Transport Layer 3-1
Chapter 3
Transport Layer
Computer Networking,
A Top Down Approach
Featuring the Internet,
2nd edition,
Jim Kurose,Keith Ross
Addison-Wesley,July
2002,
The PowerPoint Slides are based on the
material provided by
J.F Kurose and K.W,Ross.
Transport Layer 3-2
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 3-3
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 3-4
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
application
transport
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physicalnetwork
data link
physical
Transport Layer 3-5
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 3-6
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
application
transport
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physicalnetwork
data link
physical
Transport Layer 3-7
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 3-8
Multiplexing/demultiplexing
application
transport
network
link
physical
P1 application
transport
network
link
physical
application
transport
network
link
physical
P2P3 P4P1
host 1 host 2 host 3
= process= socket
delivering received segments
to correct socket
Demultiplexing at rcv host:
gathering data from multiple
sockets,enveloping data with
header (later used for
demultiplexing)
Multiplexing at send host:
Transport Layer 3-9
How demultiplexing works
? host receives IP datagrams
? each datagram has source
IP address,destination IP
address
? each datagram carries 1
transport-layer segment
? each segment has source,
destination port number
(recall,well-known port
numbers for specific
applications)
? host uses IP addresses & port
numbers to direct segment to
appropriate socket
source port # dest port #
32 bits
application
data
(message)
other header fields
TCP/UDP segment format
Transport Layer 3-10
Connectionless demultiplexing
? Create sockets with port
numbers:
DatagramSocket mySocket1 = new
DatagramSocket(99111);
DatagramSocket mySocket2 = new
DatagramSocket(99222);
? UDP socket identified by
two-tuple:
(dest IP address,dest port number)
? When host receives UDP
segment:
? checks destination port
number in segment
? directs UDP segment to
socket with that port
number
? IP datagrams with
different source IP
addresses and/or source
port numbers directed
to same socket
Transport Layer 3-11
Connectionless demux (cont)
DatagramSocket serverSocket = new DatagramSocket(6428);
Client
IP:B
P3
client
IP,A
P1P1P3
server
IP,C
SP,6428
DP,9157
SP,9157
DP,6428
SP,6428
DP,5775
SP,5775
DP,6428
SP provides,return address”
Transport Layer 3-12
Connection-oriented demux
? TCP socket identified
by 4-tuple,
? source IP address
? source port number
? dest IP address
? dest port number
? recv host uses all four
values to direct
segment to appropriate
socket
? Server host may support
many simultaneous TCP
sockets:
? each socket identified by
its own 4-tuple
? Web servers have
different sockets for
each connecting client
? non-persistent HTTP will
have different socket for
each request
Transport Layer 3-13
Connection-oriented demux
(cont)
Client
IP:B
P3
client
IP,A
P1P1P3
server
IP,C
SP,80
DP,9157
SP,9157
DP,80
SP,80
DP,5775
SP,5775
DP,80
P4
Transport Layer 3-14
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 3-15
UDP,User Datagram Protocol [RFC 768]
?,no frills,”,bare bones”
Internet transport
protocol
?,best effort” service,UDP
segments may be:
? lost
? delivered out of order
to app
? connectionless:
? no handshaking between
UDP sender,receiver
? each UDP segment
handled independently
of others
Why is there a UDP?
? no connection
establishment (which can
add delay)
? simple,no connection state
at sender,receiver
? small segment header
? no congestion control,UDP
can blast away as fast as
desired
Transport Layer 3-16
UDP,more
? often used for streaming
multimedia apps
? loss tolerant
? rate sensitive
? other UDP uses
? DNS
? SNMP
? reliable transfer over UDP,
add reliability at
application layer
? application-specific
error recovery!
source port # dest port #
32 bits
Application
data
(message)
UDP segment format
length checksum
Length,in
bytes of UDP
segment,
including
header
Transport Layer 3-17
UDP checksum
Sender:
? treat segment contents
as sequence of 16-bit
integers
? checksum,addition (1?s
complement sum) of
segment contents
? sender puts checksum
value into UDP checksum
field
Receiver:
? compute checksum of
received segment
? check if computed checksum
equals checksum field value:
? NO - error detected
? YES - no error detected,
But maybe errors
nonetheless? More later
….
Goal,detect,errors” (e.g.,flipped bits) in transmitted
segment
Transport Layer 3-18
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 3-19
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 3-20
Reliable data transfer,getting started
send
side
receive
side
rdt_send(),called from above,
(e.g.,by app.),Passed data to
deliver to receiver upper layer
udt_send(),called by rdt,
to transfer packet over
unreliable channel to receiver
rdt_rcv(),called when packet
arrives on rcv-side of channel
deliver_data(),called by
rdt to deliver data to upper
Transport Layer 3-21
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
state
1 state2
event causing state transition
actions taken on state transition
state,when in this
“state” next state
uniquely determined
by next event
event
actions
Transport Layer 3-22
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
call from
above packet = make_pkt(data)udt_send(packet)
rdt_send(data)
extract (packet,data)
deliver_data(data)
Wait for
call from
below
rdt_rcv(packet)
sender receiver
Transport Layer 3-23
Rdt2.0,channel with bit errors
? underlying channel may flip bits in packet
? recall,UDP 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
? human scenarios using ACKs,NAKs?
? new mechanisms in rdt2.0 (beyond rdt1.0):
? error detection
? receiver feedback,control msgs (ACK,NAK) rcvr->sender
Transport Layer 3-24
rdt2.0,FSM specification
Wait for
call from
above
snkpkt = make_pkt(data,checksum)
udt_send(sndpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
Wait for
ACK or
NAK
Wait for
call from
belowsender
receiver
rdt_send(data)
L
Transport Layer 3-25
rdt2.0,operation with no errors
Wait for
call from
above
snkpkt = make_pkt(data,checksum)
udt_send(sndpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
Wait for
ACK or
NAK
Wait for
call from
below
rdt_send(data)
L
Transport Layer 3-26
rdt2.0,error scenario
Wait for
call from
above
snkpkt = make_pkt(data,checksum)
udt_send(sndpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
Wait for
ACK or
NAK
Wait for
call from
below
rdt_send(data)
L
Transport Layer 3-27
rdt2.0 has a fatal flaw!
What happens if
ACK/NAK corrupted?
? sender doesn?t know what
happened at receiver!
? can?t just retransmit,
possible duplicate
What to do?
? sender ACKs/NAKs
receiver?s ACK/NAK? What
if sender ACK/NAK lost?
? retransmit,but this might
cause retransmission of
correctly received pkt!
