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computer networking
Questions and Answers of
Computer Networking
Suppose a host wants to establish the reliability of a link by sending packets and measuring the percentage that is received; routers, for example, do this. Explain the difficulty doing this over a
Suppose TCP operates over a 1-Gbps link.(a) Assuming TCP could utilize the full bandwidth continuously, how long would it take the sequence numbers to wrap around completely?(b) Suppose an added
Suppose TCP operates over a 40-Gbps STS-768 link.(a) Assuming TCP could utilize the full bandwidth continuously, how long would it take the sequence numbers to wrap around completely?(b) Suppose an
If host A receives two SYN packets from the same port from remote host B, the second may be either a retransmission of the original or else, if B has crashed and rebooted, an entirely new connection
Suppose x and y are two TCP sequence numbers. Write a function to determine whether x comes before y (in the notation of Request for Comments 793, “x =< y”) or after y; your solution should work
Suppose an idle TCP connection exists between sockets A and B. A third party has eavesdropped and knows the current sequence number at both ends.(a) Suppose the third party sends A a forged packet
Suppose party A connects to the Internet via a dial-up IP server (e.g., using SLIP or PPP), has several open Telnet connections (using TCP), and is cut off. Party B then dials in and is assigned the
Diagnostic programs are commonly available that record the first 100 bytes, say, of every TCP connection to a certain host, port. Outline what must be done with each received TCP packet, P, in order
If a packet arrives at host A with B’s source address, it could just as easily have been forged by any third host C. If, however, A accepts a TCP connection from B, then during the three-way
The Nagle algorithm, built into most TCP implementations, requires the sender to hold a partial segment’s worth of data (even if PUSHed)until either a full segment accumulates or the most recent
Suppose a client C repeatedly connects via TCP to a given port on a server S and that each time, it is C that initiates the close.(a) How many TCP connections per second can C make here before it
Explain why TIME_WAIT is a somewhat more serious problem if the server initiates the close than if the client does. Describe a situation in which this might reasonably happen.
What is the justification for the exponential increase in timeout value proposed by Karn and Partridge? Why, specifically, might a linear (or slower) increase be less desirable?
The Jacobson/Karels algorithm sets TimeOut to be 4 mean deviations above the mean. Assume that individual packet round-trip times follow a statistical normal distribution, for which 4 mean deviations
Suppose a TCP connection, with window size 1, loses every other packet. Those that do arrive have RTT = 1 second. What happens?What happens to TimeOut? Do this for two cases:(a) After a packet is
Suppose, in TCP’s adaptive retransmission mechanism, that EstimatedRTT is 4.0 at some point and subsequent measured RTTs all are 1.0.How long does it take before the TimeOut value, as calculated by
Suppose, in TCP’s adaptive retransmission mechanism, that EstimatedRTT is 90 at some point and subsequent measured RTTs all are 200.How long does it take before the TimeOut value, as calculated by
Suppose TCP’s measured RTT is 1.0 except that every Nth RTT is 4.0.What is the largest N, approximately, that does not result in timeouts in the steady state (i.e., for which the Jacobson/Karels
Suppose that TCP is measuring RTTs of 1.0 second, with a mean deviation of 0.1 second. Suddenly the RTT jumps to 5.0 seconds, with no deviation. Compare the behaviors of the original and
Suppose that when a TCP segment is sent more than once, we take SampleRTT to be the time between the original transmission and the ACK, as in Figure 5.10(a). Show that if a connection with a
Suppose that when a TCP segment is sent more than once, we take SampleRTT to be the time between the most recent transmission and the ACK, as in Figure 5.10(b). Assume, for definiteness, that TimeOut
Consult Request for Comments 793 to find out how TCP is supposed to respond if a FIN or an RST arrives with a sequence number other than NextByteExpected. Consider both when the sequence number is
One of the purposes of TIME_WAIT is to handle the case of a data packet from a first incarnation of a connection arriving very late and being accepted as data for the second incarnation.(a) Explain
Propose an extension to TCP by which one end of a connection can hand off its end to a third host; that is, if A were connected to B, and A handed off its connection to C, then afterwards, C would be
TCP’s simultaneous open feature is seldom used.(a) Propose a change to TCP in which this is disallowed. Indicate what changes would be made in the state diagram (and, if necessary, in the
TCP is a very symmetric protocol, but the client/server model is not. Consider an asymmetric TCP-like protocol in which only the server side is assigned a port number visible to the application
The following exercise is concerned with the TCP state FIN_WAIT_2(see Figure 5.7).(a) Describe how a client might leave a suitable server in state FIN_WAIT_2 indefinitely. What feature of the
Request for Comments 1122 states (of TCP):A host MAY implement a “half-duplex” TCP close sequence, so that an application that has called CLOSE cannot continue to read data from the connection.
When TCP sends a {SYN, SequenceNum = x} or {FIN, SequenceNum =x}, the consequent ACK has Acknowledgment = x +1; that is, SYNs and FINs each take up one unit in sequence number space. Is this
Find out the generic format for TCP header options from Request for Comments 793.(a) Outline a strategy that would expand the space available for options beyond the current limit of 44 bytes.(b)
The TCP header does not have a boot ID field. Why is there no problem with one end of a TCP connection crashing and rebooting then sending a message with an ID it had previously used?
