Chapter 3

Transport Layer
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Computer Networking: A
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Top-Down
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Transport Layer: 3-1
Transport layer: roadmap
▪ Transport-layer services
▪ Multiplexing and demultiplexing
▪ Connectionless transport: UDP
▪ Principles of reliable data transfer
▪ Connection-oriented transport: TCP
▪ Principles of congestion control
▪ TCP congestion control

Transport Layer: 3-2

Transport services and protocols
application
transport

▪ provide logical communication

network
mobile network
data link
physical

between application processes national or global ISP

log
running on different hosts

ica
l en
▪ transport protocols actions in end

d-e
systems:

nd
local or

tra
• sender: breaks application messages regional ISP

nsp
into segments, passes to network layer

ort
home network content
• receiver: reassembles segments into provider
network datacenter
messages, passes to application layer application
transport
network
network

▪ two transport protocols available to data link

physical

Internet applications enterprise

network
• TCP, UDP
Transport Layer: 3-3
Transport vs. network layer services and protocols
household analogy:
12 kids in Ann’s house sending
letters to 12 kids in Bill’s
house:
▪ hosts = houses
▪ processes = kids
▪ app messages = letters in
envelopes
▪ transport protocol = Ann and Bill
who demux to in-house siblings
▪ network-layer protocol = postal
service

Transport Layer: 3-4

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

Transport Layer: 3-5

Transport Layer Actions

Sender:
application ▪ is passed an application
app. msg
application-layer message
transport ▪ determines segment TThhtransport
app. msg
header fields values
network (IP) ▪ creates segment network (IP)

link ▪ passes segment to IP link

physical physical

Transport Layer: 3-6

Transport Layer Actions

Receiver:
application ▪ receives segment from IP application
▪ checks header values
transport
transport
app. msg ▪ extracts application-layer
message
network (IP)
network (IP) ▪ demultiplexes message up
link to application via socket link

physical physical
Th app. msg

Transport Layer: 3-7

Two principal Internet transport
protocols application

▪ TCP: Transmission Control Protocol

transport
network
mobile network
data link
physical
national or global ISP
• reliable, in-order delivery

log
• congestion control

ica
l en
• flow control

d-e
• connection setup

nd
▪ UDP: User Datagram Protocol
local or

tra
regional ISP

nsp
• unreliable, unordered delivery

ort
home network content
provider
• no-frills extension of “best-effort” IP network datacenter
application

▪services not available:

network
transport
network
data link

• delay guarantees physical

• bandwidth guarantees enterprise

network

Transport Layer: 3-8

Chapter 3: roadmap
▪ Transport-layer services
▪ Multiplexing and demultiplexing
▪ Connectionless transport: UDP
▪ Principles of reliable data transfer
▪ Connection-oriented transport: TCP
▪ Principles of congestion control
▪ TCP congestion control
▪ Evolution of transport-layer
functionality
Transport Layer: 3-9
Why Multiplexing & demultiplexing……

application application
HTTP msg
transport

transport network transport

network link network
link physical link
physical physical

client
HTTP server Transport Layer: 3-10
 HTTP server
client
application application
HTTP msg
transport
Ht HTTP msg

transport network transport

network link network
link physical link
physical physical

Transport Layer: 3-11

 HTTP server
client
application application
HTTP msg
transport
Ht HTTP msg

transport Hnnetwork
Ht HTTP msg transport
network link network
link physical link
physical physical

Transport Layer: 3-12

 HTTP server
client
application application

transport

transport network transport

network link network
link physical link
physical physical

Hn Ht HTTP msg

Transport Layer: 3-13

 HTTP server
client1 client2
P-clie P-clie
application nt1 nt2
application

transport

transport network transport

network link network
link physical link
physical physical

Transport Layer: 3-14

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

application

application P1 P2 application socket

P3 transport P4
process
transport network transport
network link network
link physical link
physical physical

Transport Layer: 3-15

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

Transport Layer: 3-16

Connectionless demultiplexing
Recall: when receiving host receives
UDP segment:
▪ when creating socket, must • checks destination port # in
specify host-local port #: segment
DatagramSocket mySocket1 • directs UDP segment to
= new socket with that port #
DatagramSocket(12534);
▪ when creating datagram to
send into UDP socket, must
specify IP/UDP datagrams with same dest.
port #, but different source IP
• destination IP address addresses and/or source port
• destination port # numbers will be directed to same
socket at receiving host
Transport Layer: 3-17
 Connectionless demultiplexing: an example
DatagramSocket
serverSocket = new
DatagramSocket
DatagramSocket mySocket2 = DatagramSocket mySocket1 =
new DatagramSocket (6428); new DatagramSocket (5775);
(9157); application
application application
P1
P3 P4
transport
transport transport
network
network link 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: 3-18
Connection-oriented demultiplexing
▪ TCP socket identified by ▪ server may support many
4-tuple: simultaneous TCP sockets:
• source IP address • each socket identified by its
• source port number own 4-tuple
• dest IP address • each socket associated with
• dest port number a different connecting client
▪ demux: receiver uses all
four values (4-tuple) to
direct segment to
appropriate socket
Transport Layer: 3-19
Connection-oriented demultiplexing: example
application
application P4 P5 P6 application
P1 P2 P3
transport
transport transport
network
network link network
link physical link
physical physical
server: IP
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: 3-20
Summary
▪ Multiplexing, demultiplexing: based on segment, datagram
header field values
▪ UDP: demultiplexing using destination port number (only)
▪ TCP: demultiplexing using 4-tuple: source and destination IP
addresses, and port numbers
▪ Multiplexing/demultiplexing happen at all layers

