Chapter 3 V7.01-Short

Chapter 3
Transport Layer
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7th Edition, Global Edition
Thanks and enjoy! JFK/KWR
Jim Kurose, Keith Ross
All material copyright 1996-2016 Pearson
J.F Kurose and K.W. Ross, All Rights Reserved April 2016
Transport Layer 2-1
Chapter 3: Transport Layer
our goals:
 understand principles  learn about Internet
behind transport transport layer protocols:
layer services: • UDP: connectionless
• multiplexing, transport
demultiplexing • TCP: connection-oriented
• reliable data transfer reliable transport
• flow control • TCP congestion control
• congestion control
Transport Layer 3-2

Chapter 3 outline
3.1 transport-layer 3.5 connection-oriented
services transport: TCP
3.2 multiplexing and • segment structure
demultiplexing • reliable data transfer
3.3 connectionless • flow control
transport: UDP • connection management
3.4 principles of reliable 3.6 principles of congestion
data transfer control
3.7 TCP congestion control
Transport Layer 3-3

Transport services and protocols
application
 provide logical
transport
network
data link
communication between app physical
processes running on
different hosts
lo
gi
ca
 transport protocols run in end
enl
d
systems
-e
nd
• send side: breaks app
tra
ns
messages into segments,
po
rt
passes to network layer
• rcv side: reassembles
application
transport
network
segments into messages, data link
physical
passes to app layer
 more than one transport
protocol available to apps
• Internet: TCP and UDP
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
Transport Layer 3-5

Internet transport-layer protocols
 reliable, in-order
application
transport
network
delivery (TCP) data link

physical
network
• congestion control network data link
lo
data link physical
gi
physical
• flow control
ca
network
le
data link
nd
• connection setup physical
-en
network
d
 unreliable, unordered
tra
data link
physical
ns
delivery: UDP
po
network
rt
data link
physical
• no-frills extension of network
data link application
“ best-effort” IP physical
network
data link
transport
network
data link
 services not available: physical
physical
• delay guarantees
• bandwidth guarantees
Transport Layer 3-6

Chapter 3 outline
Transport Layer 3-7

Multiplexing/demultiplexing
multiplexing at sender:
handle data from multiple demultiplexing at receiver:
sockets, add transport header use header info to deliver
(later used for demultiplexing) received segments to correct
socket
application
application P1 P2 application socket

P3 transport P4
process
transport network transport
network link network
link physical link
physical physical
Transport Layer 3-8

How demultiplexing works
 host receives IP datagrams 32 bits
• each datagram has source IP address,
destination IP address source port # dest port #
• each datagram carries one transport-layer
segment
• each segment has source, destination other header fields
port number
 host uses IP addresses & port
numbers to direct segment to application
appropriate socket
data
(payload)
TCP/UDP segment format
Transport Layer 3-9

Connectionless demux: example
DatagramSocket serverSocket
= new DatagramSocket
DatagramSocket mySocket2 (6428); DatagramSocket
= new DatagramSocket mySocket1 = new
(9157); DatagramSocket (5775);
application
application P1 application
P3 P4
transport
transport transport
network
link physical link
physical physical
source port: 6428 source port: 6428

dest port: 9157 dest port: 5775
source port: 9157 source port: 5775

dest port: 6428 dest port: 6428
Transport Layer 3-10
Connection-oriented demux
 TCP socket identified  server host may support
by 4-tuple: many simultaneous TCP
• source IP address sockets:
• source port number • each socket identified by
• dest IP address its own 4-tuple
• dest port number  web servers have
 demux: receiver uses all different sockets for each
four values to direct connecting client
segment to appropriate • non-persistent HTTP will
have different socket for
socket each request

