Week 5-6 Session 3,1,2

These Slides are based on textbook on Syllabus

Computer Networking: A Top Down Approach ,
5th edition.
Jim Kurose, Keith Ross
Addison-Wesley, April 2009

Institut Teknologi Del

By : ABS
Jl. Sisingamangaraja
-Jaringan Komputer
Sitoluama, Laguboti 22381
- 2016 -
Toba – SUMUT
http://www.del.ac.id 1
ABS/Jaringan Komputer/Week 5-6 Sesi 3
Chapter 3
Transport Layer

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All material copyright 1996-2010

J.F Kurose and K.W. Ross, All Rights Reserved
Transport Layer 3-2
Chapter 3: Transport Layer
Our goals:
 understand principles  learn about transport
behind transport
layer services:
 multiplexing/demultipl
exing
 reliable data transfer
 flow control
 congestion control
layer protocols in the
Internet:
 UDP: connectionless
transport
 TCP: connection-oriented
transport
 TCP congestion control

Transport Layer 3-3

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 reliable 3.6 Principles of
data transfer congestion control
3.7 TCP congestion control

Transport Layer 3-4

Transport services and protocols
application
transport
 provide logical communication network
data link
between app processes physical

running on different hosts

 transport protocols run in
end systems
 send side: breaks app
messages into segments,
passes to network layer
 rcv side: reassembles application
transport
segments into messages, network
data link
passes to app layer physical

 more than one transport

protocol available to apps
 Internet: TCP and UDP

Transport Layer 3-5

Transport vs. network layer
 network layer: logical Household analogy:
communication 12 kids sending letters to
between hosts 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 who demux
to in-house siblings
 network-layer protocol =
postal service
Transport Layer 3-6
Internet transport-layer protocols
 reliable, in-order application
transport
network
delivery (TCP) data link
physical
network
 congestion control data link
network
physical
data link
 flow control physical

 connection setup
 unreliable, unordered network
data link
physicalnetwork
delivery: UDP data link
physical
 no-frills extension of network
data link
“best-effort” IP
application
physical network transport
data link network
 services not available: physical data link
physical

 delay guarantees
 bandwidth guarantees

Transport Layer 3-7

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 reliable 3.6 Principles of
data transfer congestion control
3.7 TCP congestion control

Transport Layer 3-8

 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 3-9
How demultiplexing works
 host receives IP
datagrams 32 bits
 each datagram has source source port # dest port #
IP address, destination IP
address
 each datagram carries 1 other header fields
transport-layer segment
 each segment has source,
destination port number application
data
 host uses IP addresses &
(message)
port numbers to direct
segment to appropriate
socket TCP/UDP segment format

Transport Layer 3-10

Connectionless demultiplexing
 recall: create sockets with
host-local port numbers:
DatagramSocket mySocket1 = new
DatagramSocket(12534);
DatagramSocket mySocket2 = new
DatagramSocket(12535);
 recall: when creating
datagram to send into UDP
socket, must specify
(dest IP address, dest port number)
 when host receives UDP
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

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 3-13

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 3-14

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 3-15

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 reliable 3.6 Principles of
data transfer congestion control
3.7 TCP congestion control

Transport Layer 3-16

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 3-17

UDP: more
 often used for 32 bits
streaming multimedia
source port # dest port #
apps Length, in
bytes of UDP length checksum
 loss tolerant
segment,
 rate sensitive including
 other UDP uses header

 DNS
Application
 SNMP data
 reliable transfer over (message)
UDP: add reliability at
application layer
UDP segment format
 application-specific
error recovery!
Transport Layer 3-18
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 ….

Transport Layer 3-19

Internet Checksum Example

sum

Transport Layer 3-20

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 reliable 3.6 Principles of
data transfer congestion control
3.7 TCP congestion control

Transport Layer 3-21

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 3-22
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 3-23
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 3-24
 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 3-25

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

Transport Layer 3-26

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 3-27

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
How do humans recover from “errors”
 negative acknowledgements (NAKs): receiver explicitly
tells senderduring
that pktconversation?
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 3-28

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 3-29

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 3-30

rdt2.0: operation with no errors
rdt_send(data)
snkpkt = make_pkt(data, checksum)
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
below

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

Transport Layer 3-31

rdt2.0: error scenario
rdt_send(data)
snkpkt = make_pkt(data, checksum)
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
below

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

Transport Layer 3-32

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 3-33
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 3-34

 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)

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

Transport Layer 3-35

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. #

Transport Layer 3-36

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 3-37

 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 3-38
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

Transport Layer 3-39

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
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)
isACK(rcvpkt,0) ) udt_send(sndpkt)
start_timer
L

Transport Layer 3-40

rdt3.0 in action

Transport Layer 3-41

rdt3.0 in action

Transport Layer 3-42

Performance of rdt3.0

 rdt3.0 works, but performance stinks

 ex: 1 Gbps link, 15 ms prop. delay, 8000 bit packet:
L 8000bits
d trans   9
 8 microsecon ds
R 10 bps
 U sender: utilization – fraction of time sender busy sending

