Chapter 3 Computer Networks
Which of the following best completes this sentence: "In the absence of loss, TCP slow start increases the sending rate ... "
" ... faster than AIMD. In fact, slowstart increases the sending rate exponentially fast per RTT."
Which of the following actions are used in network-assisted congestion control (say versus end-end congestion control) to signal congestion. Check all that apply.
A router marks a field in the datagram header at a congested router. A router sends an ICMP message to a host telling it to slow down its sending rate.
Consider TCP's Fast Retransmit optimization (see Figure 3.37 from the text, below). Of course, the sender doesn't know for sure that the segment with sequence # 100 is actually lost (it can't see into the channel). Can a sender get three duplicate ACKs for a segment that in fact has not been lost? Which of the following statements are true? Suppose a channel can lose, but will not corrupt, messages.
If the channel can reorder messages, a triple duplicate ACK can occur even though a message is not lost; since it's possible that a message has just been reordered and has not yet arrived when the three duplicate ACKs were generated. If the channel cannot reorder messages, a triple duplicate ACK indicates to the sender that a segment loss has happened for sure. Actually (again assuming the channel cannot corrupt or reorder messages), even a single duplicate ACK would indicate that a segment loss has happed for sure.
Consider the figure below, which shows the application-to-application throughput achieved when two senders are competing at a shared bottleneck link. Suppose that when the overall arrival rate, lambdain' (for each sender) is close to R/2, the throughput to the application layer (at each receiver), lambdaout, is equal to 0.8 * lambdain'. What fraction of the packets transmitted at the sender are retransmissions?
.20
What is meant by transport-layer demultiplexing?
Receiving a transport-layer segment from the network layer, extracting the payload (data) and delivering the data to the correct socket.
True or False: On the sending side, the UDP sender will take each application-layer chunk of data written into a UDP socket and send it in a distinct UDP datagram. And then on the receiving side, UDP will deliver a segment's payload into the appropriate socket, preserving the application-defined message boundary.
True
True or False: The transport layer provides for host-to-host delivery service?
True.
Why is the UDP header length field needed?
Because the payload section can be of variable length, and this lets UDP know where the segment ends.
In SR, why does the receiver have to acknowledge packets with sequence numbers that are less than (and to the left of) those in its window, which starts at rcv_base.
Because the sender may not have received an ACK for that packet yet.
Consider the TCP Telnet scenario below (from Fig. 3.36 in text). What timer-related action does the sender take on the receipt of ACK 120?
Cancels any running timers.
What are advantages of the streams concept in QUIC? Select all that apply.
Since each stream has its own error control, if one stream experiences an error (e.g., lost or damaged segment), the other streams are unaffected. Streams allow concurrent retrieval of web objects, while avoiding Head of the Line (HOL) blocking.
Compute the Internet checksum value for these two 16-bit words: 11110101 11010011 and 10110011 01000100
01010110 11100111
Suppose a packet is 10K bits long, the channel transmission rate connecting a sender and receiver is 10 Mbps, and the round-trip propagation delay is 10 ms. How many packets can the sender transmit before it starts receiving acknowledgments back?
10
Compute the Internet checksum value for these two 16-bit words: 01000001 11000100 and 00100000 00101011
10011110 00010000
What is meant by a cumulative acknowledgment, ACK(n)?
A cumulative ACK(n) acks all packets with a sequence number up to and including n as being received.
Check all of the services below that are provided by the TCP protocol.
A flow-control service that ensures that a sender will not send at such a high rate so as to overflow receiving host buffers. In-order data delivery Reliable data delivery. A byte stream abstraction, that does not preserve boundaries between message data sent in different socket send calls at the sender. A congestion control service to ensure that multiple senders do not overload network links.
Check all of the services below that are provided by the UDP protocol.
A message abstraction, that preserves boundaries between message data sent in different socket send calls at the sender.
Consider the TCP Telnet scenario below (from Fig. 3.31 in text). Why is it that the receiver sends an ACK that is one larger than the sequence number in the received datagram?
Because the send-to receiver segment carries only one byte of data, and after that segment is received, the next expected byte of data is just the next byte (i.e., has an index that is one larger) in the data stream.