Handling duplicates,
? sender adds sequence
number to each pkt
? sender retransmits current
pkt if ACK/NAK garbled
? receiver discards (doesn?t
deliver up) duplicate pkt
Sender sends one packet,
then waits for receiver
response
stop and wait
Transport Layer 3-28
rdt2.1,sender,handles garbled ACK/NAKs
Wait for
call 0 from
above
sndpkt = make_pkt(0,data,checksum)
udt_send(sndpkt)
rdt_send(data)
Wait for
ACK or
NAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isNAK(rcvpkt) )
sndpkt = make_pkt(1,data,checksum)
udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isNAK(rcvpkt) )
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt)
Wait for
call 1 from
above
Wait for
ACK or
NAK 1
LL
Transport Layer 3-29
rdt2.1,receiver,handles garbled ACK/NAKs
Wait for
0 from
below
sndpkt = make_pkt(NAK,chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) &&
has_seq0(rcvpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK,chksum)
udt_send(sndpkt)
Wait for
1 from
below
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)
sndpkt = make_pkt(ACK,chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) &&
has_seq1(rcvpkt)
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(ACK,chksum)
udt_send(sndpkt)
sndpkt = make_pkt(NAK,chksum)
udt_send(sndpkt)
Transport Layer 3-30
rdt2.1,discussion
Sender:
? seq # added to pkt
? two seq,#?s (0,1) will
suffice,Why?
? must check if received
ACK/NAK corrupted
? twice as many states
? state must,remember”
whether,current” pkt
has 0 or 1 seq,#
Receiver:
? must check if received
packet is duplicate
? state indicates whether
0 or 1 is expected pkt
seq #
? note,receiver can not
know if its last
ACK/NAK received OK
at sender
Transport Layer 3-31
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 3-32
rdt2.2,sender,receiver fragments
Wait for
call 0 from
above
sndpkt = make_pkt(0,data,checksum)
udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
Wait for
ACK
0
sender FSM
fragment
Wait for
0 from
below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK1,chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSM
fragment
L
Transport Layer 3-33
rdt3.0,channels with errors and loss
New assumption:
underlying channel can
also lose packets (data
or ACKs)
? checksum,seq,#,ACKs,
retransmissions will be
of help,but not enough
Q,how to deal with loss?
? sender waits until
certain data or ACK
lost,then retransmits
? yuck,drawbacks?
Approach,sender waits
“reasonable” amount of
time for ACK
? retransmits if no ACK
received in this time
? if pkt (or ACK) just delayed
(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 3-34
rdt3.0 sender
sndpkt = make_pkt(0,data,checksum)
udt_send(sndpkt)
start_timer
rdt_send(data)
Wait
for
ACK0
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
Wait for
call 1 from
above
sndpkt = make_pkt(1,data,checksum)
udt_send(sndpkt)
start_timer
rdt_send(data)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,0) )
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,1)
stop_timer
stop_timer
udt_send(sndpkt)
start_timer
timeout
udt_send(sndpkt)
start_timer
timeout
rdt_rcv(rcvpkt)
Wait for
call 0from
above
Wait
for
ACK1
L
rdt_rcv(rcvpkt)
L
L
L
Transport Layer 3-35
rdt3.0 in action
Transport Layer 3-36
rdt3.0 in action
Transport Layer 3-37
Performance of rdt3.0
? rdt3.0 works,but performance stinks
? example,1 Gbps link,15 ms e-e prop,delay,1KB packet:
Ttransmit = 8kb/pkt
10**9 b/sec = 8 microsec
? U sender,utilization – fraction of time sender busy sending
? 1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link
? network protocol limits use of physical resources!
U s e n d e r =, 008
3 0, 0 0 8
= 0, 0 0 0 2 7
m i c r o s e c
o n d s
L / R
R T T + L / R
=
L (packet length in bits)
R (transmission rate,bps) =
Transport Layer 3-38
rdt3.0,stop-and-wait operation
first packet bit transmitted,t = 0
sender receiver
RTT
last packet bit transmitted,t = L / R
first packet bit arrives
last packet bit arrives,send ACK
ACK arrives,send next
packet,t = RTT + L / R
U s e n d e r =, 008
3 0, 0 0 8
= 0, 0 0 0 2 7
m i c r o s e c
o n d s
L / R
R T T + L / R
=
Transport Layer 3-39
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 3-40
Pipelining,increased utilization
first packet bit transmitted,t = 0
sender receiver
RTT
last bit transmitted,t = L / R
first packet bit arrives
last packet bit arrives,send ACK
ACK arrives,send next
packet,t = RTT + L / R
last bit of 2nd packet arrives,send ACK
last bit of 3rd packet arrives,send ACK
U s e n d e r =, 0 24
3 0, 0 0 8
= 0, 0 0 0 8
m i c r o s e c o n
ds
3 * L / R
R T T + L / R
=
Increase utilization
by a factor of 3!
Transport Layer 3-41
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 deceive duplicate ACKs (see receiver)
? timer for each in-flight pkt
? timeout(n),retransmit pkt n and all higher seq # pkts in window
Transport Layer 3-42
GBN,sender extended FSM
Wait start_timerudt_send(sndpkt[base])
udt_send(sndpkt[base+1])
…
udt_send(sndpkt[nextseqnum-1])
timeout
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 = getacknum(rcvpkt)+1
If (base == nextseqnum)
stop_timer
else
start_timer
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
base=1
nextseqnum=1
rdt_rcv(rcvpkt)
&& corrupt(rcvpkt)
L
Transport Layer 3-43
GBN,receiver extended FSM
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 #
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)
&& notcurrupt(rcvpkt)
&& hasseqnum(rcvpkt,expectedseqnum)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(expectedseqnum,ACK,chksum)
udt_send(sndpkt)
expectedseqnum++
expectedseqnum=1
sndpkt =
make_pkt(expectedseqnum,ACK,chksum)
L
Transport Layer 3-44
GBN in
action
Transport Layer 3-45
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 3-46
Selective repeat,sender,receiver windows
Transport Layer 3-47
Selective repeat
data from above,
? if next available seq # in
window,send pkt
timeout(n):
? resend pkt n,restart timer
ACK(n) in [sendbase,sendbase+N]:
? mark pkt n as received
? if n smallest unACKed pkt,
advance window base to
next unACKed seq #
sender
pkt n in [rcvbase,rcvbase+N-1]
? send ACK(n)
? out-of-order,buffer
? in-order,deliver (also
deliver buffered,in-order
pkts),advance window to
next not-yet-received pkt
pkt n in [rcvbase-N,rcvbase-1]
? ACK(n)
otherwise:
? ignore
receiver
Transport Layer 3-48
Selective repeat in action
Transport Layer 3-49
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 3-50
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 3-51
TCP,Overview 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
overwhelm receiver
? point-to-point:
? one sender,one receiver
? reliable,in-order byte
steam:
? no,message boundaries”
? pipelined:
? TCP congestion and flow
control set window size
? send & receive buffers
s oc k et
door
T CP
s en d b uf f er
T CP
r ec ei v e b uf f er
s oc k et
door
s e g m e n t
ap pl i c ati on
w r i tes da ta
ap pl i c ati on
r ea ds da ta
Transport Layer 3-52
TCP segment structure
source port # dest port #
32 bits
application
data
(variable length)
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen notused
Options (variable length)
URG,urgent data
(generally not used)
ACK,ACK #
valid
PSH,push data now
(generally not used)
RST,SYN,FIN:
connection estab
(setup,teardown
commands)
# bytes
rcvr willing
to accept
counting
by bytes
of data
(not segments!)
