chapter 3: transport layer - uoacgi.di.uoa.gr/~shadj/plh36/tcp_presentation_kurose.pdf · chapter...

1

3: Transport Layer 3a-1

Chapter 3: Transport LayerChapter goals:❒ understand principles

behind transport layerservices:

❍ multiplexing/demultiplexing

❍ reliable data transfer❍ flow control❍ congestion control

❒ instantiation andimplementation in theInternet

Chapter Overview:❒ transport layer services❒ multiplexing/demultiplexing❒ connectionless transport: UDP❒ principles of reliable data

transfer❒ connection-oriented transport:

TCP❍ reliable transfer❍ flow control❍ connection management

❒ principles of congestion control❒ TCP congestion control


Transport services and protocols

❒ provide logical communicationbetween app’ processesrunning on different hosts

❒ transport protocols run inend systems

❒ transport vs network layerservices:

❒ network layer: data transferbetween end systems

❒ transport layer: datatransfer between processes

❍ relies on, enhances, networklayer services

applicationtransportnetworkdata linkphysical


networkdata linkphysical



networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

2


Transport-layer protocols

Internet transport services:❒ reliable, in-order unicast

delivery (TCP)❍ congestion❍ flow control❍ connection setup

❒ unreliable (“best-effort”),unordered unicast ormulticast delivery: UDP

❒ services not available:❍ real-time❍ bandwidth guarantees❍ reliable multicast






networkdata linkphysicalnetwork

data linkphysical

logical end-end transport


applicationtransportnetwork

M P2applicationtransportnetwork

Multiplexing/demultiplexingRecall: segment - unit of data

exchanged betweentransport layer entities

❍ aka TPDU: transportprotocol data unit

receiver

HtHn

Demultiplexing: delivering received segments to correct app layer processes

segmentsegment M

applicationtransportnetwork

P1M

M MP3 P4

segmentheader

application-layerdata

3


Multiplexing/demultiplexing

multiplexing/demultiplexing:❒ based on sender, receiver

port numbers, IP addresses❍ source, dest port #s in

each segment❍ recall: well-known port

numbers for specificapplications

gathering data from multiple app processes, enveloping data with header (later used for demultiplexing)

source port # dest port #

32 bits

applicationdata

(message)

other header fields

TCP/UDP segment format

Multiplexing:


Multiplexing/demultiplexing: examples

host A server Bsource port: xdest. port: 23

source port:23dest. port: x

port use: simple telnet app

Web clienthost A

Webserver B

Web clienthost C

Source IP: CDest IP: B

source port: xdest. port: 80

Source IP: CDest IP: B

source port: ydest. port: 80

port use: Web server

Source IP: ADest IP: B

source port: xdest. port: 80

4


UDP: User Datagram Protocol [RFC 768]

❒ “no frills,” “bare bones”Internet transportprotocol

❒ “best effort” service, UDPsegments may be:

❍ lost❍ delivered out of order

to app❒ connectionless:

❍ no handshaking betweenUDP sender, receiver

❍ each UDP segmenthandled independentlyof others

Why is there a UDP?❒ no connection

establishment (which canadd delay)

❒ simple: no connection stateat sender, receiver

❒ small segment header❒ no congestion control: UDP

can blast away as fast asdesired


UDP: more❒ often used for streaming

multimedia apps❍ loss tolerant❍ rate sensitive

❒ other UDP uses(why?):

❍ DNS❍ SNMP

❒ reliable transfer over UDP:add reliability atapplication layer

❍ application-specificerror recover!


32 bits

Applicationdata

(message)

UDP segment format

length checksumLength, in

bytes of UDPsegment,including

header

5


UDP checksum

Sender:❒ treat segment contents

as sequence of 16-bitintegers

❒ checksum: addition (1’scomplement sum) ofsegment contents

❒ sender puts checksumvalue into UDP checksumfield

Receiver:❒ compute checksum of

received segment❒ check if computed checksum

equals checksum field value:❍ NO - error detected❍ YES - no error detected.

But maybe errorsnonethless? More later ….

Goal: detect “errors” (e.g., flipped bits) in transmittedsegment


Principles of Reliable data transfer❒ important in app., transport, link layers❒ top-10 list of important networking topics!

