Chapter 3. Transport Layer

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 3-1

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 3-2

Transport services and protocols
application
transport
▪ provide logical communication network
data link
between app processes physical
running on different hosts

▪ transport protocols run in
end systems
• send side: breaks app
messages into segments,
passes to network layer
• rcv side: reassembles application
segments into messages, transport
network
passes to app layer data link
physical
▪ more than one transport

protocol available to apps
• Internet: TCP and UDP
Transport Layer 3-3
Transport vs. network layer
▪ network layer: logical household analogy:
communication
between hosts 12 kids in Ann’s house sending
letters to 12 kids in Bill’s
▪ transport layer: house:
logical ▪ hosts = houses
communication ▪ processes = kids
between processes ▪ app messages = letters in
envelopes
• relies on, enhances, ▪ transport protocol = Ann
network layer and Bill who demux to in-
services house siblings
▪ network-layer protocol =
postal service
Transport Layer 3-4

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

physical
network
• congestion control network

data link
data link
physical
physical
• flow control network
data link
• connection setup physical
network
▪ unreliable, unordered data link

physical
delivery: UDP network

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 3-5

Chapter 3 outline
Transport Layer 3-6

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 3-7

How demultiplexing works
▪ host receives IP datagrams 32 bits
• each datagram has source IP source port # dest port #
address, destination IP
address
other header fields
• each datagram carries one
transport-layer segment
• each segment has source, application
destination port number data
▪ host uses IP addresses & (payload)
port numbers to direct
segment to appropriate
TCP/UDP segment format
socket
Transport Layer 3-8

Connectionless demultiplexing
▪ recall: created socket has ▪ recall: when creating
host-local port #: datagram to send into UDP
DatagramSocket mySocket1 socket, must specify
= new DatagramSocket(12534);
• destination IP address
• destination port #
▪ when host receives UDP IP datagrams with same

segment: dest. port #, but different
• checks destination port # source IP addresses
in segment and/or source port
numbers will be directed
• directs UDP segment to to same socket at dest
socket with that port #
Transport Layer 3-9

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

dest port: 9157 dest port: ?
source port: 9157 source port: ?

dest port: 6428 dest port: ?
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
four values to direct each connecting client
segment to appropriate • non-persistent HTTP will
socket have different socket for
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
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 ▪ streaming multimedia
protocol apps (loss tolerant, rate
▪ “best effort” service, UDP sensitive)
segments may be: ▪ DNS
• lost ▪ SNMP
• delivered out-of-order ▪ reliable transfer over
to app
UDP:
▪ connectionless:
▪ add reliability at
• no handshaking application layer
between UDP sender,
receiver ▪ application-specific error
recovery!
• each UDP segment
handled independently
of others
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
UDP segment format ▪ no congestion control:
UDP 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, received segment
as sequence of 16-bit ▪ check if computed checksum
integers
equals checksum field value:
▪ checksum: addition
(one’s complement sum) • NO - error detected
of segment contents • YES - no error detected.
▪ sender puts checksum But maybe errors
value into UDP checksum nonetheless? More later
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

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
steam: • MSS: maximum segment
size
• no “message
boundaries” ▪ connection-oriented:
▪ pipelined: • handshaking (exchange
of control msgs) inits
• TCP congestion and sender, receiver state
flow control set window before data exchange
size
▪ 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 rwnd
data
checksum urg pointer
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 ACKed yet ACKed but not usable
out-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 reaction measurements, not just
to segment loss current SampleRTT

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

RTT (milliseconds)
300
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 3-25
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
seq # that 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
oldest unacked to be ACKed
segment
• start timer if there are
• expiration interval: still unacked segments
TimeOutInterval

TCP sender (simplified)
data received from application above
create segment, seq. #: NextSeqNum
pass segment to IP (i.e., “send”)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
L start timer
NextSeqNum = InitialSeqNum wait
SendBase = InitialSeqNum for
event timeout
retransmit not-yet-acked segment
with smallest seq. #
start timer
ACK received, with ACK field value y
if (y > SendBase) {
SendBase = y
/* SendBase–1: last cumulatively ACKed byte */
if (there are currently not-yet-acked segments)
start timer
else stop timer
} Transport Layer 3-30
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

timeout
timeout
ACK=100
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

timeout
ACK=100
X
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
(“triple duplicate
duplicate ACKs”),
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
timeout
ACK=100
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 from sender
too much, 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
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");