Internet
checksum
(as in UDP)
Transport Layer 3-53
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 Host B
User
types
?C?
host ACKs
receipt
of echoed
?C?
host ACKs
receipt of
?C?,echoes
back ?C?
time
simple telnet scenario
Transport Layer 3-54
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 3-55
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 3-56
Example RTT estimation:
R TT,gai a, c s, um a s s, e du t o f a nt a s i a, e ure c om, f r
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
t i m e ( s e c onn ds )
R
TT
(
m
i
l
l
i
s
e
c
ond
s
)
Sa m p l e R T T Es t i m a t e d R T T
Transport Layer 3-57
TCP Round Trip Time and Timeout
Setting the timeout
? EstimtedRTTplus,safety margin”
? large variation in EstimatedRTT -> larger safety margin
? first estimate of how much SampleRTT deviates from
EstimatedRTT,
TimeoutInterval = EstimatedRTT + 4*DevRTT
DevRTT = (1-?)*DevRTT +
?*|SampleRTT-EstimatedRTT|
(typically,? = 0.25)
Then set timeout interval:
Transport Layer 3-58
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 3-59
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 3-60
TCP sender events:
data rcvd from app:
? Create segment with
seq #
? seq # is byte-stream
number of first data
byte in segment
? 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 3-61
TCP
sender
(simplified)
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 */
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 3-62
TCP,retransmission scenarios
Host A
time
premature timeout
Host B
Se
q=
92
tim
eou
t
Host A
losst
im
eou
t
lost ACK scenario
Host B
X
time
Se
q=
92
tim
eou
t
SendBase
= 100
SendBase
= 120
SendBase
= 120
Sendbase
= 100
Transport Layer 3-63
TCP retransmission scenarios (more)
Host A
losst
im
eou
t
Cumulative ACK scenario
Host B
X
time
SendBase
= 120
Transport Layer 3-64
TCP ACK generation [RFC 1122,RFC 2581]
Event at Receiver
Arrival of in-order segment with
expected seq #,All data up to
expected seq # already ACKed
Arrival of in-order segment with
expected seq #,One other
segment has ACK pending
Arrival of out-of-order segment
higher-than-expect seq,#,
Gap detected
Arrival of segment that
partially or completely fills gap
TCP Receiver action
Delayed ACK,Wait up to 500ms
for next segment,If no next segment,
send ACK
Immediately send single cumulative
ACK,ACKing both in-order segments
Immediately send duplicate ACK,
indicating seq,# of next expected byte
Immediate send ACK,provided that
segment startsat lower end of gap
Transport Layer 3-65
Fast Retransmit
? Time-out period often
relatively long:
? long delay before
resending lost packet
? Detect lost segments
via duplicate ACKs.
? Sender often sends
many segments back-to-
back
? If segment is lost,
there will likely be many
duplicate ACKs.
? If sender receives 3
ACKs for the same
data,it supposes that
segment after ACKed
data was lost:
? fast retransmit,resend
segment before timer
expires
Transport Layer 3-66
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
}
Fast retransmit algorithm:
a duplicate ACK for
already ACKed segment
fast retransmit
Transport Layer 3-67
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 3-68
TCP Flow Control
? receive side of TCP
connection has a
receive buffer:
? speed-matching
service,matching the
send rate to the
receiving app?s drain
rate? app process may be
slow at reading from
buffer
sender won?t overflow
receiver?s buffer by
transmitting too much,
too fast
flow control
Transport Layer 3-69
TCP Flow control,how it works
(Suppose TCP receiver
discards out-of-order
segments)
? spare room in buffer
= RcvWindow
= RcvBuffer-[LastByteRcvd -
LastByteRead]
? Rcvr advertises spare
room by including value
of RcvWindow in
segments
? Sender limits unACKed
data to RcvWindow
? guarantees receive
buffer doesn?t overflow
Transport Layer 3-70
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 3-71
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 3-72
TCP Connection Management (cont.)
Closing a connection:
client closes socket:
clientSocket.close();
Step 1,client end system
sends TCP FIN control
segment to server
Step 2,server receives
FIN,replies with ACK,
Closes connection,sends
FIN,
client server
close
close
closed
tim
ed
wa
it
Transport Layer 3-73
TCP Connection Management (cont.)
Step 3,client receives FIN,
replies with ACK,
? Enters,timed wait” -
will respond with ACK
to received FINs
Step 4,server,receives
ACK,Connection closed,
Note,with small
modification,can handle
simultaneous FINs.
client server
closing
closing
closed
tim
ed
w
ait
closed
Transport Layer 3-74
TCP Connection Management (cont)
TCP client
lifecycle
TCP server
lifecycle
Transport Layer 3-75
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 3-76
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 3-77
Causes/costs of congestion,scenario 1
? two senders,two
receivers
? one router,
infinite buffers
? no retransmission
? large delays
when congested
? maximum
achievable
throughput
unlimited shared
output link buffers
Host A l
in, original data
Host B
lout
Transport Layer 3-78
Causes/costs of congestion,scenario 2
? one router,finite buffers
? sender retransmission of lost packet
finite shared output
link buffers
Host A l
in, original
data
Host B
lout
l'in, original data,plus
retransmitted data
Transport Layer 3-79
Causes/costs of congestion,scenario 2
? always,(goodput)
?,perfect” retransmission only when loss:
? retransmission of delayed (not lost) packet makes larger
(than perfect case) for same
lin lout=
lin lout>
lin
lout
“costs” of congestion:
? more work (retrans) for given,goodput”
? unneeded retransmissions,link carries multiple copies of pkt
Transport Layer 3-80
Causes/costs of congestion,scenario 3
? four senders
? multihop paths
? timeout/retransmit
linQ,what happens as
and increase?lin
finite shared output
link buffers
Host A l
in, original data
Host B
lout
l'in, original data,plus
retransmitted data
Transport Layer 3-81
Causes/costs of congestion,scenario 3
Another,cost” of congestion:
? when packet dropped,any,upstream transmission
capacity used for that packet was wasted!
