CS计算机代考程序代写 algorithm dns case study ER Chapter 3 Transport Layer

Chapter 3 Transport Layer
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All material copyright 1996-2007
J.F Kurose and K.W. Ross, All Rights Reserved
Computer Networking: A Top Down Approach , 6th edition.
Jim Kurose, Keith Ross
Transport Layer 3-1

Chapter 3: Transport Layer
Our goals:
 understand principles behind transport layer services:
 reliable data transfer  flow control
 congestion control
 learn about transport layer protocols in the Internet:
 UDP: connectionless transport
 TCP: connection-oriented transport
 TCP congestion control
Transport Layer 3-2

Chapter 3 outline
 3.1 Transport-layer services
 3.3 Connectionless transport: UDP
 3.4 Principles of reliable data transfer
 3.5 Connection-oriented transport: TCP
 segment structure
 reliable data transfer  flow control
 3.6 Principles of congestion control
 3.7 TCP congestion control
Transport Layer 3-3

Transport services and protocols
 provide logical communication between app processes running on different hosts
 transport protocols run in end systems
 send side: breaks app messages into segments, passes to network layer
 rcv side: reassembles segments into messages, passes to app layer
 more than one transport protocol available to apps
 Internet: TCP and UDP
application
transport
network
data link
physical
application
transport
network
data link
physical
Transport Layer 3-4

Transport vs. network layer
 network layer: logical communication between hosts
 transport layer: logical communication between processes
 relies on, enhances, network layer services
Household analogy:
12 kids sending letters to 12 kids
 processes = kids
 app messages = letters
in envelopes
 hosts = houses
 transport protocol = Ann and Bill
 network-layer protocol = postal service
Transport Layer 3-5

Internet transport-layer protocols
 reliable, in-order delivery (TCP)
 congestion control  flow control
 connection setup
 unreliable, unordered delivery: UDP
 no-frills extension of “best-effort” IP
 services not available:  delay guarantees
 bandwidth guarantees
application
transport
network
data link physical
network data link
physical
network
network data link
data link
physical
physical network
data link
data link
physical
network
physical
network
data link
application
transport
network
physical
data link
physical
Transport Layer 3-6

UDP: User Datagram Protocol [RFC 768]
 “no frills,” “bare bones” Internet transport protocol
 “best effort” service, UDP segments may be:
 lost
 delivered out of order
to app
 connectionless:
 no handshaking between
UDP sender, receiver
 each UDP segment handled independently of others
Why is there a UDP?
 no connection establishment (which can add delay)
 simple: no connection state at sender, receiver
 small segment header
 no congestion control: UDP can blast away as fast as desired
Transport Layer 3-8

UDP: more
 often used for streaming multimedia apps
32 bits
source port #
dest port #
length
Application data
(message)
checksum
 loss tolerant  rate sensitive
 other UDP uses  DNS
 SNMP
Length, in bytes of UDP segment, including header
 reliable transfer over UDP: add reliability at application layer
 application-specific error recovery!
UDP segment format
Transport Layer 3-9

UDP checksum
Goal: detect “errors” (e.g., flipped bits) in transmitted segment
Sender:
 treat segment contents as sequence of 16-bit integers
 checksum: addition (1’s complement sum) of segment contents
 sender puts checksum value into UDP checksum field
Receiver:
 compute checksum of received segment
 check if computed checksum equals checksum field value:
 NO – error detected
 YES – no error detected. But maybe errors nonetheless? More later ….
Transport Layer 3-10

Internet Checksum Example
 Note
 When adding numbers, a carryout from the most significant bit needs to be added to the result
 Example: add two 16-bit integers 11110011001100110
11101010101010101 11011101110111011
sum checksum
wraparound
11011101110111100 10100010001000011
Transport Layer 3-11

Principles of Reliable data transfer
 important in app., transport, link layers
 top-10 list of important networking topics!
 characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Transport Layer 3-13

