Chapter 2, 3

Transport Layer

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Transport Layer 3-1

Part 4 outline

4.1 Transport-layer 4.5 Connection-oriented

services transport: TCP
4.2 Demultiplexing segment structure
reliable data transfer
4.3 Connectionless

flow control
transport: UDP
connection management
4.4 Principles of
4.6 Principles of
reliable data transfer
congestion control
4.7 TCP congestion
control
4.8 DNS

Transport Layer 3-2

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 data link
physical
end systems network
data link
send side: breaks app physical network
data link
messages into segments, physical

passes to network layer network

data link
physical
rcv side: reassembles
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

Internet transport-layer protocols

reliable, in-order application

transport
delivery (TCP) network
data link network
physical
congestion control network
data link
physical
data link
flow control physical
network
connection setup data link
physical network
data link
unreliable, unordered physical

delivery: UDP network

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

delay guarantees
bandwidth guarantees

Transport Layer 3-4

Part 4 outline

4.1 Transport-layer 4.5 Connection-oriented

services transport: TCP
4.2 Demultiplexing segment structure
reliable data transfer
4.3 Connectionless

flow control
transport: UDP
connection management
4.4 Principles of
4.6 Principles of
reliable data transfer
congestion control
4.7 TCP congestion
control
4.8 DNS

Transport Layer 3-5

 Demultiplexing
Demultiplexing at rcv host:
delivering received segments
to correct socket

= 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 3-6
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 3-7

 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

Transport Layer 3-8

Connection-oriented demux (cont)

P1 P4 P5 P6 P2 P1P3

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

SP: 9157 SP: 9157

client DP: 80 DP: 80 Client
server
IP: A S-IP: A
IP: C S-IP: B IP:B
D-IP:C D-IP:C

Transport Layer 3-9

Connection-oriented demux: Threaded
Web Server

P1 P4 P2 P1P3

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

SP: 9157 SP: 9157

client DP: 80 DP: 80 Client
server
IP: A S-IP: A
IP: C S-IP: B IP:B
D-IP:C D-IP:C

Transport Layer 3-10

Connectionless demultiplexing

When host receives UDP

Create sockets with port
numbers:
DatagramSocket mySocket1 = new
DatagramSocket(12534);
DatagramSocket mySocket2 = new
DatagramSocket(12535);
UDP socket identified by
two-tuple:
(dest IP address, dest port number)
segment:
checks destination port
number in segment
directs UDP segment to
socket with that port
number
IP datagrams with
different source IP
addresses and/or source
port numbers directed
to same socket

Transport Layer 3-11

 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 3-12

Part 4 outline

4.1 Transport-layer 4.5 Connection-oriented

services transport: TCP
4.2 Demultiplexing segment structure
reliable data transfer
4.3 Connectionless

flow control
transport: UDP
connection management
4.4 Principles of
4.6 Principles of
reliable data transfer
congestion control
4.7 TCP congestion
control
4.8 DNS

Transport Layer 3-13

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

Transport Layer 3-14

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

Transport Layer 3-15

UDP checksum

Goal: detect errors (e.g., flipped bits) in transmitted

segment

Sender: Receiver:
treat segment contents Run addition over checksum
as sequence of 16-bit and segment contents
integers check if computed value
checksum: addition (1s equals 0
complement sum) of NO - error detected
segment contents YES - no error detected.
sender puts checksum But maybe errors
value into UDP checksum nonetheless?
field

Transport Layer 3-16

 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
Transport Layer 3-17
Part 4 outline

4.1 Transport-layer 4.5 Connection-oriented

services transport: TCP
4.2 Demultiplexing segment structure
reliable data transfer
4.3 Connectionless

flow control
transport: UDP
connection management
4.4 Principles of
4.6 Principles of
reliable data transfer
congestion control
4.7 TCP congestion
control
4.8 DNS

Transport Layer 3-18

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 3-19
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 3-20
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 3-21
 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

Transport Layer 3-22

Reliable data transfer: getting started
Well:
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

Transport Layer 3-23

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

Transport Layer 3-24

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 3-25

rdt2.0: FSM specification
rdt_send(data)
sndpkt = 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
L
call from
sender below

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

Transport Layer 3-26

rdt2.0 has a fatal flaw!

