Description
Problems
P1. Suppose Client A initiates a Telnet session with Server S. At about the same
time, Client B also initiates a Telnet session with Server S. Provide possible
source and destination port numbers for
a. The segments sent from A to S.
b. The segments sent from B to S.
c. The segments sent from S to A.
d. The segments sent from S to B.
e. If A and B are different hosts, is it possible that the source port number in
the segments from A to S is the same as that from B to S?
f. How about if they are the same host?
P2. Consider Figure 3.5. What are the source and destination port values in the segments flowing from the server back to the clients’ processes? What are the IP
addresses in the network-layer datagrams carrying the transport-layer segments?
P3. UDP and TCP use 1s complement for their checksums. Suppose you have the
following three 8-bit bytes: 01010011, 01100110, 01110100. What is the 1s
complement of the sum of these 8-bit bytes? (Note that although UDP and
TCP use 16-bit words in computing the checksum, for this problem you are
being asked to consider 8-bit sums.) Show all work. Why is it that UDP takes
the 1s complement of the sum; that is, why not just use the sum? With the 1s
complement scheme, how does the receiver detect errors? Is it possible that a
1-bit error will go undetected? How about a 2-bit error?
P4. a. Suppose you have the following 2 bytes: 01011100 and 01100101. What is
the 1s complement of the sum of these 2 bytes?
b. Suppose you have the following 2 bytes: 11011010 and 01100101. What is
the 1s complement of the sum of these 2 bytes?
c. For the bytes in part (a), give an example where one bit is flipped in each
of the 2 bytes and yet the 1s complement doesn’t change.
4 ! RTTFE + RTTBE + processing time.
PROBLEMS 289
P5. Suppose that the UDP receiver computes the Internet checksum for the received
UDP segment and finds that it matches the value carried in the checksum field.
Can the receiver be absolutely certain that no bit errors have occurred? Explain.
P6. Consider our motivation for correcting protocol rdt2.1. Show that the
receiver, shown in Figure 3.57, when operating with the sender shown in Figure 3.11, can lead the sender and receiver to enter into a deadlock state, where
each is waiting for an event that will never occur.
P7. In protocol rdt3.0, the ACK packets flowing from the receiver to the
sender do not have sequence numbers (although they do have an ACK field
that contains the sequence number of the packet they are acknowledging).
Why is it that our ACK packets do not require sequence numbers?
P8. Draw the FSM for the receiver side of protocol rdt3.0.
P9. Give a trace of the operation of protocol rdt3.0 when data packets and
acknowledgment packets are garbled. Your trace should be similar to that
used in Figure 3.16.
P10. Consider a channel that can lose packets but has a maximum delay that is
known. Modify protocol rdt2.1 to include sender timeout and retransmit.
Informally argue why your protocol can communicate correctly over this
channel.
Wait for
0 from
below
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt)||
has_seq0(rcvpkt)))
compute chksum
make_pkt(sndpkt,NAK,chksum)
udt_send(sndpkt) rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt)||
has_seq1(rcvpkt)))
compute chksum
make_pkt(sndpkt,NAK,chksum)
udt_send(sndpkt) rdt_rvc(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
compute chksum
make_pkt(sendpkt,ACK,chksum)
udt_send(sndpkt)
rdt_rvc(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq0(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
compute chksum
make_pkt(sendpkt,ACK,chksum)
udt_send(sndpkt)
Wait for
1 from
below
Figure 3.57 ! An incorrect receiver for protocol rdt 2.1
290 CHAPTER 3 • TRANSPORT LAYER
P11. Consider the rdt2.2 receiver in Figure 3.14, and the creation of a new packet
in the self-transition (i.e., the transition from the state back to itself) in the Waitfor-0-from-below and the Wait-for-1-from-below states: sndpkt=make_
pkt(ACK,0,checksum) and sndpkt=make_pkt(ACK,0,
checksum). Would the protocol work correctly if this action were removed
from the self-transition in the Wait-for-1-from-below state? Justify your
answer. What if this event were removed from the self-transition in the Waitfor-0-from-below state? [Hint: In this latter case, consider what would happen if the first sender-to-receiver packet were corrupted.]
P12. The sender side of rdt3.0 simply ignores (that is, takes no action on)
all received packets that are either in error or have the wrong value in the
acknum field of an acknowledgment packet. Suppose that in such circumstances, rdt3.0 were simply to retransmit the current data packet. Would
the protocol still work? (Hint: Consider what would happen if there were
only bit errors; there are no packet losses but premature timeouts can occur.
Consider how many times the nth packet is sent, in the limit as n
approaches infinity.)