Reliable data transfer: getting started
rdt_send(): called from above, (e.g., by app.). Passed data to deliver to receiver upper layer
deliver_data(): called by rdt to deliver data to upper
send receive side side
udt_send(): called by rdt, to transfer packet over unreliable channel to receiver
rdt_rcv(): called when packet arrives on rcv-side of channel
Transport Layer 3-16

Reliable data transfer: getting started
We’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
event causing state transition actions taken on state transition
event state actions 2
state: when in this “state” next state uniquely determined by next event
state 1
Transport Layer 3-17

Rdt1.0: reliable transfer over a reliable channel
 underlying channel perfectly reliable  no bit errors
 no loss of packets
 separate FSMs for sender, receiver:
 sender sends data into underlying channel  receiver read data from underlying channel
Wait for call from above
rdt_send(data)
packet = make_pkt(data) udt_send(packet)
sender
Wait for call from below
rdt_rcv(packet)
extract (packet,data) deliver_data(data)
receiver
Transport Layer 3-18

Rdt2.0: channel with bit errors
 underlying channel may flip bits in packet  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
 new mechanisms in rdt2.0 (beyond rdt1.0):
 error detection
 receiver feedback: control msgs (ACK,NAK) rcvr->sender
Transport Layer 3-19

rdt2.0: FSM specification
rdt_send(data)
snkpkt = make_pkt(data, checksum)
receiver
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
udt_send(NAK)
Wait for call from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(ACK)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(sndpkt) rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
sender
Wait for call from above
Wait for ACK or NAK
Transport Layer 3-20

rdt2.0: operation with no errors
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(sndpkt)
Wait for call from above
Wait for ACK or NAK
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
udt_send(NAK)
Wait for call from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(ACK)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
Transport Layer 3-21

rdt2.0: error scenario
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(sndpkt)
Wait for call from above
Wait for ACK or NAK
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
udt_send(NAK)
Wait for call from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(ACK)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
Transport Layer 3-22

rdt2.0 has a fatal flaw!
What happens if ACK/NAK corrupted?
 sender doesn’t know what happened at receiver!
 can’t just retransmit: possible duplicate
Handling duplicates:
 sender retransmits current pkt if ACK/NAK garbled
 sender adds sequence number to each pkt
 receiver discards (doesn’t deliver up) duplicate pkt
stop and wait
Sender sends one packet, then waits for receiver response
Transport Layer 3-23

rdt2.1: sender, handles garbled ACK/NAKs
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) )
udt_send(sndpkt)
Wait for call 0 from above
Wait for ACK or NAK 0
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt)
L
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) )
udt_send(sndpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt)
L
Wait for ACK or NAK 1
rdt_send(data)
sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt)
Wait for call 1 from above
Transport Layer 3-24

rdt2.1: receiver, handles garbled ACK/NAKs
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) && has_seq1(rcvpkt)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq0(rcvpkt)
extract(rcvpkt,data) deliver_data(data)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
Wait for 0 from below
Wait for 1 from below
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) && has_seq0(rcvpkt)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt)
extract(rcvpkt,data) deliver_data(data)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
Transport Layer 3-25

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” pkt has 0 or 1 seq. #
Receiver:
 must check if received packet is duplicate
 state indicates whether 0 or 1 is expected pkt seq #
 note: receiver can not know if its last ACK/NAK received OK at sender
Transport Layer 3-26

rdt2.2: a NAK-free protocol
 same functionality as rdt2.1, using ACKs only
 instead of NAK, receiver sends ACK for last pkt received OK
 receiver must explicitly include seq # of pkt being ACKed
 duplicate ACK at sender results in same action as NAK: retransmit current pkt
Transport Layer 3-27