Transport Layer: 3-21

Chapter 3: roadmap
▪ Transport-layer services
▪ Multiplexing and demultiplexing
▪ Connectionless transport: UDP
▪ Principles of reliable data transfer
▪ Connection-oriented transport: TCP
▪ Principles of congestion control
▪ TCP congestion control
▪ Evolution of transport-layer
functionality
Transport Layer: 3-22
UDP vs TCP

Transport Layer: 3-23

UDP: User Datagram Protocol
Why is there a UDP?
▪ “no frills,” “bare bones” ▪ no connection
Internet transport protocol establishment (which can
add RTT delay)
▪ “best effort” service, UDP
▪ simple: no connection state
segments may be: at sender, receiver [Do not
• lost handle Buffer size, congestion control
parameters, Seq and Ack numbers]
• delivered out-of-order to app
▪ small header size
▪ connectionless: ▪ no congestion control
• no handshaking between UDP ▪ UDP can blast away as fast as
sender, receiver desired!
• each UDP segment handled ▪ can function in the face of
independently of others congestion
Transport Layer: 3-24
UDP: User Datagram Protocol
▪ UDP use:
▪ streaming multimedia apps (loss tolerant, rate sensitive)
▪ DNS
▪ SNMP
▪ HTTP/3
▪ if reliable transfer needed over UDP (e.g., HTTP/3):
▪ add needed reliability at application layer
▪ add congestion control at application layer

Transport Layer: 3-25

UDP: User Datagram Protocol [RFC 768]

Transport Layer: 3-26

UDP segment header
32 bits
source port # dest port #
length checksum

application length, in bytes of

data UDP segment,
(payload) including header

data to/from
UDP segment format application layer

Transport Layer: 3-27

UDP checksum
Goal: detect errors (i.e., flipped bits) in transmitted segment
1st number 2nd number sum

Transmitted: 5 6 11

Received: 4 6 11

receiver-computed sender-computed
checksum
= checksum (as received)

Transport Layer: 3-28

UDP checksum
Goal: detect errors (i.e., flipped bits) in transmitted segment
sender: receiver:
▪ treat contents of UDP ▪ compute checksum of received
segment (including UDP header segment
fields and IP addresses) as
sequence of 16-bit integers ▪ check the checksum value
▪ checksum: addition (one’s calculated by receiver:
complement sum) of segment • checksum of receiver 🡪 all zero bits -
content no error
▪ checksum value put into • checksum of receiver 🡪 any bit
non-zero – error present
UDP checksum field
Transport Layer: 3-29
Internet checksum: an example
example: add two 16-bit integers
1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0
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 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0
checksum 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: 3-30
Internet checksum: weak protection!
example: add two 16-bit integers
0 1
1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0 1 0
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 Even though
numbers have
sum 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0 changed (bit
flips), no change
checksum 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1 in checksum!

Transport Layer: 3-31

Summary: UDP
▪ “no frills” protocol:
• segments may be lost, delivered out of order
• best effort service: “send and hope for the best”
▪ UDP has its plusses:
• no setup/handshaking needed (no RTT incurred)
• can function when network service is compromised
• helps with reliability (checksum)
▪ build additional functionality on top of UDP in application layer
(e.g., HTTP/3)
Chapter 3: roadmap
▪ Transport-layer services
▪ Multiplexing and demultiplexing
▪ Connectionless transport: UDP
▪ Principles of reliable data transfer
▪ Connection-oriented transport: TCP
▪ Principles of congestion control
▪ TCP congestion control
▪ Evolution of transport-layer
functionality
Transport Layer: 3-33
 Principles of reliable data transfer

sending receiving
process process
applicatio data data
ntranspor
t reliable
channel
reliable service abstraction

