Reti di calcolatori e
Sicurezza
-- Transport Layer ---
Part of these slides are adapted from the slides of the book:
Computer Networking: A Top Down Approach Featuring the Internet,
2nd edition.
Jim Kurose, Keith Ross
Addison-Wesley, July 2002.
(copyright 1996-2002
J.F Kurose and K.W. Ross, All Rights Reserved)
Transport Layer
3-1
Chapter 3: Transport Layer
Our goals:
 understand principles
behind transport
layer services:




multiplexing/demultipl
exing
reliable data transfer
flow control
congestion control
 learn about transport
layer protocols in the
Internet:



UDP: connectionless
transport
TCP: connection-oriented
transport
TCP congestion control
Transport Layer
3-2
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
 3.4 Principles of
reliable data transfer
 3.5 Connection-oriented
transport: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer
3-3
Transport services and protocols
 provide logical communication
between app processes
running on different hosts
 transport protocols run in
end systems
 send side: breaks app
messages into segments,
passes to network layer
 rcv side: reassembles
segments into messages,
passes to app layer
 more than one transport
protocol available to apps
 Internet: TCP and UDP
application
transport
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
application
transport
network
data link
physical
Transport Layer
3-4
Transport vs. network layer
 network layer: logical
communication
between hosts
 transport layer: logical
communication
between processes

relies on, enhances,
network layer services
Household analogy:
12 kids sending letters to
12 kids
 processes = kids
 app messages = letters
in envelopes
 hosts = houses
 transport protocol =
Ann and Bill
 network-layer protocol
= postal service
Transport Layer
3-5
Internet transport-layer protocols
 reliable, in-order
delivery (TCP)



congestion control
flow control
connection setup
 unreliable, unordered
delivery: UDP

no-frills extension of
“best-effort” IP
 services not available:
 delay guarantees
 bandwidth guarantees
application
transport
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
application
transport
network
data link
physical
Transport Layer
3-6
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
 3.4 Principles of
reliable data transfer
 3.5 Connection-oriented
transport: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer
3-7
Un primo servizio
 Network = trasferimento di dati tra host
della rete

Host identificato in modo univoco da un
indirizzo IP
 Come si fa a traferire i dati ricevuti ai
processi che li hanno richiesti?

Multiplexing -- demultiplexing
Transport Layer
3-8
Porte e processi
 Porta identifica in modo univoco un
processo in esecuzione su di un host
Porte di “sistema”: 0 – 1023
 FTP – 21, HTTP – 80,
 RFC 1700

 Applicazioni di rete
 Porte maggiori di 1024
Transport Layer
3-9
Multiplexing/demultiplexing
segment – unità di
trasferimento di dati delle
entità coinvolte al livello
del trasporto
TPDU: transport protocol data
unit
application-layer
data
segment
header
segment
Ht M
Hn segment
P1
M
application
transport
network
P3
Demultiplexing: invio dei
segmenti ricevuti ai processi
receiver
M
M
application
transport
network
P4
M
P2
application
transport
network
Transport Layer 3-10
Multiplexing/demultiplexing
Multiplexing (host invio):
Raccogliere i dati,
Inviarli in rete
con l’indirizzo corretto
Demultiplexing (host recezione):
Inviare i msg ricevuti al socket
corretto
= socket
application
transport
network
link
= process
P3
P1
P1
application
transport
network
P2
P4
application
transport
network
link
link
physical
host 1
physical
host 2
physical
host 3
Transport Layer
3-11
Demultiplexing: modalità di
funzionamento
 host riceve un msg IP
(datagrams)
 Ogni datagram è
caratterizzato da una
coppia di indirizzi IP
(mittente, destinatario)
 ogni datagram come paylod
ha un msg di trasporto
 Msg del trasporto ha le
informazioni dulle porte
del mittente e del
destinatario
 Indirizzi IP & porte sono
utilizzate per inviare il msg al
socket corretto
32 bits
source port #
dest port #
other header fields
application
data
(message)
TCP/UDP segment format
Transport Layer 3-12
Connectionless demultiplexing
 Creazione del socket (in un
qualche modo)
 Socket UDP sono
identificati da:
(dest IP address, dest port number)
 Alla recezione del
segmento UDP:


Si contralla la porta di
destinazione
Invia il segmento al socket
in ascolto su quella porta
 datagrams con IP e porta
del mittente diverse
possono essere inviati
allo stesso socket.
Transport Layer 3-13
Connectionless demux
DatagramSocket serverSocket = new DatagramSocket(6428);
P3
SP: 6428
SP: 6428
DP: 9157
DP: 5775
SP: 9157
client
IP: A
P1
P1
P3
DP: 6428
SP: 5775
server
IP: C
DP: 6428
Client
IP:B
Transport Layer 3-14
Connection-oriented demux
 TCP socket sono
identificati




source IP address
source port number
dest IP address
dest port number
 Host in recezione
utilizza queste quattro
informazioni per inviare
il msg a destinazione
 Server puo’ operare in
modalità multithreading
(pertanto sono attivi
diversi socket)