rdt2.2: sender, receiver fragments
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) ||
isACK(rcvpkt,1) ) udt_send(sndpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt,0)
L
Wait for call 0 from above
Wait for ACK
rdt_rcv(rcvpkt) && (corrupt(rcvpkt) ||
has_seq1(rcvpkt)) udt_send(sndpkt)
Wait for 0 from below
receiver FSM fragment
sender FSM fragment
0
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt)
extract(rcvpkt,data) deliver_data(data)
sndpkt = make_pkt(ACK1, chksum) udt_send(sndpkt)
Transport Layer 3-28

rdt3.0: channels with errors and loss
New assumption:
underlying channel can also lose packets (data or ACKs)
 checksum, seq. #, ACKs, retransmissions will be of help, but not enough
Approach: sender waits “reasonable” amount of time for ACK
 retransmits if no ACK received in this time
 if pkt (or ACK) just delayed (not lost):
 retransmission will be duplicate, but use of seq. #’s already handles this
 receiver must specify seq # of pkt being ACKed
 requires countdown timer Transport Layer 3-29

rdt3.0 sender
rdt_send(data)
sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt)
start_timer L
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
rdt_rcv(rcvpkt)
L
Wait for call 0from above
Wait for ACK0
timeout udt_send(sndpkt) start_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt,0)
stop_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt,1)
stop_timer
timeout
udt_send(sndpkt) start_timer
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,0) )
L
Wait for ACK1
Wait for call 1 from above
rdt_rcv(rcvpkt)
L
rdt_send(data)
sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt)
start_timer
Transport Layer 3-30

rdt3.0 in action
Transport Layer 3-31

rdt3.0 in action
Transport Layer 3-32

Performance of rdt3.0
 rdt3.0 works, but performance stinks
 example: 1 Gbps link, 15 ms e-e prop. delay, 1KB packet:
T= transmit
L (packet length in bits) R (transmission rate, bps)
8kb/pkt 10**9 b/sec
=
 U sender: utilization – fraction of time sender busy sending
Usender= L / R = .008 = 0.00027 RTT + L / R 30.008 microsec
onds
 1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link
 network protocol limits use of physical resources!
Transport Layer 3-33
= 8 microsec

rdt3.0: stop-and-wait operation
sender first packet bit transmitted, t = 0
last packet bit transmitted, t = L / R RTT
ACK arrives, send next packet, t = RTT + L / R
Usender= L / R RTT + L / R
receiver
first packet bit arrives
last packet bit arrives, send ACK
=
.008 30.008
=
0.00027 microsec onds
Transport Layer 3-34

Pipelined protocols
Pipelining: 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
Transport Layer 3-35

Pipelining: increased utilization
sender
receiver
first packet bit transmitted, t = 0 last bit transmitted, t = L / R
RTT
ACK arrives, send next packet, t = RTT + L / R
first packet bit arrives
last packet bit arrives, send ACK
last bit of 2nd packet arrives, send ACK last bit of 3rd packet arrives, send ACK
Increase utilization by a factor of 3!
Usender= 3*L/R = .024 = 0.0008 RTT + L / R 30.008 microsecon
ds
Transport Layer 3-36

Go-Back-N
Sender:
 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 receive duplicate ACKs (see receiver)
 timer for each in-flight pkt
 timeout(n): retransmit pkt n and all higher seq # pkts in window
Transport Layer 3-37

GBN: sender extended FSM
L
base=1 nextseqnum=1
rdt_rcv(rcvpkt)
&& corrupt(rcvpkt)
rdt_send(data)
if (nextseqnum < base+N) { sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ } else refuse_data(data) Wait timeout start_timer udt_send(sndpkt[base]) udt_send(sndpkt[base+1]) ... udt_send(sndpkt[nextseqnum-1]) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) base = getacknum(rcvpkt)+1 If (base == nextseqnum) stop_timer else start_timer Transport Layer 3-38 GBN: receiver extended FSM L expectedseqnum=1 sndpkt = default udt_send(sndpkt) Wait rdt_rcv(rcvpkt) && notcurrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpkt) expectedseqnum++ make_pkt(expectedseqnum,ACK,chksum) 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!  Re-ACK pkt with highest in-order seq #
Transport Layer 3-39