Connection-oriented demux: example
application
application P4 P5 P6 application
P3 P2 P3
transport
transport transport
network
link physical link
physical server: IP physical
address B
host: IP source IP,port: B,80 host: IP

address A dest IP,port: A,9157 source IP,port: C,5775 address C
dest IP,port: B,80
source IP,port: A,9157
dest IP, port: B,80
source IP,port: C,9157
dest IP,port: B,80
three segments, all destined to IP address: B,
dest port: 80 are demultiplexed to different sockets Transport Layer 3-12
Connection-oriented demux: example
threaded server
application
application application
P4
P3 P2 P3
transport
transport transport
network
link physical link
physical server: IP physical
address B
host: IP source IP,port: B,80 host: IP

address A dest IP,port: A,9157 source IP,port: C,5775 address C
dest IP,port: B,80
source IP,port: A,9157
dest IP, port: B,80
source IP,port: C,9157
dest IP,port: B,80

Chapter 3 outline

UDP: User Datagram Protocol [RFC 768]
 “ no frills,” “ bare bones”  UDP use:
Internet transport protocol  streaming multimedia
 “ best effort” service, UDP apps (loss tolerant, rate
segments may be: sensitive)
• lost  DNS
• delivered out-of-order to  SNMP
app
 reliable transfer over
 connectionless:
UDP:
• no handshaking between
UDP sender, receiver  add reliability at
application layer
• each UDP segment
handled independently  application-specific error
of others recovery!

UDP: segment header
length, in bytes of
32 bits UDP segment,
source port # dest port # including header
length checksum
why is there a UDP?
 no connection
application establishment (which can
data add delay)
(payload)  simple: no connection state
at sender, receiver
 small header size
 no congestion control: UDP
UDP segment format
can blast away as fast as
desired

UDP checksum
Goal: detect “ errors” (e.g., flipped bits) in
transmitted segment
sender: receiver:
 treat segment contents,  compute checksum of
including header fields, as received segment
sequence of 16-bit integers
 check if computed checksum
 checksum: addition (one’s
complement sum) of
equals checksum field value:
segment contents • NO - error detected
 sender puts checksum • YES - no error detected
value into UDP checksum • But maybe errors
field

Internet checksum: example
example: add two 16-bit integers
1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0
1 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1
wraparound 1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1
sum 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0
checksum 1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
Note: when adding numbers, a carryout from the most

significant bit needs to be added to the result
* Check out the online interactive exercises for more

examples: http://gaia.cs.umass.edu/kurose_ross/interactive/ Transport Layer 3-18
Chapter 3 outline

Principles of reliable data transfer
 important in application, transport, link layers
• top-10 list of important networking topics!
 characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)



rdt1.0: reliable transfer over a reliable channel
 underlying channel perfectly reliable
• no bit errors
• no loss of packets
 separate operations for sender, receiver:
• sender sends data into underlying channel
• receiver reads data from underlying channel

rdt2.0: channel with bit errors
 underlying channel may flip bits in packet
• checksum to detect bit errors
 the question: how to recover from errors:
• acknowledgements (ACKs): receiver explicitly tells
sender that pkt received OK
• negative acknowledgements (NAKs): receiver explicitly
tells sender that pkt had errors
• sender retransmits pkt on receipt of NAK
 new mechanisms in rdt2.0 (beyond rdt1.0):
• error detection
• feedback: control msgs (ACK,NAK) from receiver to
sender

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

sender sends one packet,
then waits for receiver
response

rdt2.1: discussion
sender: receiver:
 seq # added to pkt  must check if received
 two seq. #’s (0,1) will packet is duplicate
suffice. Why? • state indicates whether
0 or 1 is expected pkt
 must check if received seq #
ACK/NAK corrupted

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

rdt3.0: channels with errors and loss
new assumption: approach: sender waits

underlying channel can “ reasonable” amount of
also lose packets (data, time for ACK
ACKs)  retransmits if no ACK
• checksum, seq. #, received in this time
ACKs, retransmissions  if pkt (or ACK) just delayed
(not lost):
will be of help … but
not enough • retransmission will be
duplicate, but seq. #’s
already handles this
• receiver must specify seq
# of pkt being ACKed
 requires countdown timer

