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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.

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

Consider a router that is managing three flows, on which packets of constant size arrive at the following wall clock times: flow A: \(1,3,5,6,8,9,11\) flow B: \(1,4,7,8,9,13,15\) flow C: \(1,2,4,6,7,12\) All three flows share the same outbound link, on which the router can transmit one packet per time unit. Assume that there is an infinite amount of buffer space. (a) Suppose the router implements fair queuing. For each packet, give the wall clock time when it is transmitted by the router. Arrival time ties are to be resolved in order \(\mathrm{A}, \mathrm{B}, \mathrm{C}\). Note that wall clock time \(T=2\) is FQ-clock time \(A_{i}=1.333 .\) (b) Suppose the router implements weighted fair queuing, where flows \(\mathrm{A}\) and \(\mathrm{C}\) are given an equal share of the capacity, and flow B is given twice the capacity of flow A. For each packet, give the wall clock time when it is transmitted.

Suppose host A reaches host B via routers R1 and R2: A-R1-R2-B. Fast retransmit is not used, and A calculates TimeOut as \(2 \times\) EstimatedRTT. Assume that the A-R1 and \(R 2-B\) links have infinite bandwidth; the \(R 1 \longrightarrow R 2\) link, however, introduces a 1 -second-per-packet bandwidth delay for data packets (though not ACKs). Describe a scenario in which the R1-R2 link is not \(100 \%\) utilized, even though A always has data ready to send. Hint: Suppose A's CongestionWindow increases from \(N\) to \(N+1\), where \(N\) is R1's queue size.

Give an argument why the congestion-control problem is better managed at the internet level than the ATM level, at least when only part of the internet is ATM. In an exclusively IP-over-ATM network, is congestion better managed at the cell level or at the TCP level? Why?

Consider the following two causes of a 1 -second network delay (assume ACKs return instantaneously): One intermediate router with a 1 -second outbound per-packet bandwidth delay and no competing traffic One intermediate router with a 100-ms outbound per-packet bandwidth delay and with a steadily replenished (from another source) 10 packets in the queue (a) How might a transport protocol in general distinguish between these two cases? (b) Suppose TCP Vegas sends over the above connections, with an initial CongestionWindow of 3 packets. What will happen to CongestionWindow in each case? Assume BaseRTT \(=1\) second and \(\beta\) is 1 packet per second.

Suppose two hosts \(\mathrm{A}\) and \(\mathrm{B}\) are connected via a router \(\mathrm{R}\). The \(\mathrm{A}-\mathrm{R}\) link has infinite bandwidth; the \(R-B\) link can send one packet per second. \(R\) 's queue is infinite. Load is to be measured as the number of packets per second sent from A to B. Sketch the throughput-versus-load and delay-versus-load graphs, or if a graph cannot be drawn, explain why. Would another way to measure load be more appropriate?

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