Chapter 4: Transport Layer

Magda El Zarki
Prof. of CS
UC Irvine
Chapter 4: Transport Layer
Our goals:
❒  understand principles ❒  learn about transport
behind transport layer protocols in the
layer services: Internet:
❍  multiplexing/ ❍  UDP: connectionless
demultiplexing transport
❍  reliable data transfer ❍  TCP: connection-oriented
❍  flow control transport
❍  congestion control ❍  TCP congestion control
Chapter 4 outline
❒  4.1 Transport-layer ❒  4.5 Connection-oriented
services transport: TCP
❒  4.2 Multiplexing and ❍  segment structure
demultiplexing ❍  reliable data transfer
❒  4.3 Connectionless ❍  flow control
connection management
transport: UDP ❍ 

❒  4.6 Principles of
❒  4.4 Principles of
reliable data transfer congestion control
❒  4.7 TCP congestion
control
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 vs. network layer
❒  network layer: logical
communication
between hosts
❒  transport layer: logical
communication
between processes
❍  relies on, enhances,
network layer services
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
Chapter 4 outline
❒  4.1 Transport-layer ❒  4.5 Connection-oriented
services transport: TCP
❒  4.2 Multiplexing and ❍  segment structure
demultiplexing ❍  reliable data transfer
❒  4.3 Connectionless ❍  flow control
connection management
transport: UDP ❍ 

❒  4.6 Principles of
❒  4.4 Principles of
reliable data transfer congestion control
❒  4.7 TCP congestion
control
 Multiplexing/demultiplexing
Demultiplexing at rcv host: Multiplexing at send host:
gathering data from multiple
delivering received segments
sockets, enveloping each data
to correct socket
with 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
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
port, and destination port
numbers application
❒  host uses IP addresses & port data
numbers to direct segment to (message)
appropriate socket

TCP/UDP segment format

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 same
dest port number, but
(dest IP address, dest port number)
different source IP
addresses and/or source
port numbers directed
to same socket
 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
❒  recv host uses all four different sockets for
values to direct each connecting client
segment to appropriate ❍  non-persistent HTTP will
socket have different socket for
each request
Connection-oriented demux
(cont)

P1 P4 P5 P6 P2 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
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
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
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 as ❒  compute checksum of
sequence of 16-bit received segment
integers ❒  check if computed checksum
❒  checksum: addition (1’s equals checksum field value:
complement sum) of ❍  NO - error detected
segment contents ❍  YES - no error detected.
❒  sender puts checksum But maybe errors
value into UDP checksum nonetheless? More later
field ….
 Internet Checksum Example
❒  Note
❍  When adding numbers, a carryout from the
most significant bit needs to be added to the
result
❒  Example: add two 16-bit integers

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

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

sum 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0
checksum 1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
Chapter 4 outline
❒  4.1 Transport-layer ❒  4.5 Connection-oriented
services transport: TCP
❒  4.2 Multiplexing and ❍  segment structure
demultiplexing ❍  reliable data transfer
❒  4.3 Connectionless ❍  flow control
connection management
transport: UDP ❍ 

❒  4.6 Principles of
❒  4.4 Principles of
reliable data transfer congestion control
❒  4.7 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)
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)
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 rdt
(e.g., by app.). Pass data to to deliver data to upper layer
deliver to receiver upper layer

send receive
side side

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

to transfer packet over arrives on rcv-side of channel
unreliable channel to receiver
Reliable data transfer: getting started
We’ll:
❒  incrementally develop sender, receiver sides of
reliable data transfer protocol (rdt)
❒  consider only unidirectional data transfer
❍  but control info will flow on both directions!
❒  use finite state machines (FSM) to specify
sender, receiver
event causing state transition
actions taken on state transition
state: when in this
“state” next state state state
1 event
uniquely determined 2
by next event actions
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)
sndpkt = 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
Λ

call from
sender below

rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
rdt2.0: operation with no errors
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for Wait for rdt_rcv(rcvpkt) &&
call from ACK or udt_send(sndpkt) corrupt(rcvpkt)
above NAK
udt_send(NAK)

rdt_rcv(rcvpkt) && isACK(rcvpkt)