Agreeing to establish a connection
2-way handshake:
Q: will 2-way handshake
always work in
network?
Let’s talk
ESTAB ▪ variable delays
OK
ESTAB ▪ retransmitted messages (e.g.
req_conn(x)) due to
message loss
▪ message reordering
choose x
req_conn(x)
▪ can’t “see” other side
ESTAB
acc_conn(x)
ESTAB

TCP 3-way handshake
client state server state

LISTEN 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 3-way handshake: FSM
closed
Socket connectionSocket =
welcomeSocket.accept();
L Socket clientSocket =
SYN(x) newSocket("hostname","port
number");
SYNACK(seq=y,ACKnum=x+1)
create new socket for SYN(seq=x)
communication back to client listen
SYN SYN
rcvd sent
SYNACK(seq=y,ACKnum=x+1)
ESTAB ACK(ACKnum=y+1)
ACK(ACKnum=y+1)
L

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

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!

Causes/costs of congestion: scenario 1
original data: lin throughput: lout
▪ two senders, two
receivers Host A
▪ one router, infinite buffers unlimited shared

▪ output link capacity: R output link buffers
▪ no retransmission
Host B
R/2
delay
lout
lin R/2 lin R/2

▪ maximum per-connection ❖ large delays as arrival rate, lin,
throughput: R/2 approaches capacity
▪ one router, finite buffers
▪ sender retransmission of timed-out packet
• application-layer input = application-layer output: lin =
lout
• transport-layer input includes retransmissions : l‘in lin
lin : original data

lout
l'in: original data, plus
retransmitted data
Host A
finite shared output

Host B
link buffers
R/2
idealization: perfect
knowledge
lout
▪ sender sends only when
router buffers available
lin R/2
lin : original data

lout
copy l'in: original data, plus
retransmitted data
A free buffer space!

Host B
link buffers
Idealization: known loss
packets can be lost,
dropped at router due
to full buffers
▪ sender only resends if
packet known to be lost
lin : original data

lout
copy l'in: original data, plus
retransmitted data
A
no buffer space!
Host B
Idealization: known loss R/2
packets can be lost,
dropped at router due when sending at R/2,
some packets are
lout
to full buffers retransmissions but
▪ sender only resends if

asymptotic goodput
is still R/2 (why?)
packet known to be lost lin R/2
lin : original data

lout
retransmitted data
A
free buffer space!
Host B
Realistic: duplicates R/2
▪ packets can be lost, dropped at
router due to full buffers when sending at R/2,
some packets are
lout
▪ sender times out prematurely, retransmissions
sending two copies, both of including duplicated

that are delivered!
which are delivered lin R/2
lin
timeout
copy l'in lout
A
free buffer space!
Host B
Realistic: duplicates R/2
▪ packets can be lost, dropped at
router due to full buffers when sending at R/2,
some packets are
lout
▪ sender times out prematurely, retransmissions
sending two copies, both of including duplicated

that are delivered!
which are delivered lin R/2
“costs” of congestion:
▪ more work (retrans) for given “goodput”
▪ unneeded retransmissions: link carries multiple copies of pkt
• decreasing goodput

▪ four senders Q: what happens as lin and lin’
increase ?
▪ multihop paths
A: as red lin’ increases, all arriving
▪ timeout/retransmit blue pkts at upper queue are
dropped, blue throughput g 0
Host A
lin : original data lout
Host B
retransmitted data
link buffers
Host D
Host C

C/2
lout
lin’ C/2
another “cost” of congestion:

▪ when packet dropped, any “upstream
transmission capacity used for that packet was
wasted!

Causes/costs of congestion: insights
R/2
▪ throughput can never exceed
throughput: lout
capacity
lin
delay
R/2
▪ delay increases as capacity approached

R/2 lin R/2
▪ loss/retransmission decreases
lout throughput:
effective throughput
lin R/2 R/2
▪ un-needed duplicates further
throughput: lout
decreases effective throughput
R/2 lin R/2
▪ upstream transmission capacity / lout

buffering wasted for packets lost
downstream lin’ R/2
Transport Layer: 3-57

Approaches towards congestion control
End-end congestion control:

▪ no explicit feedback from
network
▪ congestion inferred from
observed loss, delay data data
ACKs
ACKs
▪ approach taken by TCP

Approaches towards congestion control
Network-assisted congestion
control:
explicit congestion info
▪ routers provide direct feedback
to sending/receiving hosts with data data
flows passing through congested ACKs
ACKs
router
▪ may indicate congestion level or
explicitly set sending rate
▪ TCP ECN, ATM, DECbit protocols

