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Chapter 5: Problem 11

Suppose a host wants to establish the reliability of a link by sending packets and measuring the percentage that are received; routers, for example, do this. Explain the difficulty of doing this over a TCP connection.

Short Answer

Expert verified
TCP masks packet loss through retransmissions and acknowledgments, hiding true link reliability. Using simpler protocols like UDP can better measure packet loss.

Step by step solution

01

Understanding TCP

Transmission Control Protocol (TCP) is a reliable, connection-oriented protocol that ensures all packets sent are received and acknowledged. Due to its built-in error-checking and retransmission mechanisms, TCP inherently guarantees that all packets will eventually be received, making it difficult to measure lost packets directly with TCP.
02

Packet Loss Concealment

TCP masks packet loss by retransmitting lost packets until they are successfully received. As a result, while TCP provides overall reliable transmission, it does not reveal how many packets were originally lost and retransmitted, hiding the actual link reliability.
03

Acknowledgment and Timing

TCP relies on acknowledgments (ACKs) to confirm packet receipt. Timers and sequence numbers are used to detect lost packets and trigger retransmissions. This means the sender does not differentiate between originally received packets and those that were retransmitted, making direct measurement of packet loss difficult.
04

Impact of Retransmissions

Frequent retransmissions due to packet loss can significantly affect overall transmission time and efficiency. However, TCP's goal is to deliver all packets accurately, not to provide insight into the transmission quality of each original packet, obfuscating the true reliability of the link.
05

Alternative Methods

To measure the link reliability accurately, it would be more effective to use a simpler protocol like UDP (User Datagram Protocol) that does not guarantee delivery and, therefore, allows direct observation of packet loss.

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Key Concepts

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

Packet Loss Measurement
Measuring packet loss over a TCP connection can be tricky. TCP has built-in mechanisms to ensure all packets arrive reliably, potentially leading to misleading results about the actual packet loss.
When packets are lost, TCP automatically retransmits them, so the receiving end gets the complete data without ever knowing packets were lost initially.
This makes it hard to measure the real packet loss rate as TCP hides the loss through its retransmission processes.
Retransmission Mechanics
TCP achieves reliability through retransmission mechanics. Whenever TCP detects a lost packet, it retransmits it to ensure the data arrives correctly at the destination.
Here鈥檚 a breakdown of how this works:
  • TCP uses sequence numbers: Every packet is assigned a sequence number. This helps in tracking which packet has been lost or requires retransmission.
  • Acknowledgments (ACKs): The receiving end sends back ACKs for packets received. If the sender doesn鈥檛 receive an ACK in time, it assumes the packet was lost.
  • Timers: TCP uses timers to detect lost packets. If an ACK isn鈥檛 received within a certain period, the sender retransmits the packet.
This robust system ensures data reliability but masks the actual packet loss, hiding the real picture of network reliability.
TCP versus UDP
When it comes to understanding link reliability and packet loss, it鈥檚 vital to compare TCP with UDP (User Datagram Protocol). TCP and UDP serve different purposes in network communication:
  • TCP (Transmission Control Protocol): Ensures complete and reliable data transfer. It uses mechanisms like acknowledgment, retransmissions, and sequence numbers to achieve this, but it hides packet loss because it鈥檚 focused on ensuring all data arrives successfully.
  • UDP (User Datagram Protocol): Does not guarantee delivery, order, or error-checking. It sends packets without waiting for acknowledgments or retransmitting lost packets. This makes it simpler and faster, but also unreliable for data integrity. However, it allows for direct measurement of packet loss since lost packets are not retransmitted.
If accurately measuring packet loss is the goal, UDP is the better choice, as it does not hide the loss like TCP does through its reliability mechanisms.

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Most popular questions from this chapter

Suppose a client \(C\) repeatedly connects via TCP to a given port on a server \(S\), and that each time it is \(\mathrm{C}\) that initiates the close. (a) How many TCP connections a second can C make here before it ties up all its available ports in TIME_WAIT state? Assume client ephemeral ports are in the range 1024-5119, and that TIME_WAIT lasts 60 seconds. (b) Berkeley-derived TCP implementations typically allow a socket in TIME WAIT state to be reopened before TIME_WAIT expires, if the highest sequence number used by the old incarnation of the connection is less than the ISN used by the new incarnation. This solves the problem of old data accepted as new; however, TIME_WAIT also serves the purpose of handling late final FINs. What would such an implementation have to do to address this and still achieve strict compliance with the TCP requirement that a FIN sent anytime before or during a connection's TIME_WAIT receive the same response?

The Jacobson/Karels algorithm sets TimeOut to be 4 mean deviations above the mean. Assume that individual packet round-trip times follow a statistical normal distribution, for which 4 mean deviations are \(\pi\) standard deviations. Using statistical tables, for example, what is the probability that a packet will take more than TimeOut time to arrive?

Write a test program that uses the socket interface to send messages between a pair of Unix workstations connected by some LAN (e.g., Ethernet, ATM, or FDDI). Use this test program to perform the following experiments. (a) Measure the round-trip latency of TCP and UDP for different message sizes (e.g., 1 byte, 100 bytes, 200 bytes, ..., 1000 bytes). (b) Measure the throughput of TCP and UDP for 1-KB, 2-KB, 3-KB, ...,32-KB messages. Plot the measured throughput as a function of message size. (c) Measure the throughput of TCP by sending \(1 \mathrm{MB}\) of data from one host to another. Do this in a loop that sends a message of some size, for example, 1024 iterations of a loop that sends 1-KB messages. Repeat the experiment with different message sizes and plot the results.

Consult Request for Comments 793 to find out how TCP is supposed to respond if a FIN or an RST arrives with a sequence number other than NextByteExpected. Consider both when the sequence number is within the receive window and when it is not.

Explain why TIME_WAIT is a somewhat more serious problem if the server initiates the close than if the client does. Describe a situation in which this might reasonably happen.
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