rdt3.0 in action
sender receiver sender receiver
send pkt0 pkt0 send pkt0 pkt0
rcv pkt0 rcv pkt0
ack0 send ack0 ack0 send ack0
rcv ack0 rcv ack0
send pkt1 pkt1 send pkt1 pkt1
rcv pkt1 X
ack1 send ack1 loss
rcv ack1
send pkt0 pkt0
rcv pkt0 timeout
ack0 send ack0 resend pkt1 pkt1
rcv pkt1
ack1 send ack1
rcv ack1
send pkt0 pkt0
(a) no loss rcv pkt0
ack0 send ack0
(b) packet loss

rdt3.0 in action
sender receiver
sender receiver send pkt0 pkt0
send pkt0 pkt0 rcv pkt0
ack0 send ack0
rcv pkt0
send ack0 rcv ack0
ack0 send pkt1 pkt1
rcv ack0 rcv pkt1
send pkt1 pkt1
rcv pkt1 send ack1
ack1 ack1
send ack1
X
loss timeout
resend pkt1 pkt1
rcv pkt1
timeout
resend pkt1 pkt1 rcv ack1 pkt0 (detect duplicate)
rcv pkt1 send pkt0 send ack1
(detect duplicate) ack1
ack1 send ack1 rcv pkt0
rcv ack1 ack0 send ack0
send pkt0 pkt0
rcv pkt0
ack0 send ack0
(c) ACK loss (d) premature timeout/ delayed ACK

Performance of rdt3.0
 rdt3.0 is correct, but performance stinks
 e.g.: 1 Gbps link, 15 ms prop. delay, 8000 bit packet:
L 8000 bits
Dtrans = R = = 8 microsecs
109 bits/sec
 U sender: utilization – fraction of time sender busy sending

U L/R .008
sender = = = 0.00027
RTT + L / R 30.008
 if RTT=30 msec, 1KB pkt every 30 msec: 33kB/sec thruput over 1 Gbps link
 network protocol limits use of physical resources!

rdt3.0: stop-and-wait operation
sender receiver
first packet bit transmitted, t = 0
last packet bit transmitted, t = L / R
first packet bit arrives

RTT last packet bit arrives, send ACK
ACK arrives, send next

packet, t = RTT + L / R
U L/R .008
sender = = = 0.00027
RTT + L / R 30.008

Pipelining: increased utilization
sender receiver
first packet bit transmitted, t = 0
last bit transmitted, t = L / R
first packet bit arrives

RTT last packet bit arrives, send ACK
last bit of 2nd packet arrives, send ACK
last bit of 3rd packet arrives, send ACK
ACK arrives, send next
packet, t = RTT + L / R
3-packet pipelining increases
utilization by a factor of 3!
U 3L / R .0024
sender = = = 0.00081
RTT + L / R 30.008

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
Pipelined protocols: overview
Go-back-N: Selective Repeat:
 sender can have up to N  sender can have up to N
unacked packets in unack’ed packets in
pipeline pipeline
 receiver only sends  rcvr sends individual ack
cumulative ack for each packet
• doesn’t ack packet if
there’s a gap
 sender has timer for  sender maintains timer for
oldest unacked packet each unacked packet
• when timer expires, • when timer expires,
retransmit all unacked retransmit only that
packets unacked packet

Go-Back-N: sender
 k-bit seq # in pkt header
 “ window” of up to N, consecutive unack’ed pkts allowed
 ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK”

• may receive duplicate ACKs (see receiver)
 timer for oldest in-flight pkt
 timeout(n): retransmit packet n and all higher seq # pkts in window

Go-Back-N
sender receiver
data from above: pkt n at expectedseqnum
 if next available seq # in  send ACK(n)
window
 send pkt  increase expectedseqnum
 start a timer if no timer otherwise:
timeout(n):  resend last ACK
 resend all pkts from send_base
to nextseqnum
 restart timer
ACK(n):
 mark pkts up to n as received
 restart timer if there is still
unacked pkts

GBN: receiver
ACK-only: always send ACK for correctly-received
pkt with highest in-order seq #
• may generate duplicate ACKs
• need only remember expectedseqnum
 out-of-order pkt:
• discard (don’t buffer): no receiver buffering!
• re-ACK pkt with highest in-order seq #