True or False: When computing the Internet checksum for two numbers, a single flipped bit in each of the two numbers will always result in a changed checksum.
False
Consider the transport-layer flows interacting at a congested link. In the face of such congestion, what happens at this link to a transport-layer flow that does not cut back on its sending rate?
Nothing different from the other flows crossing the congested link.
Which of the following statements are true about a checksum? Hint: more than one statement is true.
The receiver of a packet with a checksum field will add up the received bytes, just as the sender did, and compare this locally-computed checksum with the checksum value in the packet header. If these two values are different then the receiverknows that one of the bits in the received packet has been changed during transmission from sender to receiver. The sender-computed checksum value is often included in a checksum field within a packet header. A checksum is computed at a sender by considering each byte within a packet as a number, and then adding these numbers (each number representing a bytes) together to compute a sum (which is known as a checksum).
What are some reasons for discarding received-but- out-of-sequence packets at the receiver in GBN? Indicate one or more of the following statements that are correct.
The sender will resend that packet in any case. The implementation at the receiver is simpler
Where is transport-layer functionality primarily implemented?
Transport layer functions are implemented primarily at the hosts at the "edge" of the network.
True or False: When multiple TCP clients send TCP segments to the same destination port number at a receiving host, those segments (from different senders) will always be directed to the same socket at the receiving host.
False
In the SR receiver window (see diagram below, taken from PPT slides and video), why haven't the red packets been delivered yet? Check the one or more reasons below that apply.
There is a packet with a lower sequence number than any of the red packets that has yet to be received, so in-order delivery of data in the red packets up to the application layer is not possible
Suppose a packet is 10K bits long, the channel transmission rate connecting a sender and receiver is 10 Mbps, and the round-trip propagation delay is 10 ms. What is the maximum channel utilization of a stop-and-wait protocol for this channel?
.1
Consider the figure below, which plots the evolution of TCP's congestion window at the beginning of each time unit (where the unit of time is equal to the RTT); see Figure 3.53 in the text. In the abstract model for this problem, TCP sends a "flight" of packets of size cwnd at the beginning of each time unit. The result of sending that flight of packets is that either (i) all packets are ACKed at the end of the time unit, (ii) there is a timeout for the first packet, or (iii) there is a triple duplicate ACK for the first packet. In this problem, you are asked to reconstruct the sequence of events (ACKs, losses) that resulted in the evolution of TCP's cwnd shown below. Consider the evolution of TCP's congestion window in the example above and answer the following questions. The initial value of cwnd is 1 and the initial value of ssthresh (shown as a red +) is 8. 1. Give the times at which TCP is in slow start. Format your answer like: 1,3,5,9 (If none submit blank) 2. Give the times at which TCP is in congestion avoidance. Format your answer like: 1,3,5,9 (If none submit blank) 3. Give the times at which TCP is in fast recovery. Format your answer like: 1,3,5,9 (If none submit blank) 4. Give the times at which packets are lost via timeout. Format your answer like: 1,3,5,9 (If none submit blank) 5. Give the times at which packets are lost via triple ACK. Format your answer like: 1,3,5,9 (If none submit blank) 6. Give the times at which the value of ssthresh changes (if it changes between t=3 and t=4, use t=4 in your answer)
1. 1,2,3,19,20,21,33,34,35 2. 4,5,6,7,8,10,11,12,13,14,15,16,17,22,23,24,25,26,28,29,30,31,32,36,37,38,39,40 3. 9,18,27 4. 18,32 5. 8,17,26 6. 9,18,19,27,33