P13. Consider the rdt 3.0 protocol. Draw a diagram showing that if the network connection between the sender and receiver can reorder messages
(that is, that two messages propagating in the medium between the sender
and receiver can be reordered), then the alternating-bit protocol will not
work correctly (make sure you clearly identify the sense in which it will
not work correctly). Your diagram should have the sender on the left and
the receiver on the right, with the time axis running down the page, showing data (D) and acknowledgment (A) message exchange. Make sure you
indicate the sequence number associated with any data or acknowledgment
segment.
P14. Consider a reliable data transfer protocol that uses only negative acknowledgments. Suppose the sender sends data only infrequently. Would a NAK-only
protocol be preferable to a protocol that uses ACKs? Why? Now suppose the
sender has a lot of data to send and the end-to-end connection experiences
few losses. In this second case, would a NAK-only protocol be preferable to a
protocol that uses ACKs? Why?
P15. Consider the cross-country example shown in Figure 3.17. How big would
the window size have to be for the channel utilization to be greater than 98
percent? Suppose that the size of a packet is 1,500 bytes, including both
header fields and data.
P16. Suppose an application uses rdt 3.0 as its transport layer protocol. As the
stop-and-wait protocol has very low channel utilization (shown in the crosscountry example), the designers of this application let the receiver keep sending back a number (more than two) of alternating ACK 0 and ACK 1 even if
PROBLEMS 291
the corresponding data have not arrived at the receiver. Would this application design increase the channel utilization? Why? Are there any potential
problems with this approach? Explain.
P17. Consider two network entities, A and B, which are connected by a perfect bidirectional channel (i.e., any message sent will be received correctly; the
channel will not corrupt, lose, or re-order packets). A and B are to deliver
data messages to each other in an alternating manner: First, A must deliver a
message to B, then B must deliver a message to A, then A must deliver a message to B and so on. If an entity is in a state where it should not attempt to
deliver a message to the other side, and there is an event like
rdt_send(data) call from above that attempts to pass data down for
transmission to the other side, this call from above can simply be ignored
with a call to rdt_unable_to_send(data), which informs the higher
layer that it is currently not able to send data. [Note: This simplifying
assumption is made so you don’t have to worry about buffering data.]
Draw a FSM specification for this protocol (one FSM for A, and one FSM for
B!). Note that you do not have to worry about a reliability mechanism here;
the main point of this question is to create a FSM specification that reflects
the synchronized behavior of the two entities. You should use the following
events and actions that have the same meaning as protocol rdt1.0 in
Figure 3.9: rdt_send(data), packet = make_pkt(data),
udt_send(packet), rdt_rcv(packet), extract
(packet,data), deliver_data(data). Make sure your protocol
reflects the strict alternation of sending between A and B. Also, make sure to
indicate the initial states for A and B in your FSM descriptions.
P18. In the generic SR protocol that we studied in Section 3.4.4, the sender transmits a message as soon as it is available (if it is in the window) without waiting for an acknowledgment. Suppose now that we want an SR protocol that
sends messages two at a time. That is, the sender will send a pair of messages
and will send the next pair of messages only when it knows that both messages in the first pair have been received correctly.
Suppose that the channel may lose messages but will not corrupt or reorder
messages. Design an error-control protocol for the unidirectional reliable
transfer of messages. Give an FSM description of the sender and receiver.
Describe the format of the packets sent between sender and receiver, and vice
versa. If you use any procedure calls other than those in Section 3.4 (for
example, udt_send(), start_timer(), rdt_rcv(), and so on),
clearly state their actions. Give an example (a timeline trace of sender and
receiver) showing how your protocol recovers from a lost packet.
P19. Consider a scenario in which Host A wants to simultaneously send packets to
Hosts B and C. A is connected to B and C via a broadcast channel—a packet
292 CHAPTER 3 • TRANSPORT LAYER
sent by A is carried by the channel to both B and C. Suppose that the broadcast channel connecting A, B, and C can independently lose and corrupt
packets (and so, for example, a packet sent from A might be correctly
received by B, but not by C). Design a stop-and-wait-like error-control protocol for reliably transferring packets from A to B and C, such that A will not
get new data from the upper layer until it knows that both B and C have correctly received the current packet. Give FSM descriptions of A and C. (Hint:
The FSM for B should be essentially the same as for C.) Also, give a description of the packet format(s) used.
