Chapter 3

Transport
Layer
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Computer Networking: A
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Top-Down Approach
Thanks and enjoy! JFK/KWR 8th edition
Jim Kurose, Keith Ross
All material copyright 1996-2020
J.F Kurose and K.W. Ross, All Rights Reserved Pearson, 2020
Transport Layer: 3-1
Transport layer: overview
Our goal:
 understand principles  learn about Internet transport
behind transport layer layer protocols:
services: • UDP: connectionless transport
• multiplexing, • TCP: connection-oriented reliable
demultiplexing transport
• reliable data transfer • TCP congestion control
• flow control
• congestion control

Transport Layer: 3-2

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
 Evolution of transport-layer
functionality
Transport Layer: 3-3
Transport services and protocols
application
transport

 provide logical communication mobile

network
network
data link
physical

between application processes national or global ISP

running on different hosts

log
ica
 transport protocols actions in end

le
n d-
systems:

e 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
messages, passes to application layer application
transport
datacenter
network
network
 two transport protocols available to data link
physical

Internet applications enterprise

network
• TCP, UDP
Transport Layer: 3-4
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: 3-5
Transport vs. network layer services and protocols

 network layer: logical household analogy:

communication between 12 kids in Ann’s house sending
hosts letters to 12 kids in Bill’s house:
 hosts = houses
 transport layer: logical
 processes = kids
communication between  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: 3-6
Transport Layer Actions

Sender:
application
 is passed an application- app. msg
application
layer message
 determines segment TThtransport
app. msg
transport h

header fields values

 creates segment network
network (IP)
 passes segment to IP (IP)
link
link
physical physical

Transport Layer: 3-7

Transport Layer Actions

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

Transport Layer: 3-8

Two principal Internet transport protocols
application
transport

 TCP: Transmission Control Protocol mobile

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

log
• congestion control

ica
le
• flow control

n d-
e nd
• connection setup local or

tra
regional ISP
 UDP: User Datagram Protocol

nsp
ort
home network
• unreliable, unordered delivery content
provider
network
• no-frills extension of “best-effort” IP application
transport
datacenter
network
network
 services not available: data link
physical

• delay guarantees enterprise

network
• bandwidth guarantees
Transport Layer: 3-9
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: 3-10
 HTTP server
client
application application
HTTP msg
transport

transport network transport

network link network
link physical link
physical physical

Transport Layer: 3-11

 HTTP server
client
application application
HTTP msg
Ht HTTP msg
transport

transport network transport

network link network
link physical link
physical physical

Transport Layer: 3-12

 HTTP server
client
application application
HTTP msg
Ht HTTP msg
transport
Hnetwork
n Ht HTTP msg
transport transport
network link network
link physical link
physical physical

Transport Layer: 3-13

 HTTP server
client
application application

transport

transport network transport

network link network
link physical link
physical physical

Hn Ht HTTP msg

Transport Layer: 3-14

 HTTP server
client1 client2
application P-client1 P-client2 application

transport

transport network transport

network link network
link physical link
physical physical

Transport Layer: 3-15

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: 3-16

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: 3-17

Connectionless demultiplexing
Recall: when receiving host receives
 when creating socket, must UDP segment:
specify host-local port #: • checks destination port # in
segment
DatagramSocket mySocket1
= new DatagramSocket(12534); • directs UDP segment to
socket with that port #
 when creating datagram to
send into UDP socket, must
IP/UDP datagrams with same dest.
specify 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: 3-18
 Connectionless demultiplexing: an example
DatagramSocket
serverSocket = new
DatagramSocket
DatagramSocket mySocket2 = DatagramSocket mySocket1 =
new DatagramSocket (6428); new DatagramSocket (5775);
(9157); application
application P1 application
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: 3-19
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: 3-20
Connection-oriented demultiplexing: example
application
application P4 P5 P6 application
P1 P2 P3
transport
transport transport
network
network link network
link physical link
physical server: physical
IP
address
B
host: IP source IP,port: B,80 host: IP
address dest IP,port: A,9157 source IP,port: C,5775 address
A dest IP,port: B,80 C
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: 3-21
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: 3-22