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

Transport Layer 3-43

 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 3-44

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 3-45
 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 3-46

Pipelined Protocols
Go-back-N: big picture:
 sender can have up to
N unacked packets in
pipeline
 rcvr only sends
cumulative acks
 doesn’t ack packet if
there’s a gap
 sender has timer for
oldest unacked packet
 if timer expires,
retransmit all unack’ed
packets
Selective Repeat: big pic
 sender can have up to
N unack’ed packets in
pipeline
 rcvr sends individual
ack for each packet
 sender maintains timer
for each unacked
packet
 when timer expires,
retransmit only
unack’ed packet
Transport Layer 3-47
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 3-48

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])
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
base = getacknum(rcvpkt)+1
If (base == nextseqnum)
stop_timer
else
start_timer
Transport Layer 3-49
GBN: receiver extended FSM
default
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 (don’t buffer) -> no receiver buffering!
 Re-ACK pkt with highest in-order seq #
Transport Layer 3-50
GBN in
action

Transport Layer 3-51

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, unACK’ed pkts

Transport Layer 3-52

Selective repeat: sender, receiver windows

Transport Layer 3-53

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 3-54

Selective repeat in action

Transport Layer 3-55

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 3-56
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 reliable 3.6 Principles of
data transfer congestion control
3.7 TCP congestion control

Transport Layer 3-57

 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
stream:  MSS: maximum segment
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 3-58

 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 3-59

 TCP seq. #’s and ACKs
Seq. #’s:
Host A Host B
 byte stream
“number” of first User
types
byte in segment’s ‘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 doesn’t
time
say, - up to
simple telnet scenario
implementor
Transport Layer 3-60
TCP Round Trip Time and Timeout
Q: how to set TCP Q: how to estimate RTT?
SampleRTT: measured time from
timeout value? 
segment transmission until ACK
 longer than RTT receipt
 but RTT varies  ignore retransmissions
 too short:  SampleRTT will vary, want
premature timeout estimated RTT “smoother”
 average several recent
 unnecessary measurements, not just
retransmissions current SampleRTT
 too long: slow
reaction to segment
loss

Transport Layer 3-61

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 3-62

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 3-63

TCP Round Trip Time and Timeout
Setting the timeout
 EstimatedRTT 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 3-64

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 reliable 3.6 Principles of
data transfer congestion control
3.7 TCP congestion control

Transport Layer 3-65

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

Transport Layer 3-66

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 3-67

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 acked 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 */

Transport Layer 3-68
 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 3-69
 TCP retransmission scenarios (more)
Host A Host B
timeout

X
loss

SendBase
= 120

time
Cumulative ACK scenario

Transport Layer 3-70

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 starts at lower end of gap

Transport Layer 3-71

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 3-72
 Host A Host B

X
timeout

time

Figure 3.37 Resending a segment after triple duplicate ACK

Transport Layer 3-73
 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

Transport Layer 3-74

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 reliable 3.6 Principles of
data transfer congestion control
3.7 TCP congestion control

Transport Layer 3-75

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 3-76

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]

Transport Layer 3-77

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 reliable 3.6 Principles of
data transfer congestion control
3.7 TCP congestion control

Transport Layer 3-78

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 3-79
TCP Connection Management (cont.)

Closing a connection: client server

close
client closes socket:
clientSocket.close();

Step 1: client end system close

sends TCP FIN control
segment to server

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

Transport Layer 3-80

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

Transport Layer 3-81

TCP Connection Management (cont)

TCP server
lifecycle

TCP client
lifecycle

Transport Layer 3-82

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 reliable 3.6 Principles of
data transfer congestion control
3.7 TCP congestion control

Transport Layer 3-83

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 3-84

 Causes/costs of congestion: scenario 1
Host A lout
 two senders, two lin : original data

receivers
one router,
Host B unlimited shared
 output link buffers

infinite buffers
 no retransmission

 large delays
when congested
 maximum
achievable
throughput

Transport Layer 3-85

Causes/costs of congestion: scenario 2
 one router, finite buffers
 sender retransmission of timed-out packet
 application-layer input = application-layer output: lin = lout
 transport-layer input includes retransmissions : l‘in lin

lin : original data

lout
l'in: original data, plus
retransmitted data
Host B

Host A

finite shared output

link buffers
Transport Layer 3-86
Congestion scenario 2a: ideal case
R/2
 sender sends
only when router

lout
buffers available

lin
R/2

lin : original data

lout
copy l'in: original data, plus
retransmitted data
Host B

free buffer space!

