Transport Layer: A Note On The Use of These PPT Slides

Chapter 3
Transport Layer
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note our copyright of this material. 2009.
Thanks and enjoy! JFK/KWR
All material copyright 1996-2009

J.F Kurose and K.W. Ross, All Rights Reserved
Transport Layer 3-1
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 3-2

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
3.6 Principles of
3.4 Principles of
reliable data transfer congestion control
3.7 TCP congestion
control
Transport Layer 3-3

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 3-4

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
network-layer protocol
= postal service
Transport Layer 3-5

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 3-6

Chapter 3 outline
flow control
3.3 Connectionless

transport: UDP
3.6 Principles of
3.4 Principles of
3.7 TCP congestion
control
Transport Layer 3-7

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 3-8
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 3-9

Connectionless demultiplexing
When host receives UDP
Create sockets with port
segment:
numbers:
DatagramSocket mySocket1 = new checks destination port
DatagramSocket(12534); number in segment
DatagramSocket mySocket2 = new directs UDP segment to
DatagramSocket(12535); socket with that port
number
UDP socket identified by
two-tuple: IP datagrams with
(dest IP address, dest port number) different source IP
addresses and/or source
port numbers directed
to same socket
Transport Layer 3-10

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

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
receiving host uses all different sockets for
four 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 P1P3
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

Connection-oriented demux:
Threaded Web Server
P1 P4 P2 P1P3
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
3.6 Principles of
3.4 Principles of
3.7 TCP congestion
control

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

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 (1s 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
Chapter 3 outline
flow control
3.3 Connectionless

transport: UDP
3.6 Principles of
3.4 Principles of
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)


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
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

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

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

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
call from
below
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
deliver_data(data)
udt_send(ACK)

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
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 doesnt know what pkt if ACK/NAK garbled
happened at receiver! sender adds sequence
cant just retransmit: number to each pkt
possible duplicate receiver discards (doesnt
deliver up) duplicate pkt
stop and wait

Sender sends one packet,
then waits for receiver
response

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)

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)

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

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) ||
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 3-36
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

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
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)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) || sndpkt = make_pkt(1, data, checksum)
isACK(rcvpkt,0) ) udt_send(sndpkt)
start_timer


rdt3.0 in action

rdt3.0 in action

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 microseconds
R 10 bps
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!

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
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

Pipelining Protocols
Go-back-N: overview Selective Repeat: overview
sender: up to N sender: up to N unACKed
unACKed pkts in packets in pipeline
pipeline receiver: ACKs individual
receiver: only sends pkts
cumulative ACKs sender: maintains timer
doesnt ACK pkt if for each unACKed pkt
theres a gap
if timer expires: retransmit
sender: has timer for only unACKed packet
oldest unACKed pkt
if timer expires:
retransmit all unACKed
packets

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
timeout(n): retransmit pkt n and all higher seq # pkts in window

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++
}
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 3-47
GBN: receiver extended FSM
default
udt_send(sndpkt) rdt_rcv(rcvpkt)
&& notcurrupt(rcvpkt)
&& 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 #
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

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

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?
Chapter 3 outline
flow control
3.3 Connectionless

transport: UDP
3.6 Principles of
3.4 Principles of
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

MSS: maximum segment
steam:
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

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 pointer
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
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
implementer
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

EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT
Exponential weighted moving average

influence of past sample decreases exponentially fast
typical value: = 0.125

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-)*DevRTT +
*|SampleRTT-EstimatedRTT|
(typically, = 0.25)
Then set timeout interval:
TimeoutInterval = EstimatedRTT + 4*DevRTT

Chapter 3 outline
flow control
3.3 Connectionless

transport: UDP
3.6 Principles of
3.4 Principles of
3.7 TCP congestion
control

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

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 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 */

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
TCP retransmission scenarios (more)
Host A Host B
timeout
X
loss
SendBase
= 120
time
Cumulative ACK scenario

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

Fast Retransmit
time-out period often If sender receives 3
relatively long: ACKs for same data, it
long delay before assumes that segment
resending lost packet after ACKed data was
detect lost segments 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 for that
segment

Host A Host B
seq # x1
seq # x2
seq # x3 ACK x1
seq # x4 X
seq # x5
ACK x1
ACK x1
ACK x1
triple
duplicate
ACKs
timeout
time

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
3.6 Principles of
3.4 Principles of
3.7 TCP congestion
control

