Chapter3[one.]

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

Transport Layer 3-1

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

Transport Layer 3-2

1

Transport services and protocols
provide logical communication application

between app processes
transport
network

running on different hosts
data link network
physical data link
network physical
transport protocols run in

lo
data link

icg
physical
end systems

a
le
network

nd
data link
send side: breaks app

-e
physical network

nd
data link
messages into segments, physical

tr
an
passes to network layer network

s
data link

po
physical
rcv side: reassembles

rt
segments into messages, application
transport
passes to app layer network
data link
more than one transport physical

protocol available to apps
Internet: TCP and UDP

Transport Layer 3-3

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

Transport Layer 3-4

2

Internet transport-layer protocols
reliable, in-order application
transport
delivery (TCP) network
data link network
physical
congestion control network
data link
physical

lo
data link

g
flow control

ic
physical

a
le
network
connection setup

nd
data link

-e
physical network

nd
data link
unreliable, unordered physical

tr
an
delivery: UDP network

s
data link

po
physical

rt
no-frills extension of
“best-effort” IP application
transport
network
services not available: data link
physical

delay guarantees
bandwidth guarantees

Transport Layer 3-5

Chapter 3 outline
flow control
3.3 Connectionless
transport: UDP
3.7 TCP congestion
control

Transport Layer 3-6

3

Multiplexing/demultiplexing
Demultiplexing at rcv host: Multiplexing at send host:
gathering data from multiple
delivering received segments
sockets, enveloping data with
to correct socket
header (later used for
demultiplexing)
= socket = process

P3 P1
P1 P2 P4 application
application application

transport transport transport

network network network

link link link

physical physical physical

host 2 host 3
host 1
Transport Layer 3-7

How demultiplexing works
host receives IP datagrams
each datagram has source 32 bits
IP address, destination IP
address source port # dest port #

each datagram carries 1
transport-layer segment other header fields
each segment has source,
destination port number
host uses IP addresses & port application
numbers to direct segment to data
appropriate socket (message)

TCP/UDP segment format

Transport Layer 3-8

4

Connectionless demultiplexing
When host receives UDP
Create sockets with port
segment:
numbers:
DatagramSocket mySocket1 = new checks destination port
DatagramSocket(99111); number in segment
DatagramSocket mySocket2 = new directs UDP segment to
DatagramSocket(99222); socket with that port
number
UDP socket identified by
two-tuple: IP datagrams with
different source IP
(dest IP address, dest port number)
addresses and/or source
port numbers directed
to same socket

Transport Layer 3-9

Connectionless demux (cont)
DatagramSocket serverSocket = new DatagramSocket(6428);

P2 P1
P1
P3

SP: 6428 SP: 6428
DP: 9157 DP: 5775

SP: 9157 SP: 5775
client DP: 6428 DP: 6428 Client
server
IP: A IP: C IP:B

SP provides “return address”

Transport Layer 3-10

5

Connection-oriented demux
TCP socket identified Server host may support
by 4-tuple: many simultaneous TCP
source IP address sockets:
source port number each socket identified by
dest IP address its own 4-tuple
dest port number Web servers have
recv host uses all four different sockets for
values to direct each connecting client
segment to appropriate non-persistent HTTP will
socket have different socket for
each request


Connection-oriented demux
(cont)

P1 P4 P5 P6 P2 P1
P3

SP: 5775
DP: 80
S-IP: B
D-IP:C

SP: 9157 SP: 9157
server
IP: A S-IP: A
IP: C S-IP: B IP:B
D-IP:C D-IP:C


6

Connection-oriented demux:
Threaded Web Server

P1 P4 P2 P1
P3

SP: 5775
DP: 80
S-IP: B
D-IP:C

SP: 9157 SP: 9157
server
IP: A S-IP: A
IP: C S-IP: B IP:B
D-IP:C D-IP:C


Chapter 3 outline
flow control
3.3 Connectionless
transport: UDP
3.7 TCP congestion
control


