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Suppose that the UDP receiver computes the Internet checksum for the received UDP segment and finds that it matches the value carried in the checksum field. Can the receiver be absolutely certain that no bit errors have occurred? Explain.

Short Answer

Expert verified
No, the receiver cannot be absolutely certain; some errors might go undetected.

Step by step solution

01

Understanding the Internet Checksum

The Internet checksum is a simple method used to detect errors in transmitted data. It works by performing a checksum computation over the data, and the result is sent along with the data. Upon receiving the data, the receiver computes the checksum again and compares it with the received checksum value.
02

Evaluating the Checksum Matching

If the receiver finds that the computed checksum matches the checksum value carried in the segment, it typically means that the data has not been altered during transmission, suggesting no bit errors occurred.
03

Limitations of the Internet Checksum

However, one must understand that the Internet checksum can detect many errors, such as single-bit and some multi-bit errors, but it is not foolproof. There are certain patterns of bit errors which might not be detected by this checksum method due to its limited capability.
04

Conclusion on Absolute Certainty

Given the limitations, if the checksum matches, the receiver can have confidence that the data is likely error-free, but not absolute certainty. There are possible cases where multiple bit errors might cancel each other out, leading to the same checksum value by coincidence.

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Key Concepts

These are the key concepts you need to understand to accurately answer the question.

UDP Protocol
The UDP protocol stands for User Datagram Protocol. It is a part of the Internet Protocol suite that allows applications to send messages (datagrams) across the network. UDP is known for being a connectionless protocol, which means that it doesn’t establish a stable connection between the sender and receiver before transmitting data. This characteristic makes it lightweight and fast compared to TCP (Transmission Control Protocol).

In UDP communication, several key aspects standout:
  • No Acknowledgments: UDP does not acknowledge the receipt of packets. Once data is sent, there's no confirmation that it was received, which can lead to loss of data without notice.
  • No Guaranteed Delivery: Since it doesn’t establish a connection, UDP can’t ensure data delivery, order, or protection from duplication.
  • Use Cases: Due to its speed, UDP is ideal in situations where timing is crucial, such as live broadcasts, online gaming, and VoIP services, where a small amount of lost data is preferable over delays caused by packet recovery processes.
Error Detection
Error detection is a crucial element in digital communication, ensuring the accuracy and integrity of data transferred over networks. One common method of error detection is through the use of a checksum.

A checksum is a computed value that depends on the contents of the data. The sender calculates a checksum value and sends it along with the data. The receiver then performs a similar calculation to verify the checksum:
  • Commonality: The Internet checksum is traditionally used in many protocols, including UDP, to detect errors.
  • Limitations: Simple checksum methods like the Internet checksum may fail to detect certain types of errors, particularly some multi-bit errors, because they might produce the same checksum result even if errors are present.
  • Improvements: For more robust error detection, other methods like Cyclic Redundancy Check (CRC) offer improved accuracy over simple checksums by using more complex polynomial calculations.
Understanding these methods helps in balancing the trade-offs between computational complexity and error detection accuracy.
Network Reliability
Network reliability refers to the network's ability to consistently perform its intended functions. It is an essential quality, particularly in data communications, affecting user satisfaction and application performance. Various factors contribute to network reliability:
  • Error Detection and Correction: Methods like checksums, CRC, and error-correcting codes help maintain reliability by ensuring that data errors are detected and, when possible, corrected.
  • Redundancy: Using multiple pathways for data travel can safeguard against path failures, benefiting protocols like TCP to automatically switch paths if one fails.
  • Scalability: Reliable networks can efficiently handle growing user numbers and data volume without a decline in performance.
Implementing effective error detection strategies enhances network reliability by minimizing erroneous data transmission, even over unreliable protocols like UDP.
Data Transmission
Data transmission is the process of sending digital information from a source to a destination over a communication channel. The effectiveness of data transmission depends on several critical components:
  • Protocols: Communication protocols like UDP and TCP set the rules for data transmission. While UDP offers faster data sending through its connectionless approach, TCP provides a more reliable transfer of data.
  • Transmission Medium: This can range from wired cables to wireless systems. Each medium has its own speed, reliability, and interference characteristics impacting transmission.
  • Compression and Encoding: Techniques used to reduce data size for transmission efficiency and to ensure data security and integrity during the transfer.
Successful data transmission requires a combination of well-designed protocols, error detection and correction mechanisms, and appropriate transmission mediums to meet the needs of various applications. An understanding of these components helps in aligning data transmission strategies with specific network requirements.

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Most popular questions from this chapter

Consider transferring an enormous file of \(L\) bytes from Host A to Host B. Assume an MSS of 536 bytes. a. What is the maximum value of \(L\) such that TCP sequence numbers are not exhausted? Recall that the TCP sequence number field has 4 bytes. b. For the \(L\) you obtain in (a), find how long it takes to transmit the file. Assume that a total of 66 bytes of transport, network, and data-link header are added to each segment before the resulting packet is sent out over a \(155 \mathrm{Mbps}\) link. Ignore flow control and congestion control so A can pump

Consider a simplified TCP's AIMD algorithm where the congestion window size is measured in number of segments, not in bytes. In additive increase, the congestion window size increases by one segment in each RTT. In multiplicative decrease, the congestion window size decreases by half (if the result is not an integer, round down to the nearest integer). Suppose that two TCP connections, \(C_{1}\) and \(C_{2}\), share a single congested link of speed 30 segments per second. Assume that both \(\mathrm{C}_{1}\) and \(\mathrm{C}_{2}\) are in the congestion avoidance phase. Connection \(\mathrm{C}_{1}\) 's RTT is \(50 \mathrm{msec}\) and connection \(\mathrm{C}_{2}\) 's RTT is \(100 \mathrm{msec}\). Assume that when the data rate in the link exceeds the link's speed, all TCP connections experience data segment loss. a. If both \(\mathrm{C}_{1}\) and \(\mathrm{C}_{2}\) at time \(\mathrm{t}_{0}\) have a congestion window of 10 segments, what are their congestion window sizes after 1000 msec? b. In the long run, will these two connections get the same share of the bandwidth of the congested link? Explain.

Host \(\mathrm{A}\) and \(\mathrm{B}\) are directly connected with a \(100 \mathrm{Mbps}\) link. There is one TCP connection between the two hosts, and Host \(\mathrm{A}\) is sending to Host \(\mathrm{B}\) an enormous file over this connection. Host A can send its application data into its TCP socket at a rate as high as \(120 \mathrm{Mbps}\) but Host B can read out of its TCP receive buffer at a maximum rate of \(50 \mathrm{Mbps}\). Describe the effect of TCP flow control.

We have said that an application may choose UDP for a transport protocol because UDP offers finer application control (than TCP) of what data is sent in a segment and when. a. Why does an application have more control of what data is sent in a segment? b. Why does an application have more control on when the segment is sent?

Suppose Host A sends two TCP segments back to back to Host B over a TCP connection. The first segment has sequence number 90 ; the second has sequence number 110 . a. How much data is in the first segment? b. Suppose that the first segment is lost but the second segment arrives at B. In the acknowledgment that Host B sends to Host A, what will be the acknowledgment number?

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