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Suppose a TCP Vegas connection measures the RTT of its first packet and sets BaseRT to that, but then a network link failure occurs and all subsequent traffic is routed via an alternative path with twice the RTT. How will TCP Vegas respond? What will happen to the value of CongestionWindow? Assume no actual timeouts occur, and that \(\beta\) is much smaller than the initial ExpectedRate.

Short Answer

Expert verified
TCP Vegas will reduce the CongestionWindow as it perceives network congestion due to the doubled RTT.

Step by step solution

01

Understand RTT and BaseRT

RTT (Round Trip Time) is the time it takes for a packet to travel from the source to the destination and back. BaseRT is the initial RTT measured by TCP Vegas.
02

Identify the Change in RTT

After a network failure, RTT doubles because packets now travel via an alternative path that takes twice as long. If the initial RTT was RTT_initial, the new RTT becomes 2 * RTT_initial.
03

Expected Rate Calculation

TCP Vegas calculates the ExpectedRate as the initial CongestionWindow (CWND) divided by BaseRT. Since BaseRT does not change, ExpectedRate remains the same.
04

Calculate Actual Rate

The ActualRate is calculated using the current RTT. Hence, ActualRate = CWND / (2 * RTT_initial).
05

Compare ExpectedRate and ActualRate

With the increased RTT, the ActualRate will be half the ExpectedRate. This causes TCP Vegas to interpret that the network is congested.
06

TCP Vegas Congestion Control

TCP Vegas reacts to perceived congestion by adjusting the CWND. Since \(\beta\) is much smaller than the initial ExpectedRate, TCP Vegas will reduce the CWND until ActualRate matches closely with the ExpectedRate.

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Key Concepts

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

Round Trip Time (RTT)
Round Trip Time (RTT) is a critical concept in TCP Vegas and network communication. It represents the total time a data packet takes to travel from the source to the destination and back again. Measuring RTT involves sending a packet to the destination and waiting for an acknowledgment.

In the provided exercise, RTT_initial is the first measured RTT before the failure occurs. Understanding RTT helps in managing data flow, ensuring smooth and efficient communication.

For example, if RTT is 100 milliseconds, it means a data packet takes 100ms to go to the destination and back.
BaseRT
In TCP Vegas, BaseRT is the first RTT measurement taken when the connection is established. It sets a baseline for calculations.

BaseRT is crucial because it does not change unless the route or the infrastructure changes significantly. In our exercise, BaseRT is recorded before the network failure and remains constant even after the RTT doubles due to the new path taken.

This fixed baseline allows TCP Vegas to detect changes in network performance and make adjustments accordingly.
CongestionWindow
The CongestionWindow (CWND) is a TCP control mechanism that limits the amount of data the sender can transmit without receiving an acknowledgment.

In TCP Vegas, CWND is dynamically adjusted based on network conditions to avoid congestion.

For instance, if TCP Vegas detects longer RTTs and interprets it as congestion, it will reduce the CWND size to minimize packet loss and alleviate congestion. Conversely, if the network is clear, TCP Vegas might increase CWND to utilize the available bandwidth better.
Expected Rate
The Expected Rate in TCP Vegas is computed by dividing the initial CWND by BaseRT. It reflects the data rate the sender aims to maintain under ideal conditions.

Since BaseRT doesn’t change in our exercise, the Expected Rate remains constant.

For instance, if the initial CWND is 10 packets and BaseRT is 100ms, the Expected Rate would be 0.1 packets/millisecond. This value serves as a benchmark to evaluate the current network performance.
Actual Rate
The Actual Rate in TCP Vegas is calculated using the current RTT, which can change due to varying network conditions.

It’s determined by dividing the current CWND by the current RTT. In our exercise, after the route change, the RTT doubles. Consequently, the Actual Rate becomes CWND divided by twice the initial RTT, halving the rate if the CWND remains unchanged.

Monitoring the Actual Rate helps TCP Vegas in adjusting CWND to match the Expected Rate, ensuring stable network performance.
Congestion Control
Congestion control in TCP Vegas aims to prevent network congestion by adjusting the data transmit rate based on RTT measurements.

In our exercise, after the network failure, the doubled RTT causes TCP Vegas to perceive congestion, as the Actual Rate drops below the Expected Rate.

TCP Vegas responds by reducing the CWND. It continues adjusting until the Actual Rate aligns closely with the Expected Rate, thereby stabilizing the network performance.

This proactive approach helps in maintaining optimal throughput while avoiding packet loss and excessive delays.

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Most popular questions from this chapter

Discuss the relative advantages and disadvantages of marking a packet (as in the DECbit mechanism) versus dropping a packet (as in RED gateways).

Suppose a router's drop policy is to drop the highest-cost packet whenever queues are full, where it defines the "cost" of a packet to be the product of its size by the time remaining that it will spend in the queue. (Note that in calculating cost it is equivalent to use the sum of the sizes of the earlier packets in lieu of remaining time.) (a) What advantages and disadvantages might such a policy offer, compared to tail drop? (b) Give an example of a sequence of queued packets for which dropping the highest-cost packet differs from dropping the largest packet. (c) Give an example where two packets exchange their relative cost ranks as time progresses.

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.

Assume that TCP implements an extension that allows window sizes much larger than \(64 \mathrm{~KB}\). Suppose that you are using this extended TCP over a 1-Gbps link with a latency of \(100 \mathrm{~ms}\) to transfer a \(10-\mathrm{MB}\) file, and the TCP receive window is \(1 \mathrm{MB}\). If TCP sends 1-KB packets (assuming no congestion and no lost packets): (a) How many RTTs does it take until slow start opens the send window to \(1 \mathrm{MB}\) ? (b) How many RTTs does it take to send the file? (c) If the time to send the file is given by the number of required RTTs multiplied by the link latency, what is the effective throughput for the transfer? What percentage of the link bandwidth is utilized?

The text states that additive increase is a necessary condition for a congestioncontrol mechanism to be stable. Outline a specific instability that might arise if all increases were exponential; that is, if TCP continued to use "slow" start after CongestionWindow increased beyond CongestionThreshold.

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