Consider the two 16-bit words (shown in binary) below. Recall that to compute the Internet checksum of a set of 16-bit words, we compute the one's complement sum [1] of the two words. That is, we add the two numbers together, making sure that any carry into the 17th bit of this initial sum is added back into the 1's place of the resulting sum); we then take the one's complement of the result. Compute the Internet checksum value for these two 16-bit words: 01110010 10011100 this binary number is 29340 decimal (base 10) 00110001 11101000 this binary number is 12776 decimal (base 10) 1. What is the sum of these two 16 bit numbers? Don't put any spaces in your answer 2.Using the sum from question 1, what is the checksum? Don't put any spaces in your answer
1. 10100100 10000100 2. 01011011 01111011
Consider the figure below in which a TCP sender and receiver communicate over a connection in which the segments can be lost. The TCP sender wants to send a total of 10 segments to the receiver and sends an initial window of 5 segments at t = 1, 2, 3, 4, and 5, respectively. Suppose the initial value of the sequence number is 143 and every segment sent to the receiver each contains 693 bytes. The delay between the sender and receiver is 7 time units, and so the first segment arrives at the receiver at t = 8, and an ACK for this segment arrives at t = 15. As shown in the figure, 2 of the 5 segments is lost between the sender and the receiver, but one of the ACKs is lost. Assume there are no timeouts and any out of order segments received are thrown out. 1. What is the 55iksequence number of the segment sent at t=1? 2. What is the sequence number of the segment sent at t=2? 3. What is the sequence number of the segment sent at t=3? 4. What is the sequence number of the segment sent at t=4? 5. What is the sequence number of the segment sent at t=5? 6. What is the value of the ACK sent at t=8? (If segment lost, write 'x') 7. What is the value of the ACK sent at t=9? (If segment lost, write 'x') 8. What is the value of the ACK sent at t=10? (If segment lost, write 'x') 9. What is the value of the ACK sent at t=11? (If segment lost, write 'x') 10. What is the value of the ACK sent at t=12? (If segment lost, write 'x') 11. What is the sequence number of the segment sent at t = 15? (If ACK never arrives, write 'x') 12. What is the sequence number of the segment sent at t = 16? (If ACK never arrives, write 'x') 13. What is the sequence number of the segment sent at t = 17? (If ACK never arrives, write 'x') 14. What is the sequence number of the segment sent at t = 18? (If ACK never arrives, write 'x') 15. What is the sequence number of the segment sent at t = 19? (If ACK never arrives, write 'x')
1. 143 2. 836 3. 1529 4. 2222 5. 2915 6. 836 7. 1529 8. 2222 9. x 10. x 11. 3608 12. x 13. 4301 14. 4994 15. x
Suppose that as shown in the figure below, a TCP sender is sending segments with 100 bytes of payload. The TCP sender sends five segments with sequence numbers 100, 200, 300, 400, and 500. Suppose that the segment with sequence number 300 is lost. The TCP receiver will buffer correctly-received but not-yet-in-order segments for later delivery to the application layer (once missing segments are later received). 1. After receiving segment 100, the receiver responds with an ACK with value: 2. After receiving segment 200, the receiver responds with an ACK with value: 3. After receiving segment 500, the receiver responds with an ACK with value: 4. After receiving the retransmitted segment 300, the receiver responds with an ACK with value: 5. The TCP receiver does not respond in the example, with an ACK with value:
1. 200 2. 300 3. 300, duplicate ack 4. 600 5. 400
Suppose that TCP's current estimated values for the round trip time (estimatedRTT) and deviation in the RTT (DevRTT) are 230 msec and 30 msec, respectively (see Section 3.5.3 for a discussion of these variables). Suppose that the next three measured values of the RTT are 360 msec, 340 msec, and 300 msec respectively. Compute TCP's new value ofDevRTT, estimatedRTT,and the TCP timeout value after each of these three measured RTT values is obtained. Use the values of α = 0.125, and β = 0.25. Round your answers to two decimal places after leading zeros. 1. What is the estimatedRTT after the first RTT? 2. What is the RTT Deviation for the the first RTT? 3. What is the TCP timeout for the first RTT? 4. What is the estimatedRTT after the second RTT? 5. What is the RTT Deviation for the the second RTT? 6. What is the TCP timeout for the second RTT? 7. What is the estimatedRTT after the third RTT? 8. What is the RTT Deviation for the the third RTT? 9. What is the TCP timeout for the third RTT?