Transport Layer: 3-34

 Principles of reliable data transfer

sending receiving sending receiving

process process process process
applicatio data data application data data
ntranspor transport
t reliable
channel
sender-side of receiver-side
reliable service abstraction reliable data of reliable data
transfer protocol transfer protocol

transport
network
unreliable channel

reliable service implementation

Transport Layer: 3-35

Principles of reliable data transfer

sending receiving
process process
application data data
transport

sender-side of receiver-side
Complexity of reliable data reliable data
transfer protocol
of reliable data
transfer protocol
transfer protocol will depend
(strongly) on characteristics of transport
network
unreliable channel (lose, unreliable channel
corrupt, reorder data?)
reliable service implementation

Transport Layer: 3-36

Principles of reliable data transfer

sending receiving
process process
application data data
transport

sender-side of receiver-side
reliable data of reliable data
Sender, receiver do not know transfer protocol transfer protocol
the “state” of each other, e.g.,
was a message received? transport

▪ unless communicated via a

network
unreliable channel

message
reliable service implementation

Transport Layer: 3-37

 Reliable data transfer protocol (rdt): interfaces
rdt_send(): called from above, deliver_data(): called by rdt
(e.g., by app.). Passed data to to deliver data to upper layer
deliver to receiver upper layer
sending receiving
process process
rdt_send() data data
deliver_data()

sender-side data receiver-side

implementation of implementation of
rdt reliable data packet rdt reliable data
transfer protocol transfer protocol

udt_send() Header data Header data rdt_rcv()

unreliable channel
udt_send(): called by rdt rdt_rcv(): called when packet
to transfer packet over arrives on receiver side of
Bi-directional communication over
unreliable channel to receiver unreliable channel channel
Transport Layer: 3-38
Reliable data transfer: getting started
We will:
▪ incrementally develop sender, receiver sides of reliable data transfer
protocol (rdt)
▪ consider only unidirectional data transfer
• but control info will flow in both directions!
▪ use finite state machines (FSM) to specify sender, receiver
event causing state transition
actions taken on state transition
state: when in this “state”
next state uniquely state state
determined by next 1 event
event
2
actions

Transport Layer: 3-39

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 reads data from underlying channel

Wait for rdt_send(data) Wait for rdt_rcv(packet)

sender call from packet = make_pkt(data) receiver call from extract (packet,data)
above udt_send(packet) below deliver_data(data)

Transport Layer: 3-40

rdt2.0: channel with bit errors
▪ underlying channel may flip bits in packet
• checksum (e.g., Internet checksum) to detect bit errors
▪ the question: how to recover from errors?

How do humans recover from “errors” during conversation?

Transport Layer: 3-41

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

stop and wait

sender sends one packet, then waits for receiver response
Transport Layer: 3-42
rdt2.0: FSM specifications
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
Wait for Wait for isNAK(rcvpkt)
sender call from ACK or udt_send(sndpkt) rdt_rcv(rcvpkt) && corrupt(rcvpkt)
above NAK
udt_send(NAK)

rdt_rcv(rcvpkt) && isACK(rcvpkt)

Wait for
Λ call from receiver
below

rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)

extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)

Transport Layer: 3-43

rdt2.0: FSM specification
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
Wait for Wait for isNAK(rcvpkt)
sender call from ACK or udt_send(sndpkt) rdt_rcv(rcvpkt) && corrupt(rcvpkt)
above NAK
udt_send(NAK)

rdt_rcv(rcvpkt) && isACK(rcvpkt)

Wait for
Λ call from receiver
below

Note: “state” of receiver (did the receiver get my

rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
message correctly?) isn’t known to sender unless
extract(rcvpkt,data)
somehow communicated from receiver to sender deliver_data(data)
▪ that’s why we need a protocol! udt_send(ACK)

Transport Layer: 3-44

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

rdt_rcv(rcvpkt) && isACK(rcvpkt)

Wait for
Λ call from receiver
below

rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)

extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)

Transport Layer: 3-45

rdt2.0: corrupted packet scenario
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
Wait for Wait for isNAK(rcvpkt)
sender call from ACK or udt_send(sndpkt) rdt_rcv(rcvpkt) && corrupt(rcvpkt)
above NAK
udt_send(NAK)

rdt_rcv(rcvpkt) && isACK(rcvpkt)

Wait for
Λ call from receiver
below

rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)

extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)

Transport Layer: 3-46

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

stop and wait

sender sends one packet, then
waits for receiver response
Transport Layer: 3-47
(EXTRA for interest)
rdt2.1: sender, handling garbled ACK/NAKs
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt) rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt) ||
Wait for Wait for isNAK(rcvpkt) )
call 0 from ACK or
NAK 0 udt_send(sndpkt)
above
rdt_rcv(rcvpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) &&
&& notcorrupt(rcvpkt)
isACK(rcvpkt)
&& isACK(rcvpkt)
Λ
Λ
Wait for Wait for
ACK or call 1 from
rdt_rcv(rcvpkt) NAK 1 above
&& (corrupt(rcvpkt) ||
isNAK(rcvpkt) ) rdt_send(data)

udt_send(sndpkt) sndpkt = make_pkt(1, data, checksum)

udt_send(sndpkt)