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: 3-23
UDP: User Datagram Protocol
 “no frills,” “bare bones” Why is there a UDP?
Internet transport protocol  no connection
establishment (which can
 “best effort” service, UDP add RTT delay)
segments may be:  simple: no connection state
• lost at sender, receiver
• delivered out-of-order to app  small header size
 connectionless:  no congestion control
 UDP can blast away as fast as
• no handshaking between UDP desired!
sender, receiver  can function in the face of
• each UDP segment handled congestion
independently of others
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: Transport Layer Actions

SNMP client SNMP server

application application

transport transport
(UDP) (UDP)

network (IP) network

(IP)
link
link
physical physical

Transport Layer: 3-27

UDP: Transport Layer Actions

SNMP client SNMP server

UDP sender actions:
application
 is passed an application- SNMP msg
application
layer message
transport  determines UDP segment UDPtransport
UDPh h SNMP msg

(UDP) header fields values (UDP)

 creates UDP segment network
network (IP)
 passes segment to IP (IP)
link
link
physical physical

Transport Layer: 3-28

UDP: Transport Layer Actions

SNMP client SNMP server

UDP receiver actions:
application  receives segment from IP application
 checks UDP checksum
transport
transport
SNMP msg header value
(UDP)  extracts application-layer (UDP)

UDP h SNMP (IP)

network msg message network
 demultiplexes message up (IP)
link to application via socket link

physical physical

Transport Layer: 3-29

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: 3-30

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
checksum
= sender-computed
checksum (as received)

Transport Layer: 3-31

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 if computed checksum equals
 checksum: addition (one’s checksum field value:
complement sum) of segment • Not equal - error detected
content • Equal - no error detected. But maybe
 checksum value put into errors nonetheless? More later ….
UDP checksum field
Transport Layer: 3-32
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: h ttp://gaia.cs.umass.edu/kurose_ross/interactive/
Transport Layer: 3-33
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: 3-34

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: 3-36
 Principles of reliable data transfer

sending receiving
process process
application data data
transport
reliable channel

reliable service abstraction

Transport Layer: 3-37

 Principles of reliable data transfer

sending receiving sending receiving

process process process process
application data data application data data
transport transport
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: 3-38

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: 3-39

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
network
 unless communicated via a unreliable channel

message
reliable service implementation

Transport Layer: 3-40

 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 Bi-directional communication over arrives on receiver side of
unreliable channel to receiver unreliable channel channel
Transport Layer: 3-41
Reliable data transfer: getting started
We will:
 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 stat state
determined by next e event
event 2
1 actions

Transport Layer: 3-42

How to recover from errors?
 acknowledgements (ACKs): receiver explicitly tells sender that
pkt received OK
 underlying channel may flip bits in packet
• checksum to detect bit errors
 sender doesn’t know what happened at receiver: possible
duplicates
• sender adds sequence number to each pkt
• receiver discards (doesn’t deliver up) duplicate pkt

stop and wait

sender sends one packet, then waits for receiver response
Transport Layer: 3-43
How to recover from errors?
 underlying channel can lose packets (data, ACKs)
• 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: 3-44

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: 3-45

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 L
L 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) L
rdt_rcv(rcvpkt) &&
sndpkt = make_pkt(1, data, checksum)
( corrupt(rcvpkt) || udt_send(sndpkt)
isACK(rcvpkt,0) ) start_timer
L

Transport Layer: 3-46

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: 3-47
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: 3-48
Performance of rdt3.0 (stop-and-wait)

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

 example: 1 Gbps link, 15 ms prop. delay, 8000 bit packet

• time to transmit packet into channel:
L 8000 bits
Dtrans = R = = 8 microsecs
109 bits/sec

Transport Layer: 3-49

rdt3.0: stop-and-wait operation
sender receiver
first packet bit transmitted, t = 0

first packet bit arrives

RTT last packet bit arrives, send ACK

ACK arrives, send next

packet, t = RTT + L / R

Transport Layer: 3-50

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: 3-51

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: 3-52

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

Transport Layer: 3-53

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: 3-54
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: 3-55
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 (re)send ack1
012345678 pkt4
rcv ack1, send receive pkt4, discard,
pkt5 (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: 3-56

Selective repeat
 receiver individually acknowledges all correctly received packets
• buffers packets, as needed, for eventual in-order delivery to upper
layer
 sender times-out/retransmits individually for unACKed packets
• sender maintains timer for each unACKed pkt
 sender window
• N consecutive seq #s
• limits seq #s of sent, unACKed packets