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
IP
(currently)
TCP data application
speed-matching
unused buffer
datagrams space
(in buffer) process service: matching
send rate to receiving
applications drain rate
app process may be
slow at reading from
buffer
TCP Flow control: how it works
IP
(currently)
TCP data application
receiver: advertises
unused buffer
datagrams space
(in buffer) process unused buffer space by
including rwnd value in
rwnd segment header
RcvBuffer
sender: limits # of
(suppose TCP receiver unACKed bytes to rwnd
discards out-of-order guarantees receivers
segments) buffer doesnt overflow
unused buffer space:
= rwnd
= RcvBuffer-[LastByteRcvd -
LastByteRead]

Chapter 3 outline
flow control
3.3 Connectionless

transport: UDP
3.6 Principles of
3.4 Principles of
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
specifies server initial
number");
seq. #
server: contacted by client
Socket connectionSocket =
Step 3: client receives SYNACK,
welcomeSocket.accept(); replies with ACK segment,
which may contain data

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

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

TCP Connection Management (cont)
TCP server
lifecycle
TCP client
lifecycle

Chapter 3 outline
flow control
3.3 Connectionless

transport: UDP
3.6 Principles of
3.4 Principles of
3.7 TCP congestion
control

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
out
two senders, two
in : original data
receivers
one router,
Host B unlimited shared
output link buffers
infinite buffers
no retransmission
large delays
when congested
maximum
achievable
throughput
one router, finite buffers

sender retransmission of lost packet
Host A in : original out

data
'in : original data, plus
retransmitted data
Host B finite shared output

link buffers

always: =
(goodput)
in out
perfect retransmission only when loss: > out
in
retransmission of delayed (not lost) packet makes larger
in
(than perfect case) for same out
R/2 R/2 R/2
R/3
out
out
out
R/4
R/2 R/2 R/2

in in in
a. b. c.
costs of congestion:
more work (retrans) for given goodput
unneeded retransmissions: link carries multiple copies of pkt
four senders
Q: what happens as
multihop paths in
and increase ?
timeout/retransmit in
Host A out
in : original data
'in : original data, plus
retransmitted data
finite shared output

link buffers
Host B

H
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!

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 senders path sent by sender, interspersed
underloaded: with data cells
sender should use bits in RM cell set by switches
available bandwidth (network-assisted)
if senders 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

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

Chapter 3 outline
flow control
3.3 Connectionless

transport: UDP
3.6 Principles of
3.4 Principles of
3.7 TCP congestion
control

TCP congestion control:
goal: TCP sender should transmit as fast as possible,
but without congesting network
Q: how to find rate just below congestion level
decentralized: each TCP sender sets its own rate,
based on implicit feedback:
ACK: segment received (a good thing!), network not
congested, so increase sending rate
lost segment: assume loss due to congested
network, so decrease sending rate

TCP congestion control: bandwidth probing
probing for bandwidth: increase transmission rate
on receipt of ACK, until eventually loss occurs, then
decrease transmission rate
continue to increase on ACK, decrease on loss (since available
bandwidth is changing, depending on other connections in
network)
ACKs being received,
X loss, so decrease rate
so increase rate
X
X
X
sending rate
TCPs
X sawtooth
behavior
time
Q: how fast to increase/decrease?

details to follow Transport Layer 3-93
TCP Congestion Control: details
sender limits rate by limiting number

of unACKed bytes in pipeline:
LastByteSent-LastByteAcked cwnd
cwnd: differs from rwnd (how, why?)
sender limited by min(cwnd,rwnd)
cwnd
roughly, bytes
cwnd
rate = bytes/sec
RTT
RTT
cwnd is dynamic, function of
perceived network congestion ACK(s)

TCP Congestion Control: more details
segment loss event: ACK received: increase

reducing cwnd cwnd
timeout: no response slowstart phase:
from receiver increase exponentially
cut cwnd to 1 fast (despite name) at
3 duplicate ACKs: at connection start, or
following timeout
least some segments
getting through (recall congestion avoidance:
increase linearly
fast retransmit)
cut cwnd in half, less
aggressively than on
timeout

TCP Slow Start
when connection begins, cwnd =
1 MSS Host A Host B
example: MSS = 500 bytes
& RTT = 200 msec
RTT
initial rate = 20 kbps
available bandwidth may be >>
MSS/RTT
desirable to quickly ramp up
to respectable rate
increase rate exponentially
until first loss event or when
threshold reached
double cwnd every RTT
time
done by incrementing cwnd
by 1 for every ACK received
Transitioning into/out of slowstart
ssthresh: cwnd threshold maintained by TCP
on loss event: set ssthresh to cwnd/2
remember (half of) TCP rate when congestion last occurred
when cwnd >= ssthresh: transition from slowstart to congestion
avoidance phase
duplicate ACK
dupACKcount++ new ACK
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s),as allowed
cwnd = 1 MSS
ssthresh = 64 KB cwnd > ssthresh
dupACKcount = 0 slow congestion
start timeout avoidance
ssthresh = cwnd/2
cwnd = 1 MSS
timeout dupACKcount = 0
retransmit missing segment
ssthresh = cwnd/2
cwnd = 1 MSS
dupACKcount = 0

