Chapter 3

Compare GBN, SR, and TCP (no delayed ACK). Assume that the timeout values for all three protocols are sufficiently long such that 5 consecutive data segments and their corresponding ACKs can be received (if not lost in the channel) by the receiving host (Host B) and the sending host (Host A) respectively. Suppose Host A sends 5 data segments to Host B, and the 2 nd segment (sent from \(\mathrm{A}\) ) is lost. In the end, all 5 data segments have been correctly received by Host B. a. How many segments has Host A sent in total and how many ACKs has Host B sent in total? What are their sequence numbers? Answer this question for all three protocols. b. If the timeout values for all three protocol are much longer than 5 RTT, then which protocol successfully delivers all five data segments in shortest time interval?

Consider sending a large file from a host to another over a TCP connection that has no loss. a. Suppose TCP uses AIMD for its congestion control without slow start. Assuming cwnd increases by 1 MSS every time a batch of ACKs is received and assuming approximately constant round-trip times, how long does it take for cwnd increase from 6 MSS to 12 MSS (assuming no loss events)? b. What is the average throughout (in terms of MSS and RTT) for this connection up through time \(=6\) RTT?

Recall the macroscopic description of TCP throughput. In the period of time from when the connection's rate varies from \(W /(2 \cdot R T T)\) to \(W / R T T\), only one packet is lost (at the very end of the period). a. Show that the loss rate (fraction of packets lost) is equal to $$ L=\text { loss rate }=\frac{1}{\frac{3}{8} W^{2}+\frac{3}{4} W} $$ b. Use the result above to show that if a connection has loss rate \(L\), then its average rate is approximately given by $$ =\frac{1.22 \cdot M S S}{R T T \sqrt{L}} $$

Consider that only a single TCP (Reno) connection uses one \(10 \mathrm{Mbps}\) link which does not buffer any data. Suppose that this link is the only congested link between the sending and receiving hosts. Assume that the TCP sender has a huge file to send to the receiver, and the receiver's receive buffer is much larger than the congestion window. We also make the following assumptions: each TCP segment size is 1,500 bytes; the two-way propagation delay of this connection is \(150 \mathrm{msec}\); and this TCP connection is always in congestion avoidance phase, that is, ignore slow start. a. What is the maximum window size (in segments) that this TCP connection can achieve? b. What is the average window size (in segments) and average throughput (in bps) of this TCP connection? c. How long would it take for this TCP connection to reach its maximum window again after recovering from a packet loss?

Consider a simplified TCP's AIMD algorithm where the congestion window size is measured in number of segments, not in bytes. In additive increase, the congestion window size increases by one segment in each RTT. In multiplicative decrease, the congestion window size decreases by half (if the result is not an integer, round down to the nearest integer). Suppose that two TCP connections, \(C_{1}\) and \(C_{2}\), share a single congested link of speed 30 segments per second. Assume that both \(\mathrm{C}_{1}\) and \(\mathrm{C}_{2}\) are in the congestion avoidance phase. Connection \(\mathrm{C}_{1}\) 's RTT is \(50 \mathrm{msec}\) and connection \(\mathrm{C}_{2}\) 's RTT is \(100 \mathrm{msec}\). Assume that when the data rate in the link exceeds the link's speed, all TCP connections experience data segment loss. a. If both \(\mathrm{C}_{1}\) and \(\mathrm{C}_{2}\) at time \(\mathrm{t}_{0}\) have a congestion window of 10 segments, what are their congestion window sizes after 1000 msec? b. In the long run, will these two connections get the same share of the bandwidth of the congested link? Explain.

Consider a modification to TCP's congestion control algorithm. Instead of additive increase, we can use multiplicative increase. A TCP sender increases its window size by a small positive constant \(a(0

In our discussion of TCP congestion control in Section 3.7, we implicitly assumed that the TCP sender always had data to send. Consider now the case that the TCP sender sends a large amount of data and then goes idle (since it has no more data to send) at \(t_{1}\). TCP remains idle for a relatively long period of time and then wants to send more data at \(t_{2}\). What are the advantages and disadvantages of having TCP use the cwnd and ssthresh values from \(t_{1}\) when starting to send data at \(t_{2}\) ? What alternative would you recommend? Why?

In this problem we investigate whether either UDP or TCP provides a degree of end-point authentication. a. Consider a server that receives a request within a UDP packet and responds to that request within a UDP packet (for example, as done by a DNS server). If a client with IP address \(\mathrm{X}\) spoofs its address with address Y, where will the server send its response? b. Suppose a server receives a SYN with IP source address Y, and after responding with a SYNACK, receives an ACK with IP source address Y with the correct acknowledgment number. Assuming the server chooses a random initial sequence number and there is no "man-in-the-middle," can the server be certain that the client is indeed at \(Y\) (and not at some other address \(\mathrm{X}\) that is spoofing \(\mathrm{Y})\) ?

In this problem, we consider the delay introduced by the TCP slow-start phase. Consider a client and a Web server directly connected by one link of rate \(R\). Suppose the client wants to retrieve an object whose size is exactly equal to \(15 S\), where \(S\) is the maximum segment size (MSS). Denote the round-trip time between client and server as RTT (assumed to be constant). Ignoring protocol headers, determine the time to retrieve the object (including TCP connection establishment) when a. \(4 S / R>S / R+R T T>2 S / R\) b. \(S / R+R T T>4 S / R\) c. \(S / R>R T T\).