Host A

finite shared output

link buffers
Transport Layer 3-87
Congestion scenario 2b: known loss
 packets may get
dropped at router due
to full buffers
 sometimes lost
 sender only resends if
packet known to be lost
(admittedly idealized)
lin : original data
lout
copy l'in: original data, plus
retransmitted data
Host B

no buffer space!
Host A

Transport Layer 3-88

Congestion scenario 2b: known loss
 packets may get
dropped at router due R/2

to full buffers when sending at

 sometimes not lost R/2, some packets

lout
are retransmissions
 sender only resends if but asymptotic
packet known to be lost goodput is still R/2
(why?)
(admittedly idealized) lin
R/2

lin : original data

lout
l'in: original data, plus
retransmitted data
Host B

free buffer space!

Host A

Transport Layer 3-89

Congestion scenario 2c: duplicates
 packets may get
dropped at router due R/2

to full buffers when sending at

 sender times out R/2, some packets

lout
are retransmissions
prematurely, sending including duplicated
two copies, both of that are delivered!
which are delivered lin
R/2

lin
timeout
copy l'in lout

Host B

free buffer space!

Host A

Transport Layer 3-90

Congestion scenario 2c: duplicates
 packets may get
dropped at router due R/2

to full buffers when sending at

 sender times out R/2, some packets

lout
are retransmissions
prematurely, sending including duplicated
two copies, both of that are delivered!
which are delivered lin
R/2

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

Transport Layer 3-91

Causes/costs of congestion: scenario 3
 four senders Q: what happens as l
 multihop paths and l increase ?
in
 timeout/retransmit in
Host A lout
lin : original data
l'in : original data, plus
retransmitted data

finite shared output

link buffers

Host B

Transport Layer 3-92

Causes/costs of congestion: scenario 3
H l
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!

Transport Layer 3-93

Approaches towards congestion control
Two broad approaches towards congestion control:

end-end congestion network-assisted

control:
 no explicit feedback from
network
 congestion inferred from
end-system observed loss,
delay
 approach taken by TCP
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 3-94

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

Transport Layer 3-95

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 3-96

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 reliable 3.6 Principles of
data transfer congestion control
3.7 TCP congestion control

Transport Layer 3-97

 TCP congestion control: additive increase,
multiplicative decrease
 approach: increase transmission rate (window size),
probing for usable bandwidth, until loss occurs
 additive increase: increase cwnd by 1 MSS every
RTT until loss detected
 multiplicative decrease: cut cwnd in half after
loss
cwnd: congestion window size

congestion
window

24 Kbytes

saw tooth
behavior: probing
16 Kbytes

for bandwidth
8 Kbytes

time
time

Transport Layer 3-98

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

Transport Layer 3-99

TCP Slow Start
 when connection Host A Host B
begins, increase rate
exponentially until

RTT
first loss event:
 initially cwnd = 1 MSS
 double cwnd every RTT
 done by incrementing
cwnd for every ACK
received
 summary: initial rate is
slow but ramps up
exponentially fast time

Transport Layer 3-100

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

Transport Layer 3-101

Refinement
Q: when should the
exponential
increase switch to
linear?
A: when cwnd gets to
1/2 of its value
before timeout.

Implementation:
 variable ssthresh
 on loss event, ssthresh is
set to 1/2 of cwnd just
before loss event

Transport Layer 3-102

Summary: TCP Congestion Control
New
New ACK!
ACK! new ACK
duplicate ACK
dupACKcount++ new ACK
.
cwnd = cwnd + MSS (MSS/cwnd)
dupACKcount = 0
cwnd = cwnd+MSS transmit new segment(s), as allowed
dupACKcount = 0
L transmit new segment(s), as allowed
cwnd = 1 MSS
ssthresh = 64 KB cwnd > ssthresh
dupACKcount = 0 slow L congestion
start timeout avoidance
ssthresh = cwnd/2
cwnd = 1 MSS duplicate ACK
timeout dupACKcount = 0 dupACKcount++
ssthresh = cwnd/2 retransmit missing segment
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
timeout
New
ACK!
ssthresh = cwnd/2
cwnd = 1 New ACK
dupACKcount = 0
cwnd = ssthresh dupACKcount == 3
dupACKcount == 3 retransmit missing segment dupACKcount = 0
ssthresh= cwnd/2 ssthresh= cwnd/2
cwnd = ssthresh + 3 cwnd = ssthresh + 3
retransmit missing segment retransmit missing segment
fast
recovery
duplicate ACK
cwnd = cwnd + MSS
transmit new segment(s), as allowed

Transport Layer 3-103

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 3-104

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 = 2·10-10 Wow – a very small loss rate!
 new versions of TCP for high-speed

Transport Layer 3-105

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 3-106

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 3-107

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 connections;
 new app asks for 1 TCP, gets
rate R/10
 new app asks for 11 TCPs,
gets R/2 !

Transport Layer 3-108

Chapter 3: Summary
 principles behind transport
layer services:
 multiplexing,
demultiplexing
 reliable data transfer
 flow control Next:
 congestion control  leaving the network
 instantiation and “edge” (application,
implementation in the transport layers)
Internet  into the network
 UDP “core”
 TCP
Transport Layer 3-109