P20. Consider a scenario in which Host A and Host B want to send messages to
Host C. Hosts A and C are connected by a channel that can lose and corrupt
(but not reorder) messages. Hosts B and C are connected by another channel (independent of the channel connecting A and C) with the same properties. The transport layer at Host C should alternate in delivering messages
from A and B to the layer above (that is, it should first deliver the data from
a packet from A, then the data from a packet from B, and so on). Design a
stop-and-wait-like error-control protocol for reliably transferring packets
from A and B to C, with alternating delivery at C as described above. Give
FSM descriptions of A and C. (Hint: The FSM for B should be essentially
the same as for A.) Also, give a description of the packet format(s) used.
P21. Suppose we have two network entities, A and B. B has a supply of data messages that will be sent to A according to the following conventions. When A
gets a request from the layer above to get the next data (D) message from B,
A must send a request (R) message to B on the A-to-B channel. Only when B
receives an R message can it send a data (D) message back to A on the B-toA channel. A should deliver exactly one copy of each D message to the layer
above. R messages can be lost (but not corrupted) in the A-to-B channel; D
messages, once sent, are always delivered correctly. The delay along both
channels is unknown and variable.
Design (give an FSM description of) a protocol that incorporates the appropriate mechanisms to compensate for the loss-prone A-to-B channel and
implements message passing to the layer above at entity A, as discussed
above. Use only those mechanisms that are absolutely necessary.
P22. Consider the GBN protocol with a sender window size of 4 and a sequence
number range of 1,024. Suppose that at time t, the next in-order packet that the
receiver is expecting has a sequence number of k. Assume that the medium
does not reorder messages. Answer the following questions:
a. What are the possible sets of sequence numbers inside the sender’s window at time t? Justify your answer.
b. What are all possible values of the ACK field in all possible messages currently propagating back to the sender at time t? Justify your answer.
PROBLEMS 293
P23. Consider the GBN and SR protocols. Suppose the sequence number space is
of size k. What is the largest allowable sender window that will avoid the
occurrence of problems such as that in Figure 3.27 for each of these protocols?
P24. Answer true or false to the following questions and briefly justify your
answer:
a. With the SR protocol, it is possible for the sender to receive an ACK for a
packet that falls outside of its current window.
b. With GBN, it is possible for the sender to receive an ACK for a packet that
falls outside of its current window.
c. The alternating-bit protocol is the same as the SR protocol with a sender
and receiver window size of 1.
d. The alternating-bit protocol is the same as the GBN protocol with a sender
and receiver window size of 1.
P25. We have said that an application may choose UDP for a transport protocol
because UDP offers finer application control (than TCP) of what data is sent
in a segment and when.
a. Why does an application have more control of what data is sent in a segment?
b. Why does an application have more control on when the segment is sent?
P26. Consider transferring an enormous file of L bytes from Host A to Host B.
Assume an MSS of 536 bytes.
a. What is the maximum value of L such that TCP sequence numbers are not
exhausted? Recall that the TCP sequence number field has 4 bytes.
b. For the L you obtain in (a), find how long it takes to transmit the file.
Assume that a total of 66 bytes of transport, network, and data-link header
are added to each segment before the resulting packet is sent out over a
155 Mbps link. Ignore flow control and congestion control so A can pump
out the segments back to back and continuously.
P27. Host A and B are communicating over a TCP connection, and Host B has
already received from A all bytes up through byte 126. Suppose Host A then
sends two segments to Host B back-to-back. The first and second segments
contain 80 and 40 bytes of data, respectively. In the first segment, the
sequence number is 127, the source port number is 302, and the destination
port number is 80. Host B sends an acknowledgment whenever it receives a
segment from Host A.
a. In the second segment sent from Host A to B, what are the sequence number, source port number, and destination port number?
b. If the first segment arrives before the second segment, in the acknowledgment of the first arriving segment, what is the acknowledgment number,
the source port number, and the destination port number?
294 CHAPTER 3 • TRANSPORT LAYER
c. If the second segment arrives before the first segment, in the acknowledgment of the first arriving segment, what is the acknowledgment
number?
d. Suppose the two segments sent by A arrive in order at B. The first acknowledgment is lost and the second acknowledgment arrives after the first timeout interval. Draw a timing diagram, showing these segments and all other
segments and acknowledgments sent. (Assume there is no additional packet
loss.) For each segment in your figure, provide the sequence number and
the number of bytes of data; for each acknowledgment that you add, provide the acknowledgment number.
P28. Host A and B are directly connected with a 100 Mbps link. There is one TCP
connection between the two hosts, and Host A is sending to Host B an enormous file over this connection. Host A can send its application data into its TCP
socket at a rate as high as 120 Mbps but Host B can read out of its TCP receive
buffer at a maximum rate of 50 Mbps. Describe the effect of TCP flow control.