Transport Layer: 3-48

(EXTRA for interest)
rdt2.1: receiver, handling garbled ACK/NAKs
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq0(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && (corrupt(rcvpkt) rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(NAK, chksum) sndpkt = make_pkt(NAK, chksum)
udt_send(sndpkt) udt_send(sndpkt)
Wait for Wait for
rdt_rcv(rcvpkt) && 0 from 1 from rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) && below below not corrupt(rcvpkt) &&
has_seq1(rcvpkt) has_seq0(rcvpkt)
sndpkt = make_pkt(ACK, chksum) sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt) udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)

extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)

Transport Layer: 3-49

rdt2.1: discussion
sender: receiver:
▪ seq # added to pkt ▪ must check if received packet
▪ two seq. #s (0,1) will suffice. is duplicate
Why? • state indicates whether 0 or 1 is
expected pkt seq #
▪ must check if received ACK/NAK
corrupted ▪ note: receiver can not know if
its last ACK/NAK received OK
▪ twice as many states at sender
• state must “remember” whether
“expected” pkt should have seq #
of 0 or 1

Transport Layer: 3-50

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

As we will see, TCP uses this approach to be NAK-free

Transport Layer: 3-51

(EXTRA for interest)
rdt2.2: sender, receiver fragments
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt) rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
Wait for Wait for
ACK isACK(rcvpkt,1) )
call 0 from
above 0 udt_send(sndpkt)
sender FSM
fragment rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && && isACK(rcvpkt,0)
(corrupt(rcvpkt) || Λ
has_seq1(rcvpkt)) Wait for receiver FSM
0 from
udt_send(sndpkt) below fragment
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK1, chksum)
udt_send(sndpkt) Transport Layer: 3-52
rdt3.0: channels with errors and loss
New channel assumption: underlying channel can also lose
packets (data, ACKs)
• checksum, sequence #s, ACKs, retransmissions will be of help …
but not quite enough

Q: How do humans handle lost

sender-to-receiver words in
conversation?
Transport Layer: 3-53
rdt3.0: channels with errors and loss
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 seq #s already handles this!
• receiver must specify seq # of packet being ACKed
▪ use countdown timer to interrupt after “reasonable” amount
of time
timeout

Transport Layer: 3-54

(EXTRA for interest)
rdt3.0 sender
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
start_timer

Wait for Wait

call 0 from for
above ACK0
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) rdt_rcv(rcvpkt)
&& isACK(rcvpkt,1) && notcorrupt(rcvpkt)
stop_timer && isACK(rcvpkt,0)
stop_timer
Wait Wait for
for call 1 from
ACK1 above

rdt_send(data)
sndpkt = make_pkt(1, data, checksum)
udt_send(sndpkt)
start_timer

Transport Layer: 3-55

(EXTRA for interest)
rdt3.0 sender
rdt_send(data)
rdt_rcv(rcvpkt) &&
sndpkt = make_pkt(0, data, checksum) ( corrupt(rcvpkt) ||
udt_send(sndpkt) isACK(rcvpkt,1) )
rdt_rcv(rcvpkt) start_timer Λ
Λ Wait for Wait
for timeout
call 0 from
ACK0 udt_send(sndpkt)
above
start_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) rdt_rcv(rcvpkt)
&& isACK(rcvpkt,1) && notcorrupt(rcvpkt)
stop_timer && isACK(rcvpkt,0)
stop_timer
Wait Wait for
timeout for call 1 from
udt_send(sndpkt) ACK1 above
start_timer rdt_rcv(rcvpkt)
rdt_send(data) Λ
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) || sndpkt = make_pkt(1, data, checksum)
isACK(rcvpkt,0) ) udt_send(sndpkt)
start_timer
Λ

Transport Layer: 3-56

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
loss
ack1 send ack1
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

Transport Layer: 3-57
rdt3.0 in action
sender receiver
sender receiver send pkt0
pkt0
rcv pkt0
send pkt0 pkt0 send ack0
ack0
rcv pkt0 rcv ack0
ack0 send ack0 send pkt1 pkt1
rcv ack0 rcv pkt1
send pkt1 pkt1 send ack1
rcv pkt1 ack1
ack1 send ack1
X timeout
loss resend pkt1
pkt1 rcv pkt1
timeout
resend pkt1 pkt1
rcv pkt1 rcv ack1 (detect duplicate)
send pkt0 pkt0 send ack1
(detect duplicate)
ack1 send ack1 ack1 rcv pkt0
rcv ack1 rcv ack1 send ack0
send pkt0 pkt0 (ignore) ack0
rcv pkt0
ack0 send ack0 pkt1

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

Transport Layer: 3-58
rdt3.0: stop-and-wait operation
sender receiver

L/R L/R
Usender=
RTT + L / R
.008 RTT
=
30.008
= 0.00027

▪ rdt 3.0 protocol performance stinks!