H
o
s
t
A
H
o
s
t
B
l
o
u
t
Transport Layer 3-82
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
Two broad approaches towards congestion control:
Transport Layer 3-83
Case study,ATM ABR congestion control
ABR,available bit rate:
?,elastic service”
? if sender?s path
“underloaded”,
? sender should use
available bandwidth
? if sender?s path
congested,
? sender throttled to
minimum guaranteed
rate
RM (resource management)
cells:
? sent by sender,interspersed
with data cells
? bits in RM cell set by switches
(“network-assisted”)
? NI bit,no increase in rate
(mild congestion)
? CI bit,congestion
indication
? RM cells returned to sender by
receiver,with bits intact
Transport Layer 3-84
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 minimum 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 3-85
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 3-86
TCP Congestion Control
? end-end control (no network
assistance)
? sender limits transmission:
LastByteSent-LastByteAcked
? CongWin
? Roughly,
? CongWin is dynamic,function
of perceived network
congestion
How does sender
perceive congestion?
? loss event = timeout or
3 duplicate acks
? TCP sender reduces
rate (CongWin) after
loss event
three mechanisms:
? AIMD
? slow start
? conservative after
timeout events
rate = CongWinRTT Bytes/sec
Transport Layer 3-87
TCP AIMD
8 K b yte s
1 6 K b yte s
2 4 K b yte s
ti m e
co n g e sti o n
w i n d o w
multiplicative decrease:
cut CongWin in half
after loss event
additive increase:
increase CongWin by
1 MSS every RTT in
the absence of loss
events,probing
Long-lived TCP connection
Transport Layer 3-88
TCP Slow Start
? When connection begins,
CongWin = 1 MSS
? Example,MSS = 500
bytes & RTT = 200 msec
? initial rate = 20 kbps
? available bandwidth may
be >> MSS/RTT
? desirable to quickly ramp
up to respectable rate
? When connection begins,
increase rate
exponentially fast until
first loss event
Transport Layer 3-89
TCP Slow Start (more)
? When connection
begins,increase rate
exponentially until
first loss event:
? double CongWin every
RTT
? done by incrementing
CongWin for every ACK
received
? Summary,initial rate
is slow but ramps up
exponentially fast
Host A
RTT
Host B
time
Transport Layer 3-90
Refinement
? After 3 dup ACKs:
?CongWin is cut in half
? window then grows
linearly
? But after timeout event:
?CongWin instead set to
1 MSS;
? window then grows
exponentially
? to a threshold,then
grows linearly
? 3 dup ACKs indicates
network capable of
delivering some segments
? timeout before 3 dup
ACKs is,more alarming”
Philosophy:
Transport Layer 3-91
Refinement (more)
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
0
2
4
6
8
10
12
14
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Tr a ns m i s s i on r ou nd
c
onge
s
t
i
on
w
i
nd
ow
s
i
z
e
(
s
e
g
m
e
nt
s
)
S e r ie s 1 S e r ie s 2
t h r e s h o ld
T C P
T a h o e
T C P
R e n o
Transport Layer 3-92
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 3-93
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
router
capacity R
TCP
connection 2
TCP Fairness
Transport Layer 3-94
Why is TCP fair?
Two competing sessions:
? Additive increase gives slope of 1,as throughout increases
? multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughput
congestion avoidance,additive increase
loss,decrease window by factor of 2
congestion avoidance,additive increase
loss,decrease window by factor of 2
Transport Layer 3-95
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 3-96
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 3-97
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 3-98
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 3-99
TCP Delay Modeling,Slow Start (1)
Now suppose window grows according to slow start
Will show that the delay for one object is:
R
S
R
SR T TP
R
OR T TL a te n c y P )12(2 ???
?
??
?
? ????
where P is the number of times TCP idles at server:
}1,{m i n ?? KQP
- 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 3-100
TCP Delay Modeling,Slow Start (2)
RTT
init iat e T C P
c onnect ion
request
objec t
f irs t w indow
= S/ R
s ec ond w indow
= 2S/ R
t hird w indow
= 4S/ R
f ourt h w indow
= 8S/ R
c om plet e
t rans m is s ion
objec t
deliv ered
t im e at
c lient
t im e at
s erv er
Example:
? O/S = 15 segments
? K = 4 windows
? Q = 2
? P = min{K-1,Q} = 2
Server idles P=2 times
Delay components:
? 2 RTT for connection
estab and request
? O/R to transmit
object
? time server idles due
to slow start
Server idles,
P = min{K-1,Q} times
Transport Layer 3-101
TCP Delay Modeling (3)
R
S
R
S
R T TPR T T
R
O
R
S
R T T
R
S
R T T
R
O
i d l e T i m eR T T
R
O
P
k
P
k
P
p
p
)12(][2
]2[2
2d e l a y
1
1
1
??????
?????
???
?
?
?
?
?
t h w i n d o w a f t e r t h e t i m ei d l e 2 1 kRSR T TRS k ??????? ?? ??
e m e n ta c k n o w l e d g r e c e i v e ss e r v e r u n t i l
s e g m e n t s e n d t os t a r t ss e r v e r w h e nf r o m t i m e?? R T TRS
w in d o w k th th e tr a n s m it toti m e2 1 ?? RSk
RTT
init iat e T C P
c onnect ion
request
objec t
f irs t w indow
= S/ R
s ec ond w indow
= 2S/ R
t hird w indow
= 4S/ R
f ourt h w indow
= 8S/ R
c om plet e
t rans m is s ion
objec t
deliv ered
t im e at
c lient
t im e at
s erv er
Transport Layer 3-102
TCP Delay Modeling (4)
?
?
?
?
?
?
??
???
???
?????
?????
?
?
)1(lo g
)}1(lo g:{m i n
}12:{m i n
}/222:{m i n
}222:{m i n
2
2
110
110
S
O
S
O
kk
S
O
k
SOk
OSSSkK
k
k
k
?
?
Calculation of Q,number of idles for infinite-size object,
is similar (see HW).
Recall K = number of windows that cover object
How do we calculate K?
Transport Layer 3-103
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 3-104
0
2
4
6
8
10
12
14
16
18
20
28
K b p s
100
K b p s
1
M b p s
10
M b p s
n on -p e r s i s te n t
p e r s i s te n t
p a r a l l e l n on -
p e r s i s te n t
HTTP Response time (in seconds)
RTT = 100 msec,O = 5 Kbytes,M=10 and X=5
For low bandwidth,connection & response time dominated by
transmission time.
Persistent connections only give minor improvement over parallel
connections.
Transport Layer 3-105
0
10
20
30
40
50
60
70
28
K b p s
100
K b p s
1
M b p s
10
M b p s
n on -p e r s i s te n t
p e r s i s te n t
p a r a l l e l n on -
p e r s i s te n t
HTTP Response time (in seconds)
RTT =1 sec,O = 5 Kbytes,M=10 and X=5
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 3-106
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”
Chapter 3
Transport Layer
Computer Networking,
A Top Down Approach
Featuring the Internet,
2nd edition,
Jim Kurose,Keith Ross
Addison-Wesley,July
2002,
The PowerPoint Slides are based on the
material provided by
J.F Kurose and K.W,Ross.