GBN in action
Transport Layer 3-40

Selective Repeat
 receiver individually acknowledges all correctly received pkts
 buffers pkts, as needed, for eventual in-order delivery to upper layer
 sender only resends pkts for which ACK not received
 sender timer for each unACKed pkt
 sender window
 N consecutive seq #’s
 again limits seq #s of sent, unACKed pkts
Transport Layer 3-41

Selective repeat: sender, receiver windows
Transport Layer 3-42

Selective repeat
sender
data from above :
 if next available seq # in window, send pkt
timeout(n):
 resend pkt n, restart timer
ACK(n) in [sendbase,sendbase+N]:
 mark pkt n as received
 if n smallest unACKed pkt, advance window base to next unACKed seq #
receiver
pkt n in [rcvbase, rcvbase+N-1]
 send ACK(n)
 out-of-order: buffer
 in-order: deliver (also deliver buffered, in-order pkts), advance window to next not-yet-received pkt
pkt n in [rcvbase-N,rcvbase-1]
 ACK(n)
otherwise:
 ignore
Transport Layer 3-43

Selective repeat in action
Transport Layer 3-44

Selective repeat: dilemma
Example:
 seq #’s: 0, 1, 2, 3
 window size=3
 receiver sees no difference in two scenarios!
 incorrectly passes duplicate data as new in (a)
Q: what relationship between seq # size and window size?
Transport Layer 3-45

TCP: Overview
 point-to-point:
 one sender, one receiver
 reliable, in-order byte steam:
 no “message boundaries”  pipelined:
 TCP congestion and flow control set window size
 send & receive buffers
RFCs: 793, 1122, 1323, 2018, 2581
 full duplex data:
 bi-directional data flow
in same connection
 MSS: maximum segment size
 connection-oriented:
 handshaking (exchange of control msgs) init’s sender, receiver state before data exchange
 flow controlled: socket  sender will not
door overwhelm receiver Transport Layer 3-47
socket door
application writes data
application reads data
TCP send buffer
segment
TCP receive buffer

TCP segment structure
32 bits
URG: urgent data (generally not used)
ACK: ACK # valid
PSH: push data now (generally not used)
RST, SYN, FIN: connection estab (setup, teardown
commands)
Internet checksum (as in UDP)
counting
by bytes
of data
(not segments!)
# bytes rcvr willing to accept
source port #
head len
not used
sequence number
acknowledgement number
dest port #
checksum
U
A
P
R
S
F
Options (variable length)
application data
(variable length)
Receive window
Urg data pnter
Transport Layer 3-48

TCP seq. #’s and ACKs
Seq. #’s:
 byte stream “number” of first byte in segment’s data
ACKs:
 seq # of next byte expected from other side
 cumulative ACK Q: how receiver handles
out-of-order segments
 A: TCP spec doesn’t say, – up to implementor
Host A
Host B
host ACKs receipt of ‘C’, echoes back ‘C’
User types ‘C’
host ACKs receipt
of echoed ‘C’
simple telnet scenario
Transport Layer 3-49
time

TCP Round Trip Time and Timeout
Q: how to set TCP timeout value?
 longer than RTT  but RTT varies
 too short: premature timeout
 unnecessary retransmissions
 too long: slow reaction to segment loss
Q: how to estimate RTT?
 SampleRTT: measured time from segment transmission until ACK receipt
 ignore retransmissions  SampleRTT will vary, want
estimated RTT “smoother”
 average several recent measurements, not just current SampleRTT
Transport Layer 3-50

TCP Round Trip Time and Timeout
EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT
 Exponential weighted moving average
 influence of past sample decreases exponentially fast  typical value:  = 0.125
Transport Layer 3-51

Example RTT estimation:
350
300
250
200
150
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
100
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
SampleRTT Estimated RTT
Transport Layer 3-52
RTT (milliseconds)

