance is essential for diagnosing and resolving network issues effectively.

To find the number of bits needed for the subnets, we use the formula \(2^n \geq \text{number of subnets}\), where \(n\) is the number of bits borrowed from the host portion. Since the company requires at least 10 subnets, we calculate: \[ 2^n \geq 10 \implies n \geq 4 \quad (\text{since } 2^4 = 16 \text{ and } 2^3 = 8) \] Thus, we need to borrow 4 bits from the host portion of the address. The original subnet mask of /24 has 8 bits for the host portion (32 total bits – 24 bits for the network). By borrowing 4 bits, the new subnet mask becomes /28 (24 + 4 = 28). Next, we calculate the number of usable IP addresses per subnet. The formula for usable addresses is \(2^h – 2\), where \(h\) is the number of host bits remaining. After borrowing 4 bits, we have: \[ h = 32 – 28 = 4 \] Thus, the number of usable addresses is: \[ 2^4 – 2 = 16 – 2 = 14 \] This means each subnet can accommodate 14 usable IP addresses, which meets the requirement of at least 25 hosts. However, since we need at least 25 hosts per subnet, we realize that the subnet mask of /28 is insufficient. Instead, if we use a subnet mask of /26 (which corresponds to 255.255.255.192), we have: \[ h = 32 – 26 = 6 \] Calculating the usable addresses gives us: \[ 2^6 – 2 = 64 – 2 = 62 \] This allows for 62 usable addresses per subnet, which satisfies the requirement for at least 25 hosts. Therefore, the correct subnet mask to use is 255.255.255.192, providing ample room for the required number of hosts and subnets.

However, the process of establishing a secure connection through HTTPS involves a handshake mechanism that requires additional computational resources. This handshake includes the exchange of cryptographic keys and the establishment of a secure session, which can introduce a slight latency compared to HTTP. Therefore, while HTTPS enhances security, it may result in marginally slower performance due to this overhead. It is also important to note that while HTTPS provides a strong layer of security, it does not eliminate the need for other security measures. For instance, developers must still implement secure coding practices, validate user inputs, and regularly update their software to protect against vulnerabilities. Additionally, transitioning to HTTPS often requires changes to server configurations, such as obtaining and installing SSL/TLS certificates, which is essential for enabling secure connections. In summary, while HTTPS significantly improves the security of data in transit, it may introduce some latency due to the encryption process, and it does not negate the necessity for comprehensive security practices and proper server configuration.

To mitigate these issues, configuring the POP3 client to leave a copy of messages on the server for a specified duration allows users to access their emails from different devices without losing any data. This setting ensures that emails remain on the server for a certain period, allowing for retrieval from other clients or devices. Additionally, it is crucial to implement SSL/TLS encryption to secure the connection between the client and the server, protecting sensitive email data during transmission. While IMAP is indeed a better choice for users who require synchronization across multiple devices, the question specifically asks about the best configuration for POP3. Therefore, the most effective approach is to leave copies of messages on the server, which balances the need for local storage with the ability to access emails from various locations. This configuration not only preserves email integrity but also enhances user experience by preventing data loss.

In IPv6, the number of available addresses in a subnet can be calculated using the formula: $$ \text{Number of Hosts} = 2^{(128 – \text{prefix length})} – 2 $$ The subtraction of 2 accounts for the network and broadcast addresses, which are not usable for hosts. Starting with a /64 prefix, if we were to subnet further, we would reduce the number of bits available for hosts. For example: – A /65 prefix would provide: $$ 2^{(128 – 65)} – 2 = 2^{63} – 2 \approx 9.22 \times 10^{18} \text{ hosts} $$ – A /66 prefix would provide: $$ 2^{(128 – 66)} – 2 = 2^{62} – 2 \approx 4.61 \times 10^{18} \text{ hosts} $$ – A /67 prefix would provide: $$ 2^{(128 – 67)} – 2 = 2^{61} – 2 \approx 2.30 \times 10^{18} \text{ hosts} $$ – A /68 prefix would provide: $$ 2^{(128 – 68)} – 2 = 2^{60} – 2 \approx 1.15 \times 10^{18} \text{ hosts} $$ – A /69 prefix would provide: $$ 2^{(128 – 69)} – 2 = 2^{59} – 2 \approx 5.75 \times 10^{17} \text{ hosts} $$ From this analysis, we can see that even a /66 prefix provides an enormous number of addresses, far exceeding the requirement of 50 hosts. However, best practices suggest that subnetting should be done in a way that allows for future growth while still being efficient. Using a /66 prefix allows for a manageable number of addresses while still providing ample room for expansion. A /67 or /68 prefix would also suffice, but they are less efficient in terms of address space utilization. Therefore, the most appropriate choice that balances the need for current hosts and future scalability is a /66 prefix. This approach not only meets the immediate requirement but also adheres to the best practices of IPv6 addressing, which emphasize efficient use of address space while allowing for future growth.