Transport Layer 3-2
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 3-3
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 3-4
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
application
transport
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physicalnetwork
data link
physical
Transport Layer 3-5
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 3-6
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
application
transport
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physicalnetwork
data link
physical
Transport Layer 3-7
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 3-8
Multiplexing/demultiplexing
application
transport
network
link
physical
P1 application
transport
network
link
physical
application
transport
network
link
physical
P2P3 P4P1
host 1 host 2 host 3
= process= socket
delivering received segments
to correct socket
Demultiplexing at rcv host:
gathering data from multiple
sockets,enveloping data with
header (later used for
demultiplexing)
Multiplexing at send host:
Transport Layer 3-9
How demultiplexing works
? host receives IP datagrams
? each datagram has source
IP address,destination IP
address
? each datagram carries 1
transport-layer segment
? each segment has source,
destination port number
(recall,well-known port
numbers for specific
applications)
? host uses IP addresses & port
numbers to direct segment to
appropriate socket
source port # dest port #
32 bits
application
data
(message)
other header fields
TCP/UDP segment format
Transport Layer 3-10
Connectionless demultiplexing
? Create sockets with port
numbers:
DatagramSocket mySocket1 = new
DatagramSocket(99111);
DatagramSocket mySocket2 = new
DatagramSocket(99222);
? UDP socket identified by
two-tuple:
(dest IP address,dest port number)
? When host receives UDP
segment:
? checks destination port
number in segment
? directs UDP segment to
socket with that port
number
? IP datagrams with
different source IP
addresses and/or source
port numbers directed
to same socket
Transport Layer 3-11
Connectionless demux (cont)
DatagramSocket serverSocket = new DatagramSocket(6428);
Client
IP:B
P3
client
IP,A
P1P1P3
server
IP,C
SP,6428
DP,9157
SP,9157
DP,6428
SP,6428
DP,5775
SP,5775
DP,6428
SP provides,return address”
Transport Layer 3-12
Connection-oriented demux
? TCP socket identified
by 4-tuple,
? source IP address
? source port number
? dest IP address
? dest port number
? recv host uses all four
values to direct
segment to appropriate
socket
? Server host may support
many simultaneous TCP
sockets:
? each socket identified by
its own 4-tuple
? Web servers have
different sockets for
each connecting client
? non-persistent HTTP will
have different socket for
each request
Transport Layer 3-13
Connection-oriented demux
(cont)
Client
IP:B
P3
client
IP,A
P1P1P3
server
IP,C
SP,80
DP,9157
SP,9157
DP,80
SP,80
DP,5775
SP,5775
DP,80
P4
Transport Layer 3-14
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 3-15
UDP,User Datagram Protocol [RFC 768]
?,no frills,”,bare bones”
Internet transport
protocol
?,best effort” service,UDP
segments may be:
? lost
? delivered out of order
to app
? connectionless:
? no handshaking between
UDP sender,receiver
? each UDP segment
handled independently
of others
Why is there a UDP?
? no connection
establishment (which can
add delay)
? simple,no connection state
at sender,receiver
? small segment header
? no congestion control,UDP
can blast away as fast as
desired
Transport Layer 3-16
UDP,more
? often used for streaming
multimedia apps
? loss tolerant
? rate sensitive
? other UDP uses
? DNS
? SNMP
? reliable transfer over UDP,
add reliability at
application layer
? application-specific
error recovery!
source port # dest port #
32 bits
Application
data
(message)
UDP segment format
length checksum
Length,in
bytes of UDP
segment,
including
header
Transport Layer 3-17
UDP checksum
Sender:
? treat segment contents
as sequence of 16-bit
integers
? checksum,addition (1?s
complement sum) of
segment contents
? sender puts checksum
value into UDP checksum
field
Receiver:
? compute checksum of
received segment
? check if computed checksum
equals checksum field value:
? NO - error detected
? YES - no error detected,
But maybe errors
nonetheless? More later
….
Goal,detect,errors” (e.g.,flipped bits) in transmitted
segment
Transport Layer 3-18
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 3-19
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 3-20
Reliable data transfer,getting started
send
side
receive
side
rdt_send(),called from above,
(e.g.,by app.),Passed data to
deliver to receiver upper layer
udt_send(),called by rdt,
to transfer packet over
unreliable channel to receiver
rdt_rcv(),called when packet
arrives on rcv-side of channel
deliver_data(),called by
rdt to deliver data to upper
Transport Layer 3-21
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
state
1 state2
event causing state transition
actions taken on state transition
state,when in this
“state” next state
uniquely determined
by next event
event
actions
Transport Layer 3-22
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
call from
above packet = make_pkt(data)udt_send(packet)
rdt_send(data)
extract (packet,data)
deliver_data(data)
Wait for
call from
below
rdt_rcv(packet)
sender receiver
Transport Layer 3-23
Rdt2.0,channel with bit errors
? underlying channel may flip bits in packet
? recall,UDP 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
? human scenarios using ACKs,NAKs?
? new mechanisms in rdt2.0 (beyond rdt1.0):
? error detection
? receiver feedback,control msgs (ACK,NAK) rcvr->sender
Transport Layer 3-24
rdt2.0,FSM specification
Wait for
call from
above
snkpkt = make_pkt(data,checksum)
udt_send(sndpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
Wait for
ACK or
NAK
Wait for
call from
belowsender
receiver
rdt_send(data)
L
Transport Layer 3-25
rdt2.0,operation with no errors
Wait for
call from
above
snkpkt = make_pkt(data,checksum)
udt_send(sndpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
Wait for
ACK or
NAK
Wait for
call from
below
rdt_send(data)
L
Transport Layer 3-26
rdt2.0,error scenario
Wait for
call from
above
snkpkt = make_pkt(data,checksum)
udt_send(sndpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
Wait for
ACK or
NAK
Wait for
call from
below
rdt_send(data)
L
Transport Layer 3-27
rdt2.0 has a fatal flaw!
What happens if
ACK/NAK corrupted?
? sender doesn?t know what
happened at receiver!
? can?t just retransmit,
possible duplicate
What to do?
? sender ACKs/NAKs
receiver?s ACK/NAK? What
if sender ACK/NAK lost?
? retransmit,but this might
cause retransmission of
correctly received pkt!