Suppose we were to implement remote file system mounting using an unreliable RPC protocol that offers zero-or-more semantics. If a message reply is received, this improves to at-least-once
The RPC-based “NFS” remote file system is sometimes considered to have slower-than-expected write performance. In NFS, a server’s RPC reply to a client write request means that the data are
Consider a client and server using an RPC mechanism that includes a channel abstraction and boot IDs.(a) Give a scenario involving server reboot in which an RPC request is sent twice by the client
Suppose an RPC request is of the form “increment the value of field X of disk block N by 10%.” Specify a mechanism to be used by the executing server to guarantee that an arriving request is
Consider a SunRPC client sending a request to a server.(a) Under what circumstances can the client be sure its request has executed exactly once?(b) Suppose we wished to add at-most-once semantics to
Suppose TCP were to be used as the underlying transport in an RPC protocol; each TCP connection is to carry a sequential stream of requests and replies. What analog, if any, would TCP have for:(a)
Write a test program that uses the socket interface to send messages between a pair of Unix workstations connected by some LAN (e.g., Ethernet, ATM, or FDDI). Use this test program to perform the
Try to find situations where an RTP application might reasonably do the following:■ Send multiple packets at essentially the same time that need different timestamps.■ Send packets at different
Having the timestamp clock count time in units of one frame time or one voice sample time would be the minimum resolution to ensure accurate playback. But the time unit is usually considerably
Suppose we want returning RTCP reports from receivers to amount to no more than 5% of the outgoing primary RTP stream. If each report is 84 bytes, the RTP traffic is 20 kBps, and there are 1000
RFC 3550 specifies that the time interval between receiver RTCP reports include a randomization factor to avoid having all the receivers send at the same time. If all the receivers sent in the same
What might a server actually do with the packet loss rate data and jitter data in receiver reports?
Video applications typically run over UDP rather than TCP because they cannot tolerate retransmission delays. However, this means video applications are not constrained by TCP’s congestion control
Propose a mechanism for deciding when to report an RTP packet as lost. How does your mechanism compare with the TCP adaptive retransmission mechanisms?
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
TCP uses a host-centric, feedback-based, window-based resource allocation model. How might TCP have been designed to use instead the following models?(a) Host-centric, feedback-based, and
Sketch curves for throughput, delay, and power, each as a function of load, for the following networks. Throughput is to be measured as a percentage of the maximum. Load is to be measured (somewhat
Suppose two hosts A and B are connected via a router R. The A–R link has infinite bandwidth; the R–B link can send one packet per second.R’s queue is infinite. Load is to be measured as the
Is it possible for TCP Reno to reach a state with the congestion window size much larger than (e.g., twice as large as) RTT × bandwidth?Is it likely?
Consider the arrangement of host H and routers R and R1 in Figure 6.28. All links are full duplex, and all routers are faster than their links. Show that R1 cannot become congested, and for any other
Suppose a congestion control scheme results in a collection of competing flows that achieve the following throughput rates: 100 kBps, 60 kBps, 110 kBps, 95 kBps, and 150 kBps.(a) Calculate the
In fair queuing, the value Fi was interpreted as a timestamp: the time when the ith packet would finish transmitting. Give an interpretation of Fi for weighted fair queuing, and also give a formula
Give an example of how nonpreemption in the implementation of fair queuing leads to a different packet transmission order from bitby-bit round-robin service.
Suppose a router has three input flows and one output. It receives the packets listed in Table 6.1 all at about the same time, in the order listed, during a period in which the output port is busy
Suppose a router has three input flows and one output. It receives the packets listed in Table 6.2 all at about the same time, in the order listed, during a period in which the output port is busy
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
Two users, one using Telnet and one sending files with FTP, both send their traffic out via router R. The outbound link from R is slow enough that both users keep packets in R’s queue at all times.
Consider a router that is managing three flows, on which packets of constant size arrive at the following wall clock times:All three flows share the same outbound link, on which the router can
Consider a router that is managing three flows, on which packets of constant size arrive at the following wall clock times:All three flows share the same outbound link, on which the router can
Assume that TCP implements an extension that allows window sizes much larger than 64 kB. Suppose that you are using this extended TCP over a 1-Gbps link with a latency of 100 ms to transfer a 10-MB
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
For the situation given in the previous problem, compute the effective throughput achieved by this connection. Assume that each packet holds 1 kB of data and that the RTT = 100 ms.Exercise
During linear increase, TCP computes an increment to the congestion window asExplain why computing this increment each time an ACK arrives may not result in the correct increment. Give a more precise
Under what circumstances may coarse-grained timeouts still occur in TCP even when the fast retransmit mechanism is being used?