Wait for
Λ
call from
below

rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
extract(rcvpkt,data)
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)
Wait for Wait for rdt_rcv(rcvpkt) &&
call from ACK or udt_send(sndpkt) corrupt(rcvpkt)
above NAK
udt_send(NAK)

rdt_rcv(rcvpkt) && isACK(rcvpkt)

Wait for
Λ
call from
below

rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
rdt2.0 has a fatal flaw!
What happens if ACK/ Handling duplicates:
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 – rdt 3.0
rdt2.1: sender, handles garbled ACK/NAKs
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt) rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
Wait for Wait for
ACK or
isNAK(rcvpkt) )
call 0 from
NAK 0 udt_send(sndpkt)
above
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) rdt_rcv(rcvpkt)
&& isACK(rcvpkt) && notcorrupt(rcvpkt)
&& isACK(rcvpkt)
Λ

Λ

Wait for Wait for
ACK or call 1 from
rdt_rcv(rcvpkt) && NAK 1 above
( corrupt(rcvpkt) ||
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)
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)
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 cannot
❒  twice as many states know if its last ACK/
❍  state must “remember” NAK received OK at
whether “current” pkt
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) &&
( 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) || Λ

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)
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 Λ

Λ
Wait for Wait
for timeout
call 0from
ACK0 udt_send(sndpkt)
above
start_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) rdt_rcv(rcvpkt)
&& isACK(rcvpkt,1) && notcorrupt(rcvpkt)
stop_timer && isACK(rcvpkt,0)
stop_timer
Wait Wait for
timeout for call 1 from
udt_send(sndpkt) ACK1 above
start_timer rdt_rcv(rcvpkt)
rdt_send(data) Λ

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

rdt3.0 in action
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
dtrans = = 9
= 8 microseconds
R 10 bps
❍  U sender: utilization – fraction of time sender busy sending

L/R .008
U = = = 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!
 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

L/R .008
U = = = 0.00027
sender 30.008
RTT + L / R microsec
onds
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!
3*L/R .024
U = = = 0.0008
sender 30.008
RTT + L / R microsecon
ds
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 in packet timer for each
reception (missing)
unacked packet
❒  Sender has timer for ❍  When timer expires,
oldest unacked packet retransmit only unack
❍  If timer expires, packet
retransmit all unacked
packets
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
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
Go-Back-N contd
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 #
GBN: sender extended FSM
rdt_send(data)
if (nextseqnum < base+N) {
sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum)
udt_send(sndpkt[nextseqnum])
if (base == nextseqnum)
start_timer
nextseqnum++
}
Λ
else
refuse_data(data)
base=1
nextseqnum=1
timeout
start_timer
Wait udt_send(sndpkt[base])
rdt_rcv(rcvpkt) udt_send(sndpkt[base+1])
&& corrupt(rcvpkt) …
udt_send(sndpkt[nextseqnum-1]
)
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
base = getacknum(rcvpkt)+1
If (base == nextseqnum)
stop_timer
else
start_timer
GBN: receiver extended FSM
default
udt_send(sndpkt) rdt_rcv(rcvpkt)
&& notcurrupt(rcvpkt)
Λ
&& hasseqnum(rcvpkt,expectedseqnum)
expectedseqnum=1 Wait extract(rcvpkt,data)
sndpkt = deliver_data(data)
make_pkt(expectedseqnum,ACK,chksum) sndpkt = make_pkt(expectedseqnum,ACK,chksum)
udt_send(sndpkt)
expectedseqnum++

ACK-only: always send ACK for correctly-received pkt

with highest in-order seq #
❍  may generate duplicate ACKs
❍  need only remember expectedseqnum
❒  out-of-order pkt:
❍  discard (don’t buffer) -> no receiver buffering!
❍  Re-ACK pkt with highest in-order seq #
GBN in
action
Selective Repeat
❒  receiverindividually 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 4 outline
❒  4.1 Transport-layer ❒  4.5 Connection-oriented
services transport: TCP
❒  4.2 Multiplexing and ❍  segment structure
demultiplexing ❍  reliable data transfer
❒  4.3 Connectionless ❍  flow control
connection management
transport: UDP ❍ 