1. 246.25 2. 55 3. 466.25 4. 257.97 5. 64.69 6. 516.72 7. 263.22 8. 59.02 9. 499.32
Suppose a UDP segment (A in the figure below) arrives at a host with an IP address of 128.119.40.186. The source port in the UDP segment is 4829 and the destination port is 3546. The IP address of the sending host is 60.54.75.24. Now consider the UDP datagram (and the IP datagram that will encapsulate it) sent in reply by the application on host 128.119.40.186 to the original sender host, labeled B in the figure above. Complete the sentences below ... What are the source and destination port numbers and IP addresses? (Enter the integer port number or the 4-part dotted decimal IP address, included the period) The source port number of the UDP segment (B) sent in reply is: The source IP address of the IP datagram containing the UDP segment (B) sent in reply is: The destination port number of the UDP segment (B) sent in reply is: The destination IP address of the IP datagram containing the UDP segment (B) sent in reply is: QUESTION LIST: 1. The source port number of the UDP segment (B) sent in reply is: 2. The source IP address of the IP datagram containing the UDP segment (B) sent in reply is: 3. The destination port number of the UDP segment (B) sent in reply is: 4. The destination IP address of the IP datagram containing the UDP segment (B) sent in reply is:
1. 3546 2. 128.119.40.186 3. 4829 4. 60.54.75.24
Consider the figure below in which a TCP sender and receiver communicate over a connection in which the sender->receiver segments may be lost. The TCP sender sends an initial window of 5 segments. Suppose the initial value of the sender->receiver sequence number is 41 and the first 5 segments each contain 407 bytes. The delay between the sender and receiver is 7 time units, and so the first segment arrives at the receiver at t=8. As shown in the figure below, 3 of the 5 segment(s) are lost between the segment and receiver. 1. Give the sequence numbers associated with each of the 5 segments sent by the sender. Format your answer as: a,b,c,... 2. Give the ACK numbers the receiver sends in response to each of the segments. If a segment never arrives use 'x' to denote it, and format your answer as: a,b,c,...
1. 41,448,855,1262,1669 2. x,41,x,x,41
In the scenario below, the left and right TCP clients communicate with a TCP server using TCP sockets. The Python code used to create a single welcoming socket in the server is shown in the figure (the welcoming socket itself is not shown graphically); code is also shown for the client sockets as well. The three sockets shown in server were created as a result of the server accepting connection requests on this welcoming socket from the two clients (one connection from the client on the left, and two connections from the client on the right). 1. What is the source port # for packet A? 2. What is the destination port # for packet A? 3. What is the source port # for packet D? 4. What is the destination port # for packet D? 5. What is the source port # for packet C? 6. What is the destination port # for packet C? 7. What is the source port # for packet B? 8. What is the destination port # for packet B?
1. 5744 2. 5119 3. 5257 4. 5119 5. 5397 6. 5119 7. 5119 8. 5744
In the scenario below, the left and right clients communicate with a server using UDP sockets. The same socket at the server is used to communicate with both clients. The Python code used to create the sockets is shown in the figure. Consider the four transport-layer packets - A, B, C and D - shown in the figure below. 1. What is the source port # for packet D? 2. What is the destination port # for packet D? 3. What is the source port # for packet A? 4. What is the destination port # for packet A? 5. What is the source port # for packet B? 6. What is the destination port # for packet B? 7. What is the source port # for packet C? 8. What is the destination port # for packet C?
1. 7029 2. 5386 3. 7028 4. 7029 5. 7029 6. 7028 7. 5386 8. 7029
Consider the purposes/goals/use of different reliable data transfer protocol mechanisms. For the given purpose/goal/use match it to the RDT mechanism that is used to implement the given purpose/goal/use. 1. Lets the sender know that a packet was NOT received correctly at the receiver. 2. Used by sender or receiver to detect bits flipped during a packet's transmission. 3. Allows for duplicate detection at receiver. 4. Lets the sender know that a packet was received correctly at the receiver. 5. Allows the receiver to eventually receive a packet that was corrupted or lost in an earlier transmission.