P29. SYN cookies were discussed in Section 3.5.6.
a. Why is it necessary for the server to use a special initial sequence number
in the SYNACK?
b. Suppose an attacker knows that a target host uses SYN cookies. Can the
attacker create half-open or fully open connections by simply sending an
ACK packet to the target? Why or why not?
c. Suppose an attacker collects a large amount of initial sequence numbers sent
by the server. Can the attacker cause the server to create many fully open
connections by sending ACKs with those initial sequence numbers? Why?
P30. Consider the network shown in Scenario 2 in Section 3.6.1. Suppose both
sending hosts A and B have some fixed timeout values.
a. Argue that increasing the size of the finite buffer of the router might possibly decrease the throughput (“out).
b. Now suppose both hosts dynamically adjust their timeout values (like
what TCP does) based on the buffering delay at the router. Would increasing the buffer size help to increase the throughput? Why?
P31. Suppose that the five measured SampleRTT values (see Section 3.5.3) are
106 ms, 120 ms, 140 ms, 90 ms, and 115 ms. Compute the EstimatedRTT
after each of these SampleRTT values is obtained, using a value of α = 0.125
and assuming that the value of EstimatedRTT was 100 ms just before the
first of these five samples were obtained. Compute also the DevRTT after
each sample is obtained, assuming a value of β = 0.25 and assuming the
value of DevRTT was 5 ms just before the first of these five samples was
obtained. Last, compute the TCP TimeoutInterval after each of these
samples is obtained.
PROBLEMS 295
P32. Consider the TCP procedure for estimating RTT. Suppose that # = 0.1. Let
SampleRTT1 be the most recent sample RTT, let SampleRTT2 be the next
most recent sample RTT, and so on.
a. For a given TCP connection, suppose four acknowledgments have been
returned with corresponding sample RTTs: SampleRTT4, SampleRTT3,
SampleRTT2, and SampleRTT1. Express EstimatedRTT in terms of
the four sample RTTs.
b. Generalize your formula for n sample RTTs.
c. For the formula in part (b) let n approach infinity. Comment on why this
averaging procedure is called an exponential moving average.
P33. In Section 3.5.3, we discussed TCP’s estimation of RTT. Why do you think
TCP avoids measuring the SampleRTT for retransmitted segments?
P34. What is the relationship between the variable SendBase in Section 3.5.4
and the variable LastByteRcvd in Section 3.5.5?
P35. What is the relationship between the variable LastByteRcvd in Section
3.5.5 and the variable y in Section 3.5.4?
P36. In Section 3.5.4, we saw that TCP waits until it has received three
duplicate ACKs before performing a fast retransmit. Why do you think the
TCP designers chose not to perform a fast retransmit after the first duplicate
ACK for a segment is received?
P37. 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 2nd segment
(sent from 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?
P38. In our description of TCP in Figure 3.53, the value of the threshold,
ssthresh, is set as ssthresh=cwnd/2 in several places and
ssthresh value is referred to as being set to half the window size when a
loss event occurred. Must the rate at which the sender is sending when the
loss event occurred be approximately equal to cwnd segments per RTT?
Explain your answer. If your answer is no, can you suggest a different
manner in which ssthresh should be set?
296 CHAPTER 3 • TRANSPORT LAYER
P39. Consider Figure 3.46(b). If “$in increases beyond R/2, can “out increase
beyond R/3? Explain. Now consider Figure 3.46(c). If “$in increases
beyond R/2, can “out increase beyond R/4 under the assumption that a
packet will be forwarded twice on average from the router to the receiver?
Explain.
P40. Consider Figure 3.58. Assuming TCP Reno is the protocol experiencing the
behavior shown above, answer the following questions. In all cases, you
should provide a short discussion justifying your answer.
a. Identify the intervals of time when TCP slow start is operating.
b. Identify the intervals of time when TCP congestion avoidance is
operating.
c. After the 16th transmission round, is segment loss detected by a triple
duplicate ACK or by a timeout?
d. After the 22nd transmission round, is segment loss detected by a triple
duplicate ACK or by a timeout?
e. What is the initial value of ssthresh at the first transmission round?
f. What is the value of ssthresh at the 18th transmission round?
g. What is the value of ssthresh at the 24th transmission round?
h. During what transmission round is the 70th segment sent?
i. Assuming a packet loss is detected after the 26th round by the receipt of a
triple duplicate ACK, what will be the values of the congestion window
size and of ssthresh?