❒  4.6 Principles of
❒  4.4 Principles of
reliable data transfer congestion control
(rdt) ❒  4.7 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) init’s
control set window size sender, receiver state
❒  send & receive buffers before data exchange
❒  flow controlled:
application application
socket
writes data reads data
socket ❍  sender will not
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 U A P R S F Receive window
(generally not used) # bytes
checksum Urg data pnter rcvr willing
RST, SYN, FIN: to accept
Options (variable length)
connection estab
(setup, teardown
commands)
application
Internet data
checksum (variable length)
(as in UDP)
 TCP seq. #’s and ACKs
Seq. #’s:
Host A Host B
❍  byte stream
“number” of first User Seq=4
2, ACK
types =79, d
byte in segment’s ‘C’
ata =
‘C’
data host ACKs
receipt of
ACKs: ’
C ‘C’, echoes
, d a ta = ‘
3
❍  seq # of next byte 7 9 , A CK=4 back ‘C’
S e q =
expected from
other side host ACKs
❍  cumulative ACK receipt Seq=4
of echoed 3, ACK
=80
Q: how receiver handles ‘C’
out-of-order segments
❍  A: TCP spec
doesn’t say, - up to time
simple telnet scenario
implementor
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
TCP Round Trip Time and Timeout
EstimatedRTT = (1- α)*EstimatedRTT + α*SampleRTT

❒  Exponential weighted moving average

❒  influence of past sample decreases exponentially fast
❒  typical value: α = 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

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

(typically, β = 0.25)

Then set timeout interval:

TimeoutInterval = EstimatedRTT + 4*DevRTT

Chapter 4 outline
❒  4.1 Transport-layer ❒  4.5 Connection-oriented
services transport: TCP
❒  4.2 Multiplexing and ❍  segment structure
demultiplexing ❍  reliable data transfer
❒  4.3 Connectionless ❍  flow control
connection management
transport: UDP ❍ 

❒  4.6 Principles of
❒  4.4 Principles of
reliable data transfer congestion control
❒  4.7 TCP congestion
control
TCP reliable data transfer
❒  TCP creates rdt ❒  Retransmissions are
service on top of IP’s triggered by:
unreliable service ❍  timeout events
❒  Pipelined segments ❍  duplicate acks
❒  Cumulative acks ❒  Initially consider
❒  TCP uses single
simplified TCP sender:
ignore duplicate acks
retransmission timer ❍ 
❍  ignore flow control,
congestion control
TCP sender events:
data rcvd from app: timeout:
❒  Create segment with ❒  retransmit segment
seq # that caused timeout
❒  seq # is byte-stream ❒  restart timer
number of first data Ack rcvd:
byte in segment ❒  If acknowledges
❒  start timer if not previously unacked
already running (think segments
of timer as for oldest ❍  update what is known to
unacked segment) be acked
❒  expiration interval: ❍  start timer if there are
TimeOutInterval outstanding segments
NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum

loop (forever) { TCP

sender
switch(event)

event: data received from application above

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

} /* end of loop forever */

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

Seq=9 Seq=9
2, 8 b 2, 8 b
ytes d ytes d
ata Seq= ata

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

ta
=100
ACK
X
loss
Seq=9 Seq=9
2, 8 b
2, 8 b
ytes d Sendbase ytes d
ata
ata
= 100

Seq=92 timeout
SendBase
= 120
= 100
ACK

SendBase
= 100 SendBase
= 120 premature timeout
time time
lost ACK scenario
 TCP retransmission scenarios (more)
Host A Host B

Seq=9
2, 8 b
ytes d
ata

=100
timeout

Seq=1 A C K
00, 20
bytes
data
X
loss

SendBase = 120
ACK
= 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 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.
 Host A Host B