1. NAK 2. Checksum 3. Sequence numbers 4. ACK 5.Retransmission
Consider the rdt2.1 sender and receiver FSMs shown below, with labeled transitions S1 through S6 at the sender, and transitions R1 through R6 at the receiver. The sender and receiver start in the "Wait for call 0 from above" and "Wait for 0 from below" states, respectively. Suppose that no channel errors occur. A sequence of interleaved sender and receiver transitions is given below. Transitions S1 and S4 are already provided. Choose the sender or receiver transition for the unlabeled transitions x1, x2, x3, and x4 below to indicate the time-ordered sequence of transitions (interleaved sender and receiver transitions) that will result in two messages being delivered at the receiver, with the sender and receiver returning to their initial states (again, given that no channel errors occur). S1, x1, x2, S4, x3, x4 QUESTION LIST: 1. transition x1 2. transition x2 3. transition x3 4. transition x4
1. R1 2. S3 3. R4 4. S6
Consider the rdt2.1 sender and receiver FSMs shown below, with labeled transitions S1 through S6 at the sender, and transitions R1 through R6 at the receiver. The sender and receiver start in the "Wait for call 0 from above" and "Wait for 0 from below" states, respectively. Suppose that the first packet from the sender is correctly received at the receiver but that ACK message sent from receiver-to-sender is corrupted; all other messages (before or after that ACK) are transmitted error-free. Match the unlabeled transitions x1, x2, x3, x4, x5 in the time-ordered sequence of transitions below (interleaved sender and receiver transitions) that will occur following the initial S1 transition, which is followed by a corrupted ACK transmission, that will result in a message being delivered at the receiver, with the sender and receiver returning to their initial states. Note that some transitions are already provided below. S1, x1 (ACK corrupted), x2, x3, x4, S4, x5, S6. 1. transition x1 2. transition x2 3. transition x3 4. transition x4 5. transition x5
1. R1 (ACK corrupted) 2. S2 3. R3 4. S3 5.R4
Consider the rdt2.1 sender and receiver FSMs shown below, with labeled transitions S1 through S6 at the sender, and transitions R1 through R6 at the receiver. The sender and receiver start in the "Wait for call 0 from above" and "Wait for 0 from below" states, respectively. Suppose that the initial message transmission by the sender is corrupted, but that no other message transmissions are corrupted. Match the unlabeled transitions x1, x2, x3, x4, x5 in the time-ordered sequence of transitions below (interleaved sender and receiver transitions) that will occur following the initial S1 transition (which is corrupted), that will result in two messages being delivered at the receiver, with the sender and receiver returning to their initial states (again, given that the initial message transmission by the sender is corrupted). Note that transitions S1, S4, and S6 are already provided below. S1 (message corrupted), x1, x2, x3, x4, S4, x5, S6. 1. transition x1 2. transition x2 3. transition x3 4. transition x4 5. transition x5
1. R6 2. S2 3. R1 4. S3 5. R4
Consider the rdt2.2 protocol from the text (pages 209-212). The FSMs for the sender and receiver are shown below: Suppose that the channel connecting the sender and receiver can corrupt but not lose or reorder packets. Now consider the figure below, which shows four data packets and three corresponding ACKs being exchanged between an rdt 2.2 sender and receiver. The actual corruption or successful transmission/reception of a packet is indicated by the corrupt and OK labels, respectively, shown above the packets in the figure below. 1. At time t=0, what is the sender state? 2. At time t=0, what is the receiver state? 3. At time t=0, what is the sequence/ack # of the packet? 4. At time t=1, what is the sender state? 5. At time t=1, what is the receiver state? 6. At time t=1, what is the sequence/ack # of the packet? 7. At time t=2, what is the sender state? 8. At time t=2, what is the receiver state? 9. At time t=2, what is the sequence/ack # of the packet? 10. At time t=3, what is the sender state? 11. At time t=3, what is the receiver state? 12. At time t=3, what is the sequence/ack # of the packet? 13. How many times is the payload of the received packet passed up to the higher layer?
1. Wait for ACK 0 2. Wait for 0 from below 3. 0 4. Wait for ACK 0 5. Wait for 1 from below 6. 0 7. Wait for ACK 1 8. Wait for 1 from below 9. 1 10. Wait for ACK 1 11. Wait for 0 from below 12. 1 13. 3
Use the pulldown menu to match a congestion control approach to how the sender detects congestion. 1. The sender infers segment loss from the absence of an ACK from the receiver. 2. Bits are set at a congested router in a sender-to-receiver datagram, and bits are in the returned to the sender in a receiver-to sender ACK, to indicate congestion to the sender. 3. The sender measures RTTs and uses the current RTT measurement to infer the level of congestion.
