Chapter 3

Transport Layer

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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
 3.3 Connectionless  flow control
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

lo
gi
ca
 transport protocols run in

le
nd
end systems

-e
nd
 send side: breaks app

tr
a
messages into segments,

ns
po
passes to network layer

rt
 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

lo
network
physical

gi
data link
flow control

ca
 physical

le
connection setup

nd


-e
nd
 unreliable, unordered network

tr
data link

a
physicalnetwork
delivery: UDP

ns
data link

po
physical

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

 delay guarantees
 bandwidth guarantees

Transport Layer 3-6

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

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

Transport Layer 3-7

 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 IP
address, destination IP address 32 bits
 each datagram carries 1
source port # dest port #
transport-layer segment
 each segment has source,
destination port number other header fields
 host uses IP addresses & port
numbers to direct segment to
appropriate socket
application
data
(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
 UDP socket identified by number
 IP datagrams with
two-tuple:
(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”

Transport Layer 3-11

 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 3-12

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 3-13

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 3-14

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

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

Transport Layer 3-15

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

Transport Layer 3-16

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 3-17

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

Sender: Receiver:
 treat segment contents  compute checksum of received
as sequence of 16-bit 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 3-18

 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 3-19
Chapter 3 outline
 3.1 Transport-layer  3.5 Connection-oriented
services transport: TCP
 3.2 Multiplexing and  segment structure
demultiplexing  reliable data transfer
 3.3 Connectionless  flow control
connection management
transport: UDP


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

Transport Layer 3-20

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

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

 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 3-24

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 3-25

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 3-26

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 3-27

rdt2.0: FSM specification
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
receiver
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)

Transport Layer 3-28

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
 call from
below

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

Transport Layer 3-29

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
 call from
below

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

Transport Layer 3-30

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

stop and wait

Sender sends one packet,
then waits for receiver
response

Transport Layer 3-31

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
( corrupt(rcvpkt) ||
isNAK(rcvpkt) ) rdt_send(data)

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

udt_send(sndpkt)

Transport Layer 3-32

 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) && rdt_rcv(rcvpkt) &&
(corrupt(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 3-33

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 3-34

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 3-35

 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) || 
has_seq1(rcvpkt)) Wait for
0 from
receiver FSM
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 3-36
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

Transport Layer 3-37

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

Transport Layer 3-38

rdt3.0 in action

Transport Layer 3-39

rdt3.0 in action

Transport Layer 3-40

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 microsec
onds
 1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link
 network protocol limits use of physical resources!

Transport Layer 3-41

 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 3-42

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 3-43
 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 3-44
Pipelining Protocols
Go-back-N: big picture: Selective Repeat: big pic
 Sender can have up to  Sender can have up to
N unacked packets in N unacked packets in
pipeline pipeline
 Rcvr only sends  Rcvr acks individual
cumulative acks packets
 Doesn’t ack packet if  Sender maintains
there’s a gap timer for each
 Sender has timer for unacked packet
oldest unacked packet  When timer expires,
 If timer expires, retransmit only unack
retransmit all unacked packet
packets

Transport Layer 3-45

Selective repeat: big picture
 Sender can have up to N unacked packets
in pipeline
 Rcvr acks individual packets
 Sender maintains timer for each unacked
packet
 When timer expires, retransmit only unack
packet

Transport Layer 3-46

Go-Back-N
Sender:
 k-bit seq # in pkt header
 “window” of up to N, consecutive unack’ed pkts allowed
 send_base: oldest unacked packet
 nextseqnum: smallest unused seqnum


nextseqnum - send_base ≤Sender Window Size (SWS=N)
 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 3-47

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-
 rdt_rcv(rcvpkt) &&
1])
notcorrupt(rcvpkt)
base = getacknum(rcvpkt)+1
If (base == nextseqnum)
stop_timer
else
restart_timer Transport Layer 3-48
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 (don’t buffer) -> no receiver buffering!
• Is it bad?
 Re-ACK pkt with highest in-order seq #
Transport Layer 3-49
GBN in
action

Transport Layer 3-50

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 3-51

Selective repeat: sender, receiver windows

Transport Layer 3-52

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), why?
advance window base to
next unACKed seq # otherwise:
 ignore

Transport Layer 3-53

Selective repeat in action

Transport Layer 3-54

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 3-55

Relationship b/w seq# &
window sizes
 LAF – LFR <= RWS
where LAF: Largest Acceptable frame
LFR: recv_base – 1: Last Frame Received
 RWS=1 for Go-Back-N protocol
 RWS <= SWS for Selective Repeat proto
 When RWS=1, SWS <= N -1 (N=size of seq# space=
2^k for k-bit seq no field)
 When RWS=SWS, SWS <= N/2

SW protocol alternates b/w two halves of seq# space

It also continually slides b/w two halves rather than discretely
alternating b/w them
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
 3.3 Connectionless  flow control
connection management
transport: UDP


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

Transport Layer 3-57

 TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581
 full duplex data:
 bi-directional data flow in same
 point-to-point: connection
 MSS: maximum segment size
 one sender, one receiver
(536, 1460)
 reliable, in-order byte  MSS is set based on MTU (MSS
= MTU – 40)
steam:  Path MTU Discovery

 no “message boundaries”  connection-oriented:

 pipelined:  handshaking (exchange of control

msgs) init’s sender, receiver
 TCP congestion and flow state before data exchange
control set window size  flow controlled:
 sender will not overwhelm
 send & receive buffers
receiver

a p p lic a t io n a p p lic a t io n
w r ite s d a ta re a d s d a ta
socket socket
door door
TC P TC P
s e n d b u ffe r r e c e iv e b u f f e r
segm ent