Transport Layer: 3-57

Selective repeat: sender, receiver windows

Transport Layer: 3-58

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

Transport Layer: 3-59

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 send ack3
012345678 pkt4
rcv ack1, send
receive pkt4, buffer,
pkt5
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: 3-60

Selective repeat: sender window
(after receipt)
receiver window
(after receipt)

0123012 pkt0
pkt1

a dilemma!
0123012 0123012
0123012 pkt2 0123012
0123012
example: 0123012 pkt3
X
 seq #s: 0, 1, 2, 3 (base 4 counting) 0123012
pkt0 will accept packet

 window size=3 (a) no problem

with seq number 0

0123012 pkt0
0123012 pkt1 0123012
0123012 pkt2 X 0123012
X 0123012
X
timeout
retransmit pkt0
0123012 pkt0
will accept packet
with seq number 0
(b) oops!
Transport Layer: 3-61
Selective repeat: sender window
(after receipt)
receiver window
(after receipt)

0123012 pkt0
pkt1

a dilemma!
0123012 0123012
0123012 pkt2 0123012
0123012
example: 0123012 pkt3
X
 seq #s: 0, 1, 2, 3 (base 4 counting)  receiver can’t
0123012
pkt0 will accept packet
see sender side
 window size=3 (a) no problem
 receiver
with seq number 0

behavior
identical in both
cases!
0something’s
123012 pkt0
0(very)
1 2 3 0 1wrong!
Q: what relationship is needed 2 pkt1
pkt2
0123012
X
between sequence # size and 0123012 0123012
X 0123012
window size to avoid problem timeout
X
in scenario (b)? retransmit pkt0
0123012 pkt0
will accept packet
with seq number 0
(b) oops!
Transport Layer: 3-62
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: 3-63
TCP: overview RFCs: 793,1122, 2018, 5681, 7323

 point-to-point:  cumulative ACKs

• one sender, one receiver  pipelining:
 reliable, in-order byte • TCP congestion and flow control
stream: set window size
• no “message boundaries"  connection-oriented:
 full duplex data: • handshaking (exchange of control
• bi-directional data flow in messages) initializes sender,
same connection receiver state before data exchange
• MSS: maximum segment size  flow controlled:
• sender will not overwhelm receiver

Transport Layer: 3-64

 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
length (of TCP header) head not
len used C EUAP R SF receive window flow control: # bytes
Internet checksum checksum Urg data pointer receiver willing to accept

options (variable
C, E: congestion notification length)
TCP options
application data sent by
RST, SYN, FIN: connection data application into
management (variable length) TCP socket

Transport Layer: 3-65

TCP sequence 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 data checksum

rwnd
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

ACKed yet ACKed but not usable
Q: how receiver handles out-of- (“in-flight”) yet sent

order segments outgoing segment from receiver

• A: TCP spec doesn’t say, - up

source port # dest port #
sequence number

to implementor acknowledgement number

A rwnd
checksum urg pointer
Transport Layer: 3-66
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: 3-67
TCP Sender (simplified)
event: data received from event: timeout
application  retransmit segment that
 create segment with seq # caused timeout
 restart timer
 seq # is byte-stream number
of first data byte in segment
event: ACK received
 start timer if not already
 if ACK acknowledges
running
• think of timer as for oldest
previously unACKed segments
unACKed segment • update what is known to be
ACKed
• expiration interval:
TimeOutInterval • start timer if there are still
unACKed segments
Transport Layer: 3-68
TCP Receiver: ACK generation [RFC 5681]

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

Transport Layer: 3-69

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
timeout

timeout
Seq=100, 20 bytes of data
ACK=100
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: 3-70

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: 3-71
TCP fast retransmit
Host A Host B
TCP fast retransmit
if sender receives 3 additional
ACKs for same data (“triple Se q= 9
2, 8 by
Seq= data tes of
duplicate ACKs”), resend unACKed 100, 2
data
0 b yt e
s of
segment with smallest seq # X
 likely that unACKed segment lost,
=100
so don’t wait for timeout ACK

timeout
=100
ACK
CK =100
A
= 10 0
Receipt of three duplicate ACKs ACK

indicates 3 segments received Seq=100, 20 bytes of data

after a missing segment – lost

segment is likely. So retransmit!