TCP Round Trip Time and Timeout
Setting the timeout
 EstimtedRTT plus “safety margin”
 large variation in EstimatedRTT -> larger safety margin
 first estimate of how much SampleRTT deviates from EstimatedRTT:
DevRTT = (1-)*DevRTT + *|SampleRTT-EstimatedRTT|
(typically,  = 0.25) Then set timeout interval:
TimeoutInterval = EstimatedRTT + 4*DevRTT
Transport Layer 3-53

TCP reliable data transfer
 TCP creates rdt service on top of IP’s unreliable service
 Pipelined segments
 Cumulative acks
 TCP uses single retransmission timer
 Retransmissions are triggered by:
 timeout events  duplicate acks
 Initially consider simplified TCP sender:
 ignore duplicate acks  ignore flow control,
congestion control
Transport Layer 3-55

TCP sender events:
data rcvd from app:
 Create segment with seq #
 seq # is byte-stream number of first data byte in segment
 start timer if not already running (think of timer as for oldest unacked segment)
 expiration interval: TimeOutInterval
timeout:
 retransmit segment that caused timeout
 restart timer Ack rcvd:
 If acknowledges previously unacked segments
 update what is known to be acked
 start timer if there are outstanding segments
Transport Layer 3-56

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum
loop (forever) {
switch(event)
event: data received from application above
create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timer
pass segment to IP
NextSeqNum = NextSeqNum + length(data)
event: timer timeout
retransmit not-yet-acknowledged segment with
smallest sequence number start timer
event: ACK received, with ACK field value of y if (y > SendBase) {
SendBase = y
if (there are currently not-yet-acknowledged segments)
start timer }
} /* end of loop forever */
TCP
sender (simplified)
Comment:
• SendBase-1: last cumulatively ack’ed byte Example:
• SendBase-1 = 71; y= 73, so the rcvr wants 73+ ;
y > SendBase, so that new data is acked
Transport Layer 3-57

TCP: retransmission scenarios
Host A
Host B
Host A
Host B
X
loss
SendBase = 100
Sendbase
= 100 SendBase
= 120
SendBase = 120
time
premature timeout Transport Layer 3-58
time
lost ACK scenario
Seq=92 timeout
Seq=92 timeout
timeout

TCP retransmission scenarios (more)
Host A
Host B
X
loss
SendBase = 120
time
Cumulative ACK scenario
Transport Layer 3-59
timeout

TCP ACK generation [RFC 1122, RFC 2581]
Event at Receiver
Arrival of in-order segment with expected seq #. All data up to expected seq # already ACKed
Arrival of in-order segment with expected seq #. One other segment has ACK pending
Arrival of out-of-order segment higher-than-expect seq. # . Gap detected
Arrival of segment that partially or completely fills gap
TCP Receiver action
Delayed ACK. Wait up to 500ms
for next segment. If no next segment, send ACK
Immediately send single cumulative ACK, ACKing both in-order segments
Immediately send duplicate ACK, indicating seq. # of next expected byte
Immediate send ACK, provided that segment startsat lower end of gap
Transport Layer 3-60

Fast Retransmit
 Time-out period often  relatively long:
 long delay before resending lost packet
 Detect lost segments via duplicate ACKs.
 Sender often sends many segments back-to- back
 If segment is lost, there will likely be many duplicate ACKs.
If sender receives 3 ACKs for the same data, it supposes that segment after ACKed data was lost:
 fast retransmit: resend segment before timer expires
Transport Layer 3-61

Fast retransmit algorithm:
event: ACK received, with ACK field value of y if (y > SendBase) {
SendBase = y
if (there are currently not-yet-acknowledged segments)
start timer }
else {
increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) {
resend segment with sequence number y }
a duplicate ACK for already ACKed segment
fast retransmit
Transport Layer 3-62

