1. TCP waits until it has received three duplicate ACK 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? - PowerPoint PPT Presentation

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1. TCP waits until it has received three duplicate ACK 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?

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Why do you think the TCP designers chose not to perform a fast retransmit after ... the indicated link costs, use Dijkstra's shortest path algorithm to compute the ... – PowerPoint PPT presentation

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Title: 1. TCP waits until it has received three duplicate ACK 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?


1
  • 1. TCP waits until it has received three
    duplicate ACK 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?

The designer probably felt that waiting for two
subsequent packets (rather than 1) was the right
tradeoff between triggering a quick
retransmission when needed, but not
retransmitting prematurely in the face of packet
reordering.
2
  • Refer to the figure that illustrates the
    convergence of TCPs additive increase,
    multiplicative decease algorithm. Suppose that
    instead of a multiplicative decrease, TCP
    decreased the window size by a constant amount.
    Would the resulting additive increase additive
    decrease converge to an equal share algorithm?
    Justify your answer using a diagram similar to
    the figure.

equal bandwidth share
R
loss decrease window by factor of 2
congestion avoidance additive increase
Connection 2 throughput
loss decrease window by factor of 2
congestion avoidance additive increase
Connection 1 throughput
R
3
  • Refer to the following figure. In Figure
    (a), the ratio of the linear decrease on loss
    connection 1 and connection2 is the same as
    ratio of the linear increases unity. In this
    case the throughput never move off of the AB line
    segment. In Figure (b) the ratio of the linear
    decrease on loss between connection 1 and
    connection2 is 21. That is, whenever there is a
    loss, connection 1 decreases its window by twice
    the amount of connection 2. We see that
    eventually, after enough losses, and subsequent
    increases, that connection1s throughput will go
    to 0, and the full link bandwidth will be
    allocated to connection 2.

4
  • 3. Recall the idealized model for the
    steady-state dynamics of TCP. In the period of
    time from when the connections window size
    varies from (WMSS)/2 to WMSS, only one packet
    is lost (at the very end of the period).
  • a. Show that the loss rate is equal to
  • L loss rate 1/(3/8w23/4w)
  • b. Use the above result to show that if a
  • connection has loss rate L, then its
    average
  • bandwidth is approximately given by
  • Avg. BW of connection 1.22MSS/RTTsqrt(L)

5
  • Consider the following network. With the
    indicated link costs, use Dijkstras shortest
    path algorithm to compute the shortest path from
    F to all network nodes.

2
14
4
1
9
1
6
2
3
1
1
4
1
3
6
Dijsktras Algorithm
1 Initialization 2 N A 3 for all
nodes v 4 if v adjacent to A 5 then
D(v) c(A,v) 6 else D(v) infty 7 8
Loop 9 find w not in N such that D(w) is a
minimum 10 add w to N 11 update D(v) for
all v adjacent to w and not in N 12 D(v)
min( D(v), D(w) c(w,v) ) 13 / new cost
to v is either old cost to v or known 14
shortest path cost to w plus cost from w to v /
15 until all nodes in N
7
6
8
  • Recall the two FEC schemes for Internet phone.
    Suppose that the first scheme generates a
    redundant chunk for every four original chunks.
    Suppose the second scheme uses a low-bit-rate
    encoding whose transmission rate is 25 of the
    transmission rate of the nominal stream.
  • a. How much additional bandwidth does each
    scheme require? How much playback delay does
    each scheme add?
  • b. How do the two schemes perform if the
    first packet is lost in every group of five
    packets? Which scheme will have better audio
    quality?
  • c. How do the two schemes perform if the
    first packet is lost in every group of two
    packets? Which scheme will have better audio
    quality?

9
  • 6. What would be preemptive priority queueing?
    Does preemptive priority queueing make sense for
    computer networks?
  • 7. Give an example of scheduling discipline that
    is not work conserving.
  • 8. What are some of the difficulties associated
    with the Intserv model and per flow reservation
    of resources?

10
  • 9. A connection has a mean rate of 1 Mbps, a peak
    rate of 10 Mbps, and a delay jitter of 500 ms.
    What is the amount of buffer needed at the
    receiver to remove the delay jitter?
  • 10. Compute the max-min fair allocation for
    sources A, B, C, D and E, when their demands are
    2, 3, 4, 4, 5, and the resource size is 15.
  • 11. Connections A and B are continuously
    backlogged during time 0,1 and have weights 1
    and 4. A receivers 4 Kbits of service in 0,1.
    What service is B guaranteed to receive with FCFS
    and GPS disciplines?

11
  • 12. Packets of length 100 and 200 bits from
    connections A and B arrive at an empty FQ
    scheduler at time 0. If the line rate is 100
    bps, (a) at what real time do the packets
    complete service? (b) what is the corresponding
    round number (virtual time) when each packet
    completes service? (c) If a packet of length 10
    arrives on connection A at real time 1.5 s, what
    would be its finish number?
  • 13. If connections A and B in the above have
    weights of 2 and 5, respectively, recompute parts
    (a), (b) and (c).
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