❒ characteristics of unreliable channel will determinecomplexity of reliable data transfer protocol (rdt)

6


Reliable data transfer: getting started

sendside

receiveside

rdt_send(): called from above,(e.g., by app.). Passed data to

deliver to receiver upper layer

udt_send(): called by rdt,to transfer packet over

unreliable channel to receiver

rdt_rcv(): called when packetarrives on rcv-side of channel

deliver_data(): called byrdt to deliver data to upper


Reliable data transfer: getting startedWe’ll:❒ incrementally develop sender, receiver sides of

reliable data transfer protocol (rdt)❒ consider only unidirectional data transfer

❍ but control info will flow on both directions!❒ use finite state machines (FSM) to specify

sender, receiver

state1

state2

event causing state transitionactions taken on state transition

state: when in this“state” next state

uniquely determinedby next event

eventactions

7


Rdt1.0: reliable transfer over a reliable channel

❒ underlying channel perfectly reliable❍ no bit erros❍ no loss of packets

❒ separate FSMs for sender, receiver:❍ sender sends data into underlying channel❍ receiver read data from underlying channel


Rdt2.0: channel with bit errors

❒ underlying channel may flip bits in packet❍ recall: UDP checksum to detect bit errors

❒ the question: how to recover from errors:❍ acknowledgements (ACKs): receiver explicitly tells sender

that pkt received OK❍ negative acknowledgements (NAKs): receiver explicitly

tells sender that pkt had errors❍ sender retransmits pkt on receipt of NAK❍ human scenarios using ACKs, NAKs?

❒ new mechanisms in rdt2.0 (beyond rdt1.0):❍ error detection❍ receiver feedback: control msgs (ACK,NAK) rcvr->sender

8


rdt2.0: FSM specification

sender FSM receiver FSM


rdt2.0: in action (no errors)


9


rdt2.0: in action (error scenario)



rdt2.0 has a fatal flaw!

What happens ifACK/NAK corrupted?

❒ sender doesn’t know whathappened at receiver!

❒ san’t just retransmit:possible duplicate

What to do?❒ sender ACKs/NAKs

receiver’s ACK/NAK? Whatif sender ACK/NAK lost?

❒ retransmit, but this mightcause retransmission ofcorrectly received pkt!

Handling duplicates:❒ sender adds sequence

number to each pkt❒ sender retransmits current

pkt if ACK/NAK garbled❒ receiver discards (doesn’t

deliver up) duplicate pkt

Sender sends one packet, then waits for receiver response

stop and wait

10


rdt2.1: sender, handles garbled ACK/NAKs


rdt2.1: receiver, handles garbled ACK/NAKs

11


rdt2.1: discussion

Sender:❒ seq # added to pkt❒ two seq. #’s (0,1) will

suffice. Why?❒ must check if received

ACK/NAK corrupted❒ twice as many states

❍ state must “remember”whether “current” pkthas 0 or 1 seq. #

Receiver:❒ must check if received

packet is duplicate❍ state indicates whether

0 or 1 is expected pktseq #

❒ note: receiver can notknow if its lastACK/NAK received OKat sender


rdt2.2: a NAK-free protocol

❒ same functionality asrdt2.1, using NAKs only

❒ instead of NAK,receiver sends ACK forlast pkt received OK

❍ receiver must explicitlyinclude seq # of pktbeing ACKed

❒ duplicate ACK atsender results in sameaction as NAK:retransmit current pkt

senderFSM

!

12


rdt3.0: channels with errors and loss

New assumption:underlying channel canalso lose packets (dataor ACKs)

❍ checksum, seq. #, ACKs,retransmissions will beof help, but not enough

Q: how to deal with loss?❍ sender waits until

certain data or ACKlost, then retransmits

❍ yuck: drawbacks?

Approach: sender waits“reasonable” amount oftime for ACK

❒ retransmits if no ACKreceived in this time

❒ if pkt (or ACK) just delayed(not lost):

❍ retransmission will beduplicate, but use of seq.#’s already handles this

❍ receiver must specify seq# of pkt being ACKed

❒ requires countdown timer


rdt3.0 sender

13


rdt3.0 in action


rdt3.0 in action

14


Performance of rdt3.0

❒ rdt3.0 works, but performance stinks❒ example: 1 Gbps link, 15 ms e-e prop. delay, 1KB packet:

Ttransmit = 8kb/pkt10**9 b/sec = 8 microsec

Utilization = U = = 8 microsec30.016 msec

fraction of timesender busy sending = 0.00015

❍ 1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link❍ network protocol limits use of physical resources!