TCP: congestion avoidance
when cwnd > ssthresh AIMD
grow cwnd linearly
ACKs: increase cwnd
increase cwnd by 1
by 1 MSS per RTT:
MSS per RTT additive increase
approach possible loss: cut cwnd in half
congestion slower (non-timeout-detected
than in slowstart loss ): multiplicative
implementation: cwnd decrease
= cwnd + MSS/cwnd
for each ACK received AIMD: Additive Increase
Multiplicative Decrease

TCP congestion control FSM: overview
slow cwnd > ssthresh

congestion
start avoidance
loss:
timeout
loss:
timeout
loss: new ACK loss:

timeout 3dupACK
fast
loss: recovery
3dupACK

TCP congestion control FSM: details
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
transmit new segment(s),as allowed
cwnd = 1 MSS
ssthresh = 64 KB cwnd > ssthresh
dupACKcount = 0 slow congestion
start timeout avoidance
ssthresh = cwnd/2
cwnd = 1 MSS duplicate ACK
timeout dupACKcount = 0
dupACKcount++
ssthresh = cwnd/2
cwnd = 1 MSS
dupACKcount = 0
timeout
ssthresh = cwnd/2
cwnd = 1 New ACK
dupACKcount = 0
retransmit missing segment cwnd = ssthresh dupACKcount == 3
dupACKcount == 3 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

Popular flavors of TCP
cwnd window size (in segments)
TCP Reno
ssthresh
ssthresh
TCP Tahoe
Transmission round

Summary: TCP Congestion Control
when cwnd < ssthresh, sender in slow-start

phase, window grows exponentially.
when cwnd >= ssthresh, sender is in congestion-
avoidance phase, window grows linearly.
when triple duplicate ACK occurs, ssthresh set
to cwnd/2, cwnd set to ~ ssthresh
when timeout occurs, ssthresh set to cwnd/2,

cwnd set to 1 MSS.

TCP throughput
Q: whats average throughout of TCP as
function of window size, RTT?
ignoring slow start
let W be 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: 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

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
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

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 !

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

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Chapter 3

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All material copyright 1996-2009

Transport Layer 3-2

Transport Layer 3-3

running on different hosts

more than one transport

Transport Layer 3-4

Transport Layer 3-5

Transport Layer 3-6

Transport Layer 3-7

transport transport transport

network network network

link link link

physical physical physical

each datagram carries 1

TCP/UDP segment format

Transport Layer 3-9

Transport Layer 3-10

SP: 6428 SP: 6428

SP: 9157 SP: 5775

SP provides return address

Transport Layer 3-11

Transport Layer 3-12

SP: 9157 SP: 9157

Transport Layer 3-13

SP: 9157 SP: 9157

Transport Layer 3-14

Transport Layer 3-15

Transport Layer 3-16

loss tolerant Length, in source port # dest port #

Transport Layer 3-17

Transport Layer 3-18

Transport Layer 3-20

characteristics of unreliable channel will determine

characteristics of unreliable channel will determine

characteristics of unreliable channel will determine

udt_send(): called by rdt, rdt_rcv(): called when packet

Transport Layer 3-24

Transport Layer 3-25

underlying channel perfectly reliable

separate FSMs for sender, receiver:

Wait for rdt_send(data) Wait for rdt_rcv(packet)

Transport Layer 3-26

the question: how to recover from errors:

Transport Layer 3-27

rdt_rcv(rcvpkt) && isACK(rcvpkt)

Transport Layer 3-28

rdt_rcv(rcvpkt) && isACK(rcvpkt)

Transport Layer 3-29

rdt_rcv(rcvpkt) && isACK(rcvpkt)

Transport Layer 3-30

stop and wait

Transport Layer 3-31

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

Transport Layer 3-32

Transport Layer 3-33

Transport Layer 3-34

same functionality as rdt2.1, using ACKs only

Transport Layer 3-35

New assumption: Approach: sender waits

Transport Layer 3-37

Transport Layer 3-38