Handling duplicates,
? sender adds sequence
number to each pkt
? sender retransmits current
pkt if ACK/NAK garbled
? receiver discards (doesn?t
deliver up) duplicate pkt
Sender sends one packet,
then waits for receiver
response
stop and wait
Transport Layer 3-28
rdt2.1,sender,handles garbled ACK/NAKs
Wait for
call 0 from
above
sndpkt = make_pkt(0,data,checksum)
udt_send(sndpkt)
rdt_send(data)
Wait for
ACK or
NAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isNAK(rcvpkt) )
sndpkt = make_pkt(1,data,checksum)
udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isNAK(rcvpkt) )
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt)
Wait for
call 1 from
above
Wait for
ACK or
NAK 1
LL
Transport Layer 3-29
rdt2.1,receiver,handles garbled ACK/NAKs
Wait for
0 from
below
sndpkt = make_pkt(NAK,chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) &&
has_seq0(rcvpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK,chksum)
udt_send(sndpkt)
Wait for
1 from
below
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)
sndpkt = make_pkt(ACK,chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) &&
has_seq1(rcvpkt)
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(ACK,chksum)
udt_send(sndpkt)
sndpkt = make_pkt(NAK,chksum)
udt_send(sndpkt)
Transport Layer 3-30
rdt2.1,discussion
Sender:
? seq # added to pkt
? two seq,#?s (0,1) will
suffice,Why?
? must check if received
ACK/NAK corrupted
? twice as many states
? state must,remember”
whether,current” pkt
has 0 or 1 seq,#
Receiver:
? must check if received
packet is duplicate
? state indicates whether
0 or 1 is expected pkt
seq #
? note,receiver can not
know if its last
ACK/NAK received OK
at sender
Transport Layer 3-31
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 3-32
rdt2.2,sender,receiver fragments
Wait for
call 0 from
above
sndpkt = make_pkt(0,data,checksum)
udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
Wait for
ACK
0
sender FSM
fragment
Wait for
0 from
below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK1,chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSM
fragment
L
Transport Layer 3-33
rdt3.0,channels with errors and loss
New assumption:
underlying channel can
also lose packets (data
or ACKs)
? checksum,seq,#,ACKs,
retransmissions will be
of help,but not enough
Q,how to deal with loss?
? sender waits until
certain data or ACK
lost,then retransmits
? yuck,drawbacks?
Approach,sender waits
“reasonable” amount of
time for ACK
? retransmits if no ACK
received in this time
? if pkt (or ACK) just delayed
(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 3-34
rdt3.0 sender
sndpkt = make_pkt(0,data,checksum)
udt_send(sndpkt)
start_timer
rdt_send(data)
Wait
for
ACK0
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
Wait for
call 1 from
above
sndpkt = make_pkt(1,data,checksum)
udt_send(sndpkt)
start_timer
rdt_send(data)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,0) )
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,1)
stop_timer
stop_timer
udt_send(sndpkt)
start_timer
timeout
udt_send(sndpkt)
start_timer
timeout
rdt_rcv(rcvpkt)
Wait for
call 0from
above
Wait
for
ACK1
L
rdt_rcv(rcvpkt)
L
L
L
Transport Layer 3-35
rdt3.0 in action
Transport Layer 3-36
rdt3.0 in action
Transport Layer 3-37
Performance of rdt3.0
? rdt3.0 works,but performance stinks
? example,1 Gbps link,15 ms e-e prop,delay,1KB packet:
Ttransmit = 8kb/pkt
10**9 b/sec = 8 microsec
? U sender,utilization – fraction of time sender busy sending
? 1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link
? network protocol limits use of physical resources!
U s e n d e r =, 008
3 0, 0 0 8
= 0, 0 0 0 2 7
m i c r o s e c
o n d s
L / R
R T T + L / R
=
L (packet length in bits)
R (transmission rate,bps) =
Transport Layer 3-38
rdt3.0,stop-and-wait operation
first packet bit transmitted,t = 0
sender receiver
RTT
last packet bit transmitted,t = L / R
first packet bit arrives
last packet bit arrives,send ACK
ACK arrives,send next
packet,t = RTT + L / R
U s e n d e r =, 008
3 0, 0 0 8
= 0, 0 0 0 2 7
m i c r o s e c
o n d s
L / R
R T T + L / R
=
Transport Layer 3-39
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 3-40
Pipelining,increased utilization
first packet bit transmitted,t = 0
sender receiver
RTT
last bit transmitted,t = L / R
first packet bit arrives
last packet bit arrives,send ACK
ACK arrives,send next
packet,t = RTT + L / R
last bit of 2nd packet arrives,send ACK
last bit of 3rd packet arrives,send ACK
U s e n d e r =, 0 24
3 0, 0 0 8
= 0, 0 0 0 8
m i c r o s e c o n
ds
3 * L / R
R T T + L / R
=
Increase utilization
by a factor of 3!
Transport Layer 3-41
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 deceive duplicate ACKs (see receiver)
? timer for each in-flight pkt
? timeout(n),retransmit pkt n and all higher seq # pkts in window
Transport Layer 3-42
GBN,sender extended FSM
Wait start_timerudt_send(sndpkt[base])
udt_send(sndpkt[base+1])
…
udt_send(sndpkt[nextseqnum-1])
timeout
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 = getacknum(rcvpkt)+1
If (base == nextseqnum)
stop_timer
else
start_timer
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
base=1
nextseqnum=1
rdt_rcv(rcvpkt)
&& corrupt(rcvpkt)
L
Transport Layer 3-43
GBN,receiver extended FSM
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 #
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)
&& notcurrupt(rcvpkt)
&& hasseqnum(rcvpkt,expectedseqnum)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(expectedseqnum,ACK,chksum)
udt_send(sndpkt)
expectedseqnum++
expectedseqnum=1
sndpkt =
make_pkt(expectedseqnum,ACK,chksum)
L
Transport Layer 3-44
GBN in
action
Transport Layer 3-45
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 3-46
Selective repeat,sender,receiver windows
Transport Layer 3-47
Selective repeat
data from above,
? if next available seq # in
window,send pkt
timeout(n):
? resend pkt n,restart timer
ACK(n) in [sendbase,sendbase+N]:
? mark pkt n as received
? if n smallest unACKed pkt,
advance window base to
next unACKed seq #
sender
pkt n in [rcvbase,rcvbase+N-1]
? send ACK(n)
? out-of-order,buffer
? in-order,deliver (also
deliver buffered,in-order
pkts),advance window to
next not-yet-received pkt
pkt n in [rcvbase-N,rcvbase-1]
? ACK(n)
otherwise:
? ignore
receiver
Transport Layer 3-48
Selective repeat in action
Transport Layer 3-49
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 3-50
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 3-51
TCP,Overview 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
overwhelm receiver
? point-to-point:
? one sender,one receiver
? reliable,in-order byte
steam:
? no,message boundaries”
? pipelined:
? TCP congestion and flow
control set window size
? send & receive buffers
s oc k et
door
T CP
s en d b uf f er
T CP
r ec ei v e b uf f er
s oc k et
door
s e g m e n t
ap pl i c ati on
w r i tes da ta
ap pl i c ati on
r ea ds da ta
Transport Layer 3-52
TCP segment structure
source port # dest port #
32 bits
application
data
(variable length)
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen notused
Options (variable length)
URG,urgent data
(generally not used)
ACK,ACK #
valid
PSH,push data now
(generally not used)
RST,SYN,FIN:
connection estab
(setup,teardown
commands)
# bytes
rcvr willing
to accept
counting
by bytes
of data
(not segments!)