Transport Layer: 3-72

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: 3-73
TCP flow control
application
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: 3-74

TCP flow control
application
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: 3-75

TCP flow control
application
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: 3-76

TCP flow control
application
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: 3-77

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: connection state:

ESTAB ESTAB
connection variables: connection Variables:
seq # client-to- seq # client-to-
server server
server-to-client server-to-client
rcvBuffer size rcvBuffer size
network
at server,client at network
server,client

Socket clientSocket = Socket connectionSocket =

newSocket("hostname","port number"); welcomeSocket.accept();
Transport Layer: 3-78
Agreeing to establish a connection
2-way handshake:

Q: will 2-way handshake always

Let’s talk work in network?
ESTAB
OK  variable delays
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

Transport Layer: 3-79

 2-way handshake scenarios
choose x
req_conn(x)
ESTAB
acc_conn(x)

ESTAB
data(x+1) accept
data(x+1
ACK(x+1)
)
connection
x completes

No problem!

Transport Layer: 3-80

2-way handshake scenarios

choose x
req_conn(x)
ESTAB
retransmit acc_conn(x)
req_conn(
x)
ESTAB
req_conn(x)

connection
client x completes server
terminat forgets x
es

ESTAB
acc_conn(x)
Problem: half open
connection! (no client)
Transport Layer: 3-81
2-way handshake scenarios
choose x
req_conn(x)
ESTAB
retransmit acc_conn(x)
req_conn(
x)
ESTAB
data(x+1) accept
data(x+1
retransmit )
data(x+1)
connection
x completes server
client
terminat forgets x
es req_conn(x)
ESTAB
data(x+1) accept
data(x+1
)
Problem: dup data
accepted!
 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: 3-83

A human 3-way handshake protocol

1. On belay?

2. Belay on.
3. Climbing.

Transport Layer: 3-84

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: 3-85

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: 3-86
Principles of congestion control
Congestion:
 informally: “too many sources sending too much data too fast for
network to handle”
 Indicators:
• long delays (queueing in router buffers)
• packet loss (buffer overflow at routers)
 different from flow control! congestion control:
 a top-10 problem! too many senders,
sending too fast

flow control: one sender

too fast for one receiver
Transport Layer: 3-87
Causes/costs of congestion: scenario 1
original data: lin throughput: lout
Simplest scenario:
Host A
 one router, infinite buffers
 input, output link capacity: R infinite shared
output link buffers
 two flows
R R
 no retransmissions needed

Host B

R/2
Q: What happens as
lout

delay
arrival rate lin
throughput:

approaches R/2?
lin R/2 lin R/2
maximum per-connection large delays as arrival rate
throughput: R/2 lin approaches capacity
Transport Layer: 3-88
 Causes/costs of congestion: scenario 2
 Realistic scenario: one router, finite buffers R/2

 sender retransmits lost, timed-out packet

lout
“wasted” capacity due
to un-needed
retransmissions
• application-layer input = application-layer output: lin = lout

throughput:
when sending at
• transport-layer input includes retransmissions : l’in lin R/2, some packets
are retransmissions,
including needed
and un-needed
Host A lin : original data lin
copy R/2 duplicates, that are
timeo
ut l'in: original data, plus delivered!
retransmitted data

free buffer space!

R R

Host B finite shared output

link buffers
Transport Layer: 3-89
Causes/costs of congestion: scenario 2
Realistic scenario: un-needed duplicates R/2
 packets can be lost, dropped at router due to

lout
“wasted” capacity due
full buffers – requiring retransmissions to un-needed
retransmissions
 but sender times can time out prematurely,

throughput:
sending two copies, both of which are delivered when sending at
R/2, some packets
are retransmissions,
including needed
and un-needed
lin R/2 duplicates, that are
delivered!
“costs” of congestion:
 more work (retransmission) for given receiver throughput
 unneeded retransmissions: link carries multiple copies of a packet
• decreasing maximum achievable throughput

Transport Layer: 3-90

Causes/costs of congestion: scenario 3
 four senders Q: what happens as lin and lin’ increase ?
 multi-hop paths
A: as red lin’ increases, all arriving blue pkts at upper
 timeout/retransmit queue are dropped, blue throughput g 0
Host A lin : original data
Host B
l'in: original data, plus
retransmitted data
finite shared
output link buffers

Host D
lout
Host C

Transport Layer: 3-91

Causes/costs of congestion: scenario 3
R/2
lout

lin’ R/2

another “cost” of congestion:

 when packet dropped, any upstream transmission capacity and
buffering used for that packet was wasted!