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 byte in User Seq=4
2, ACK=
segment’s data types 79, da
t a = ‘C
‘C’ ’
ACKs: host ACKs
 seq # of next byte receipt of
= ‘C’ ‘C’, echoes
expected from other 43 , data
, ACK
= back ‘C’
side Seq =79
 cumulative ACK

Q: how receiver handles host ACKs

out-of-order segments receipt Seq=4
of echoed 3, ACK
=80
 A: TCP spec doesn’t
‘C’
say, - up to
implementor
time
simple telnet scenario

Transport Layer 3-60

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”
 average several recent
retransmissions
 too long: slow reaction measurements, not just
to segment loss current SampleRTT

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
 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 3-64

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

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

Transport Layer 3-65

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 ack’ed byte
smallest sequence number Example:
start timer • SendBase-1 = 71;
y= 73, so the rcvr
event: ACK received, with ACK field value of y wants 73+ ;
if (y > SendBase) {
y > SendBase, so
SendBase = y
if (there are currently not-yet-acknowledged segments)
that new data is
start timer acked
}

} /* end of loop forever */

Transport Layer 3-68
 TCP: retransmission scenarios
Host A Host B Host A Host B

Seq=9 Seq=9
2, 8 b 2, 8 b y te
y te s d a s data
ta Seq=

Seq=92 timeout
1 00, 2
0 by t
es da
timeout

t a
=100
ACK 0
10
X CK=
A AC K = 120
loss
Seq=9 Seq=9
2, 8 b
2, 8 b
y t es da Sendbase y t es da
ta
ta
= 100

Seq=92 timeout
SendBase
= 120 = 120
0 K
=10 AC
ACK

SendBase
= 100 SendBase
= 120 premature timeout
time time
lost ACK scenario
Transport Layer 3-69
 TCP retransmission scenarios (more)
Host A Host B

Seq=9
2, 8 byte
s data

=100
timeout

Seq=1 A CK
00, 20
bytes
data
X
loss

SendBase C K =120
A
= 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  If sender receives 3
relatively long: ACKs for the same
 long delay before data, it supposes that
resending lost packet segment after ACKed
 Detect lost segments data was lost:
via duplicate ACKs.  fast retransmit: resend
 Sender often sends segment before timer
many segments back-to- expires
back
 If segment is lost,
there will likely be many
duplicate ACKs.

Transport Layer 3-72

 Host A Host B

X
timeout

resen
d 2 nd s
egme
nt

time

Figure 3.37 Resending a segment after triple duplicate ACK Layer

Transport 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
 3.3 Connectionless  flow control
connection management
transport: UDP


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

Transport Layer 3-75

TCP Flow Control
flow control
sender won’t overflow
 receive side of TCP
receiver’s buffer by
connection has a transmitting too
receive buffer: 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
 3.3 Connectionless  flow control
connection management
transport: UDP


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

Transport Layer 3-78

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

Transport Layer 3-79

TCP Connection Management (cont.)

Closing a connection: client server

client closes socket: close

FIN
clientSocket.close();

Step 1: client end system

ACK
sends TCP FIN control close
segment to server FIN

Step 2: server receives FIN,

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

Note: with small modification,

timed wait
ACK
can handle simultaneous FINs.
closed

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
 3.3 Connectionless  flow control
connection management
transport: UDP


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

Transport Layer 3-83

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 out
 two senders, two in : original data

receivers
unlimited shared
 one router,
Host B
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 lost packet

Host A in : original data out

'in : original data, plus

retransmitted data

Host B finite shared output

link buffers

Transport Layer 3-86

 Causes/costs of congestion: scenario 2
 always: = 
 (goodput)
in out
 “perfect” retransmission only when loss:  > out
in
 retransmission of delayed (not lost) packet makes 
in larger
(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
Transport Layer 3-87
Causes/costs of congestion: scenario 3
 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

Transport Layer 3-88

Causes/costs of congestion: scenario 3
H 
o
s o
t
u
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 3-89

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

Transport Layer 3-90

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 indication

minimum guaranteed  RM cells returned to sender by

rate receiver, with bits intact

Transport Layer 3-91

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 3-92

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

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

Transport Layer 3-93

 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 c o n g e s tio n
w in d o w
congestion window size

2 4 K b y te s

Saw tooth
behavior: probing
1 6 K b y te s

for bandwidth
8 K b y te s

time
tim e

Transport Layer 3-94

 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 of loss event
perceived network congestion three mechanisms:
 AIMD
 slow start
 conservative after
timeout events

Transport Layer 3-95

 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 3-96

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

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

Transport Layer 3-97

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

Transport Layer 3-98

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 3-99

FSM: TCP Congestion Control
New
New ACK!
ACK!
duplicate ACK
dupACKcount++ new ACK
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 retransmit missing segment
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
timeout
New
ACK!
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

Transport Layer 3-100

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 3-101

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
ACK for segment being acked not changed

Transport Layer 3-102

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 3-103

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
 New versions of TCP for high-speed

Transport Layer 3-104

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 3-105

Why is TCP fair?
Two competing sessions:
 Additive increase gives slope of 1, as throughout increases
 multiplicative decrease decreases throughput proportionally

R equal bandwidth share

Connection 2 throughput

loss: decrease window by factor of 2

congestion avoidance: additive increase
loss: decrease window by factor of 2
congestion avoidance: additive increase

Connection 1 throughput R

Transport Layer 3-106

Fairness (more)
Fairness and UDP Fairness and parallel TCP
 Multimedia apps often connections
 nothing prevents app from
do not use TCP
 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
 Research area: TCP
rate R/10
friendly  new app asks for 11 TCPs,
gets R/2 !

Transport Layer 3-107

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

Transport Layer 3-108

Useful Links
 http://www.internetweekly.org/llarrow/mt
umss.html

Transport Layer 3-109