Ogni socket attivo è
identificato in modo
univoco da quadrupla di
valori
Transport Layer 3-15
Connection-oriented demux
(cont)
P3
P3
SP: 80
SP: 80
DP: 9157
DP: 5775
SP: 9157
client
IP: A
DP: 80
P1
P1
P4
SP: 5775
server
IP: C
DP: 80
Client
IP:B
Transport Layer 3-16
Multiplexing/demultiplexing
host A
source port: x
dest. port: 23
server B
source port:23
dest. port: x
Source IP: C
Dest IP: B
source port: y
dest. port: 80
telnet app
Web client
host A
Web client
host C
Source IP: A
Dest IP: B
source port: x
dest. port: 80
Source IP: C
Dest IP: B
source port: x
dest. port: 80
Web
server B
Web server
Transport Layer 3-17
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
 3.4 Principles of
reliable data transfer
 3.5 Connection-oriented
transport: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-18
UDP: User Datagram Protocol [RFC 768]
 Protocollo di trasporto
essenziale
 Servizio di tipo “best
effort”
 Segmenti UDP :
 perdere
 senza ordine
 connectionless:
 no handshaking
 Ogni segmento UDP è
gestito in modo
totalmente
independente dagli altri
segmenti
Quando è necessario
utilizzare UDP?
 Non c’è la necessità di
stabilire una connessione
 Applicazioni semplici: non
abbiamo bisogno di
informazioni di stato
 Header dei segmenti piccoli
 Nessun controllo per
evitare i problemi di
congestione
Transport Layer 3-19
UDP
 Multimedia

Possono permettersi
una perdita di
Length, in
informazione
bytes of UDP
segment,
 Chi utilizza UDP
including
 DNS
header
 Aggiungere un meccanismo
di affidabilità ad UDP
 error recover al livello
delle applicazioni
32 bits
source port #
dest port #
length
checksum
Application
data
(message)
UDP segment format
Transport Layer 3-20
UDP checksum
Obiettivo: scoprire eventuali errori di trasmissione
Sender:
 Segmenti sono visti come
sequenze di interi di 16bit
 checksum: complemento
ad 1 della somma di tutti
gli interi che compongono
il segmento
 Checksum-value viene
inserito nel campo del
segmento denominato
checksum
Receiver:
 Calcola il valore di checksum del
segmento
 Controlla se il valore calcolato
corrisponde al valore
memorizzato nel campo
checksum del segmento:
 NO - error detected
 YES - no error detected.
 Potrebbero essere presenti
altri errori
Transport Layer 3-21
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
 3.4 Principles of
reliable data transfer
 3.5 Connection-oriented
transport: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-22
Trasferimento affidabilke
 Punto importante nei livelli app., transport, link.
 Una delle problematiche piu’ importanti del networking
Transport Layer 3-23
Reliable data transfer: RDT
rdt_send(): invocata dal livello
delle applicazioni
send
side
udt_send(): trasferire
pacchetti su di un canale non
affidabile
deliver_data(): invia i
dati ai livelli superiori
receive
side
rdt_rcv(): chiamata al momento
della recezione dei pacchetti
Transport Layer 3-24
Automi e trasferimento affidabile
 Trasferimento di dati (unidirezionale)
 Ma con meccanismi di controllo del flusso.
 Sender e receiver sono specificati da automi a
stati finiti.
event causing state transition
actions taken on state transition
state: when in this
“state” next state
uniquely determined
by next event
state
1
event
actions
state
2
Transport Layer 3-25
Rdt1.0: reliable transfer over a reliable channel
 Il canale di trasmissione è affidabile
 no bit erros
 no loss of packets
 Sender e receiver (fanno le ovvie cose!!)
Wait for
call from
above
rdt_send(data)
packet = make_pkt(data)
udt_send(packet)
sender
Wait for
call from
below
rdt_rcv(packet)
extract (packet,data)
deliver_data(data)
receiver
Transport Layer 3-26
Rdt2.0: channel with bit errors
 Il canale puo’ modificare il valore dei bit presenti
nei pacchetti

UDP checksum
 Come vengono determinati questi errori di
trasmissione:



acknowledgements (ACKs): receiver invia al sender un
messaggio di recezione (senza errori) del pacchetto
negative acknowledgements (NAKs): receiver invia al
sender un messaggio di errore
Sender trasmette nuovamente il pacchetto quando riceve
un messaggio NAK
 Novità (protocolli ARQ)
 error detection
 receiver feedback (ACK,NAK)
 Ritrasmissione
Transport Layer 3-27
rdt2.0: la specifica (FSM)
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for
Wait for
call from
ACK or
udt_send(sndpkt)
above
NAK
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
sender
receiver
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
udt_send(NAK)
Wait for
call from
below
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
Transport Layer 3-28
rdt2.0: assenza di errori
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for
Wait for
call from
ACK or
udt_send(sndpkt)
above
NAK
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
udt_send(NAK)
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: Scenario con errore
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for
Wait for
call from
ACK or
udt_send(sndpkt)
above
NAK
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
Anche detti protocolli Stop-and-Wait
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
udt_send(NAK)
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 ha un fatal flaw!
Cosa succede quando I
messaggi ACK/NAK
sono corrotti?
 sender non è in grado di
sapere cosa è successo al
receiver
 Si possono duplicare i
messaggi trasmessi
Come rimediare?
 sender ACKs/NAKs e
receiver’s ACK/NAK? Cosa
accade in caso di perdita
del sender ACK/NAK?
Gestione dei messaggi
duplicati:
 sender aggiunge ad ogni
pacchetto il sequence
number
 receiver non trasmette alle
applicazioni i pacchetti
duplicati
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
isNAK(rcvpkt) )
ACK or
call 0 from
udt_send(sndpkt)
NAK 0
above
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt)
L
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isNAK(rcvpkt) )
udt_send(sndpkt)
L
Wait for
ACK or
NAK 1
Wait for
call 1 from
above
rdt_send(data)
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)
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(NAK, chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) &&
has_seq1(rcvpkt)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)
Duplicato!!!!
sndpkt = make_pkt(NAK, chksum)
udt_send(sndpkt)
Wait for
0 from
below
Wait for
1 from
below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) &&
has_seq0(rcvpkt)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)
Transport Layer 3-33
rdt2.1
Sender:
 Pacchetti con numero
di sequenza (basta un
solo bit di
informazione)
 Stati:

In ogni stato si deve
ricordare il numero di
sequenz (0 o 1)
Receiver:
 Deve controllare se il
pacchetto è duplicato

Ogni stato mantiene
informazione su quale
numero di sequenza si
attende
Transport Layer 3-34
rdt2.2: Protocollo NAK-free
 Come rdt 2.1 ma ...
 Invece di inviare un msg NAK, il receiver invia un msg di ACK
per l’ultimo pacchetto ricevuto correttamente


Informazioni di stato
“receiver must explicitly include seq # of pkt being ACKed”
 ACK duplicato (lato sender) = NAK: ristrasmissione del pkt
corrente
Transport Layer 3-35
rdt2.2: sender e receiver (in pillole)
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
Wait for
Wait for
isACK(rcvpkt,1) )
ACK
call 0 from
0
udt_send(sndpkt)
above
sender FSM
fragment
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt) ||
has_seq1(rcvpkt))
sndpkt = make_pkt(ACK1,
chksum)
udt_send(sndpkt)
Wait for
0 from
below
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
receiver FSM
fragment
L
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq0(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK0, chksum)
udt_send(sndpkt)
Transport Layer 3-36
rdt3.0: Canali con errori e perdita di
pacchetti
Ipotesi aggiuntive: Il canale di
trasmissione puo’ perdere
pacchetti (dati o ACK)

checksum, seq. #, ACK,
sono sufficienti?
Come si gestisce la perdita di
pacchetti?

sender attende un certo
periodo di tempo prima di
trasmettere nuovamente
l’informazione
Una possibile soluzione: sender
rimane in attesa per un
periodo di tempo ragionevole
 I pkt vengono trasmessi
nuovamente se non viene ricevuto
un ACK in questo periodo di
tempo
 Se la consegna del pkt (ACK) è
solo ritardata (non avviene
perdita)
 Duplicazione: gestita tramite
il # di sequenza
 receiver specifica il # di
sequenza del pkt ricevuto
 countdown timer
Transport Layer 3-37
rdt3.0 sender
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
start_timer
rdt_rcv(rcvpkt)
L
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,1)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,0) )
timeout
udt_send(sndpkt)
start_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
stop_timer
stop_timer
timeout
udt_send(sndpkt)
start_timer
L
Wait
for
ACK0
Wait for
call 0from
above
L
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
Wait
for
ACK1
Wait for
call 1 from
above
rdt_send(data)
rdt_rcv(rcvpkt)
L
sndpkt = make_pkt(1, data, checksum)
udt_send(sndpkt)
start_timer
Transport Layer 3-38
rdt3.0: esempio
Transport Layer 3-39
rdt3.0: esempio
Transport Layer 3-40
rdt3.0
 rdt3.0 funziona correttamente ma esibisce dei problemi di
efficienza relativi all’uso della banda di trasmissione
Transport Layer 3-41
rdt3.0: stop-and-wait
sender
receiver
first packet bit transmitted, t = 0
last packet bit transmitted, t = L / R
first packet bit arrives
last packet bit arrives, send ACK
RTT
ACK arrives, send next
packet, t = RTT + L / R
U
=
sender
L/R
RTT + L / R
=
.008
30.008
= 0.00027
microsec
onds
Transport Layer 3-42
Pipeline
Pipelining: il mittente invia un certo numero di
pacchetti senza attendere il relativo ACK


Operare correttamente con i # di sequenza
Buffer (mittente e destinatario)
 Due tipi di protocolli: 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
last packet bit arrives, send ACK
last bit of 2nd packet arrives, send ACK
last bit of 3rd packet arrives, send ACK
RTT
ACK arrives, send next
packet, t = RTT + L / R
Increase utilization
by a factor of 3!
U
sender
=
3*L/R
RTT + L / R
=
.024
30.008
= 0.0008
microsecon
ds
Transport Layer 3-44
Protocolli “sliding window”
Transport Layer 3-45
Go-Back-N
Sender:
 Header; k-bit per memorizzare i numeri di sequenza dei pkt.
 Si permette di avere una “finestra fino ad N”, di pkt consecutivi in cui
non è stato ricevuto il relativo ack
 ACK(n): ACK cumulativo dei pkt con # minore di n
 timer per i pkt in trasmissione
 timeout(n): trasmettere il pkt n e tutti i pkt nella parte superiore
della finestra di trasmissione
Transport Layer 3-46
GBN: lato sender
rdt_send(data)
L
base=1
nextseqnum=1
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)
Wait
rdt_rcv(rcvpkt)
&& corrupt(rcvpkt)
timeout
start_timer
udt_send(sndpkt[base])
udt_send(sndpkt[base+1])
…
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: lato receiver
default
udt_send(sndpkt)
L
Wait
expectedseqnum=1
sndpkt =
make_pkt(expectedseqnum,ACK,chksum)
rdt_rcv(rcvpkt)
&& notcurrupt(rcvpkt)
&& hasseqnum(rcvpkt,expectedseqnum)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(expectedseqnum,ACK,chksum)
udt_send(sndpkt)
expectedseqnum++
ACK viene inviato per i pkt corretti aventi il piu’ slto
numero seq #


ACK duplicati
Variabile di stato: expectedseqnum
 out-of-order pkt:
 discard -> no buffering!
 ACK pkt con il piu’ alto seq #
Transport Layer 3-48
GBN in
action
Transport Layer 3-49
Selective Repeat
 receiver invia ACK di tutti i pkt ricevuti
correttamente.