GBN in action
sender window (N=4) sender receiver
012345678 send pkt0
012345678 send pkt1
send pkt2 receive pkt0, send ack0
012345678
send pkt3 Xloss receive pkt1, send ack1
012345678
(wait)
receive pkt3, discard,
012345678 rcv ack0, send pkt4 (re)send ack1
012345678 rcv ack1, send pkt5 receive pkt4, discard,
(re)send ack1
ignore duplicate ACK receive pkt5, discard,
(re)send ack1
pkt 2 timeout
012345678 send pkt2
012345678 send pkt3
012345678 send pkt4 rcv pkt2, deliver, send ack2
012345678 send pkt5 rcv pkt3, deliver, send ack3
rcv pkt4, deliver, send ack4
rcv pkt5, deliver, send ack5

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
• limits seq #s of sent, unACKed pkts

Selective repeat: sender, receiver windows

Selective repeat
sender receiver
data from above: pkt n in [rcvbase, rcvbase+N-1]
 if next available seq # in  send ACK(n)
window, send pkt  out-of-order: buffer
timeout(n):  in-order: deliver (also
 resend pkt n, restart timer deliver buffered, in-order
pkts), advance window to
ACK(n) in [sendbase,sendbase+N-1]: next not-yet-received pkt
 mark pkt n as received
pkt n in [rcvbase-N,rcvbase-1]
 if n smallest unACKed
 ACK(n)
pkt, advance window base
to next unACKed seq # otherwise:
 ignore

Selective repeat in action
sender window (N=4) sender receiver
012345678 send pkt0
012345678 send pkt1
send pkt2 receive pkt0, send ack0
012345678
send pkt3 Xloss receive pkt1, send ack1
012345678
(wait)
receive pkt3, buffer,
012345678 rcv ack0, send pkt4 send ack3
012345678 rcv ack1, send pkt5 receive pkt4, buffer,
send ack4
record ack3 arrived receive pkt5, buffer,
send ack5
pkt 2 timeout
012345678 send pkt2
012345678 record ack4 arrived
012345678 rcv pkt2; deliver pkt2,
record ack5 arrived
012345678 pkt3, pkt4, pkt5; send ack2
Q: what happens when ack2 arrives?

Chapter 3 outline

TCP: Overview RFCs: 793,1122,1323, 2018, 2581
 point-to-point:  full duplex data:

• one sender, one receiver • bi-directional data flow
 reliable, in-order byte in same connection
• MSS: maximum segment
steam: size
• no “ message boundaries
”  connection-oriented:
• handshaking (exchange
 pipelined: of control msgs) inits
• TCP congestion and flow sender, receiver state
control set window size before data exchange
 flow controlled:
• sender will not
overwhelm receiver
TCP segment structure
32 bits
URG: urgent data counting
(generally not used) source port # dest port #
by bytes
sequence number of data
ACK: ACK #
valid acknowledgement number (not segments!)
head not
PSH: push data now len used
UAP R S F receive window
(generally not used) # bytes
checksum Urg data pointer
rcvr willing
RST, SYN, FIN: to accept
options (variable length)
connection estab
(setup, teardown
commands)
application
Internet data
checksum (variable length)
(as in UDP)

TCP seq. numbers, ACKs
outgoing segment from sender
sequence numbers: source port # dest port #
sequence number
• byte stream “ number” of acknowledgement number
first byte in segment’s checksum

rwnd
urg pointer
data window size
acknowledgements: N
• seq # of next byte

expected from other side sender sequence number space
• cumulative ACK
sent sent, not- usable not
Q: how receiver handles out- ACKed yet ACKed but not usable
of-order segments (“in-flight”) yet sent
• A: TCP spec doesn’t say, incoming segment to sender

- up to implementor source port # dest port #
sequence number
acknowledgement number
A rwnd
checksum urg pointer

TCP seq. numbers, ACKs
Host A Host B
User
types
‘C’
Seq=42, ACK=79, data = ‘C’
host ACKs
receipt of
‘C’, echoes
Seq=79, ACK=43, data = ‘C’ back ‘C’
host ACKs
receipt
of echoed
‘C’ Seq=43, ACK=80
simple telnet scenario

