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Consider a simple congestion-control algorithm that uses linear increase and multiplicative decrease but not slow start, that works in units of packets rather than bytes, and that starts each connection with a congestion window equal to one packet. Give a detailed sketch of this algorithm. Assume the delay is latency only, and that when a group of packets is sent, only a single ACK is returned. Plot the congestion window as a function of round-trip times for the situation in which the following packets are lost: \(9,25,30,38\), and 50 . For simplicity, assume a perfect timeout mechanism that detects a lost packet exactly 1 RTT after it is transmitted.

Short Answer

Expert verified
Plot the congestion window by increasing CWND linearly each RTT until loss, then halve CWND at losses of packets 9, 25, 30, 38, and 50.

Step by step solution

01

Initialize the Congestion Window

At the start of the connection, set the congestion window (CWND) to 1 packet.
02

Linear Increase

For each round-trip time (RTT) without packet loss, increase the CWND by one packet. This represents the linear increase part of the algorithm.
03

Handle Packet Loss

When packet loss is detected (in this case, packets 9, 25, 30, 38, and 50), immediately reduce the CWND by half (multiplicative decrease). Assume the CWND after reduction must be at least 1 packet.
04

Plot Congestion Window

Create a plot of the congestion window as a function of RTTs. Mark the RTTs where packet loss occurs and adjust the CWND accordingly.

Key Concepts

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

Linear Increase
The concept of linear increase in congestion control refers to the gradual growth of the congestion window (CWND) over time. Here's how it works in simple terms. When a data connection starts, it begins cautiously, usually with a CWND set to one packet. This is to avoid overwhelming the network. For each Round Trip Time (RTT) where no packet loss occurs, the CWND is increased by one packet. This slow and steady increase helps to probe the available bandwidth without causing sudden spikes in network traffic. Imagine a driver slowly accelerating on an empty road, going one mile per hour faster every minute. Similarly, linear increase helps data flow smoothly, accommodating more data packets as the network proves that it can handle them.
Multiplicative Decrease
Multiplicative decrease is the counterbalance to linear increase in congestion control. When packet loss is detected, it indicates that the network is congested and cannot handle the current load. In response, the algorithm cuts the congestion window (CWND) by half immediately. This aggressive reduction helps to quickly reduce the load on the network to alleviate congestion. For example, if the CWND was 16 packets just before packet loss, it will be reduced to 8 packets right after the loss is detected. This approach resembles a driver hitting the brakes hard when noticing a traffic jam ahead; the idea is to quickly scale back to prevent further issues.
Congestion Window
The congestion window (CWND) is a crucial parameter in congestion control algorithms. It signifies the maximum number of data packets the sender can have unacknowledged at any given time. Think of it as a measuring cup for data. A larger CWND means more data can be sent before waiting for an acknowledgment. Conversely, a smaller CWND slows down the data flow. Ideally, the CWND adjusts dynamically based on the network conditions. Starting with a small CWND helps ensure that the network is not overwhelmed from the get-go. As the network proves it can handle more traffic (i.e., no packet loss), the CWND is increased. When packet loss occurs, the CWND is reduced to throttle down the data flow.
RTT (Round Trip Time)
Round Trip Time (RTT) is the duration it takes for a signal to go from the sender to the receiver and back again. In the context of congestion control, RTT is a vital measure. Suppose it takes 100 milliseconds to send a packet and receive an acknowledgment. This 100 milliseconds is the RTT. The congestion control algorithm adjusts the CWND based on the RTT. For instance, in linear increase, the CWND is incrementally increased every RTT without packet loss. On the flip side, if a packet loss is detected, the CWND is reduced upon the next RTT completion. Monitoring RTT helps the algorithm to react appropriately to the current network conditions.
Packet Loss Detection
Packet loss detection is a key component in congestion control algorithms. When a data packet is sent but not acknowledged within a certain timeframe (equal to one RTT for our algorithm), it indicates packet loss. In our specific exercise, this mechanism kicks in perfectly after one RTT. Packet loss often signals network congestion or errors, prompting the algorithm to reduce the CWND multiplicatively (by half) to alleviate the congestion. Detecting packet loss in a timely and accurate manner is essential for maintaining robust and efficient network performance. It ensures that the sender reduces its load on the network quickly, helping to restore normal data flow. This timely detection aids in not just managing but also preventing prolonged congestive states on the network.

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Most popular questions from this chapter

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?

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?

Consider a router that is managing three flows, on which packets of constant size arrive at the following wall clock times: flow A: \(1,2,4,6,7,9,10\) flow B: \(2,6,8,11,12,15\) flow C: \(1,2,3,5,6,7,8\) 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 \(A_{i}=1.5\). (b) Suppose the router implements weighted fair queuing, where flows \(\mathrm{A}\) and \(\mathrm{B}\) are given an equal share of the capacity, and flow \(\mathrm{C}\) is given twice the capacity of flow A. For each packet, give the wall clock time when it is transmitted.

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.

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.

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