X
timeout

resen
d 2 nd s
egme
nt

time

Figure 3.37 Resending a segment after triple duplicate ACK

 Fast retransmit algorithm:
don’t wait for timeout!
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 4 outline
❒  4.1 Transport-layer ❒  4.5 Connection-oriented
services transport: TCP
❒  4.2 Multiplexing and ❍  segment structure
demultiplexing ❍  reliable data transfer
❒  4.3 Connectionless ❍  flow control
connection management
transport: UDP ❍ 

❒  4.6 Principles of
❒  4.4 Principles of
reliable data transfer congestion control
❒  4.7 TCP congestion
control
TCP Flow Control
flow control
sender won’t overflow
❒  receive side of TCP receiver’s buffer by
connection has a transmitting too much,
receive buffer: too fast

❒  speed-matching
service: matching the
send rate to the
receiving app’s drain
rate
❒  app process may be
slow at reading from
buffer
TCP Flow control: how it works
❒  Rcvr advertises spare
room by including value
of RcvWindow in
segments
❒  Sender limits unACKed
(Suppose TCP receiver data to RcvWindow
discards out-of-order ❍  guarantees receive
segments) buffer doesn’t overflow
❒  spare room in buffer
= RcvWindow
= RcvBuffer-[LastByteRcvd -
LastByteRead]
Chapter 4 outline
❒  4.1 Transport-layer ❒  4.5 Connection-oriented
services transport: TCP
❒  4.2 Multiplexing and ❍  segment structure
demultiplexing ❍  reliable data transfer
❒  4.3 Connectionless ❍  flow control
connection management
transport: UDP ❍ 

❒  4.6 Principles of
❒  4.4 Principles of
reliable data transfer congestion control
❒  4.7 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 seq.
number");
#
❒  server: contacted by client
Step 3: client receives SYNACK,
Socket connectionSocket =
welcomeSocket.accept(); replies with ACK segment,
which may contain data
TCP Connection Management (cont.)

Closing a connection: client server

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

Step 1: client end system ACK

close
sends TCP FIN control FIN
segment to server

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

timed wait
ACK
.
closed

closed
TCP Connection Management (cont)

TCP server
lifecycle

TCP client
lifecycle
Chapter 4 outline
❒  4.1 Transport-layer ❒  4.5 Connection-oriented
services transport: TCP
❒  4.2 Multiplexing and ❍  segment structure
demultiplexing ❍  reliable data transfer
flow control
❒  4.3 Connectionless
❍ 
connection management
transport: UDP ❍ 

❒  4.6 Principles of
❒  4.4 Principles of
reliable data transfer congestion control
❒  4.7 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 λout
❒  two senders, two
λin : original data

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

infinite buffers
❒  no retransmission

❒  large delays
when congested
❒  maximum
achievable
throughput
 Causes/costs of congestion: scenario 2

❒  one router,finite buffers

❒  sender retransmission of lost packet

Host A λin : original data λout

λ'in : original data, plus

retransmitted data

Host B finite shared output

link buffers
 Causes/costs of congestion: scenario 2
❒  always: = λ

λ
(goodput)
in out
❒  “perfect” retransmission only when loss: λ
> λ

in out
❒  retransmission of delayed (not lost) packet makes λ
larger
in
(than perfect case) for same λ

out
R/2 R/2 R/2

R/3
λout

λout

λout
R/4

R/2 R/2 R/2

λin λin λin

a. b. c.
“costs” of congestion:
❒  more work (retrans) for given “goodput”
❒  unneeded retransmissions: link carries multiple copies of pkt
Causes/costs of congestion: scenario 3
❒  four senders
Q: what happens as λ

❒  multihop paths in
and λ
increase ?
❒  timeout/retransmit in
Host A λout
λin : original data
λ'in : original data, plus
retransmitted data

finite shared output

link buffers

Host B
Causes/costs of congestion: scenario 3
H λ
o
o
st u
A t

H
o
st
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 (TCP/IP
delay ECN)
❒  approach taken by TCP ❍  explicit rate sender
should send at
Case study: ATM ABR congestion control

ABR: available bit rate: RM (resource management)