Suppose that between A and B there is a router R. The A–R bandwidth is infinite (that is, packets are not delayed), but the R–B link introduces a bandwidth delay of one packet per second (that
Suppose A, R, and B are as in the previous exercise, except that R’s queue now has a size of three packets in addition to the one being transmitted. A starts a connection using slow start with an
Suppose the R–B link in the previous exercise changes from a bandwidth delay to a propagation delay so that two packets now take 1 second to send. List what is sent and received during the first 8
Suppose host A reaches host B via routers R1 and R2: A–R1–R2–B.Fast retransmit is not used, and A calculates TimeOut as 2 × EstimatedRTT. Assume that the A–R1 and R2–B links have infinite
You are an Internet service provider; your client hosts connect directly to your routers. You know some hosts are using experimental TCPs and suspect some may be using a “greedy” TCP with no
Defeating TCP congestion control mechanisms usually requires explicit cooperation of the sender. However, consider the receiving end of a large data transfer using a TCP modified to ACK packets that
Consider the TCP trace in Figure 6.29. Identify time intervals representing slow start on startup, slow start after timeout, and linearincrease congestion avoidance. Explain what is going on from T
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
Suppose TCP is used over a lossy link that loses on average one segment in four. Assume the bandwidth × delay window size is considerably larger than four segments.(a) What happens when we start a
Suppose two TCP connections share a path through a router R. The router’s queue size is six segments; each connection has a stable congestion window of three segments. No congestion control is used
Suppose a TCP connection has a window size of eight segments and an RTT of 800 ms, the sender sends segments at a regular rate of one every 100 ms, and the receiver sends ACKs back at the same rate
The text states that additive increase is a necessary condition for a congestion control mechanism to be stable. Outline a specific instability that might arise if all increases were exponential;
Discuss the relative advantages and disadvantages of marking a packet (as in the DECbit mechanism) versus dropping a packet (as in RED gateways).
Consider a RED gateway with MaxP = 0.02 and with an average queue length halfway between the two thresholds.(a) Find the drop probability Pcount for count = 1 and count = 50.(b) Calculate the
Consider a RED gateway with MaxP = p and with an average queue length halfway between the two thresholds.(a) Calculate the probability that none of the first n packets are dropped.(b) Find p such
Explain the intuition behind setting MaxThreshold = 2 × MinThreshold in RED gateways.
In RED gateways, explain why MaxThreshold is actually less than the actual size of the available buffer pool.
Explain the fundamental conflict between tolerating burstiness and controlling network congestion.
Why do you think that the drop probability P of a RED gateway does not simply increase linearly from P = 0 at MinThresh to P = 1 at MaxThresh?
In TCP Vegas, the calculation of ActualRate is done by dividing the amount of data transmitted in one RTT interval by the length of the RTT.(a) Show that for any TCP, if the window size remains
Suppose a TCP Vegas connection measures the RTT of its first packet and sets BaseRTT to that, but then a network link failure occurs and all subsequent traffic is routed via an alternative path with
Consider the following two causes of a 1-second network delay(assume ACKs return instantaneously):■ one intermediate router with a 1-second outbound per-packet bandwidth delay and no competing
Give an argument why the congestion control problem is better managed at the internet level than the ATM level, at least when only part of the internet is ATM. In an exclusively IP-over-ATM network,
Consider the taxonomy of Figure 6.24.(a) Give an example of a real-time application that is intolerant/rate adaptive.(b) Explain why you might expect a loss-tolerant application to be at least
The transmission schedule (Table 6.3) for a given flow lists for each second the number of packets sent between that time and the following second. The flow must stay within the bounds of a token
The transmission schedule (Table 6.4) for a given flow is for each second the number of packets sent between that time and the following second. The flow must stay within the bounds of a token bucket
Suppose a router has accepted flows with the TSpecs shown in Table 6.5, described in terms of token bucket filters with token rate r packets per second and bucket depth B packets. All flows are in
Suppose an RSVP router suddenly loses its reservation state but otherwise remains running.(a) What will happen to the existing reserved flows if the router handles reserved and nonreserved flows via
Consider the following C code:where num_raises + 1 corresponds to the number of valid entries in array salary_history. Show the on-the-wire representation of employee0 that is generated by XDR.
Show the on-the-wire representation of employee1 from the previous problem that is generated by XDR.Exercise 1where num_raises + 1 corresponds to the number of valid entries in array salary_history.
For the data structures given in the previous problem, give the XDR routine that encodes/decodes these structures. If you have XDR available to you, run this routine and measure how long it takes to
Using library functions like htonl and Unix’s bcopy or Windows’CopyMemory, implement a routine that generates the same onthe-wire representation of the structures given in Exercise 1 as XDR does.
Use XDR and htonl to encode a 1000-element array of integers. Measure and compare the performance of each. How do these compare to a simple loop that reads and writes a 1000-element array of
Write your own implementation of htonl. Using both your own htonl and (if little-endian hardware is available) the standard library version, run appropriate experiments to determine how much longer
Give the ASN.1 encoding for the following three integers. Note that ASN.1 integers, like those in XDR, are 32 bits in length.(a) 101.(b) 10,120.(c) 16,909,060.
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