7

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


UDP: more
often used for streaming
multimedia apps 32 bits

loss tolerant Length, in source port # dest port #
rate sensitive bytes of UDP length checksum
segment,
other UDP uses including
DNS header
SNMP
reliable transfer over UDP: Application
add reliability at data
application layer (message)
application-specific
error recovery!
UDP segment format


8

UDP checksum
Goal: detect “errors” (e.g., flipped bits) in transmitted
segment

Sender: Receiver:
treat segment contents compute checksum of
as sequence of 16-bit received segment
integers check if computed checksum
checksum: addition (1’s equals checksum field value:
complement sum) of NO - error detected
segment contents YES - no error detected.
sender puts checksum But maybe errors
value into UDP checksum nonetheless? More later
field ….


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

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0
1 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

wraparound 1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

sum 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0
checksum 1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

9

Chapter 3 outline
flow control
3.3 Connectionless
transport: UDP
3.7 TCP congestion
control


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)

10





11

Reliable data transfer: getting started
rdt_send(): called from above, deliver_data(): called by
(e.g., by app.). Passed data to rdt to deliver data to upper
deliver to receiver upper layer

send receive
side side

udt_send(): called by rdt, rdt_rcv(): called when packet
to transfer packet over arrives on rcv-side of channel
unreliable channel to receiver


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
state: when in this
“state” next state state state
1 event
uniquely determined 2
by next event actions


12

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 rdt_send(data) Wait for rdt_rcv(packet)
call from call from extract (packet,data)
above packet = make_pkt(data) below deliver_data(data)
udt_send(packet)

sender receiver


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


13

rdt2.0: FSM specification
rdt_send(data)
snkpkt = make_pkt(data, checksum) receiver
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for Wait for rdt_rcv(rcvpkt) &&
call from ACK or udt_send(sndpkt) corrupt(rcvpkt)
above NAK
udt_send(NAK)

rdt_rcv(rcvpkt) && isACK(rcvpkt)
Wait for
Λ
call from
sender below

rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)


rdt2.0: operation with no errors
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
above NAK
udt_send(NAK)

Wait for
Λ call from
below

rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
deliver_data(data)
udt_send(ACK)


14

rdt2.0: error scenario
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
above NAK
udt_send(NAK)

Wait for
Λ call from
below

rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
deliver_data(data)
udt_send(ACK)


rdt2.0 has a fatal flaw!
What happens if Handling duplicates:
ACK/NAK corrupted? sender retransmits current
sender doesn’t know what pkt if ACK/NAK garbled
happened at receiver! sender adds sequence
can’t just retransmit: number to each pkt
possible duplicate receiver discards (doesn’t
deliver up) duplicate pkt

stop and wait
Sender sends one packet,
then waits for receiver
response


15

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

udt_send(sndpkt) sndpkt = make_pkt(1, data, checksum)
udt_send(sndpkt)


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

deliver_data(data)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)


16

rdt2.1: discussion
Sender: Receiver:
seq # added to pkt must check if received
two seq. #’s (0,1) will packet is duplicate
suffice. Why? state indicates whether
0 or 1 is expected pkt
must check if received seq #
ACK/NAK corrupted
note: receiver can not
twice as many states know if its last
state must “remember” ACK/NAK received OK
whether “current” pkt
at sender
has 0 or 1 seq. #


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


17

rdt2.2: sender, receiver fragments
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
Wait for Wait for
ACK isACK(rcvpkt,1) )
call 0 from
above 0 udt_send(sndpkt)
sender FSM
fragment rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && && isACK(rcvpkt,0)
(corrupt(rcvpkt) || Λ
has_seq1(rcvpkt)) Wait for receiver FSM
0 from
udt_send(sndpkt) below fragment
&& has_seq1(rcvpkt)
deliver_data(data)
sndpkt = make_pkt(ACK1, chksum)
udt_send(sndpkt) Transport Layer 3-35

rdt3.0: channels with errors and loss

New assumption: Approach: sender waits
underlying channel can “reasonable” amount of
also lose packets (data time for ACK
or ACKs) retransmits if no ACK
checksum, seq. #, ACKs, received in this time
retransmissions will be if pkt (or ACK) just delayed
of help, but not enough (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