Buffer per gestire l’ordine dei pacchetti
Sender invia nuovamente i pkt senza ACK
Sender attiva timer per ogni pkt senza ACK
 La finestra del sender:
 N # di sequenza consecutivi
 Limite superiore alla dimensione della finestra
Transport Layer 3-50
Selective repeat: sender, receiver windows
Transport Layer 3-51
Selective repeat
sender
data from above :
receiver
pkt n in [rcvbase, rcvbase+N-1]
 if next available seq # in
 send ACK(n)
timeout(n):
 in-order: deliver (also
window, send pkt
 resend pkt n, restart timer
ACK(n) in [sendbase,sendbase+N]:
 mark pkt n as received
 if n smallest unACKed pkt,
advance window base to
next unACKed seq #
 out-of-order: buffer
deliver buffered, in-order
pkts), advance window to
next not-yet-received pkt
pkt n in
[rcvbase-N,rcvbase-1]
 ACK(n)
otherwise:
 ignore
Transport Layer 3-52
Selective repeat
Transport Layer 3-53
Selective repeat
 seq #’s: 0, 1, 2, 3
 window size=3
 receiver non riesce a
discriminare i due
comportamenti
 Window di dimensione
inferiore allo spazio
dei numeri di sequenza
Transport Layer 3-54
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
 3.4 Principles of
reliable data transfer
 3.5 Connection-oriented
transport: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-55
TCP: Overview
 point-to-point:
 one sender, one receiver
 reliable, in-order byte
stream:

no “message boundaries”
 pipelined:
 TCP congestion and flow
control set window size
 send & receive buffers
socket
door
application
writes data
application
reads data
TCP
send buffer
TCP
receive buffer
RFCs: 793, 1122, 1323, 2018, 2581
 full duplex data:
 bi-directional data flow
in same connection
 MSS: maximum segment
size
 connection-oriented:
 handshaking (exchange
of control msgs) init’s
sender, receiver state
before data exchange
 flow controlled:
 sender will not
socket
door
overwhelm receiver
segment
Transport Layer 3-56
TCP segment structure
32 bits
URG: urgent data
(generally not used)
ACK: ACK #
valid
PSH: push data now
(generally not used)
RST, SYN, FIN:
connection estab
(setup, teardown
commands)
Internet
checksum
(as in UDP)
source port #
dest port #
sequence number
acknowledgement number
head not
UA P R S F
len used
checksum
Receive window
Urg data pnter
Options (variable length)
counting
by bytes
of data
(not segments!)
# bytes
rcvr willing
to accept
application
data
(variable length)
Transport Layer 3-57
Il segment TCP
 Connessione:
 (SrcPort, SrcIPAddr, DsrPort, DstIPAddr)
 window + flow control
 acknowledgment, SequenceNum, RcvdWinow
Data(SequenceNum)
Sender
Receiver
Acknowledgment +
RcvdWindow
 Flags
 SYN, FIN, RESET, PUSH, URG, ACK
 Checksum
 pseudo header + TCP header + data
Transport Layer 3-58
TCP seq. #’s and ACKs
Seq. #’s:
 byte stream
“number” of first
byte in segment’s
data
ACKs:
 seq # of next byte
expected from
other side
 cumulative ACK
Q: how receiver handles
out-of-order segments
 A: TCP spec doesn’t
say, - up to
implementor
Host A
User
types
‘C’
Host B
host ACKs
receipt of
‘C’, echoes
back ‘C’
host ACKs
receipt
of echoed
‘C’
simple telnet scenario
time
Transport Layer 3-59
TCP Round Trip Time and Timeout
Q: how to set TCP
timeout value?
 longer than RTT

but RTT varies
 too short: premature
timeout
 unnecessary
retransmissions
 too long: slow reaction
to segment loss
Q: how to estimate RTT?
 SampleRTT: measured time from
segment transmission until ACK
receipt
 ignore retransmissions
 SampleRTT will vary, want
estimated RTT “smoother”
 average several recent
measurements, not just
current SampleRTT
Transport Layer 3-60
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 3-61
Example RTT estimation:
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
350
RTT (milliseconds)
300
250
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 3-62
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 3-63
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
 3.4 Principles of
reliable data transfer
 3.5 Connection-oriented
transport: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-64
TCP:
Sender
Pseudo Codice
00 sendbase = initial_sequence number
01 nextseqnum = initial_sequence number
02
03 loop (forever) {
04
switch(event)
05
event: data received from application above
06
create TCP segment with sequence number nextseqnum
07
start timer for segment nextseqnum
08
pass segment to IP
09
nextseqnum = nextseqnum + length(data)
10
event: timer timeout for segment with sequence number y
11
retransmit segment with sequence number y
12
compute new timeout interval for segment y
13
restart timer for sequence number y
14
event: ACK received, with ACK field value of y
15
if (y > sendbase) { /* cumulative ACK of all data up to y */
16
cancel all timers for segments with sequence numbers < y
17
sendbase = y
18
}
19
else { /* a duplicate ACK for already ACKed segment */
20
increment number of duplicate ACKs received for y
21
if (number of duplicate ACKS received for y == 3) {
22
/* TCP fast retransmit */
23
resend segment with sequence number y
24
restart timer for segment y
25
}
26
} /* end of loop forever */
Transport Layer 3-65
TCP reliable data transfer
 TCP creates rdt
service on top of IP’s
unreliable service
 Pipelined segments
 Cumulative acks
 TCP uses single
retransmission timer
 Retransmissions are
triggered by:


timeout events
duplicate acks
 Initially consider
simplified TCP sender:


ignore duplicate acks
ignore flow control,
congestion control
Transport Layer 3-66
TCP sender events:
data rcvd from app:
 Create segment with
seq #
Se scade un timer, lo
 seq # is
byte-stream
rifaccio
ripartire con
valore
time-out
doppio.
number
ofdifirst
data
Se ok, risettato al valore
byte in segment
ottenuto con
estimatedRTT e devRTT
 start timer
if not
already running (think
of timer as for oldest
unacked segment)
 expiration interval:
TimeOutInterval
timeout:
 retransmit segment
that caused timeout
 restart timer
Ack rcvd:
 If acknowledges
previously unacked
segments


update what is known to
be acked
start timer if there are
outstanding segments
Transport Layer 3-67
NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
loop (forever) {
switch(event)
event: data received from application above
create TCP segment with sequence number NextSeqNum
if (timer currently not running)
start timer
pass segment to IP
NextSeqNum = NextSeqNum + length(data)
event: timer timeout
retransmit not-yet-acknowledged segment with
smallest sequence number
start timer
event: ACK received, with ACK field value of y
if (y > SendBase) {
SendBase = y
if (there are currently not-yet-acknowledged segments)
start timer
}
} /* end of loop forever */
TCP
sender
(simplified)
Comment:
• SendBase-1: last
cumulatively
ack’ed byte
Example:
• SendBase-1 = 71;
y= 73, so the rcvr
wants 73+ ;
y > SendBase, so
that new data is
acked
Transport Layer 3-68
TCP: retransmission scenarios
Host A
X
loss
Sendbase
= 100
SendBase
= 120
SendBase
= 100
time
SendBase
= 120
lost ACK scenario
Host B
Seq=92 timeout
Host B
Seq=92 timeout
timeout
Host A
time
premature timeout
Transport Layer 3-69
TCP retransmission scenarios (more)
timeout
Host A
Host B
X
loss
SendBase
= 120
time
Cumulative ACK scenario
Transport Layer 3-70
TCP ACK generation
[RFC 1122, RFC 2581]
Event at Receiver
TCP Receiver action
Arrival of in-order segment with
expected seq #. All data up to
expected seq # already ACKed
Delayed ACK. Wait up to 500ms
for next segment. If no next segment,
send ACK
Arrival of in-order segment with
expected seq #. One other
segment has ACK pending
Immediately send single cumulative
ACK, ACKing both in-order segments
Arrival of out-of-order segment
higher-than-expect seq. # .
Gap detected
Immediately send duplicate ACK,
indicating seq. # of next expected byte
Arrival of segment that
partially or completely fills gap
Immediate send ACK, provided that
segment startsat lower end of gap
Transport Layer 3-71
Fast Retransmit
 Time-out period often
relatively long:

long delay before
resending lost packet
 Detect lost segments
via duplicate ACKs.


Sender often sends
many segments back-toback
If segment is lost,
there will likely be many
duplicate ACKs.
 If sender receives 3
ACKs for the same
data, it supposes that
segment after ACKed
data was lost:

fast retransmit: resend
segment before timer
expires
Transport Layer 3-72
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
already ACKed segment
fast retransmit
Transport Layer 3-73
Commenti
 ACK comulativi
 Sender
 Sendbase = piu’ piccolo numero di sequenza dei
segmenti trasmessi ma di cui non si è ancora
ricevuto ACK
 Nextseqnum = numero di sequenza del prossimo
dato da inviare
Transport Layer 3-74
TCP vs GBN
 Sender invia i segmenti 1, 2, …, N.
Assumiamo che i segmenti arrivino
correttamente al destinatario.
 ACK(n) viene perduto (unico ACK perduto)
 GBN trasmette nuovamente i segmenti ??
 TCP trasmette nuovamente i segmenti ??
Transport Layer 3-75
TCP vs GBN
 Sender invia i segmenti 1, 2, …, N.
Assumiamo che i segmenti arrivino
correttamente al destinatario.
 ACK(n) viene perduto (unico ACK perduto)
 GBN trasmette nuovamente i segmenti n,
n+1 , …, N
 TCP trasmette nuovamente al piu’ il
segmento n (se il timeout di n scatta prima
dell’arrivo di ACK(n+1))
Transport Layer 3-76
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
 3.4 Principles of
reliable data transfer
 3.5 Connection-oriented
transport: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-77
TCP Flow Control
 receive side of TCP
connection has a
receive buffer:
flow control
sender won’t overflow
receiver’s buffer by
transmitting too much,
too fast
 speed-matching
 app process may be
service: matching the
send rate to the
receiving app’s drain
rate
slow at reading from
buffer
Transport Layer 3-78
TCP Flow control: how it works
 Rcvr advertises spare
(Suppose TCP receiver
discards out-of-order
segments)
 spare room in buffer
room by including value
of RcvWindow in
segments
 Sender limits unACKed
data to RcvWindow