1. end-end 2. network-assisted 3. delay-based
Suppose a packet is 10K bits long, the channel transmission rate connecting a sender and receiver is 10 Mbps, and the round-trip propagation delay is 10 ms. What is the channel utilization of a pipelined protocol with an arbitrarily high level of pipelining for this channel?
1.0
Which of the following statements about TCP's Additive-increase-multiplicative-decrease (AIMD) algorithm are true? Check all that are true.
AIMD cuts the congestion window size, cwnd, in half whenever loss is detected by a triple duplicate ACK. AIMD is a end-end approach to congestion control. AIMD cuts the congestion window size,cwnd, i to 1 whenever a timeout occurs.
What are advantages of implementing transport-layer functionality in QUIC at the application layer? Select all that apply.
QUIC can establish all connection parameters (security, reliability, flow and congestion control)in just one handshake rather than separately in two. As an application-layer protocol, QUIC can be updated/modified at "app frequency" rather than at the frequency of operating system updates.
Consider the rdt 2.0 sender and receiver shown below, with FSM transitions at the sender labeled S1, S2, and S3; and receiver transitions labeled R1 and R2.Which of the following sequences of transitions could possibly occur as a result of an initial rdt_send() call at the sender, and possible later message corruption and subsequent error recovery.
S1, R2, S3 S1, R1, S2 S1, R1, S2, R2,S3
Consider the RDT 3.0 protocol, for reliably communicating data from a sender to receiver over a channel that can lose or corrupt packets in either direction, and when the maximum delay from sender to receiver and back is not known. The FSMs for the sender and receiver are shown below, with their transitions labeled as SX and RY, respectively. Now let's consider the sequence of sender and receiver transitions that would happen when one or more of the following complications occur: a packet (data or ACK) is lost, a timer times out (prematurely or not), or a message is corrupted. One or more of these events has occurred to produce the sequence of transitions below. In the sequence below, one transition has been omitted and replaced with a "*". Transition Sequence: S0, R3, S1, S2, R0, *, S5, R1, S6, S7, R2, S8 What is the missing transition? To indicate the missing transition, enter S or R, followed by an index.
S3
Which of the fields below are in a UDP segment header? [Hint: note the use of the word "header" in this question statement.]
Source port number Length (of UDP header plus payload) Internet checksum Destination port number
What is meant by transport-layer multiplexing?
Taking data from one socket (one of possibly many sockets), encapsulating a data chuck with header information - thereby creating a transport layer segment - and eventually passing this segment to the network layer.
Over what set of bytes is the checksum field in the UDP header computed over?
The entire UDP segment, except the checksum field itself, and the IP sender and receive address fields
True or False: When computing the Internet checksum for two numbers, a single flipped bit (i.e., in just one of the two numbers) will always result in a changed checksum.
True
True or False: When multiple UDP clients send UDP segments to the same destination port number at a receiving host, those segments (from different senders) will always be directed to the same socket at the receiving host.
True
True or False: with TCP's flow control mechanism, where the receiver tells the sender how much free buffer space it has (and the sender always limits the amount of outstanding, unACKed, in-flight data to less than this amount), it is not possible for the sender to send more data than the receiver has room to buffer.
True
Which of the following statements about pipelining are true? One or more statements may be true.
With a pipelined sender, there may be transmitted packets "in flight" - propagating through the channel - packets that the sender has sent but that the receiver has not yet received. A pipelined sender can have transmitted multiple packets for which the sender has yet to receive an ACK from the receiver.
Assuming that the congestion window size,cwnd, has not yet reached Wmax, TCP CUBIC will ... (check all that apply)
... always have a window size, cwnd, and hence a sending rate, higher than that of AIMD (assuming a given window size, Wmax, at which loss would occur). ... increase its sending rate faster than AIMD when cwnd is far away from Wmax, but increase slower than AIMD when cwnd is closer to Wmax
Match the description of a TCP connection management message with the name of the message used to accomplish that function. 1. A message from client to server initiating a connection request. 2. A message from server to client ACKing receipt of a SYN message and indicating the willingness of the server to establish a TCP connection with the client. 3. A message indicating that the sending side is initiating the protocol to terminate a connection. 4. A message sent in response to a request to terminate a connection, ACKing that the side receiving this message is also willing to terminate the connection 5. A general purpose error message used during connection set up or tear down to let the other side know that an error has occurred, and that the referenced connection should be shut down.