What happens if Handling duplicates:

ACK/NAK corrupted? sender retransmits current
sender doesnt know what pkt even if ACK/NAK
happened at receiver! garbled
cant just retransmit: sender adds sequence
possible duplicate number to each pkt
receiver discards (doesnt
deliver up) duplicate pkt

stop and wait

Sender sends one packet,
then waits for receiver
response

Transport Layer 3-27

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)
L
L
Wait for Wait for
ACK or call 1 from
rdt_rcv(rcvpkt) && NAK 1 above
( corrupt(rcvpkt) ||
isNAK(rcvpkt) ) rdt_send(data)

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

udt_send(sndpkt)

Transport Layer 3-28

 rdt2.1: receiver, handles garbled ACK/NAKs
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq0(rcvpkt)
extract(rcvpkt,data)
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)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
Duplicate packet
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)

Transport Layer 3-29

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 ACK/NAK received OK
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
at sender
Transport Layer 3-30
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 3-31

 rdt2.2: sender, receiver fragments
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(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) || L
has_seq1(rcvpkt)) Wait for receiver FSM
0 from
udt_send(sndpkt) below fragment
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK1, chksum)
udt_send(sndpkt) Transport Layer 3-32
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 3-33
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 L
Moved
L 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) L
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) || sndpkt = make_pkt(1, data, checksum) Stray packet
isACK(rcvpkt,0) ) udt_send(sndpkt)
start_timer
L

Transport Layer 3-34

rdt3.0 in action

Transport Layer 3-35

rdt3.0 in action

Transport Layer 3-36

 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
sender
= = = 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 3-37

 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
sender
= = = 0.00027
RTT + L / R 30.008 microsec
onds

Transport Layer 3-38

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 3-39
 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
sender
= = = 0.0008
RTT + L / R 30.008 microsecon
ds
Transport Layer 3-40
Go-Back-N
Sender:
k-bit seq # in pkt header
window of up to N, consecutive unacked 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 (ideally). In practice, timer for
oldest un-acked packet
timeout(n): retransmit pkt n and all higher seq # pkts in window
Transport Layer 3-41
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++
}
L 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])
L
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
Cumulative Ack: Received all
base = getacknum(rcvpkt)+1 packets up to new base
If (base == nextseqnum) value
stop_timer
Not optimal. Should ideally else
just increment the timeout start_timer Transport Layer 3-42
 GBN: receiver extended FSM
Keep repeating the default
last ACK packet
udt_send(sndpkt) rdt_rcv(rcvpkt)
&& notcurrupt(rcvpkt)
L && 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 (dont buffer) -> no receiver buffering!
Re-ACK pkt with highest in-order seq #
Transport Layer 3-43
GBN in
action

Transport Layer 3-44

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 3-45

Selective repeat: sender, receiver windows

Different

Transport Layer 3-46

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) Ack the
advance window base to acknowledged
next unACKed seq # otherwise: packets again, so as
to prevent sender
ignore getting stuck
because of lost
ACKs

Transport Layer 3-47

Selective repeat in action

Transport Layer 3-48

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?
Window size must be less than half the
sequence number space

But what happens with old stray packets? Transport Layer 3-49
Part 4 outline

4.1 Transport-layer 4.5 Connection-oriented

services transport: TCP
4.2 Demultiplexing segment structure
reliable data transfer
4.3 Connectionless

flow control
transport: UDP
connection management
4.4 Principles of
4.6 Principles of
reliable data transfer
congestion control
4.7 TCP congestion
control
4.8 DNS

Transport Layer 3-50

 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

MSS: maximum segment
stream:
size
no message boundaries
connection-oriented:
pipelined:
handshaking (exchange
TCP congestion and flow
of control msgs) inits
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

Transport Layer 3-51

 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)