guarantees receive
buffer doesn’t overflow
= RcvWindow
= RcvBuffer-[LastByteRcvd LastByteRead]
Transport Layer 3-79
Sliding Window
Sending application
Receiving application
TCP
LastByteWritten
LastByteAcked
LastByteSent
 Sending side
 LastByteAcked < =
LastByteSent
 LastByteSent < =
LastByteWritten
 buffer bytes between
LastByteAcked and
LastByteWritten
TCP
LastByteRead
NextByteExpected
LastByteRcvd
 Receiving side
 LastByteRead <
NextByteExpected
 NextByteExpected < =
LastByteRcvd +1
 buffer bytes between
NextByteRead and
LastByteRcvd
Transport Layer 3-80
TCP Flow Control: variabili di stato
 Send buffer size: MaxSendBuffer
 Receive buffer size: MaxRcvBuffer
 Receiving side
 LastByteRcvd - LastByteRead < = MaxRcvBuffer
 AdvertisedWindow = MaxRcvBuffer - (NextByteExpected NextByteRead)
 Sending side
 LastByteSent - LastByteAcked < = AdvertisedWindow
 EffectiveWindow = AdvertisedWindow - (LastByteSent LastByteAcked)
 LastByteWritten - LastByteAcked < = MaxSendBuffer
 block sender if (LastByteWritten - LastByteAcked) + y >
MaxSenderBuffer
Transport Layer 3-81
TCP Controllo del flusso: azioni
 Inviare ACK all’arrivo di segmenti
 Se ho finito di spedire e ho
AdvertisedWindow = 0?
 Problema:
il ricevente non sapra’ mai se ho di
nuovo spazio nel buffer
  il ricevente se ha AdvertisedWindow = 0
continua a spedire ack fino a che si libera il
buffer
Transport Layer 3-82
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
 3.4 Principles of
reliable data transfer
 3.5 Connection-oriented
transport: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-83
TCP Connection Management
Recall: TCP sender, receiver
establish “connection”
before exchanging data
segments
 initialize TCP variables:
 seq. #s
 buffers, flow control
info (e.g. RcvWindow)
 client: connection initiator
Socket clientSocket = new
Socket("hostname","port
number");
 server: contacted by client
Socket connectionSocket =
welcomeSocket.accept();
Three way handshake:
Step 1: client host sends TCP
SYN segment to server
 specifies initial seq #
 no data
Step 2: server host receives
SYN, replies with SYNACK
segment
server allocates buffers
 specifies server initial
seq. #
Step 3: client receives SYNACK,
replies with ACK segment,
which may contain data

Transport Layer 3-84
TCP Connection Management (cont.)
Closing a connection:
client closes socket:
clientSocket.close();
client
close
Step 1: client end system
close
FIN, replies with ACK.
Closes connection, sends
FIN.
timed wait
sends TCP FIN control
segment to server
Step 2: server receives
server
closed
Transport Layer 3-85
TCP Connection Management (cont.)
Step 3: client receives FIN,
replies with ACK.

client
server
closing
Enters “timed wait” will respond with ACK
to received FINs
closing
Step 4: server, receives
Note: with small
modification, can handle
simultaneous FINs.
timed wait
ACK. Connection closed.
closed
closed
Transport Layer 3-86
TCP Connection Management (cont)
TCP server
lifecycle
TCP client
lifecycle
Transport Layer 3-87
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
 3.4 Principles of
reliable data transfer
 3.5 Connection-oriented
transport: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-88
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 3-89
Le caratteristiche del problema


Risorse allocate per evitare la congestione
Controllo della congestione se (e quando) si manifesta
Router
1.5-Mbps T1 link
Destination
Source
2
 Implementazione
 Host (protocolli del livello di trasporto)
 Router (politiche per la gestione delle code)
 Quale modello di servizio
 best-effort (Internet)
 QoS quality of service (Futuro)
Transport Layer 3-90
Contesto


Sequenze di pacchetti che viaggiono nella rete
Router hanno poca informazione sullo stato della rete
Source
1
Router
Destination
1
Router
Source
2
Router
Destination
2
Source
3
Transport Layer 3-91
Causes/costs of congestion: scenario 1
Host A
 two senders, two
receivers
 one router,
infinite buffers
 no retransmission
Host B
lout
lin : original data
unlimited shared
output link buffers
 large delays
when congested
 maximum
achievable
throughput
Transport Layer 3-92
Troughput per la connessione
 Throughput per la connessione = numero di
byte al secondo al receiver in funzione
della velocità di spedizione
 Grandi ritardi quando la velocità dei
pacchetti in arrivo è prossima alla capacità
del router
Transport Layer 3-93
Causes/costs of congestion: scenario 2
 one router, finite buffers
 sender retransmission of lost packet
Host A
Host B
lin : original
data
l'in : original data, plus
retransmitted data
lout
finite shared output
link buffers
Transport Layer 3-94
Congestione
 La velocità del sender è uguale al carico
offerto dalla rete
 Sender deve ristramettere pacchetti per
compensare le perdite
Costo della congestione:
 Maggiore carico per la trasmissione dei pacchetti
Transport Layer 3-95
Cause della Congestione
 Quattro sender
 multihop
 Timeout + ritrasmissione
Cosa succede quando
aumenta il carico
offerto dalle rete?
Quando il carico offerto a
B è elevato
il troughput della
connessione A-C
risulta zero:
il buffer in R2 è sempre pieno
Transport Layer 3-96
Costo della Congestione
 Quando un pacchetto è perso lungo un
percorso la capacità di trasmissione dei
router lungo il percorso è sprecata!!
Transport Layer 3-97
Causes/costs of congestion: scenario 3
H
o
s
t
A
l
o
u
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 3-98
Politiche di gestione delle code
 First-In-First-Out (FIFO)
 Non abbiamo alcuna politica di gestione che dipende dalle
caratteristiche dei pacchetti
 Fair Queuing (FQ)
 Meccanismi di strutturazione del flusso dei pacchetti
 Un pacchetto non puo’ mai superare la capacità del router
 Code con priorità (WFQ)
Flow 1
Flow 2
Round-robin
service
Flow 3
Flow 4
Transport Layer 3-99
Approaches towards congestion control
Two broad approaches towards congestion control:
End-end congestion
control:
 no explicit feedback from
network
 congestion inferred from
end-system observed loss,
delay
 approach taken by TCP
Network-assisted
congestion control:
 routers provide feedback
to end systems
 single bit indicating
congestion (SNA,
DECbit, TCP/IP ECN,
ATM)
 explicit rate sender
should send at
Transport Layer 3-100
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
 3.4 Principles of
reliable data transfer
 3.5 Connection-oriented
transport: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-101
TCP: Controllo della Congestione
 Idea di base: si controlla la velocità di
trasmissione controllando il numero dei
segmenti trasmessi ma di cui non si è
ancora ricevuto ACK: W
 Maggiore è il valore di W maggiore è il
throughput della connessione.
 Quando si verifica una perdita di segmento
allora si diminuisce il valore W
Transport Layer 3-102
TCP Controllo della congestione
 Due fasi
slow start (partenza lenta)
 congestion avoidance (annullamento della
congestione)