TCP round trip time, timeout
Q: how to set TCP Q: how to estimate RTT?
timeout value?  SampleRTT: measured
time from segment
 longer than RTT transmission until ACK
• but RTT varies receipt
 too short: premature • ignore retransmissions
timeout, unnecessary  SampleRTT will vary, want
retransmissions estimated RTT “ smoother”
• average several recent
 too long: slow measurements, not just
reaction to segment current SampleRTT
loss

EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT
 exponential weighted moving average
 influence of past sample decreases exponentially fast
 typical value:  = 0.125 RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
350
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
300
(milliseconds)
RTT
250
RTT (milliseconds)
200
sampleRTT
150
EstimatedRTT
100
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
time (seconds) Transport Layer 3-50
SampleRTT Estimated RTT
 timeout interval: EstimatedRTT plus “ safety margin”
• large variation in EstimatedRTT -> larger safety margin
 estimate SampleRTT deviation from EstimatedRTT:
DevRTT = (1-)*DevRTT +
*|SampleRTT-EstimatedRTT|
(typically,  = 0.25)
TimeoutInterval = EstimatedRTT + 4*DevRTT
estimated RTT “safety margin”

Chapter 3 outline

TCP reliable data transfer
 TCP creates rdt service
on top of IP’s unreliable
service
• pipelined segments
• cumulative acks let’s initially consider
• single retransmission simplified TCP sender:
timer • ignore duplicate acks
 retransmissions • ignore flow control,
triggered by: congestion control
• timeout events
• duplicate acks

TCP sender events:
data rcvd from app: timeout:
 create segment with  retransmit segment that
seq # caused timeout
 seq # is byte-stream  restart timer
number of first data ack rcvd:
byte in segment  if ack acknowledges
 start timer if not previously unacked
already running segments
• think of timer as for • update what is known to
oldest unacked segment be ACKed
• expiration interval: • start timer if there are
TimeOutInterval still unacked segments

TCP: retransmission scenarios
Host A Host B Host A Host B
SendBase=92
Seq=92, 8 bytes of data Seq=92, 8 bytes of data
Seq=100, 20 bytes of data

timeo
timeo
ACK=100
ut
ut
X
ACK=100
ACK=120
Seq=92, 8 bytes of data Seq=92, 8

SendBase=100 bytes of data
SendBase=120
ACK=100
ACK=120
SendBase=120
lost ACK scenario premature timeout

TCP: retransmission scenarios
Host A Host B

ACK=100
timeo
X
ut
ACK=120
cumulative ACK
TCP ACK generation [RFC 1122, RFC 2581]
event at receiver TCP receiver action
arrival of in-order segment with delayed ACK. Wait up to 500ms
expected seq #. All data up to for next segment. If no next segment,
expected seq # already ACKed send ACK
arrival of in-order segment with immediately send single cumulative

expected seq #. One other ACK, ACKing both in-order segments
segment has ACK pending
arrival of out-of-order segment immediately send duplicate ACK,

higher-than-expect seq. # . indicating seq. # of next expected byte
Gap detected
arrival of segment that immediate send ACK, provided that

partially or completely fills gap segment starts at lower end of gap

TCP fast retransmit
 time-out period often
relatively long: TCP fast retransmit
• long delay before if sender receives 3
resending lost packet ACKs for same data
 detect lost segments (“
(“triple
tripleduplicate ACKs”
duplicate ),
ACKs” ),
via duplicate ACKs. resend unacked
• sender often sends segment with smallest
many segments back- seq #
to-back  likely that unacked
• if segment is lost, there segment lost, so don’t
will likely be many wait for timeout
duplicate ACKs.