Pipelined protocolsPipelining: sender allows multiple, “in-flight”, yet-to-

be-acknowledged pkts❍ range of sequence numbers must be increased❍ buffering at sender and/or receiver

❒ Two generic forms of pipelined protocols: go-Back-N,selective repeat

15


Go-Back-NSender:❒ k-bit seq # in pkt header❒ “window” of up to N, consecutive unack’ed pkts allowed

❒ ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK”❍ may deceive duplicate ACKs (see receiver)

❒ timer for each in-flight pkt❒ timeout(n): retransmit pkt n and all higher seq # pkts in window


GBN: sender extended FSM

16


GBN: receiver extended FSM

receiver simple:❒ ACK-only: always send ACK for correctly-received

pkt with highest in-order seq #❍ may generate duplicate ACKs❍ need only remember expectedseqnum

❒ out-of-order pkt:❍ discard (don’t buffer) -> no receiver buffering!❍ ACK pkt with highest in-order seq #


GBN inaction

17


Selective Repeat

❒ receiver individually acknowledges all correctlyreceived pkts

❍ buffers pkts, as needed, for eventual in-order deliveryto upper layer

❒ sender only resends pkts for which ACK notreceived

❍ sender timer for each unACKed pkt❒ sender window

❍ N consecutive seq #’s❍ again limits seq #s of sent, unACKed pkts


Selective repeat: sender, receiver windows

18


Selective repeat

data from above :❒ if next available seq # in

window, send pkttimeout(n):❒ resend pkt n, restart timerACK(n) in [sendbase,sendbase+N]:

❒ mark pkt n as received❒ if n smallest unACKed pkt,

advance window base tonext unACKed seq #

senderpkt n in [rcvbase, rcvbase+N-1]

❒ send ACK(n)❒ out-of-order: buffer❒ in-order: deliver (also

deliver buffered, in-orderpkts), advance window tonext not-yet-received pkt

pkt n in [rcvbase-N,rcvbase-1]

❒ ACK(n)otherwise:❒ ignore

receiver


Selective repeat in action

19


Selective repeat: dilemmaExample:❒ seq #’s: 0, 1, 2, 3❒ window size=3

❒ receiver sees nodifference in twoscenarios!

❒ incorrectly passesduplicate data as newin (a)

Q: what relationshipbetween seq # sizeand window size?

1

3: Transport Layer 3b-1

TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581

❒ full duplex data:❍ bi-directional data flow

in same connection❍ MSS: maximum segment

size❒ connection-oriented:

❍ handshaking (exchangeof control msgs) init’ssender, receiver statebefore data exchange

❒ flow controlled:❍ sender will not

overwhelm receiver

❒ point-to-point:❍ one sender, one receiver

❒ reliable, in-order bytesteam:

❍ no “message boundaries”❒ pipelined:

❍ TCP congestion and flowcontrol set window size

❒ send & receive buffers

socketdoor

TCPsend buffer

TCPreceive buffer

socketdoor

segment

applicationwrites data

applicationreads data


TCP segment structure


32 bits

applicationdata

(variable length)

sequence numberacknowledgement number

rcvr window sizeptr urgent datachecksum

FSRPAUheadlen

notused

Options (variable length)

URG: urgent data (generally not used)

ACK: ACK #valid

PSH: push data now(generally not used)

RST, SYN, FIN:connection estab(setup, teardown

commands)

# bytes rcvr willingto accept

countingby bytes of data(not segments!)