18

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


rdt3.0 in action


19

rdt3.0 in action


Performance of rdt3.0

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

Ttransmit = L (packet length in bits) 8kb/pkt
= = 8 microsec
R (transmission rate, bps) 10**9 b/sec

U sender: utilization – fraction of time sender busy sending

U L/R .008
= = = 0.00027
sender 30.008
RTT + L / R
1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link
network protocol limits use of physical resources!


20

rdt3.0: stop-and-wait operation
sender receiver
first packet bit transmitted, t = 0
last packet bit transmitted, t = L / R

first packet bit arrives
RTT last packet bit arrives, send
ACK

ACK arrives, send next
packet, t = RTT + L / R

U L/R .008
= = = 0.00027
sender 30.008
RTT + L / R


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

21

Pipelining: increased utilization
sender receiver
first packet bit transmitted, t = 0
last bit transmitted, t = L / R

first packet bit arrives
RTT last packet bit arrives, send ACK
last bit of 2nd packet arrives, send ACK
last bit of 3rd packet arrives, send ACK
ACK arrives, send next
packet, t = RTT + L / R

Increase utilization
by a factor of 3!

U 3*L/R .024
= = = 0.0008
sender 30.008
RTT + L / R


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


22

GBN: sender extended FSM
rdt_send(data)
if (nextseqnum < base+N) {
sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum)
udt_send(sndpkt[nextseqnum])
if (base == nextseqnum)
start_timer
nextseqnum++
}
Λ else
refuse_data(data)
base=1
nextseqnum=1
timeout
start_timer
Wait udt_send(sndpkt[base])
rdt_rcv(rcvpkt) udt_send(sndpkt[base+1])
&& corrupt(rcvpkt) …
udt_send(sndpkt[nextseqnum-1])
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
base = getacknum(rcvpkt)+1
If (base == nextseqnum)
stop_timer
else
start_timer Transport Layer 3-45

GBN: receiver extended FSM
default
udt_send(sndpkt) rdt_rcv(rcvpkt)
&& notcurrupt(rcvpkt)
Λ && hasseqnum(rcvpkt,expectedseqnum)
expectedseqnum=1 Wait extract(rcvpkt,data)
sndpkt = deliver_data(data)
make_pkt(expectedseqnum,ACK,chksum) sndpkt = make_pkt(expectedseqnum,ACK,chksum)
udt_send(sndpkt)
expectedseqnum++

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 #

23

GBN in
action


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


24

Selective repeat: sender, receiver windows


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


25

Selective repeat in action


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?

26

Chapter 3 outline
flow control
3.3 Connectionless
transport: UDP
3.7 TCP congestion
control


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

point-to-point: full duplex data:
one sender, one receiver bi-directional data flow
reliable, in-order byte in same connection
steam: MSS: maximum segment
size
no “message boundaries”
connection-oriented:
pipelined:
handshaking (exchange
TCP congestion and flow of control msgs) init’s
control set window size sender, receiver state
send & receive buffers before data exchange
flow controlled:
sender will not
application application
writes data reads data
socket socket

overwhelm receiver
door door
TCP TCP
send buffer receive buffer
segment


27

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


TCP seq. #’s and ACKs
Seq. #’s:
Host A Host B
byte stream
“number” of first User Seq=4
2, AC
types
byte in segment’s
K=79,
d ata =
‘C’ ‘C’
data host ACKs
receipt of
ACKs: = ‘C’ ‘C’, echoes
data
=43,
seq # of next byte 79, A
CK back ‘C’
Seq=
expected from
other side host ACKs
cumulative ACK receipt Seq=4
of echoed 3, ACK
Q: how receiver handles ‘C’
=80

out-of-order segments
A: TCP spec doesn’t
time
say, - up to
simple telnet scenario
implementor