Transport Layer 3-52

 TCP seq. #s and ACKs
Seq. #s:
Host A Host B
byte stream
number of first User
types
byte in segments C
data host ACKs
receipt of
ACKs: C, echoes
seq # of next byte back C
expected from
other side host ACKs
cumulative ACK receipt
of echoed
Q: how receiver handles C
out-of-order segments
A: TCP spec doesnt
time
say, - up to
simple telnet scenario
implementor
Transport Layer 3-53
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
ignore retransmissions
too short: premature
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

Transport Layer 3-54

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 3-55

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

Transport Layer 3-56

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 3-57

Part 4 outline

4.1 Transport-layer 4.5 Connection-oriented

services transport: TCP
4.2 Demultiplexing segment structure
reliable data transfer
4.3 Connectionless

flow control
transport: UDP
connection management
4.4 Principles of
4.6 Principles of
reliable data transfer
congestion control
4.7 TCP congestion
control
4.8 DNS

Transport Layer 3-58

TCP reliable data transfer

TCP creates rdt Retransmissions are

service on top of IPs 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

Transport Layer 3-59

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

Transport Layer 3-60

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

Seq=92 timeout
timeout

X
loss

Sendbase
= 100

Seq=92 timeout
SendBase
= 120

SendBase
= 100 SendBase
= 120 premature timeout
time time
lost ACK scenario
Transport Layer 3-61
 TCP retransmission scenarios (more)
Host A Host B
timeout

X
loss

SendBase
= 120

time
Cumulative ACK scenario

Transport Layer 3-62

Fast Retransmit

Time-out period often If sender receives 3

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.
ACKs for the same
data, it supposes that
segment after ACKed
data was lost:
fast retransmit: resend
segment before timer
expires

Transport Layer 3-63

Part 4 outline

4.1 Transport-layer 4.5 Connection-oriented

services transport: TCP
4.2 Demultiplexing segment structure
reliable data transfer
4.3 Connectionless

flow control
transport: UDP
connection management
4.4 Principles of
4.6 Principles of
reliable data transfer
congestion control
4.7 TCP congestion
control
4.8 DNS

Transport Layer 3-64

TCP Flow Control
flow control
sender wont overflow
receive side of TCP receivers buffer by
connection has a transmitting too much,
receive buffer: too fast

speed-matching
service: matching the
send rate to the
receiving apps drain
rate
app process may be
slow at reading from
buffer
Transport Layer 3-65
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 doesnt overflow
spare room in buffer
= RcvWindow
= RcvBuffer-[LastByteRcvd -
LastByteRead]

Transport Layer 3-66

Part 4 outline

4.1 Transport-layer 4.5 Connection-oriented

services transport: TCP
4.2 Demultiplexing segment structure
reliable data transfer
4.3 Connectionless

flow control
transport: UDP
connection management
4.4 Principles of
4.6 Principles of
reliable data transfer
congestion control
4.7 TCP congestion
control
4.8 DNS

Transport Layer 3-67

TCP Connection Management
Recall: TCP sender, receiver
establish connection
before exchanging data
segments
initialize TCP variables:
seq. #s
buffers, flow control
info (e.g. RcvWindow)
client: connection initiator
Socket clientSocket = new
Socket("hostname","port
number");
server: contacted by client
Socket connectionSocket =
welcomeSocket.accept();
Three way handshake:
Step 1: client host sends TCP
SYN segment to server
specifies initial seq #
no data
Step 2: server host receives
SYN, replies with SYNACK
segment
server allocates buffers
specifies server initial
seq. #
Step 3: client receives SYNACK,
replies with ACK segment,
which may contain data
Transport Layer 3-68
TCP Connection Management (cont.)

Closing a connection: client server

close
client closes socket:
clientSocket.close();
close
Step 1: client end system
sends TCP FIN control
segment to server

timed wait
Step 2: server receives
FIN, replies with ACK.
Closes connection, sends
FIN. closed

Transport Layer 3-69

TCP Connection Management (cont.)