▪ Protocol limits performance of underlying infrastructure (channel)

Transport Layer: 3-59

rdt3.0: pipelined protocols operation
pipelining: sender allows multiple, “in-flight”, yet-to-be-acknowledged
packets
• range of sequence numbers must be increased
• buffering at sender and/or receiver

Transport Layer: 3-60

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!

Transport Layer: 3-61

Go-Back-N: sender
▪ sender: “window” of up to N, consecutive transmitted but unACKed pkts
• k-bit seq # in pkt header

▪ cumulative ACK: ACK(n): ACKs all packets up to, including seq # n

• on receiving ACK(n): move window forward to begin at n+1
▪ timer for oldest in-flight packet
▪ timeout(n): retransmit packet n and all higher seq # packets in window
Transport Layer: 3-62
Go-Back-N: receiver
▪ ACK-only: always send ACK for correctly-received packet so far, with
highest in-order seq #
• may generate duplicate ACKs
• need only remember rcv_base
▪ on receipt of out-of-order packet:
• can discard (don’t buffer) or buffer: an implementation decision
• re-ACK pkt with highest in-order seq #

Receiver view of sequence number space:

received and ACKed

… … Out-of-order: received but not ACKed

rcv_base
Not received
Transport Layer: 3-63
Go-Back-N 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

Transport Layer: 3-64

Selective repeat
▪A single packet error can cause GBN to retransmit a large
number of packets when the packet size is very large.
▪Receiver individually acknowledges all correctly received
packets
• Buffers packets, as needed, for eventual in-order delivery to the upper
layer
▪Sender times-out/retransmits individually for unACKed packets
• Sender maintains a timer for each unACKed pkt
▪sender window
• N consecutive seq #s
• limits seq #s of sent, unACKed packets

Transport Layer: 3-65

Selective repeat: sender, receiver windows

Transport Layer: 3-66

Selective Repeat in action
sender window (N=4) sender receiver
012345678 send pkt0
012345678 send pkt1
012345678 send pkt2 receive pkt0, send ack0
012345678 send pkt3 Xloss receive pkt1, send ack1
(wait)
receive pkt3, buffer,
012345678 rcv ack0, send pkt4 send ack3
012345678 rcv ack1, send pkt5
receive pkt4, buffer,
record ack3 arrived send ack4
receive pkt5, buffer,
pkt 2 timeout send ack5
012345678 send pkt2
012345678 (but not 3,4,5)
012345678 rcv pkt2; deliver pkt2,
012345678 pkt3, pkt4, pkt5; send ack2

Q: what happens when ack2 arrives?

Transport Layer: 3-67

3-68
Transport Layer
sender window receiver window
(after receipt) (after receipt)
Selective repeat: dilemma 0123012 pkt0
0123012 pkt1 0123012
0123012 pkt2 0123012
example: pkt3
0123012
0123012
▪ seq #’s: 0, 1, 2, 3 0123012
X

▪ window size=3
pkt0 will accept packet
with seq number 0
▪ receiver sees no difference in two (a) no problem

scenarios! receiver can’t see sender side.

▪ duplicate data accepted as new in receiver behavior identical in both cases!
something’s (very) wrong!
(b)
0123012 pkt0
pkt1 0123012
Q: what relationship between seq # 0123012
0123012 pkt2 0123012
size and window size to avoid X 0123012
problem in (b)? X
timeout
retransmit pkt0 X
0123012 pkt0
will accept packet
with seq number 0
less than or equal to the half of the sequence (b) oops!
number
Chapter 3: roadmap
▪ Transport-layer services
▪ Multiplexing and demultiplexing
▪ Connectionless transport: UDP
▪ Principles of reliable data transfer
▪ Connection-oriented transport: TCP
• segment structure
• reliable data transfer
• flow control
• connection management
▪ Principles of congestion control
▪ TCP congestion control
Transport Layer: 3-69
TCP: overview RFCs: 793,1122, 2018, 5681, 7323
▪ point-to-point: ▪ cumulative ACKs
• one sender, one receiver
▪ pipelining:
▪ reliable, in-order byte steam:
• no “message boundaries"
• TCP congestion and flow control
set window size
▪ full duplex data:
• bi-directional data flow in same ▪ connection-oriented:
connection • handshaking (exchange of control
• MSS: maximum segment size messages) initializes sender,
MSS: maximum amount of receiver state before data
application layer data in the exchange
segment.
▪ flow controlled:
• sender will not overwhelm receiver
Transport Layer: 3-70
TCP Seq numbers and Acks
▪ Suppose Host A wants to send a stream of data to a process in Host B over
a TCP connection. Assume that, the data stream consists of a file
consisting of 500,000 bytes, and that the MSS is 1,000 bytes. Then the
segment looks like this:

Transport Layer: 3-71

 TCP segment structure
32 bits

source port # dest port # segment seq #: counting

ACK: seq # of next expected sequence number bytes of data into bytestream
byte; A bit: this is an ACK (not segments!)
acknowledgement number
head not
length (of TCP header) len used C EUAP R SF receive window flow control: # bytes
Internet checksum checksum Urg data pointer receiver willing to accept
options (variable length)
C, E: congestion notification
TCP options
application data sent by
RST, SYN, FIN: connection data application into
management (variable length) TCP socket

Transport Layer: 3-72

 TCP sequence numbers, ACKs
Sequence numbers: outgoing segment from sender
source port # dest port #
• byte stream “number” of first sequence number
byte in segment’s data acknowledgement number
rwnd
Acknowledgements: checksum urg pointer

• seq # of next byte expected from window size

N
other side
• cumulative ACK
sender sequence number space
Q: how the receiver handles
sent sent, usable not
out-of-order segments ACKed not-yet but not usable
ACKed yet sent
• A: Discards out-of-order segments (“in-flight”)
• B: Keeps the out-of-order bytes in the outgoing segment from receiver
buffer source port # dest port #
sequence number
• C: TCP spec doesn’t say, - up to the acknowledgement number
implementor A
checksum
rwnd
urg pointer
Transport Layer: 3-73
TCP sequence numbers, ACKs (Some Scenarios)
▪ Host A has received all data from 0 to 535 and Host A is expecting data
536 and all subsequent byte streams from B.
▪ Host A received one segment from 0 through 535 and another segment
from 900 to 1000.
• It has not received any segment from 536 to 899. Therefore, there has a gap
• Cumulative acknowledgments.
▪ Host A received the segment from 900 to 1000 before receiving bytes
536 to 899. Therefore, out of order.

Transport Layer: 3-74

TCP sequence numbers, ACKs
Host A Host B

User types‘C’
Seq=42, ACK=79, data = ‘C’
host ACKs receipt of‘C’,
echoes back ‘C’
Seq=79, ACK=43, data = ‘C’
host ACKs receipt
of echoed ‘C’
Seq=43, ACK=80

simple telnet scenario

Transport Layer: 3-75
 TCP round trip time, timeout
Q: how to set TCP timeout Q: how to estimate RTT?
value? ▪ SampleRTT:measured time
▪ longer than RTT, but RTT varies! from segment transmission until
ACK receipt
▪ too short: premature timeout,
• ignore retransmissions
unnecessary retransmissions
▪ SampleRTT will vary, want
▪ too long: slow reaction to estimated RTT “smoother”
segment loss • average several recent
measurements, not just current
SampleRTT
SampleRTT values will fluctuate from segment to segment due to congestion
and load on the end systems.
Transport Layer: 3-76
TCP round trip time, timeout
EstimatedRTT = (1- α)*EstimatedRTT + α*SampleRTT
▪ exponential weighted moving average (EWMA)
▪ influence of past sample decreases exponentially fast
▪ typical value: α = 0.125
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr

(milliseconds)
RTT

sampleRTT
EstimatedRTT

time (seconds)
Transport Layer: 3-77
TCP round trip time, timeout
▪ DevRTT: EWMA of SampleRTT deviation from EstimatedRTT:
DevRTT = (1-β)*DevRTT + β*|SampleRTT-EstimatedRTT|
(typically, β = 0.25)

▪ timeout interval: EstimatedRTT plus “safety margin”

• large variation in EstimatedRTT: want a larger safety margin
TimeoutInterval = EstimatedRTT + 4*DevRTT

estimated RTT “safety margin”

* Check out the online interactive exercises for more examples: http://gaia.cs.umass.edu/kurose_ross/interactive/
Transport Layer: 3-78
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 send cumulative
SendBase=120 ACK for 120
ACK=100
ACK=120

SendBase=120

lost ACK scenario premature timeout

Transport Layer: 3-79

TCP: retransmission scenarios
Host A Host B

Seq=92, 8 bytes of data

Seq=100, 20 bytes of data

ACK=100
X
ACK=120

Seq=120, 15 bytes of data

cumulative ACK covers

for earlier lost ACK

Transport Layer: 3-80

TCP fast retransmit
Host A Host B
TCP fast retransmit
if sender receives 3 additional
Seq=92
ACKs for same data (“triple Seq=1
, 8 byte
s of da
ta
00, 20
duplicate ACKs”), resend unACKed bytes
of data
segment with smallest seq # X
▪ likely that unACKed segment lost,
=100
so don’t wait for timeout ACK

timeout
C K =100
A
=100
ACK
C K =100
A
Receipt of three duplicate ACKs
Seq=100, 20 bytes of data
indicates 3 segments received
after a missing segment – lost
segment is likely. So retransmit!
Transport Layer: 3-81
Chapter 3: roadmap
▪ Transport-layer services
▪ Multiplexing and demultiplexing
▪ Connectionless transport: UDP
▪ Principles of reliable data transfer
▪ Connection-oriented transport: TCP
• segment structure
• reliable data transfer
• flow control
• connection management
▪ Principles of congestion control
▪ TCP congestion control
Transport Layer: 3-82
TCP flow control
applicati
on
Q: What happens if network Application removing process
layer delivers data faster than data from TCP socket
buffers
application layer removes TCP socket
data from socket buffers? receiver buffers