 Valori da considerare:
 Congwin
 threshold: soglia che segnala il passaggio tra
le due fasi
Transport Layer 3-103
Controllo della congestione
 Limite superiore alle trasmissioni dei
segmenti
LastByteSent-LastByteAcked
 CongWin
 In formula
rate =
CongWin
Bytes/sec
RTT
 CongWin è il valore dinamico della funzione
che misura la congestione della rete
Transport Layer 3-104
TCP: Controllo della Congestione
 La banda di trasmissione è limitata dalla dimensione
della finestra di congestione Congwin:
Congwin
 w segmenti di dimensione MSS trasmessi in un RTT:
throughput =
w * MSS
Bytes/sec
RTT
Transport Layer 3-105
Additive Increase/Multiplicative
Decrease (AIMD)
 Modificare dinamicamente il carico offerto
 Variabile di stato (della connessione):
CongestionWindow


increase CongestionWindow when congestion goes down
decrease CongestionWindow when congestion goes up
 Informazioni di stato che cambiano in modo
dinamico
Transport Layer 3-106
AIMD
 Come si manifesta la congestione?
 Timeout
 timeout è il segnale di perdita di qualche
pacchetto.
 Perso pacchetto  decremento moltiplicativo
della finestra
 Ok  incremento additivo della finestra
Transport Layer 3-107
TCP AIMD
multiplicative decrease:
cut CongWin in half
after loss event
congestion
window
additive increase:
increase CongWin by
1 MSS every RTT in
the absence of loss
events: probing
24 Kbytes
16 Kbytes
8 Kbytes
time
Long-lived TCP connection
Transport Layer 3-108
TCP Slow Start
 When connection begins,
CongWin = 1 MSS


Example: MSS = 500
bytes & RTT = 200 msec
initial rate = 20 kbps
 When connection begins,
increase rate
exponentially fast until
first loss event
 available bandwidth may
be >> MSS/RTT

desirable to quickly ramp
up to respectable rate
Transport Layer 3-109
TCP Slowstart
Host A
initialize: Congwin = 1
for (each segment ACKed)
Congwin++
until (loss event OR
CongWin > threshold)
RTT
Slowstart algorithm
Host B
 Incremento esponenziale
(in termini del RTT) della
finestra
 Perdita di pacchetti:
timeout (Tahoe TCP), ACK
triplicati (Reno TCP)
time
Transport Layer 3-110
Un raffinamento del servizio
Idea:
 Dopo la ricezione di tre
ACK duplicati:
 CongWin viene
dimezzata
 La finestra viene fatta
crescere in modo
lineare
 Ma dopo un timeout:
 CongWin diventa 1;
 La finestra cresce
esponenzialmente fino
al raggiungimento della
soglia.
• 3 ACK dup. sono una
indicazione che la rete è
in grado di trasmettere
segmenti
• timeout dopo tre ack
duplicati è un evento
preoccupante sullo stato
della congestione della
rete
Transport Layer 3-111
TCP Congestion Avoidance
Congestion avoidance
/* slowstart is over
*/
/* Congwin > threshold */
Until (loss event) {
every w segments ACKed:
Congwin++
}
threshold = Congwin/2
Congwin = 1
1
perform slowstart
Transport Layer 3-112
Refinement (more)
14
congestion window size
(segments)
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
12 threshold
10
8
6
4
2
0
1
TCP
Tahoe
TCP
Reno
2 3
6 7
4 5
8 9 10 11 12 13 14 15
Transmission round
Series1
Series2
set to 1/2 of CongWin just
before loss event
Transport Layer 3-113
Conclusione
 CongWin ha un volore minore di Threshold, allora
in sender è nella fase di slow-start e la finestra
cresce in modo esponenziale.
 CongWin ha un volore maggiore di Threshold, il
sendere è nella fase di congestion-avoidance e la
finestra cresce in modo lineare.
 Al manifestarsi di ACK triplicato il valore di,
Threshold diviene CongWin/2 e CongWin diviene
Threshold.
 Al manifestarsi di un timeout, Threshold diviene
CongWin/2 e CongWin diviene 1 MSS.
Transport Layer 3-114
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
TCP
connection 2
bottleneck
router
capacity R
Transport Layer 3-115
Why is TCP fair?
Two competing sessions:
 Additive increase gives slope of 1, as throughout increases
 multiplicative decrease decreases throughput proportionally
R
equal bandwidth share
loss: decrease window by factor of 2
congestion avoidance: additive increase
loss: decrease window by factor of 2
congestion avoidance: additive increase
Connection 1 throughput R
Transport Layer 3-116
Fairness (more)
Fairness and UDP
 Multimedia apps often
do not use TCP