In contrast, a star topology, while easy to manage and troubleshoot, relies on a central hub or switch. If this central device fails, the entire network goes down, creating a single point of failure. Although it allows for straightforward addition of new devices, it does not provide the redundancy that a mesh topology offers. A bus topology connects all devices to a single central cable, which can lead to performance issues as more devices are added and also presents a risk of failure if the main cable is damaged. Similarly, a ring topology connects devices in a circular fashion, where each device is connected to two others. While it can provide efficient data transmission, if one device fails, it can disrupt the entire network unless a dual ring is implemented, which adds complexity and cost. Given the requirements for high availability and efficient communication among the departments, the mesh topology stands out as the most suitable choice. It not only meets the current needs but also allows for scalability as the company grows, ensuring that the network can adapt to future demands without compromising reliability. This topology is particularly advantageous in environments where uninterrupted service is critical, such as in corporate settings where departments frequently communicate and share data.

The formula for calculating the number of usable IP addresses in a subnet is given by: $$ \text{Usable IPs} = 2^{(32 – \text{Subnet Bits})} – 2 $$ The “-2” accounts for the network and broadcast addresses, which cannot be assigned to hosts. To accommodate at least 30 usable addresses, we need to find the smallest power of 2 that is greater than or equal to 32 (30 + 2). The closest power of 2 is 32, which corresponds to a subnet mask of 255.255.255.192 (or /26), since: $$ 32 = 2^{(32 – 26)} \implies 32 \text{ usable IPs} $$ This means we can create 4 subnets from the original /24 network, as each /26 subnet can accommodate 64 addresses (62 usable). Now, we can calculate the ranges for each subnet. The first subnet (for HR) will start at 192.168.1.0 and end at 192.168.1.63. The usable IP addresses for HR will therefore range from 192.168.1.1 to 192.168.1.62. The subsequent subnets will be: – IT: 192.168.1.64 to 192.168.1.127 – Sales: 192.168.1.128 to 192.168.1.191 – Marketing: 192.168.1.192 to 192.168.1.255 Thus, the correct subnet mask is 255.255.255.192, and the range of IP addresses for the HR department is from 192.168.1.1 to 192.168.1.62. This approach ensures that each department has sufficient IP addresses while adhering to the principles of subnetting.

In addition to the policy, enforcing multi-factor authentication (MFA) is a robust measure that significantly enhances security. MFA requires users to provide two or more verification factors to gain access, making it much more difficult for unauthorized individuals to access accounts, even if they have obtained a user’s password. This dual-layered approach not only deters credential sharing but also protects against various attack vectors, such as phishing and brute-force attacks. While increasing password complexity and conducting training sessions (as suggested in option b) are beneficial practices, they do not directly address the issue of credential sharing. Moreover, limiting access based on tenure (option c) does not effectively mitigate the risk of credential sharing and could lead to operational inefficiencies. Monitoring user activity (option d) is a reactive measure that may help identify incidents after they occur but does not prevent the initial risk. In summary, a comprehensive approach that combines a strict anti-sharing policy with the implementation of MFA is the most effective strategy to protect sensitive data and ensure compliance with security best practices. This approach not only addresses the immediate risk but also fosters a culture of security awareness among employees.