Step 3: client receives FIN, client server

replies with ACK. closing
Enters timed wait -
will respond with ACK
to received FINs
closing
Step 4: server, receives
ACK. Connection closed.

timed wait closed

closed

Transport Layer 3-70

TCP Connection Management (cont)

TCP server
lifecycle

TCP client
lifecycle

Transport Layer 3-71

Part 4 outline

4.1 Transport-layer 4.5 Connection-oriented

services transport: TCP
4.2 Demultiplexing segment structure
reliable data transfer
4.3 Connectionless

flow control
transport: UDP
connection management
4.4 Principles of
4.6 Principles of
reliable data transfer
congestion control
4.7 TCP congestion
control
4.8 DNS

Transport Layer 3-72

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 3-73

 Causes/costs of congestion
Host A lout
two senders, two
lin : original data

receivers
one router
Host B shared output
link buffers

large delays when congested

more work (retrans) for given goodput
unneeded retransmissions: link carries multiple copies
of pkt
when packet dropped, any transmission capacity used
for that packet was wasted! 3-74
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 (TCP/IP
delay ECN)
approach taken by TCP explicit rate sender
should send at

Transport Layer 3-75

Part 4 outline

4.1 Transport-layer 4.5 Connection-oriented

services transport: TCP
4.2 Demultiplexing segment structure
reliable data transfer
4.3 Connectionless

flow control
transport: UDP
connection management
4.4 Principles of
4.6 Principles of
reliable data transfer
congestion control
4.7 TCP congestion
control
4.8 DNS

Transport Layer 3-76

 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
congestion window size

24 Kbytes

Saw tooth
behavior: probing
16 Kbytes

for bandwidth
8 Kbytes

time
time

Transport Layer 3-77

 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
Transport Layer 3-78
 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

Transport Layer 3-79

TCP Slow Start (more)

When connection Host A Host B

begins, increase rate
exponentially until

RTT
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 time

Transport Layer 3-80

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
Philosophy:
window then grows
linearly 3 dup ACKs indicates
Typically inferred network capable of
before timeouts delivering some segments
But after timeout event: timeout indicates a
CongWin instead set to more alarming
1 MSS; congestion scenario
window then grows
exponentially
to a threshold, then
grows linearly 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 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

Transport Layer 3-84

TCP throughput

Whats 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 Futures: TCP over long, fat pipes

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 = 210-10 Wow
New versions of TCP for high-speed needed!

Transport Layer 3-86

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

Transport Layer 3-87

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 3-88

Fairness (more)
Fairness and UDP
Multimedia apps often
do not use TCP
do not want rate
throttled by congestion
control
Instead use UDP:
pump audio/video at
constant rate, tolerate
packet loss
Research area: TCP
friendly
Fairness and parallel TCP
connections
nothing prevents app from
opening parallel
connections between 2
hosts.
Web browsers do this
Example: link of rate R
supporting 9 cnctions;
new app asks for 1 TCP, gets
rate R/10
new app asks for 11 TCPs,
gets 11R/20 !
Transport Layer 3-89
 Delay modeling
Q: How long does it take to
receive an object from a
Web server after sending
a request?
Ignoring congestion, delay is
influenced by:
TCP connection establishment
data transmission delay
slow start
Notation, assumptions:
Assume one link between
client and server of rate R
S: MSS (bits)
O: object size (bits)
no retransmissions (no loss,
no corruption)
Window size:
First assume: fixed
congestion window, W
segments
Then dynamic window,
modeling slow start
Transport Layer 3-90
Fixed congestion window (1)

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

delay = 2RTT + O/R

Transport Layer 3-91

Fixed congestion window (2)

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

delay = 2RTT + O/R

+ (K-1)[S/R + RTT - WS/R]

Transport Layer 3-92

 TCP Delay Modeling: Slow Start
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

third window
= 4S/R

fourth window
= 8S/R

complete
object transmission
delivered
time at
time at server
client

Transport Layer 3-93

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

Transport Layer 3-94

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.
Transport Layer 3-95
HTTP Response time (in seconds)
RTT =1 sec, O = 5 Kbytes, M=10 and X=5
70
60
50
non-persistent
40
30 persistent
20
parallel non-
10 persistent
0
28 100 1 10
Kbps Kbps Mbps Mbps
For larger RTT, response time dominated by TCP establishment
& slow start delays. Persistent connections now give important
improvement: particularly in high delaybandwidth networks.
Transport Layer 3-96