Internetchecksum

(as in UDP)

2


TCP seq. #’s and ACKsSeq. #’s:

❍ byte stream“number” of firstbyte in segment’sdata

ACKs:❍ seq # of next byte

expected fromother side

❍ cumulative ACKQ: how receiver handles

out-of-order segments❍ A: TCP spec doesn’t

say, - up toimplementor

Host A Host B

Seq=42, ACK=79, data = ‘C’

Seq=79, ACK=43, data = ‘C’

Seq=43, ACK=80

Usertypes

‘C’

host ACKsreceipt

of echoed‘C’

host ACKsreceipt of‘C’, echoes

back ‘C’

timesimple telnet scenario


TCP: reliable data transfer

simplified sender, assuming

waitfor

event

waitfor

event

event: data received from application above

event: timer timeout for segment with seq # y

event: ACK received,with ACK # y

create, send segment

retransmit segment

ACK processing

•one way data transfer•no flow, congestion control

3


TCP:reliabledatatransfer

00 sendbase = initial_sequence number 01 nextseqnum = initial_sequence number 0203 loop (forever) { 04 switch(event) 05 event: data received from application above 06 create TCP segment with sequence number nextseqnum 07 start timer for segment nextseqnum 08 pass segment to IP 09 nextseqnum = nextseqnum + length(data) 10 event: timer timeout for segment with sequence number y 11 retransmit segment with sequence number y 12 compue new timeout interval for segment y 13 restart timer for sequence number y 14 event: ACK received, with ACK field value of y 15 if (y > sendbase) { /* cumulative ACK of all data up to y */ 16 cancel all timers for segments with sequence numbers < y 17 sendbase = y 18 } 19 else { /* a duplicate ACK for already ACKed segment */ 20 increment number of duplicate ACKs received for y 21 if (number of duplicate ACKS received for y == 3) { 22 /* TCP fast retransmit */ 23 resend segment with sequence number y 24 restart timer for segment y 25 } 26 } /* end of loop forever */

SimplifiedTCPsender


TCP ACK generation [RFC 1122, RFC 2581]

Event

in-order segment arrival, no gaps,everything else already ACKed

in-order segment arrival, no gaps,one delayed ACK pending

out-of-order segment arrivalhigher-than-expect seq. #gap detected

arrival of segment that partially or completely fills gap

TCP Receiver action

delayed ACK. Wait up to 500msfor next segment. If no next segment,send ACK

immediately send singlecumulative ACK

send duplicate ACK, indicating seq. #of next expected byte

immediate ACK if segment startsat lower end of gap

4


TCP: retransmission scenariosHost A

Seq=92, 8 bytes data

ACK=100

loss

tim

eout

time lost ACK scenario

Host B

X


ACK=100

Host A


ACK=100

Seq=

92 t

imeo

ut

time premature timeout,cumulative ACKs

Host B


ACK=120


Seq=

100

tim

eout

ACK=120


TCP Flow Controlreceiver: explicitly

informs sender of(dynamically changing)amount of free bufferspace

❍ RcvWindow field inTCP segment

sender: keeps the amountof transmitted,unACKed data less thanmost recently receivedRcvWindow

sender won’t overrunreceiver’s buffers by

transmitting too much, too fast

flow control

receiver buffering

RcvBuffer = size or TCP Receive Buffer

RcvWindow = amount of spare room in Buffer

5


TCP Round Trip Time and Timeout

Q: how to set TCPtimeout value?

❒ longer than RTT❍ note: RTT will vary

❒ too short: prematuretimeout

❍ unnecessaryretransmissions

❒ too long: slow reactionto segment loss

Q: how to estimate RTT?❒ SampleRTT: measured time from

segment transmission until ACKreceipt

❍ ignore retransmissions,cumulatively ACKed segments

❒ SampleRTT will vary, wantestimated RTT “smoother”

❍ use several recentmeasurements, not justcurrent SampleRTT


TCP Round Trip Time and Timeout

EstimatedRTT = (1-x)*EstimatedRTT + x*SampleRTT

❒ Exponential weighted moving average❒ influence of given sample decreases exponentially fast❒ typical value of x: 0.1

Setting the timeout❒ EstimtedRTT plus “safety margin”❒ large variation in EstimatedRTT -> larger safety margin

Timeout = EstimatedRTT + 4*Deviation

Deviation = (1-x)*Deviation + x*|SampleRTT-EstimatedRTT|

6


TCP Connection Management

Recall: TCP sender, receiverestablish “connection”before exchanging datasegments

❒ initialize TCP variables:❍ seq. #s❍ buffers, flow control

info (e.g. RcvWindow)❒ client: connection initiator Socket clientSocket = new

Socket("hostname","portnumber");

❒ server: contacted by client Socket connectionSocket =

welcomeSocket.accept();

Three way handshake:Step 1: client end system

sends TCP SYN controlsegment to server

❍ specifies initial seq #

Step 2: server end systemreceives SYN, replies withSYNACK control segment

❍ ACKs received SYN❍ allocates buffers❍ specifies server->

receiver initial seq. #


TCP Connection Management (cont.)