TCP fast retransmit
Host A Host B

X
ACK=100
ACK=100
timeo
ut
ACK=100
ACK=100
fast retransmit after sender

receipt of triple duplicate ACK
Chapter 3 outline

TCP flow control
application
application may process
remove data from application
TCP socket buffers ….
TCP socket OS
receiver buffers
… slower than TCP
receiver is delivering
(sender is sending) TCP
code
IP
flow control code
receiver controls sender, so
sender won’t overflow receiver’s
buffer by transmitting too much, from sender
too fast
receiver protocol stack

TCP flow control
 receiver “ advertises” free
buffer space by including to application process
rwnd value in TCP header
of receiver-to-sender
segments RcvBuffer buffered data
• RcvBuffer size set via
socket options (typical default rwnd free buffer space
is 4096 bytes)
• many operating systems
autoadjust RcvBuffer TCP segment payloads
 sender limits amount of
unacked (“ in-flight” ) data to receiver-side buffering
receiver’s rwnd value
 guarantees receive buffer
will not overflow
Chapter 3 outline

Connection Management
before exchanging data, sender/receiver “ handshake” :
 agree to establish connection (each knowing the other willing to
establish connection)
 agree on connection parameters
application application
connection state: ESTAB connection state: ESTAB

connection variables: connection Variables:
seq # client-to-server seq # client-to-server
server-to-client server-to-client
rcvBuffer size rcvBuffer size
at server,client at server,client
network network
Socket clientSocket = Socket connectionSocket =

newSocket("hostname","port welcomeSocket.accept();
number");
TCP 3-way handshake
client state server state

CLOSED LISTEN
choose init seq num, x
send TCP SYN msg
SYNSENT SYNbit=1, Seq=x
choose init seq num, y
send TCP SYNACK
msg, acking SYN SYN RCVD
SYNbit=1, Seq=y
ACKbit=1; ACKnum=x+1
received SYNACK(x)
ESTAB indicates server is live;
send ACK for SYNACK;
this segment may contain ACKbit=1, ACKnum=y+1
client-to-server data
received ACK(y)
indicates client is live
ESTAB

TCP: closing a connection
 client, server each close their side of connection
• send TCP segment with FIN bit = 1
 respond to received FIN with ACK
• on receiving FIN, ACK can be combined with own FIN
 simultaneous FIN exchanges can be handled

TCP: closing a connection
client state server state
ESTAB ESTAB
clientSocket.close()
FIN_WAIT_1 can no longer FINbit=1, seq=x
send but can
receive data CLOSE_WAIT
ACKbit=1; ACKnum=x+1
can still
FIN_WAIT_2 wait for server send data
close
LAST_ACK
FINbit=1, seq=y
TIMED_WAIT can no longer
send data
ACKbit=1; ACKnum=y+1
timed wait
for 2*max CLOSED
segment lifetime
CLOSED

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

Chapter 3 outline

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!

Chapter 3 outline

TCP congestion control: additive increase
multiplicative decrease
 approach: sender increases transmission rate (window
size), probing for usable bandwidth, until loss occurs
• additive increase: increase cwnd by 1 MSS every
RTT until loss detected
• multiplicative decrease: cut cwnd in half after loss
additively increase window size …
…. until loss occurs (then cut window in half)
congestion window size
cwnd: TCP sender
AIMD saw tooth

behavior: probing
for bandwidth
time
TCP Congestion Control: details
sender sequence number space
cwnd TCP sending rate:
 roughly: send cwnd
bytes, wait RTT for
last byte last byte ACKS, then send more
ACKed sent, not-
yet ACKed
sent bytes
(“in-flight”)
cwnd
 sender limits transmission: rate ~
~ bytes/sec
RTT
LastByteSent- < cwnd
LastByteAcked
 cwnd is dynamic, function of
perceived network congestion

TCP Slow Start
Host A Host B
 when connection begins,
increase rate
exponentially until first one segm
ent
loss event:
RTT
• initially cwnd = 1 MSS two segm
ents
• double cwnd every RTT
• done by incrementing
cwnd for every ACK four segm
ents
received
 summary: initial rate is
slow but ramps up
exponentially fast time

TCP: detecting, reacting to loss
 loss indicated by timeout:
• cwnd set to 1 MSS;
• window then grows exponentially (as in slow start) to threshold, then grows linearly
 loss indicated by 3 duplicate ACKs: TCP RENO
• dup ACKs indicate network capable of delivering some segments
• cwnd is cut in half window then grows linearly
 TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)