TCP
code
Network layer
delivering IP datagram
payload into TCP
IP
socket buffers code

from sender

receiver protocol stack

Transport Layer: 3-83

TCP flow control
applicati
on
Q: What happens if network Application removing process
layer delivers data faster than data from TCP socket
buffers
application layer removes TCP socket
data from socket buffers? receiver buffers

TCP
code
Network layer
delivering IP datagram
payload into TCP
IP
socket buffers code

from sender

receiver protocol stack

Transport Layer: 3-84

TCP flow control
applicati
on
Q: What happens if network Application removing process
layer delivers data faster than data from TCP socket
buffers
application layer removes TCP socket
data from socket buffers? receiver buffers

TCP
code

receive window
flow control: # bytes
receiver willing to accept IP
code

from sender

receiver protocol stack

Transport Layer: 3-85

TCP flow control
applicati
on
Q: What happens if network Application removing process
layer delivers data faster than data from TCP socket
buffers
application layer removes TCP socket
data from socket buffers? receiver buffers

TCP
flow control code

receiver controls sender, so

sender won’t overflow IP
code
receiver’s buffer by
transmitting too much, too fast
from sender

receiver protocol stack

Transport Layer: 3-86

TCP flow control
▪ TCP receiver “advertises” free buffer
space in rwnd field in TCP header to application process
• RcvBuffer size set via socket
options (typical default is 4096 bytes) RcvBuffer buffered data
• many operating systems autoadjust
RcvBuffer
rwnd free buffer space

▪ sender limits amount of unACKed

(“in-flight”) data to received rwnd TCP segment payloads

▪ guarantees receive buffer will not TCP receiver-side buffering

overflow
LastByteSent – LastByteAcked <= rwnd

Transport Layer: 3-87

3-88
Transport Layer

TCP flow control

▪ LastByteRead: data stream read from the buffer
▪ LastByteRcvd: the data stream that has arrived
from the network
▪ LastByteRcvd – LastByteRead <= RcvBuffer
▪A
▪ Initially rwnd = RcvBuffer
▪A
▪ Host A to continue to send segments with one
data byte when B’s receive window is zero. These
segments will be acknowledged by the receiver
TCP connection management
before exchanging data, sender/receiver “handshake”:
▪ agree to establish connection (each knowing the other willing to establish connection)
▪ agree on connection parameters (e.g., starting seq #s)

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 number"); welcomeSocket.accept();
Transport Layer: 3-89
 TCP 3-way handshake
Server state
serverSocket = socket(AF_INET,SOCK_STREAM)
Client state serverSocket.bind((‘’,serverPort))
serverSocket.listen(1)
clientSocket = socket(AF_INET, SOCK_STREAM) connectionSocket, addr = serverSocket.accept()
LISTEN
clientSocket.connect((serverName,serverPort)) 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

Transport Layer: 3-90

A human 3-way handshake protocol

1. On belay?

2. Belay on.
3. Climbing.

Transport Layer: 3-91

TCP 3-way handshake: FSM

closed

Socket connectionSocket =
welcomeSocket.accept();

Λ
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)
Λ

Transport Layer 3-92

Closing a TCP 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

Transport Layer: 3-93

TCP: closing a connection
client state server state
ESTAB ESTAB
clientSocket.close(
) FINbit=1, seq=x
FIN_WAIT_1 can no longer
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

Transport Layer 3-94

TCP: closing a connection

Fig: A typical sequence of TCP states visited

by a client TCP

Fig: A typical sequence of TCP states visited

by a server TCP

Transport Layer 3-95

Chapter 3: roadmap
▪ Transport-layer services
▪ Multiplexing and demultiplexing
▪ Connectionless transport: UDP
▪ Principles of reliable data transfer
▪ Connection-oriented transport: TCP
▪ Principles of congestion control
▪ TCP congestion control
▪ Evolution of transport-layer
functionality
Transport Layer: 3-96
TCP congestion control: AIMD
▪ approach: senders can increase sending rate until packet loss
(congestion) occurs, then decrease sending rate on loss event
Additive Increase Multiplicative Decrease
increase sending rate by 1 cut sending rate in half at each
maximum segment size every loss event
RTT until loss detected
TCP sender Sending rate