do not want rate
throttled by congestion
control
 Instead use UDP:
 pump audio/video at
constant rate, tolerate
packet loss
 Research area: TCP
friendly
Fairness and parallel TCP
connections
 nothing prevents app from
opening parallel cnctions
between 2 hosts.
 Web browsers do this
 Example: link of rate R
supporting 9 cnctions;


new app asks for 1 TCP, gets
rate R/10
new app asks for 11 TCPs,
gets R/2 !
Transport Layer 3-117
Delay modeling
Q: How long does it take to
receive an object from a
Web server after sending
a request?
Ignoring congestion, delay is
influenced by:
 TCP connection establishment
 data transmission delay
 slow start
Notation, assumptions:
 Assume one link between
client and server of rate R
 S: MSS (bits)
 O: object size (bits)
 no retransmissions (no loss,
no corruption)
Window size:
 First assume: fixed
congestion window, W
segments
 Then dynamic window,
modeling slow start
Transport Layer 3-118
Fixed congestion window (1)
First case:
WS/R > RTT + S/R: ACK for
first segment in window
returns before window’s
worth of data sent
delay = 2RTT + O/R
Transport Layer 3-119
Fixed congestion window (2)
Second case:
 WS/R < RTT + S/R: wait
for ACK after sending
window’s worth of data
sent
delay = 2RTT + O/R
+ (K-1)[S/R + RTT - WS/R]
Transport Layer 3-120
TCP Delay Modeling: Slow Start (1)
Now suppose window grows according to slow start
Will show that the delay for one object is:
Latency  2 RTT 
O
S
S

 P  RTT    ( 2 P  1)
R
R
R

where P is the number of times TCP idles at server:
P  min {Q, K  1}
- where Q is the number of times the server idles
if the object were of infinite size.
- and K is the number of windows that cover the object.
Transport Layer 3-121
TCP Delay Modeling: Slow Start (2)
Delay components:
• 2 RTT for connection
estab and request
• O/R to transmit
object
• time server idles due
to slow start
initiate TCP
connection
request
object
first window
= S/R
RTT
Server idles:
P = min{K-1,Q} times
Example:
• O/S = 15 segments
• K = 4 windows
•Q=2
• P = min{K-1,Q} = 2
Server idles P=2 times
second window
= 2S/R
third window
= 4S/R
fourth window
= 8S/R
complete
transmission
object
delivered
time at
client
time at
server
Transport Layer 3-122
TCP Delay Modeling (3)
S
 RTT  time from when server starts to send segment
R
until server receives acknowledg ement
initiate TCP
connection
2k 1
S
 time to transmit the kth window
R

request
object
S
k 1 S 

RTT

2
 idle time after the kth window
 R
R 
first window
= S/R
RTT
second window
= 2S/R
third window
= 4S/R
P
O
delay   2 RTT   idleTime p
R
p 1
P
O
S
S
  2 RTT   [  RTT  2 k 1 ]
R
R
k 1 R
O
S
S
  2 RTT  P[ RTT  ]  (2 P  1)
R
R
R
fourth window
= 8S/R
complete
transmission
object
delivered
time at
client
time at
server
Transport Layer 3-123
TCP Delay Modeling (4)
Recall K = number of windows that cover object
How do we calculate K ?
K  min {k : 2 0 S  21 S    2 k 1 S  O}
 min {k : 2 0  21    2 k 1  O / S }
O
 min {k : 2  1  }
S
O
 min {k : k  log 2 (  1)}
S
O


 log 2 (  1)
S


k
Calculation of Q, number of idles for infinite-size object,
is similar (see HW).
Transport Layer 3-124
HTTP Modeling
 Assume Web page consists of:
1 base HTML page (of size O bits)
 M images (each of size O bits)
 Non-persistent HTTP:
 M+1 TCP connections in series
 Response time = (M+1)O/R + (M+1)2RTT + sum of idle times
 Persistent HTTP:
 2 RTT to request and receive base HTML file
 1 RTT to request and receive M images
 Response time = (M+1)O/R + 3RTT + sum of idle times
 Non-persistent HTTP with X parallel connections
 Suppose M/X integer.
 1 TCP connection for base file
 M/X sets of parallel connections for images.
 Response time = (M+1)O/R + (M/X + 1)2RTT + sum of idle times

Transport Layer 3-125
HTTP Response time (in seconds)
RTT = 100 msec, O = 5 Kbytes, M=10 and X=5
20
18
16
14
12
10
8
6
4
2
0
non-persistent
persistent
parallel nonpersistent
28
100
1
10
Kbps Kbps Mbps Mbps
For low bandwidth, connection & response time dominated by
transmission time.
Persistent connections only give minor improvement over parallel
connections.
Transport Layer 3-126
HTTP Response time (in seconds)
RTT =1 sec, O = 5 Kbytes, M=10 and X=5
70
60
50
non-persistent
40
persistent
30
20
parallel nonpersistent
10
0
28
100
1
10
Kbps Kbps Mbps Mbps
For larger RTT, response time dominated by TCP establishment
& slow start delays. Persistent connections now give important
improvement: particularly in high delaybandwidth networks.
Transport Layer 3-127
Chapter 3: Summary
 principles behind transport
layer services:
 multiplexing,
demultiplexing
 reliable data transfer
 flow control
 congestion control
 instantiation and
implementation in the
Internet
 UDP
 TCP
Next:
 leaving the network
“edge” (application,
transport layers)
 into the network
“core”
Transport Layer 3-128