Chapter 3 outline
 3.1 Transport-layer services
 3.2 Multiplexing and demultiplexing
 3.3 Connectionless transport: UDP
 3.4 Principles of reliable data transfer
 3.5 Connection-oriented transport: TCP
 segment structure
 reliable data transfer  flow control
 connection management
 3.6 Principles of congestion control
 3.7 TCP congestion control
Transport Layer 3-63

TCP Flow Control
 receive side of TCP connection has a receive buffer:
flow control
sender won’t overflow receiver’s buffer by transmitting too much,
too fast
 speed-matching service: matching the send rate to the receiving app’s drain rate
 app process may be slow at reading from buffer
Transport Layer 3-64

TCP Flow control: how it works
(Suppose TCP receiver discards out-of-order segments)
 spare room in buffer
= RcvWindow
= RcvBuffer-[LastByteRcvd – LastByteRead]
 Rcvr advertises spare room by including value of RcvWindow in
segments
 Sender limits unACKed
data to RcvWindow  guarantees receive
buffer doesn’t overflow
Transport Layer 3-65

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!
Transport Layer 3-67

Causes/costs of congestion: scenario 1
 two senders, two receivers
 one router, infinite buffers
 no retransmission
Host A
lin : original data
unlimited shared
output link buffers
lout
Host B
 large delays when congested
 maximum achievable throughput
Transport Layer 3-68

Causes/costs of congestion: scenario 2
 one router, finite buffers
 sender retransmission of lost packet
Host A
lin : original lout data
l’in : original data, plus retransmitted data
finite shared output link buffers
Host B
Transport Layer 3-69

Causes/costs of congestion: scenario 2
 always: l = lout (goodput)
 “perfect” retransmission only when loss: in
in l>l
 retransmission of delayed (not lost) packet makes lin larger
(than perfect case) for same lout R/2 R/2
R/3
R/2
R/4
out
l R/2
a. b. c.
“costs” of congestion:
 more work (retrans) for given “goodput”
 unneeded retransmissions: link carries multiple copies of pkt Transport Layer 3-70
l R/2
in in in
l R/2
lout
lout
lout

Causes/costs of congestion: scenario 3
 four senders
 multihop paths
 timeout/retransmit
Q: what happens as lin and l increase ?
Host A
in
lin : original data
l’in : original data, plus
retransmitted data
lout
finite shared output link buffers
Host B
Transport Layer 3-71

Causes/costs of congestion: scenario 3
H
A
l
o s
t
o u
t
H o s t B
Another “cost” of congestion:
 when packet dropped, any “upstream transmission capacity used for that packet was wasted!
Transport Layer 3-72

Approaches towards congestion control Two broad approaches towards congestion control:
End-end congestion control:
 no explicit feedback from network
 congestion inferred from end-system observed loss, delay
 approach taken by TCP
Network-assisted congestion control:
 routers provide feedback to end systems
 single bit indicating congestion (SNA, DECbit, TCP/IP ECN, ATM)
 explicit rate sender should send at
Transport Layer 3-73

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 guaranteed rate
RM (resource management) cells:
 sent by sender, interspersed with data cells
 bits in RM cell set by switches (“network-assisted”)
 NI bit: no increase in rate (mild congestion)
 CI bit: congestion indication
 RM cells returned to sender by receiver, with bits intact
Transport Layer 3-74

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 maximum 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
Transport Layer 3-75

TCP Congestion Control: details
 sender limits transmission: LastByteSent-LastByteAcked  CongWin
 Roughly,
 CongWin is dynamic, function of perceived network congestion
How does sender perceive congestion?
 loss event = timeout or 3 duplicate acks
 TCP sender reduces rate (CongWin) after
loss event
three mechanisms:
 AIMD
 slow start
 conservative after timeout events
rate =
CongWin RTT
Bytes/sec
Transport Layer 3-77