AIMD sawtooth
behavior: probing
for bandwidth

time Transport Layer: 3-97

TCP AIMD: more
Multiplicative decrease detail: sending rate is
▪ Cut in half on loss detected by triple duplicate ACK (TCP Reno)
▪ Cut to 1 MSS (maximum segment size) when loss detected by
timeout (TCP Tahoe)

Why AIMD?
▪ AIMD – a distributed, asynchronous algorithm – has been
shown to:
• optimize congested flow rates network wide!
• have desirable stability properties

Transport Layer: 3-98

TCP congestion control: details
sender sequence number space
cwnd TCP sending behavior:
▪ roughly: send cwnd bytes,
wait RTT for ACKS, then
send more bytes
last byte
available but cwnd
ACKed sent, but TCP rate ~
~ bytes/sec
not-yet ACKed not used RTT
(“in-flight”) last byte sent

▪ TCP sender limits transmission: LastByteSent- LastByteAcked < cwnd

▪ cwnd is dynamically adjusted in response to observed

network congestion (implementing TCP congestion control)
Transport Layer: 3-99
TCP Slow Start
Host A Host B
▪ when connection begins,
increase rate exponentially
one segm
until first loss event: ent

RTT
• initially cwnd = 1 MSS two segm
ents
• double cwnd every RTT
• done by incrementing cwnd
four segm
for every ACK received ents

▪ summary: initial rate is

slow, but ramps up
time
exponentially fast
Transport Layer: 3-100
TCP Slow Start

▪loss indicated by timeout:

• cwnd set to 1 MSS;
• window then grows exponentially (as in slow start) to threshold,
then grows linearly
• Set the threshold value ssthresh is equal to cwnd/2
▪When the value of cwnd >= ssthresh, Slow Start ends and
Congestion Avoidance (CA) starts.
▪loss indicated by 3 duplicate ACKs: TCP enters in the fast
recovery mode.
Transport Layer: 3-101
TCP: Congestion Avoidance (CA)
▪ Rather than doubling the cwnd value, cwnd is increased by just a
single MSS every RTT.
▪ TCP sender increase cwnd by MSS bytes (MSS/cwnd)
▪ When the congestion avoidance ends?
▪ Depends on the timeout events and triple duplicates
▪ dup ACKs indicate network capable of delivering some segments
▪Fast Recovery: 3 dup ACKs
▪ TCP Tahoe always sets cwnd to 1 then grows exponentially (timeout or 3
duplicate acks) [Earlier Style]
▪ TCP Reno cut the cwnd in half window then grows linearly [New
Version]

Transport Layer: 3-102

TCP: from slow start to congestion
avoidance
Q: when should the exponential
increase switch to linear?
X
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

* Check out the online interactive exercises for more examples: http://gaia.cs.umass.edu/kurose_ross/interactive/
Transport Layer: 3-103
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
Λ transmit new segment(s), as allowed
cwnd = 1 MSS
ssthresh = 64 KB cwnd > ssthresh
dupACKcount = 0 slow Λ 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
dupACKcount == 3 cwnd = ssthresh 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

Transport Layer: 3-104

 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 + 3.

❑ When timeout occurs, Threshold set to CongWin/2 and

CongWin is set to 1 MSS.

Transport Layer: 3-105

3-106
Transport Layer

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 + 3.

❑ When timeout occurs, Threshold set to CongWin/2 and

CongWin is set to 1 MSS.
3-107
Transport Layer

Summary: TCP Congestion Control

TCP sender congestion control
TCP CUBIC
▪ Is there a better way than AIMD to “probe” for usable bandwidth?
▪ Insight/intuition:
• Wmax: sending rate at which congestion loss was detected
• congestion state of bottleneck link probably (?) hasn’t changed much
• after cutting rate/window in half on loss, initially ramp to to Wmax faster, but then
approach Wmax more slowly

Wmax classic TCP

TCP CUBIC - higher

Wmax/2 throughput in this
example

Transport Layer: 3-108

TCP CUBIC
▪ K: point in time when TCP window size will reach Wmax
• K itself is tunable
▪ increase W as a function of the cube of the distance between current
time and K
• larger increases when further away from K
• smaller increases (cautious) when nearer K
▪ TCP CUBIC default Wmax
in Linux, most
TCP Reno
popular TCP for TCP CUBIC
popular Web TCP
sending
servers rate

time
t0 t1 t2 t3 t4
Transport Layer: 3-109
TCP CUBIC
TCP Cubic is a congestion control algorithm that adjusts the congestion window (cwnd)
using a cubic function to optimize performance over long-distance high-bandwidth
networks. The congestion window (cwnd) is adjusted according to the following cubic
function:

Transport Layer: 3-110

TCP CUBIC
Calculate the congestion Determine the time at which
window size (cwnd) at t=8 the congestion window size
seconds after the last (cwnd) reaches 50.
congestion event.

Transport Layer: 3-111

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

Transport Layer: 3-112