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
ACKed sent, not-
yet ACKed
sent more bytes
(“in-
flight”) cwnd
▪ sender limits transmission: rate ~
~
RTT
bytes/sec
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
loss event:
RTT
• initially cwnd = 1 MSS
• double cwnd every RTT
• done by incrementing
cwnd for every ACK
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

Summary: TCP Congestion Control
New
New ACK!
ACK! new ACK
duplicate ACK
dupACKcount++ new ACK
.
cwnd = cwnd + MSS (MSS/cwnd)
dupACKcount = 0
cwnd = cwnd+MSS transmit new segment(s), as allowed
dupACKcount = 0
L transmit new segment(s), as allowed
cwnd = 1 MSS
ssthresh = 64 KB cwnd > ssthresh
dupACKcount = 0 slow L congestion
start timeout avoidance
ssthresh = cwnd/2
cwnd = 1 MSS duplicate ACK
timeout dupACKcount = 0 dupACKcount++
ssthresh = cwnd/2 retransmit missing segment
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
timeout
New
ACK!
ssthresh = cwnd/2
cwnd = 1 New ACK
dupACKcount = 0
cwnd = ssthresh dupACKcount == 3
dupACKcount == 3 retransmit missing segment dupACKcount = 0
ssthresh= cwnd/2 ssthresh= cwnd/2
cwnd = ssthresh + 3 cwnd = ssthresh + 3
retransmit missing segment retransmit missing segment
fast
recovery
duplicate ACK
cwnd = cwnd + MSS
transmit new segment(s), as allowed

TCP Fairness
fairness goal: if K TCP sessions share same
bottleneck link of bandwidth R, each should have
average rate of R/K
TCP connection 1
bottleneck
router
capacity R
TCP connection 2

Why is TCP fair?
two competing sessions:
▪ additive increase gives slope of 1, as throughout increases
▪ multiplicative decrease decreases throughput proportionally
R equal bandwidth share
loss: decrease window by factor of 2

congestion avoidance: additive increase
loss: decrease window by factor of 2
congestion avoidance: additive increase
Connection 1 throughput R
Fairness (more)
Fairness and UDP Fairness, parallel TCP
▪ multimedia apps often connections
do not use TCP ▪ application can open
• do not want rate multiple parallel
throttled by congestion connections between
control
two hosts
▪ instead use UDP:
• send audio/video at
▪ web browsers do this
constant rate, tolerate ▪ e.g., link of rate R with 9
packet loss existing connections:
• new app asks for 1 TCP, gets
rate R/10
• new app asks for 11 TCPs,
gets R/2

Chapter 3: summary
▪ principles behind transport
layer services: next:
• multiplexing, ▪ leaving the network
demultiplexing “edge” (application,
• reliable data transfer transport layers)
• flow control ▪ into the network
• congestion control “core”
▪ instantiation, ▪ two network layer
implementation in the chapters:
Internet • data plane
• UDP • control plane
• TCP

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Chapter 3: Transport Layer

Transport Layer 3-1

Transport Layer 3-2

running on different hosts

▪ more than one transport

Transport Layer 3-4

delivery (TCP) data link

• congestion control network

• connection setup physical

▪ unreliable, unordered data link

delivery: UDP network

Transport Layer 3-5

Transport Layer 3-6

application P1 P2 application socket

Transport Layer 3-7

Transport Layer 3-8

▪ when host receives UDP IP datagrams with same

Transport Layer 3-9

source port: 6428 source port: ?

source port: 9157 source port: ?

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 3-16

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

▪ point-to-point: ▪ full duplex data:

Transport Layer 3-21

first byte in segment’s rwnd

• seq # of next byte

• A: TCP spec doesn’t say, incoming segment to sender

Transport Layer 3-22

simple telnet scenario

Transport Layer 3-23

Transport Layer 3-24

RTT: gaia.cs.umass.edu to fantasia.eurecom.fr

TimeoutInterval = EstimatedRTT + 4*DevRTT

estimated RTT “safety margin”

* Check out the online interactive exercises for more

Transport Layer 3-27

Transport Layer 3-28

Transport Layer 3-29

Seq=100, 20 bytes of data

Seq=92, 8 bytes of data Seq=92, 8

lost ACK scenario premature timeout

Seq=92, 8 bytes of data

Seq=100, 20 bytes of data

Seq=120, 15 bytes of data

event at receiver TCP receiver action

arrival of in-order segment with immediately send single cumulative

arrival of out-of-order segment immediately send duplicate ACK,

arrival of segment that immediate send ACK, provided that

Transport Layer 3-33

Transport Layer 3-34

Seq=92, 8 bytes of data