TCP congestion control: additive increase, multiplicative decrease
 Approach: increase transmission rate (window size), probing for usable bandwidth, until loss occurs
 additive increase: increase CongWin by 1 MSS every RTT until loss detected
 multiplicative decrease: cut CongWin in half after
loss
Saw tooth behavior: probing for bandwidth
congestion window
24 Kbytes
16 Kbytes
8 Kbytes
time
time
Transport Layer 3-78
congestion window size

TCP Slow Start
 When connection begins,  When connection begins,
CongWin = 1 MSS
 Example: MSS = 500
bytes & RTT = 200 msec  initial rate = 20 kbps
 available bandwidth may be >> MSS/RTT
 desirable to quickly ramp up to respectable rate
increase rate exponentially fast until first loss event
Transport Layer 3-79

TCP Slow Start (more)
 When connection begins, increase rate exponentially until first loss event:
 double CongWin every RTT
 done by incrementing CongWin for every ACK received
 Summary: initial rate is slow but ramps up exponentially fast
Host A
Host B
time Transport Layer 3-80
RTT

Refinement
Q: When should the exponential
increase switch to linear?
A: When CongWin gets to 1/2 of its value before timeout.
Implementation:
 Variable Threshold
 At loss event, Threshold is set to 1/2 of CongWin just before loss event
Transport Layer 3-81

Refinement: inferring loss
 After 3 dup ACKs:
 CongWin is cut in half
 window then grows linearly
 But after timeout event:  CongWin instead set to
1 MSS;
 window then grows exponentially
 to a threshold, then grows linearly
Philosophy:
❑ 3 dup ACKs indicates network capable of delivering some segments ❑ timeout indicates a “more alarming” congestion scenario
Transport Layer 3-82

Summary: TCP Congestion Control
 When CongWin is below Threshold, sender in slow-start phase, window grows exponentially.
 When CongWin is above Threshold, sender is in congestion-avoidance phase, window grows linearly.
 When a triple duplicate ACK occurs, Threshold set to CongWin/2 and CongWin set to Threshold.
 When timeout occurs, Threshold set to CongWin/2 and CongWin is set to 1 MSS.
Transport Layer 3-83

TCP sender congestion control
State
Event
TCP Sender Action
Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS, If (CongWin > Threshold)
set state to “Congestion Avoidance”
Resulting in a doubling of CongWin every RTT
Congestion Avoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS * (MSS/CongWin)
Additive increase, resulting in increase of CongWin by 1 MSS every RTT
SS or CA
Loss event detected by triple duplicate ACK
Threshold = CongWin/2, CongWin = Threshold, Set state to “Congestion Avoidance”
Fast recovery, implementing multiplicative decrease. CongWin will not drop below 1 MSS.
SS or CA
Timeout
Threshold = CongWin/2, CongWin = 1 MSS,
Set state to “Slow Start”
Enter slow start
SS or CA
Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Transport Layer 3-84

TCP throughput
What’s the average throughout of TCP as a function of window size and RTT?
 Ignore slow start
Let W be the window size when loss occurs.
When window is W, throughput is W/RTT
Just after loss, window drops to W/2, throughput to W/2RTT.
Average throughout: .75 W/RTT
Transport Layer 3-85

TCP Fairness
Fairness goal: if K TCP sessions share same bottleneck link of bandwidth R, each should have average rate of R/K
TCP connection 1
bottleneck router capacity R
TCP connection 2
Transport Layer 3-86

Why is TCP fair?
Two competing sessions:
 Additive increase gives slope of 1, as throughout increases  multiplicative decrease decreases throughput proportionally
R
equal bandwidth share
loss: decrease window by factor of 2 congestion avoidance: additive increase
loss: decrease window by factor of 2 congestion avoidance: additive increase
Connection 1 throughput
R
Transport Layer 3-87

Chapter 3: Summary
 principles behind transport layer services:
 multiplexing, demultiplexing
 reliable data transfer
 flow control
 congestion control
 instantiation and implementation in the Internet
 UDP  TCP
Next:
 leaving the network “edge” (application, transport layers)
 into the network “core”
Transport Layer 3-88

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