Transport Layer: 3-92

Causes/costs of congestion: insights
 throughput can never exceed capacity

 delay increases as capacity approached

 loss/retransmission decreases effective

throughput
 un-needed duplicates further decreases
effective throughput
 upstream transmission capacity / buffering
wasted for packets lost downstream
Transport Layer: 3-93
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: 3-94
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
maximum segment size every each loss event
RTT until loss detected
TCP sender Sending rate

AIMD sawtooth
behavior: probing
for bandwidth

time Transport Layer: 3-95

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: 3-96

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 not-
not used
TCP rate ~
~ bytes/sec
yet ACKed 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: 3-97
TCP slow start
Host A Host B
 when connection begins,
increase rate exponentially
until first loss event:
one s e gm
ent

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

 summary: initial rate is

slow, but ramps up
exponentially fast time

Transport Layer: 3-98

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: h ttp://gaia.cs.umass.edu/kurose_ross/interactive/
Transport Layer: 3-99
 Self Study
8 Slides only
Transport Layer: 3-100
Summary: TCP congestion control
New
New ACK!
new ACK
duplicate ACK
dupACKcount++
ACK!
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

Transport Layer: 3-101

Go-Back-N: sender extended FSM
rdt_send(data)
if (nextseqnum < base+N) {
sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum)
udt_send(sndpkt[nextseqnum])
if (base == nextseqnum)
start_timer
nextseqnum++
}
L else
refuse_data(data)
base=1
nextseqnum=1
timeout
start_timer
Wait udt_send(sndpkt[base])
rdt_rcv(rcvpkt) udt_send(sndpkt[base+1])
&& corrupt(rcvpkt) …
udt_send(sndpkt[nextseqnum-
1])
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
base = getacknum(rcvpkt)+1
If (base == nextseqnum)
stop_timer
else
start_timer
Transport Layer: 3-102
Go-Back-N: receiver extended FSM
any other event
udt_send(sndpkt) rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
L && hasseqnum(rcvpkt,expectedseqnum)
expectedseqnum=1 Wait extract(rcvpkt,data)
sndpkt = deliver_data(data)
make_pkt(expectedseqnum,ACK,chksum) sndpkt = make_pkt(expectedseqnum,ACK,chksum)
udt_send(sndpkt)
expectedseqnum++

ACK-only: always send ACK for correctly-received packet with highest

in-order seq #
• may generate duplicate ACKs
• need only remember expectedseqnum
 out-of-order packet:
• discard (don’t buffer): no receiver buffering!
• re-ACK pkt with highest in-order seq # Transport Layer: 3-103
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. #
ACK received, with ACK field value y start timer

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: 3-104
TCP 3-way handshake FSM
closed
Socket connectionSocket =
welcomeSocket.accept();
L Socket clientSocket =
newSocket("hostname","port number");
SYN(x)
SYNACK(seq=y,ACKnum=x+1) SYN(seq=x)
create new socket for
communication back to client
listen

SYN
SYN sent
rcvd
SYNACK(seq=y,ACKnum=x+1)
ESTAB
ACK(ACKnum=y+1) ACK(ACKnum=y+1)
L

Transport Layer: 3-105

Closing a TCP 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

Transport Layer: 3-106

TCP throughput
 avg. TCP thruput as function of window size, RTT?
• ignore slow start, assume there is always data to send
 W: window size (measured in bytes) where loss occurs
• avg. window size (# in-flight bytes) is ¾ W
• avg. thruput is 3/4W per RTT
3 W
avg TCP thruput = bytes/sec
4 RTT
W

W/2
TCP over “long, fat pipes”
 example: 1500 byte segments, 100ms RTT, want 10 Gbps throughput
 requires W = 83,333 in-flight segments
 throughput in terms of segment loss probability, L [Mathis 1997]:

1.22 . MSS
TCP throughput =
RTT L
➜ to achieve 10 Gbps throughput, need a loss rate of L = 2·10-10 – a very
small loss rate!
 versions of TCP for long, high-speed scenarios

Transport Layer: 3-108