28

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


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

Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value: α = 0.125


29

Example RTT estimation:
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr

350

300

250
RTT (milliseconds)

200

150

100
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)

SampleRTT Estimated RTT


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-β)*DevRTT +
β*|SampleRTT-EstimatedRTT|

(typically, β = 0.25)

Then set timeout interval:

TimeoutInterval = EstimatedRTT + 4*DevRTT


30

Chapter 3 outline
flow control
3.3 Connectionless
transport: UDP
3.7 TCP congestion
control


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


31

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


NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum

loop (forever) { TCP
sender
switch(event)

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

} /* end of loop forever */

32

TCP: retransmission scenarios
Host A Host B Host A Host B

Seq=9 Seq=9
2, 8 b 2, 8 b
yte s data ytes d
ata

Seq=92 timeout
Seq=
100,
20 by
tes d
timeout

ata
=100
ACK
00
X =1 0
CK CK=12
loss A A

Seq=9 Seq=9
2, 8 b
2, 8 b
yte s data
Sendbase yte s data
= 100

Seq=92 timeout
SendBase
= 120 20
K=1
=100 AC
ACK

SendBase
= 100 SendBase
= 120 premature timeout
time time
lost ACK scenario

TCP retransmission scenarios (more)
Host A Host B

Seq=9
2, 8 b
yte s data

=100
timeout

Seq=1 ACK
0 0, 20
bytes
data
X
loss

SendBase =120
ACK
= 120

time
Cumulative ACK scenario


33

TCP ACK generation [RFC 1122, RFC 2581]

Event at Receiver TCP Receiver action
Arrival of in-order segment with Delayed ACK. Wait up to 500ms
expected seq #. All data up to for next segment. If no next segment,
expected seq # already ACKed send ACK

Arrival of in-order segment with Immediately send single cumulative
expected seq #. One other ACK, ACKing both in-order segments
segment has ACK pending

Arrival of out-of-order segment Immediately send duplicate ACK,
higher-than-expect seq. # . indicating seq. # of next expected byte
Gap detected

Arrival of segment that Immediate send ACK, provided that
partially or completely fills gap segment startsat lower end of gap


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


34

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 fast retransmit
already ACKed segment


Chapter 3 outline
flow control
3.3 Connectionless
transport: UDP
3.7 TCP congestion
control


35

TCP Flow Control
flow control
sender won’t overflow
receive side of TCP receiver’s buffer by
connection has a transmitting too much,
receive buffer: 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

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


36

Chapter 3 outline
flow control
3.3 Connectionless
transport: UDP
3.7 TCP congestion
control


TCP Connection Management
Recall: TCP sender, receiver Three way handshake:
establish “connection”
before exchanging data Step 1: client host sends TCP
segments SYN segment to server
initialize TCP variables: specifies initial seq #
seq. #s no data
buffers, flow control Step 2: server host receives
info (e.g. RcvWindow) SYN, replies with SYNACK
client: connection initiator segment
Socket clientSocket = new
server allocates buffers
Socket("hostname","port
number");
specifies server initial
seq. #
server: contacted by client
Socket connectionSocket =
Step 3: client receives SYNACK,
welcomeSocket.accept(); replies with ACK segment,
which may contain data


37

TCP Connection Management (cont.)

Closing a connection: client server

close
client closes socket: FIN
clientSocket.close();

Step 1: client end system ACK
close
sends TCP FIN control FIN
segment to server

Step 2: server receives

timed wait
ACK

FIN, replies with ACK.
Closes connection, sends
FIN.
closed


TCP Connection Management (cont.)