Closing a connection:client closes socket:

clientSocket.close();

Step 1: client end systemsends TCP FIN controlsegment to server

Step 2: server receivesFIN, replies with ACK.Closes connection, sendsFIN.

client

FIN

server

ACK

ACK

FIN

close

close

closed

tim

ed w

ait

7


TCP Connection Management (cont.)

Step 3: client receives FIN,replies with ACK.

❍ Enters “timed wait” -will respond with ACKto received FINs

Step 4: server, receivesACK. Connection closed.

Note: with smallmodification, can handlysimultaneous FINs.

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med

wai

t

closed


TCP Connection Management (cont)

TCP clientlifecycle

TCP serverlifecycle

8


Principles of Congestion Control

Congestion:❒ informally: “too many sources sending too much

data too fast for network to handle”❒ different from flow control!❒ manifestations:

❍ lost packets (buffer overflow at routers)❍ long delays (queueing in router buffers)

❒ a top-10 problem!


Causes/costs of congestion: scenario 1

❒ two senders, tworeceivers

❒ one router,infinite buffers

❒ no retransmission

❒ large delayswhen congested

❒ maximumachievablethroughput

9



❒ one router, finite buffers❒ sender retransmission of lost packet


Causes/costs of congestion: scenario 2❒ always: (goodput)❒ “perfect” retransmission only when loss:

❒ retransmission of delayed (not lost) packet makes larger(than perfect case) for same

�in

�out=

�in

�out>�

in�out

“costs” of congestion:❒ more work (retrans) for given “goodput”❒ unneeded retransmissions: link carries multiple copies of pkt

10


Causes/costs of congestion: scenario 3❒ four senders❒ multihop paths❒ timeout/retransmit

�in

Q: what happens asand increase ?�

in



Another “cost” of congestion:❒ when packet dropped, any “upstream transmission

capacity used for that packet was wasted!

11


Approaches towards congestion control

End-end congestioncontrol:

❒ no explicit feedback fromnetwork

❒ congestion inferred fromend-system observed loss,delay

❒ approach taken by TCP

Network-assistedcongestion control:

❒ routers provide feedbackto end systems

❍ single bit indicatingcongestion (SNA,DECbit, TCP/IP ECN,ATM)

❍ explicit rate sendershould send at

Two broad approaches towards congestion control:


Case study: ATM ABR congestion control

ABR: available bit rate:❒ “elastic service”❒ if sender’s path

“underloaded”:❍ sender should use

available bandwidth❒ if sender’s path

congested:❍ sender throttled to

minimum guaranteedrate

RM (resource management)cells:

❒ sent by sender, interspersedwith data cells

❒ bits in RM cell set by switches(“network-assisted”)

❍ NI bit: no increase in rate(mild congestion)

❍ CI bit: congestionindication

❒ RM cells returned to sender byreceiver, with bits intact

12


Case study: ATM ABR congestion control

❒ two-byte ER (explicit rate) field in RM cell❍ congested switch may lower ER value in cell❍ sender’ send rate thus minimum supportable rate on path

❒ EFCI bit in data cells: set to 1 in congested switch❍ if data cell preceding RM cell has EFCI set, sender sets CI

bit in returned RM cell


TCP Congestion Control❒ end-end control (no network assistance)❒ transmission rate limited by congestion window

size, Congwin, over segments:

❒ w segments, each with MSS bytes sent in one RTT:

throughput = w * MSS

RTT Bytes/sec

Congwin

13


TCP congestion control:

❒ two “phases”❍ slow start❍ congestion avoidance

❒ important variables:❍ Congwin

❍ threshold: definesthreshold between twoslow start phase,congestion controlphase

❒ “probing” for usablebandwidth:

❍ ideally: transmit as fastas possible (Congwin aslarge as possible)without loss

❍ increase Congwin untilloss (congestion)

❍ loss: decrease Congwin,then begin probing(increasing) again


TCP Slowstart

❒ exponential increase (perRTT) in window size (not soslow!)

