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Chapter 6: Problem 29

Suppose you are downloading a large file over a 3-KBps phone link. Your software displays an average-bytes-per-second counter. How will TCP congestion control and occasional packet losses cause this counter to fluctuate? Assume that only a third, say, of the total RTT is spent on the phone link.

Short Answer

Expert verified
TCP congestion control and packet losses will cause fluctuations in the average-bytes-per-second counter due to adjustments in the congestion window size.

Step by step solution

01

Understand the Role of TCP Congestion Control

TCP congestion control helps maintain smooth data transmission over a network. It adjusts the data transfer rate based on network conditions, particularly when packet losses occur. This mechanism prevents network congestion.
02

Identify the Download Speed

The download speed is given as 3 KBps (Kilobytes per second). This is the maximum speed at which data can be downloaded when there is no congestion or packet loss.
03

TCP Congestion Window

The TCP congestion window controls the number of packets that can be sent before an acknowledgment is received. Initially, it starts small and increases exponentially during the slow start phase until it encounters packet loss.
04

Impact of Packet Loss on Congestion Window

When packet loss occurs, TCP treats it as a sign of network congestion and reduces the congestion window size. This reduces the data transfer rate temporarily.
05

Analyze RTT Contribution

RTT (Round-Trip Time) significantly affects the data transfer rate. In this scenario, only a third of the RTT is spent on the phone link. However, the total download speed will still be impacted by the overall RTT including other network segments.
06

Counter Fluctuation Explanation

Due to TCP congestion control and packet losses, the average-bytes-per-second counter will not remain steady. It will fluctuate, reflecting the changes in the congestion window size and the occurrences of packet losses.

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Key Concepts

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

Network Congestion
Network congestion happens when too much data is sent through a network, overwhelming its capacity. This results in slower data transfer rates and can cause packet loss, where data packets are dropped because the network can't handle them.

TCP congestion control is designed to prevent these issues. When a network starts to become congested, TCP adjusts the data transfer rate. It begins with a slow start, increasing the data rate until packet loss is detected. Then, it slows down the transmission to relieve congestion.

The cycle of increasing and decreasing the data rate because of network congestion is why you see fluctuations in download speeds when downloading a large file.
Packet Loss
Packet loss occurs when one or more data packets traveling across a network fail to reach their destination. This can happen because of network congestion, faulty hardware, or other issues.

When TCP detects packet loss, usually via missing acknowledgments, it interprets this as a sign of congestion and reduces the congestion window size. This reduction lowers the data transfer rate temporarily to help manage the congestion.

As you download a large file, occasional packet losses will cause the average-bytes-per-second counter to drop temporarily. When packet loss subsides, the counter increases again as TCP ramps up the data rate.
Round-Trip Time (RTT)
Round-Trip Time (RTT) is the time it takes for a data packet to travel from the sender to the receiver and back again. RTT includes all segments of the network, such as phone links and other internet paths.

In this scenario, only a third of the total RTT is spent on the phone link, but the overall RTT still influences the download speed. Higher RTT means each packet takes longer to acknowledge, slowing down the effective data transfer rate.

Because RTT affects how quickly the TCP congestion window can increase, it plays a part in the fluctuations of the average-bytes-per-second counter. A larger RTT means slower recovery from packet loss and more noticeable dips in data transfer rates.

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Most popular questions from this chapter

Suppose TCP is used over a lossy link that loses on average one segment in four. Assume the bandwidth \(x\) delay window size is considerably larger than four segments. (a) What happens when we start a connection? Do we ever get to the linearincrease phase of congestion avoidance? (b) Without using an explicit feedback mechanism from the routers, would TCP have any way to distinguish such link losses from congestion losses, at least over the short term? (c) Suppose TCP senders did reliably get explicit congestion indications from routers. Assuming links as above were common, would it be feasible to support window sizes much larger than four segments? What would TCP have to do?

In fair queuing, the value \(F_{i}\) was interpreted as a timestamp: the time when the \(i\) th packet would finish transmitting. Give an interpretation of \(F_{i}\) for weighted fair queuing, and also give a formula for it in terms of \(F_{i-1}\), arrival time \(A_{i}\), packet size \(P_{i}\), and weight \(w\) assigned to the flow.

It is possible to define flows on either a host-to-host basis or a process-to- process basis. (a) Discuss the implications of each approach to application programs. (b) IPv6 includes a FlowLabel field, for supplying hints to routers about individual flows. The originating host is to put here a pseudorandom hash of all the other fields serving to identify the flow; the router can thus use any subset of these bits as a hash value for fast lookup of the flow. What exactly should the FlowLabel be based on, for each of these two approaches?

Two users, one using Telnet and one sending files with FTP, both send their traffic out via router \(R\). The outbound link from \(R\) is slow enough that both users keep packets in R's queue at all times. Discuss the relative performance seen by the Telnet user if \(\mathrm{R}\) 's queuing policy for these two flows is (a) round-robin service (b) fair queuing (c) modified fair queuing, where we count the cost only of data bytes, and not IP or TCP headers Consider outbound traffic only. Assume Telnet packets have 1 byte of data, FTP packets have 512 bytes of data, and all packets have 40 bytes of headers.

Defeating TCP congestion-control mechanisms usually requires the explicit cooperation of the sender. However, consider the receiving end of a large data transfer using a TCP modified to ACK packets that have not yet arrived. It may do this either because not all of the data is necessary or because data that is lost can be recovered in a separate transfer later. What effect does this receiver behavior have on the congestion-control properties of the session? Can you devise a way to modify TCP to avoid the possibility of senders being taken advantage of in this manner?
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