TCP: switching from slow start to CA
Q: when should the
exponential increase
switch to linear?
A: when cwnd gets to
1/2 of its value before
timeout.
Implementation:
 variable ssthresh
 on loss event, ssthresh
is set to 1/2 of cwnd just
before loss event

Reaction to Timeout Events
 After timeout event:
• CongWin set to 1 MSS;
• window then grows exponentially
• to a threshold, then grows linearly Philosophy:
 Fast Recovery
• After 3 dup ACKs:  3 dup ACKs indicates
• CongWin is cut in half
• window then grows linearly
network capable of
• TCP Reno version only delivering some segments
 timeout indicates a
“more alarming”
congestion scenario

Summary: TCP Congestion Control
 When CongWin is below Threshold, sender in slow-
start phase, window grows exponentially.
 When CongWin is above Threshold, sender is in
congestion-avoidance phase, window grows linearly.
 When a triple duplicate ACK occurs, Threshold set to
CongWin/2 and CongWin set to Threshold.
 When timeout occurs, Threshold set to CongWin/2
and CongWin is set to 1 MSS.

TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start ACK receipt CongWin = CongWin + MSS, Resulting in a doubling of
(SS) for previously If (CongWin > Threshold) CongWin every RTT
unacked set state to “Congestion
data Avoidance”
Congestion ACK receipt CongWin = CongWin+MSS * Additive increase, resulting
Avoidance for previously (MSS/CongWin) in increase of CongWin by
(CA) unacked 1 MSS every RTT
data
SS or CA Loss event Threshold = CongWin/2, Fast recovery,
detected by CongWin = Threshold, implementing multiplicative
triple Set state to “Congestion decrease. CongWin will not
duplicate Avoidance” drop below 1 MSS.
ACK
SS or CA Timeout Threshold = CongWin/2, Enter slow start
CongWin = 1 MSS,
Set state to “Slow Start”
SS or CA Duplicate Increment duplicate ACK count CongWin and Threshold
ACK for segment being acked not changed

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Chapter 3

A note on the use of these Powerpoint slides:

Transport Layer 3-2

Transport Layer 3-3

Transport Layer 3-5

delivery (TCP) data link

Transport Layer 3-6

Transport Layer 3-7

application P1 P2 application socket

Transport Layer 3-8

TCP/UDP segment format

Transport Layer 3-9

source port: 6428 source port: 6428

source port: 9157 source port: 5775

Transport Layer 3-11

host: IP source IP,port: B,80 host: IP

host: IP source IP,port: B,80 host: IP

Transport Layer 3-13

Transport Layer 3-14

Transport Layer 3-15

Transport Layer 3-16

Transport Layer 3-17

Note: when adding numbers, a carryout from the most

* Check out the online interactive exercises for more

Transport Layer 3-19

Transport Layer 3-20

Transport Layer 3-21

Transport Layer 3-22

Transport Layer 3-23

Transport Layer 3-24

stop and wait

Transport Layer 3-25

Transport Layer 3-26

Transport Layer 3-27

new assumption: approach: sender waits

Transport Layer 3-28

(b) packet loss

(c) ACK loss (d) premature timeout/ delayed ACK

Transport Layer 3-30

 U sender: utilization – fraction of time sender busy sending

Transport Layer 3-31

first packet bit arrives

ACK arrives, send next

Transport Layer 3-32

first packet bit arrives

Transport Layer 3-33

 two generic forms of pipelined protocols: go-Back-N,

Transport Layer 3-35

 ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK”

Transport Layer 3-36

Transport Layer 3-37

Transport Layer 3-38

Transport Layer 3-39

Transport Layer 3-40

Transport Layer 3-41

Transport Layer 3-42

Q: what happens when ack2 arrives?

Transport Layer 3-43

Transport Layer 3-44

 point-to-point:  full duplex data:

Transport Layer 3-46

first byte in segment’s checksum

• seq # of next byte