Internet
checksum
(as in UDP)
Transport Layer 3-53
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 Host B
User
types
?C?
host ACKs
receipt
of echoed
?C?
host ACKs
receipt of
?C?,echoes
back ?C?
time
simple telnet scenario
Transport Layer 3-54
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 3-55
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 3-56
Example RTT estimation:
R TT,gai a, c s, um a s s, e du t o f a nt a s i a, e ure c om, f r
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
t i m e ( s e c onn ds )
R
TT
(
m
i
l
l
i
s
e
c
ond
s
)
Sa m p l e R T T Es t i m a t e d R T T
Transport Layer 3-57
TCP Round Trip Time and Timeout
Setting the timeout
? EstimtedRTTplus,safety margin”
? large variation in EstimatedRTT -> larger safety margin
? first estimate of how much SampleRTT deviates from
EstimatedRTT,
TimeoutInterval = EstimatedRTT + 4*DevRTT
DevRTT = (1-?)*DevRTT +
?*|SampleRTT-EstimatedRTT|
(typically,? = 0.25)
Then set timeout interval:
Transport Layer 3-58
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 3-59
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 3-60
TCP sender events:
data rcvd from app:
? Create segment with
seq #
? seq # is byte-stream
number of first data
byte in segment
? 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 3-61
TCP
sender
(simplified)
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 */
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 3-62
TCP,retransmission scenarios
Host A
time
premature timeout
Host B
Se
q=
92
tim
eou
t
Host A
losst
im
eou
t
lost ACK scenario
Host B
X
time
Se
q=
92
tim
eou
t
SendBase
= 100
SendBase
= 120
SendBase
= 120
Sendbase
= 100
Transport Layer 3-63
TCP retransmission scenarios (more)
Host A
losst
im
eou
t
Cumulative ACK scenario
Host B
X
time
SendBase
= 120
Transport Layer 3-64
TCP ACK generation [RFC 1122,RFC 2581]
Event at Receiver
Arrival of in-order segment with
expected seq #,All data up to
expected seq # already ACKed
Arrival of in-order segment with
expected seq #,One other
segment has ACK pending
Arrival of out-of-order segment
higher-than-expect seq,#,
Gap detected
Arrival of segment that
partially or completely fills gap
TCP Receiver action
Delayed ACK,Wait up to 500ms
for next segment,If no next segment,
send ACK
Immediately send single cumulative
ACK,ACKing both in-order segments
Immediately send duplicate ACK,
indicating seq,# of next expected byte
Immediate send ACK,provided that
segment startsat lower end of gap
Transport Layer 3-65
Fast Retransmit
? Time-out period often
relatively long:
? long delay before
resending lost packet
? Detect lost segments
via duplicate ACKs.
? Sender often sends
many segments back-to-
back
? If segment is lost,
there will likely be many
duplicate ACKs.
? If sender receives 3
ACKs for the same
data,it supposes that
segment after ACKed
data was lost:
? fast retransmit,resend
segment before timer
expires
Transport Layer 3-66
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
}
Fast retransmit algorithm:
a duplicate ACK for
already ACKed segment
fast retransmit
Transport Layer 3-67
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 3-68
TCP Flow Control
? receive side of TCP
connection has a
receive buffer:
? speed-matching
service,matching the
send rate to the
receiving app?s drain
rate? app process may be
slow at reading from
buffer
sender won?t overflow
receiver?s buffer by
transmitting too much,
too fast
flow control
Transport Layer 3-69
TCP Flow control,how it works
(Suppose TCP receiver
discards out-of-order
segments)
? spare room in buffer
= RcvWindow
= RcvBuffer-[LastByteRcvd -
LastByteRead]
? Rcvr advertises spare
room by including value
of RcvWindow in
segments
? Sender limits unACKed
data to RcvWindow
? guarantees receive
buffer doesn?t overflow
Transport Layer 3-70
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 3-71
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 3-72
TCP Connection Management (cont.)
Closing a connection:
client closes socket:
clientSocket.close();
Step 1,client end system
sends TCP FIN control
segment to server
Step 2,server receives
FIN,replies with ACK,
Closes connection,sends
FIN,
client server
close
close
closed
tim
ed
wa
it
Transport Layer 3-73
TCP Connection Management (cont.)
Step 3,client receives FIN,
replies with ACK,
? Enters,timed wait” -
will respond with ACK
to received FINs
Step 4,server,receives
ACK,Connection closed,
Note,with small
modification,can handle
simultaneous FINs.
client server
closing
closing
closed
tim
ed
w
ait
closed
Transport Layer 3-74
TCP Connection Management (cont)
TCP client
lifecycle
TCP server
lifecycle
Transport Layer 3-75
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 3-76
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 3-77
Causes/costs of congestion,scenario 1
? two senders,two
receivers
? one router,
infinite buffers
? no retransmission
? large delays
when congested
? maximum
achievable
throughput
unlimited shared
output link buffers
Host A l
in, original data
Host B
lout
Transport Layer 3-78
Causes/costs of congestion,scenario 2
? one router,finite buffers
? sender retransmission of lost packet
finite shared output
link buffers
Host A l
in, original
data
Host B
lout
l'in, original data,plus
retransmitted data
Transport Layer 3-79
Causes/costs of congestion,scenario 2
? always,(goodput)
?,perfect” retransmission only when loss:
? retransmission of delayed (not lost) packet makes larger
(than perfect case) for same
lin lout=
lin lout>
lin
lout
“costs” of congestion:
? more work (retrans) for given,goodput”
? unneeded retransmissions,link carries multiple copies of pkt
Transport Layer 3-80
Causes/costs of congestion,scenario 3
? four senders
? multihop paths
? timeout/retransmit
linQ,what happens as
and increase?lin
finite shared output
link buffers
Host A l
in, original data
Host B
lout
l'in, original data,plus
retransmitted data
Transport Layer 3-81
Causes/costs of congestion,scenario 3
Another,cost” of congestion:
? when packet dropped,any,upstream transmission
capacity used for that packet was wasted!