1. SYN message 2. SYNACK message 3. FIN message 4. FINACK message 5. RESET message
For the given function of a field in the TCP segment, select the name of that field from the pull-down list. 1. This field contains the port number associated with the sending socket for this TCP segment. 2. This field contains application data that was written into a socket by the sender of this TCP segment. 3. This field contains the index in the sender-to-receiver byte stream of the first byte of that data in the payload carried in this segment. 4. This field contains the index in the byte stream of the next in-order byte expected at the receiver 5. If set, this segment cumulatively ACKs all data bytes up to, but not including, the byte index in the ACK value field of this segment. 6. This field contains the number of available bytes in the TCP receiver's buffer. 7. This field contains the Internet checksum of the TCP segment and selected fields in the IP datagram header. 8. This field contains the number of bytes in the TCP header.
1. Souce Port Number 2. Data (or payload) 3. Sequence number 4. ACK number field 5. ACK bit 6. Receiver advertised window 7. Checksum 8. Header length field
For delay-based congestion control, match the sender action to the relationship of the currently measured throughput to the value of cwnd/RTTmin 1. The currently measured throughput is greater than cwnd/RTTmin 2. The currently measured throughput is equal to or a bit less than than cwnd/RTTmin 3. The currently measured throughput is much less that than cwnd/RTTmin
1. This should never happen. 2. increase the sending rate 3. decrease the sending rate
Which of the following actions are associated with end-end congestion control (say versus network-assisted congestion control). Check all that apply.
A datagram experiences delay at a congested network router, which is then measured by the sender and used to decrease the sending rate. A sender decreases its sending rate in response to packet loss detected via its transport-layer ACKing. The transport-layer sender decreases its sending rate in response to a measured increase in the RTT. A router drops a packet at a congested router, which causes the transport-layer sender to infer that there is congestion due to the missing ACK for the lost packet.
How is the sending rate typically regulated in a TCP implementation?
By keeping a window of size cwnd over the sequence number space, and making sure that no more than cwnd bytes of data are outstanding (i.e, unACKnowledged). The size of cwnd is regulated by AIMD.
True or False: On the sending side, the TCP sender will take each application-layer chunk of data written into a TCP socket and send it in a distinct TCP segment. And then on the receiving side, TCP will deliver a segment's payload into the appropriate socket, preserving the application-defined message boundary.
False
The transport layer sits on top of the network layer, and provides its services using the services provided to it by the network layer. Thus it's important that we know what is meant by the network layer's "best effort" delivery service. True or False: The network layer's best-effort delivery service means that IP makes its "best effort" to deliver segments between communicating hosts, but it makes no guarantees. In particular, it does not guarantee segment delivery, it does not guarantee orderly delivery of segments, and it does not guarantee the integrity of the data in the segments.
Correct! The network layer's best effort service doesn't really provide much service at all, does it?
What are some reasons for not discarding received-but- out-of-sequence packets at the receiver in GBN? Indicate one or more of the following statements that are correct.
Even though that packet will be retransmitted, its next retransmission could be corrupted, so don't discard a perfectly well-received packet, silly!
Consider TCP use of an exponentially weighted moving average (EWMA) to compute the nth value of the estimated RTT:EstimatedRTTn = (1- a)*EstimatedRTTn-1 + a*SampleRTTnTrue or False: with this EWMA algorithm the value of EstimatedRTTn has no dependence on the earlier sample, SampleRTTn-1
False
True or False: It is possible for two TCP segments with source port 80 to be sent by the sending host to different clients.
True
True or False: It is possible for two UDP segments to be sent from the same socket with source port 5723 at a server to two different clients.
True
Consider the five images below. Indicate which of these images suggest the need for flow control (the others would suggest the need for congestion control).
A talking head A glass overflowing