0
0 2 4 6 8 10 12
Transmission round
14 16 18 20 22 24 26
5
10
15
20
25
Congestion window size (segments)
30
35
40
45
Figure 3.58 ! TCP window size as a function of time
VideoNote
Examining the
behavior of TCP
PROBLEMS 297
j. Suppose TCP Tahoe is used (instead of TCP Reno), and assume that triple
duplicate ACKs are received at the 16th round. What are the ssthresh
and the congestion window size at the 19th round?
k. Again suppose TCP Tahoe is used, and there is a timeout event at 22nd
round. How many packets have been sent out from 17th round till 22nd
round, inclusive?
P41. Refer to Figure 3.56, which illustrates the convergence of TCP’s AIMD
algorithm. Suppose that instead of a multiplicative decrease, TCP decreased
the window size by a constant amount. Would the resulting AIAD algorithm
converge to an equal share algorithm? Justify your answer using a diagram
similar to Figure 3.56.
P42. In Section 3.5.4, we discussed the doubling of the timeout interval after a
timeout event. This mechanism is a form of congestion control. Why does
TCP need a window-based congestion-control mechanism (as studied in
Section 3.7) in addition to this doubling-timeout-interval mechanism?
P43. Host A is sending an enormous file to Host B over a TCP connection.
Over this connection there is never any packet loss and the timers never
expire. Denote the transmission rate of the link connecting Host A to the
Internet by R bps. Suppose that the process in Host A is capable of sending
data into its TCP socket at a rate S bps, where S = 10 · R. Further suppose
that the TCP receive buffer is large enough to hold the entire file, and the
send buffer can hold only one percent of the file. What would prevent the
process in Host A from continuously passing data to its TCP socket at rate S
bps? TCP flow control? TCP congestion control? Or something else?
Elaborate.
P44. 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?
P45. Recall the macroscopic description of TCP throughput. In the period of time
from when the connection’s rate varies from W/(2 · RTT) to W/RTT, 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 = loss rate = 1
3
8
W2 +
3
4
W
298 CHAPTER 3 • TRANSPORT LAYER
b. Use the result above to show that if a connection has loss rate L, then its
average rate is approximately given by
P46. Consider that only a single TCP (Reno) connection uses one 10Mbps 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 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?
P47. Consider the scenario described in the previous problem. Suppose that the
10Mbps link can buffer a finite number of segments. Argue that in order for
the link to always be busy sending data, we would like to choose a buffer size
that is at least the product of the link speed C and the two-way propagation
delay between the sender and the receiver.
P48. Repeat Problem 43, but replacing the 10 Mbps link with a 10 Gbps link. Note
that in your answer to part c, you will realize that it takes a very long time for
the congestion window size to reach its maximum window size after recovering from a packet loss. Sketch a solution to solve this problem.
P49. Let T (measured by RTT) denote the time interval that a TCP connection
takes to increase its congestion window size from W/2 to W, where W is the
maximum congestion window size. Argue that T is a function of TCP’s average throughput.
P50. 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, C1 and C2, share a single congested link of speed 30 segments
per second. Assume that both C1 and C2 are in the congestion avoidance
! 1.22 ! MSS
RTT 2L
PROBLEMS 299
phase. Connection C1’s RTT is 50 msec and connection C2’s RTT is
100 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 C1 and C2 at time t0 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.
P51. Consider the network described in the previous problem. Now suppose that
the two TCP connections, C1 and C2, have the same RTT of 100 msec. Suppose that at time t0, C1’s congestion window size is 15 segments but C2’s
congestion window size is 10 segments.
a. What are their congestion window sizes after 2200msec?
b. In the long run, will these two connections get about the same share of the
bandwidth of the congested link?
c. We say that two connections are synchronized, if both connections reach
their maximum window sizes at the same time and reach their minimum
window sizes at the same time. In the long run, will these two connections
get synchronized eventually? If so, what are their maximum window sizes?
d. Will this synchronization help to improve the utilization of the shared
link? Why? Sketch some idea to break this synchronization.
P52. 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 < a < 1) whenever it
receives a valid ACK. Find the functional relationship between loss rate L
and maximum congestion window W. Argue that for this modified TCP,
regardless of TCP’s average throughput, a TCP connection always spends the
same amount of time to increase its congestion window size from W/2 to W.
P53. In our discussion of TCP futures in Section 3.7, we noted that to achieve a
throughput of 10 Gbps, TCP could only tolerate a segment loss probability of
2 · 10-10 (or equivalently, one loss event for every 5,000,000,000 segments).
Show the derivation for the values of 2 · 10-10 (1 out of 5,000,000) for the
RTT and MSS values given in Section 3.7. If TCP needed to support a 100
Gbps connection, what would the tolerable loss be?
P54. 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?