Step 3: client receives FIN, client server
replies with ACK. closing
FIN
Enters “timed wait” -
will respond with ACK
to received FINs ACK
closing
Step 4: server, receives FIN

ACK. Connection closed.
timed wait

ACK
Note: with small
closed
modification, can handle
simultaneous FINs.
closed


38

TCP Connection Management (cont)

TCP server
lifecycle

TCP client
lifecycle


Chapter 3 outline
flow control
3.3 Connectionless
transport: UDP
3.7 TCP congestion
control


39

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
Host A λout
two senders, two λin : original data

receivers
one router,
Host B unlimited shared
output link buffers

infinite buffers
no retransmission

large delays
when congested
maximum
achievable
throughput

40


one router, finite buffers
sender retransmission of lost packet

Host A λin : original data λout

λ'in : original data, plus
retransmitted data

Host B finite shared output
link buffers


always: λ= λ (goodput)
in out
“perfect” retransmission only when loss: λ > λout
in
retransmission of delayed (not lost) packet makes λ larger
in
(than perfect case) for same λout
R/2 R/2 R/2

R/3
λout

λout
λout

R/4

R/2 R/2 R/2
λin λin λin

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

41

four senders Q: what happens as λ
multihop paths and λ increase ?
in
timeout/retransmit in
Host A λout
λin : original data
λ'in : original data, plus
retransmitted data

finite shared output
link buffers

Host B


H λ
o
o
s
u
t
A t

H
o
s
t
B

Another “cost” of congestion:
when packet dropped, any “upstream transmission
capacity used for that packet was wasted!


42

Approaches towards congestion control
Two broad approaches towards congestion control:

End-end congestion Network-assisted
control: congestion control:
no explicit feedback from routers provide feedback
network to end systems
congestion inferred from single bit indicating
end-system observed loss, congestion (SNA,
delay DECbit, TCP/IP ECN,
approach taken by TCP ATM)
explicit rate sender
should send at


Case study: ATM ABR congestion control

ABR: available bit rate: RM (resource management)
“elastic service” cells:
if sender’s path sent by sender, interspersed
“underloaded”: with data cells
sender should use bits in RM cell set by switches
available bandwidth (“network-assisted”)
if sender’s path NI bit: no increase in rate
congested: (mild congestion)
sender throttled to CI bit: congestion
minimum guaranteed indication
rate RM cells returned to sender by
receiver, with bits intact


43

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


Chapter 3 outline
flow control
3.3 Connectionless
transport: UDP
3.7 TCP congestion
control


44

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 congestion window size congestion
window

24 Kbytes

Saw tooth
behavior: probing
16 Kbytes

for bandwidth
8 Kbytes

time
time


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

45

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


TCP Slow Start (more)
When connection Host A Host B
begins, increase rate
exponentially until
one segm
ent
RTT

first loss event:
double CongWin every two segm
ents
RTT
done by incrementing
CongWin for every ACK four segm
ents
received
Summary: initial rate
is slow but ramps up
exponentially fast time


46

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


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


47

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.


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


48

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


TCP Futures
Example: 1500 byte segments, 100ms RTT, want 10
Gbps throughput
Requires window size W = 83,333 in-flight
segments
Throughput in terms of loss rate:
1.22 ⋅ MSS
RTT L
➜ L = 2·10-10 Wow
New versions of TCP for high-speed needed!


49

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
TCP
router
connection 2
capacity R


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
Connection 2 throughput

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


50

Fairness (more)
Fairness and UDP Fairness and parallel TCP
Multimedia apps often connections
do not use TCP nothing prevents app from
do not want rate opening parallel
throttled by congestion connections between 2
control hosts.
Instead use UDP: Web browsers do this
pump audio/video at Example: link of rate R
constant rate, tolerate
packet loss
supporting 9 cnctions;
new app asks for 1 TCP, gets
Research area: TCP rate R/10
friendly new app asks for 11 TCPs,
gets R/2 !