❒  “elastic service” cells:
❒  if sender’s path ❒  sent by sender, interspersed
“underloaded”: with data cells
❍  sender should use ❒  bits in RM cell set by switches
available bandwidth (“network-assisted”)
❒  if sender’s path ❍  NI bit: no increase in rate
congested: (mild congestion)
❍  sender throttled to ❍  CI bit: congestion
minimum guaranteed indication
rate ❒  RM cells returned to sender by
receiver, with bits intact
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 4 outline
❒  4.1 Transport-layer ❒  4.5 Connection-oriented
services transport: TCP
❒  4.2 Multiplexing and ❍  segment structure
demultiplexing ❍  reliable data transfer
❒  4.3 Connectionless ❍  flow control
connection management
transport: UDP ❍ 

❒  4.6 Principles of
❒  4.4 Principles of
reliable data transfer congestion control
❒  4.7 TCP congestion
control
TCP congestion control: additive increase,
multiplicative decrease (AIMD)
❒  Approach: increase transmission rate (window size),
probing for usable bandwidth, until loss occurs
❍  additive increase: increase CongWin by 1 MSS
every RTT until loss detected
❍  multiplicative decrease: cut CongWin in half after
loss congestion
window
congestion window size

24 Kbytes

Saw tooth
behavior: probing
16 Kbytes

for bandwidth
8 Kbytes

time
time
 TCP Congestion Control: details
❒  sender limits transmission: How does sender
LastByteSent-LastByteAcked perceive congestion?
≤ CongWin ❒  loss event = timeout or
❒  Roughly, 3 duplicate acks
CongWin ❒  TCP sender reduces
rate = Bytes/sec
RTT rate (CongWin) after
❒  CongWin is dynamic, function
loss event
of perceived network three mechanisms:
congestion ❍  AIMD
❍  slow start
❍  conservative after
timeout events
 TCP Slow Start
❒  When connection begins, ❒  When connection begins,
CongWin = 1 MSS increase rate
❍  Example: MSS = 500 exponentially fast until
bytes & RTT = 200 msec first loss event
❍  initial rate = 20 kbps
❒  available bandwidth may
be >> MSS/RTT
❍  desirable to quickly ramp
up to respectable rate
TCP Slow Start (more)
❒  When connection Host A Host B
begins, increase rate
exponentially until
one segm
ent

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

❒  When CongWin is below Threshold, sender in

slow-start phase, window grows exponentially.

❒  When CongWin is above Threshold, sender is in

congestion-avoidance phase, window grows linearly.

❒  When a triple duplicate ACK occurs, Threshold

set to CongWin/2 and CongWin set to
Threshold.

❒  When timeout occurs, Threshold set to

CongWin/2 and CongWin is set to 1 MSS.
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start ACK receipt CongWin = CongWin + MSS, Resulting in a doubling of
(SS) for previously If (CongWin > Threshold) CongWin every RTT
unacked set state to “Congestion
data Avoidance”
Congestion ACK receipt CongWin = CongWin+MSS * Additive increase, resulting
Avoidance for previously (MSS/CongWin) in increase of CongWin by
(CA) unacked 1 MSS every RTT
data
SS or CA Loss event Threshold = CongWin/2, Fast recovery,
detected by CongWin = Threshold, implementing multiplicative
triple Set state to “Congestion decrease. CongWin will not
duplicate Avoidance” drop below 1 MSS.
ACK
SS or CA Timeout Threshold = CongWin/2, Enter slow start
CongWin = 1 MSS,
Set state to “Slow Start”
SS or CA Duplicate Increment duplicate ACK count CongWin and Threshold not
ACK for segment being acked changed
TCP throughput
❒  What’s the average throughout of TCP as
a function of window size and RTT?
❍  Ignore slow start
❒  Let W be the window size when loss occurs.
❒  When window is W, throughput is W/RTT
❒  Just after loss, window drops to W/2,
throughput to W/2RTT.
❒  Average throughout: .75 W/RTT
TCP Futures: 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
Wow
❒  ➜ L = 2·10-10
❒  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

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
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
constant rate, tolerate
packet loss
❒  Research area: TCP
friendly