H
o
s
t
A
H
o
s
t
B
l
o
u
t
Transport Layer 3-82
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
Two broad approaches towards congestion control:
Transport Layer 3-83
Case study,ATM ABR congestion control
ABR,available bit rate:
?,elastic service”
? if sender?s path
“underloaded”,
? sender should use
available bandwidth
? if sender?s path
congested,
? sender throttled to
minimum guaranteed
rate
RM (resource management)
cells:
? sent by sender,interspersed
with data cells
? bits in RM cell set by switches
(“network-assisted”)
? NI bit,no increase in rate
(mild congestion)
? CI bit,congestion
indication
? RM cells returned to sender by
receiver,with bits intact
Transport Layer 3-84
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 minimum 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 3-85
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 3-86
TCP Congestion Control
? end-end control (no network
assistance)
? sender limits transmission:
LastByteSent-LastByteAcked
? CongWin
? Roughly,
? CongWin is dynamic,function
of perceived network
congestion
How does sender
perceive congestion?
? loss event = timeout or
3 duplicate acks
? TCP sender reduces
rate (CongWin) after
loss event
three mechanisms:
? AIMD
? slow start
? conservative after
timeout events
rate = CongWinRTT Bytes/sec
Transport Layer 3-87
TCP AIMD
8 K b yte s
1 6 K b yte s
2 4 K b yte s
ti m e
co n g e sti o n
w i n d o w
multiplicative decrease:
cut CongWin in half
after loss event
additive increase:
increase CongWin by
1 MSS every RTT in
the absence of loss
events,probing
Long-lived TCP connection
Transport Layer 3-88
TCP Slow Start
? When connection begins,
CongWin = 1 MSS
? Example,MSS = 500
bytes & RTT = 200 msec
? initial rate = 20 kbps
? available bandwidth may
be >> MSS/RTT
? desirable to quickly ramp
up to respectable rate
? When connection begins,
increase rate
exponentially fast until
first loss event
Transport Layer 3-89
TCP Slow Start (more)
? When connection
begins,increase rate
exponentially until
first loss event:
? double CongWin every
RTT
? done by incrementing
CongWin for every ACK
received
? Summary,initial rate
is slow but ramps up
exponentially fast
Host A
RTT
Host B
time
Transport Layer 3-90
Refinement
? After 3 dup ACKs:
?CongWin is cut in half
? window then grows
linearly
? But after timeout event:
?CongWin instead set to
1 MSS;
? window then grows
exponentially
? to a threshold,then
grows linearly
? 3 dup ACKs indicates
network capable of
delivering some segments
? timeout before 3 dup
ACKs is,more alarming”
Philosophy:
Transport Layer 3-91
Refinement (more)
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
0
2
4
6
8
10
12
14
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Tr a ns m i s s i on r ou nd
c
onge
s
t
i
on
w
i
nd
ow
s
i
z
e
(
s
e
g
m
e
nt
s
)
S e r ie s 1 S e r ie s 2
t h r e s h o ld
T C P
T a h o e
T C P
R e n o
Transport Layer 3-92
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 3-93
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
router
capacity R
TCP
connection 2
TCP Fairness
Transport Layer 3-94
Why is TCP fair?
Two competing sessions:
? Additive increase gives slope of 1,as throughout increases
? multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughput
congestion avoidance,additive increase
loss,decrease window by factor of 2
congestion avoidance,additive increase
loss,decrease window by factor of 2
Transport Layer 3-95
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 3-96
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 3-97
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 3-98
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 3-99
TCP Delay Modeling,Slow Start (1)
Now suppose window grows according to slow start
Will show that the delay for one object is:
R
S
R
SR T TP
R
OR T TL a te n c y P )12(2 ???
?
??
?
? ????
where P is the number of times TCP idles at server:
}1,{m i n ?? KQP
- 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 3-100
TCP Delay Modeling,Slow Start (2)
RTT
init iat e T C P
c onnect ion
request
objec t
f irs t w indow
= S/ R
s ec ond w indow
= 2S/ R
t hird w indow
= 4S/ R
f ourt h w indow
= 8S/ R
c om plet e
t rans m is s ion
objec t
deliv ered
t im e at
c lient
t im e at
s erv er
Example:
? O/S = 15 segments
? K = 4 windows
? Q = 2
? P = min{K-1,Q} = 2
Server idles P=2 times
Delay components:
? 2 RTT for connection
estab and request
? O/R to transmit
object
? time server idles due
to slow start
Server idles,
P = min{K-1,Q} times
Transport Layer 3-101
TCP Delay Modeling (3)
R
S
R
S
R T TPR T T
R
O
R
S
R T T
R
S
R T T
R
O
i d l e T i m eR T T
R
O
P
k
P
k
P
p
p
)12(][2
]2[2
2d e l a y
1
1
1
??????
?????
???
?
?
?
?
?
t h w i n d o w a f t e r t h e t i m ei d l e 2 1 kRSR T TRS k ??????? ?? ??
e m e n ta c k n o w l e d g r e c e i v e ss e r v e r u n t i l
s e g m e n t s e n d t os t a r t ss e r v e r w h e nf r o m t i m e?? R T TRS
w in d o w k th th e tr a n s m it toti m e2 1 ?? RSk
RTT
init iat e T C P
c onnect ion
request
objec t
f irs t w indow
= S/ R
s ec ond w indow
= 2S/ R
t hird w indow
= 4S/ R
f ourt h w indow
= 8S/ R
c om plet e
t rans m is s ion
objec t
deliv ered
t im e at
c lient
t im e at
s erv er
Transport Layer 3-102
TCP Delay Modeling (4)
?
?
?
?
?
?
??
???
???
?????
?????
?
?
)1(lo g
)}1(lo g:{m i n
}12:{m i n
}/222:{m i n
}222:{m i n
2
2
110
110
S
O
S
O
kk
S
O
k
SOk
OSSSkK
k
k
k
?
?
Calculation of Q,number of idles for infinite-size object,
is similar (see HW).
Recall K = number of windows that cover object
How do we calculate K?
Transport Layer 3-103
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 3-104
0
2
4
6
8
10
12
14
16
18
20
28
K b p s
100
K b p s
1
M b p s
10
M b p s
n on -p e r s i s te n t
p e r s i s te n t
p a r a l l e l n on -
p e r s i s te n t
HTTP Response time (in seconds)
RTT = 100 msec,O = 5 Kbytes,M=10 and X=5
For low bandwidth,connection & response time dominated by
transmission time.
Persistent connections only give minor improvement over parallel
connections.
Transport Layer 3-105
0
10
20
30
40
50
60
70
28
K b p s
100
K b p s
1
M b p s
10
M b p s
n on -p e r s i s te n t
p e r s i s te n t
p a r a l l e l n on -
p e r s i s te n t
HTTP Response time (in seconds)
RTT =1 sec,O = 5 Kbytes,M=10 and X=5
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 3-106
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”