❒ loss event: timeout (TahoeTCP) and/or or threeduplicate ACKs (Reno TCP)

initialize: Congwin = 1for (each segment ACKed) Congwin++until (loss event OR CongWin > threshold)

Slowstart algorithmHost A

one segment

RTT

Host B

time

two segments

four segments

14


TCP Congestion Avoidance

/* slowstart is over *//* Congwin > threshold */Until (loss event) { every w segments ACKed: Congwin++ }threshold = Congwin/2Congwin = 1perform slowstart

Congestion avoidance

1

1: TCP Reno skips slowstart (fast recovery) after three duplicate ACKs


TCP FairnessFairness goal: if N TCP

sessions share samebottleneck link, eachshould get 1/N of linkcapacity

TCP congestionavoidance:

❒ AIMD: additiveincrease,multiplicativedecrease

❍ increase window by 1per RTT

❍ decrease window byfactor of 2 on lossevent

AIMD

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

15


Why is TCP fair?Two competing sessions:❒ Additive increase gives slope of 1, as throughout increases❒ multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughput

Conn

ect i

on 2

thr

ough

put

congestion avoidance: additive increaseloss: decrease window by factor of 2

congestion avoidance: additive increaseloss: decrease window by factor of 2


TCP latency modeling

Q: How long does it take toreceive an object from aWeb server after sendinga request?

❒ TCP connection establishment❒ data transfer delay

Notation, assumptions:❒ Assume one link between

client and server of rate R❒ Assume: fixed congestion

window, W segments❒ S: MSS (bits)❒ O: object size (bits)❒ no retransmissions (no loss,

no corruption)Two cases to consider:❒ WS/R > RTT + S/R: ACK for first segment in

window returns before window’s worth of datasent

❒ WS/R < RTT + S/R: wait for ACK after sendingwindow’s worth of data sent

16


TCP latency Modeling

Case 1: latency = 2RTT + O/R Case 2: latency = 2RTT + O/R+ (K-1)[S/R + RTT - WS/R]

K:= O/WS


TCP Latency Modeling: Slow Start

❒ Now suppose window grows according to slow start.❒ Will show that the latency of one object of size O is:

RS

RSRTTP

RORTTLatency P )12(2 ��

��

��

where P is the number of times TCP stalls at server:

}1,{min �� KQP

- where Q is the number of times the server would stall if the object were of infinite size.

- and K is the number of windows that cover the object.

17


TCP Latency Modeling: Slow Start (cont.)

RTT

initiate TCPconnection

requestobject

first window= S/R

second window= 2S/R

third window= 4S/R

fourth window= 8S/R

completetransmissionobject

delivered

time atclient

time atserver

Example:

O/S = 15 segments

K = 4 windows

Q = 2

P = min{K-1,Q} = 2

Server stalls P=2 times.


TCP Latency Modeling: Slow Start (cont.)

RS

RSRTTPRTT

RO

RSRTT

RSRTT

RO

stallTimeRTTRO

P

kP

k

P

pp

)12(][2

]2[2

2latency

1

1

1

��

��

��

�

�

�

�

�

windowth after the timestall 2 1 kRSRTT

RS k ��

��

��

�

�

ementacknowledg receivesserver until

segment send tostartsserver whenfrom time�� RTTRS

window kth the transmit totime2 1�

�

RSk

RTT

initiate TCPconnection

requestobject

first window= S/R

second window= 2S/R

third window= 4S/R

fourth window= 8S/R

completetransmissionobject

delivered

time atclient

time atserver

18


Chapter 3: Summary

❒ principles behindtransport layer services:

❍ multiplexing/demultiplexing❍ reliable data transfer❍ flow control❍ congestion control

❒ instantiation andimplementation in the Internet

❍ UDP❍ TCP

Next:❒ leaving the network

“edge” (applicationtransport layer)

❒ into the network “core”

chapter 3: transport layer - uoacgi.di.uoa.gr/~shadj/plh36/tcp_presentation_kurose.pdf · chapter...

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