Delay modeling
Notation, assumptions:
Q: How long does it take to Assume one link between
receive an object from a client and server of rate R
Web server after sending S: MSS (bits)
a request? O: object size (bits)
Ignoring congestion, delay is no retransmissions (no loss,
influenced by: no corruption)
TCP connection establishment Window size:
data transmission delay First assume: fixed
slow start congestion window, W
segments
Then dynamic window,
modeling slow start


51

Fixed congestion window (1)

First case:
WS/R > RTT + S/R: ACK for
first segment in window
returns before window’s
worth of data sent

delay = 2RTT + O/R


Fixed congestion window (2)

Second case:
WS/R < RTT + S/R: wait
for ACK after sending
window’s worth of data
sent

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


52

TCP Delay Modeling: Slow Start (1)
Now suppose window grows according to slow start

Will show that the delay for one object is:
O ⎡ S⎤ S
Latency = 2 RTT + + P ⎢ RTT + ⎥ − ( 2 P − 1)
R ⎣ R⎦ R
where P is the number of times TCP idles at server:

P = min{Q, K − 1}

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

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


TCP Delay Modeling: Slow Start (2)
Delay components: initiate TCP
connection
• 2 RTT for connection
estab and request request

• O/R to transmit
object
first window

object
= S/R

• time server idles due RTT
second window
to slow start = 2S/R

Server idles: third window
= 4S/R
P = min{K-1,Q} times

Example: fourth window
= 8S/R
• O/S = 15 segments
• K = 4 windows
•Q=2
• P = min{K-1,Q} = 2 object
complete
transmission
delivered

Server idles P=2 times time at
time at
server
client


53

TCP Delay Modeling (3)
S
+ RTT = time from when server starts to send segment
R
until server receives acknowledgement
initiate TCP
connection
S
2 k −1 = time to transmit the kth window request
R object
first window
= S/R
+
⎡S k −1 S ⎤
RTT

⎢ R + RTT − 2 R ⎥ = idle time after the kth window
second window
= 2S/R
⎣ ⎦
third window
= 4S/R

P
O
delay = + 2 RTT + ∑ idleTime p fourth window
= 8S/R
R p =1
P
O S S
= + 2 RTT + ∑ [ + RTT − 2 k −1 ]
R k =1 R R object
complete
transmission
delivered
O S S
= + 2 RTT + P[ RTT + ] − (2 P − 1) time at
R R R time at
client
server


TCP Delay Modeling (4)
Recall K = number of windows that cover object

How do we calculate K ?

K = min{k : 2 0 S + 21 S + L + 2 k −1 S ≥ O}
= min{k : 2 0 + 21 + L + 2 k −1 ≥ O / S }
O
= min{k : 2 k − 1 ≥ }
S
O
= min{k : k ≥ log 2 ( + 1)}
S
⎡ O ⎤
= ⎢log 2 ( + 1)⎥
⎢ S ⎥
Calculation of Q, number of idles for infinite-size object,
is similar (see HW).


54

HTTP Modeling
Assume Web page consists of:
1 base HTML page (of size O bits)
M images (each of size O bits)
Non-persistent HTTP:
M+1 TCP connections in series
Response time = (M+1)O/R + (M+1)2RTT + sum of idle times
Persistent HTTP:
2 RTT to request and receive base HTML file
1 RTT to request and receive M images
Response time = (M+1)O/R + 3RTT + sum of idle times
Non-persistent HTTP with X parallel connections
Suppose M/X integer.
1 TCP connection for base file
M/X sets of parallel connections for images.
Response time = (M+1)O/R + (M/X + 1)2RTT + sum of idle times


HTTP Response time (in seconds)
RTT = 100 msec, O = 5 Kbytes, M=10 and X=5
20
18
16
14
non-persistent
12
10
persistent
8
6
4 parallel non-
persistent
2
0
28 100 1 10
Kbps Kbps Mbps Mbps
For low bandwidth, connection & response time dominated by
transmission time.
Persistent connections only give minor improvement over parallel
connections.

55

Chapter3[one.]

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