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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
All material copyright 1996-2023
Jim Kurose, Keith Ross
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: 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 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
lo g
ica
l en
§ transport protocols actions in end
d-e
systems:
n
dt
local or
• sender: breaks application messages
ran
regional ISP
spo
into segments, passes to network layer home network
rt
content
• receiver: reassembles segments into provider
network
messages, passes to application layer datacenter
applicationnetwork
transport
network
§ two transport protocols available to data link

physical
Internet applications enterprise

network
• TCP, UDP
Transport Layer Actions
Sender:
application § is passed an application- application
app. msg
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 Actions
Receiver:
application § receives segment from IP application
§ checks header values
app. msg
transport § extracts application-layer transport
message
network (IP) network (IP)
§ demultiplexes message up
link to application via socket link
physical physical
Th app. msg

Chapter 3: roadmap
functionality
Multiplexing/demultiplexing
multiplexing as sender: demultiplexing as 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

HTTP server
client
application application
HTTP msg
transport
Ht HTTP msg
transport Hnnetwork
Ht HTTP msg transport
Hn Hnetwork
t HTTP msg
link network
link physical link
physical physical
Hn Ht HTTP msg

Q: how did transport layer know to deliver message to Firefox
browser process rather then Netflix process or Skype process?
client
HTTP msg
HTTP msg transport
Ht HTTP msg

link physical link
physical physical

?
de-multiplexing
application
? transport
de-multiplexing
Demultiplexing
multiplexing
application
transport
multiplexing
Multiplexing
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

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 DatagramSocket(12534);
socket with that port #
§ 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
Connectionless demultiplexing: an example
mySocket =
socket(AF_INET,SOCK_DGRAM)
mySocket.bind(myaddr,6428);
mySocket = mySocket =
socket(AF_INET,SOCK_STREAM) socket(AF_INET,SOCK_STREAM)
mySocket.bind(myaddr,9157); mySocket.bind(myaddr,5775);
application
P1
P3 P4
transport
transport transport
network
link physical link
physical physical
B D
source port: 6428 source port: ?
dest port: 9157 dest port: ?
A C
source port: 9157 source port: ?
dest port: 6428 dest port: ?
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
Connection-oriented demultiplexing: example
application
application P4 P5 P6 application
P1 P2 P3
transport
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
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

Chapter 3: roadmap
functionality
UDP: User Datagram Protocol
Why is there a UDP?
§ “no frills,” “bare bones”
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
• no handshaking between UDP § UDP can blast away as fast as
desired!
sender, receiver
§ can function in the face of
• each UDP segment handled congestion
independently of others
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

UDP: User Datagram Protocol [RFC 768]

UDP: Transport Layer Actions
SNMP client SNMP server
transport transport
(UDP) (UDP)
network (IP) network (IP)
link link
physical physical


UDP sender actions:
application § is passed an application- application
SNMP msg
layer message
transport transport
§ determines UDP segment UDP
UDPhh SNMP msg
(UDP) header fields values (UDP)
network (IP) § creates UDP segment network (IP)
link
§ passes segment to IP link
physical physical


UDP receiver actions:
application § receives segment from IP application
§ checks UDP checksum
transport transport
SNMP msg header value
(UDP) (UDP)
§ extracts application-layer
network
UDPh SNMP(IP)
msg message network (IP)
§ demultiplexes message up
link to application via socket link
physical physical

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

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)

Internet 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
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/
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!

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
functionality
Principles of reliable data transfer
sending receiving
process process
application data data
transport
reliable channel
reliable service abstraction

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

sending receiving
process process
transport
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

sending receiving
process process
transport
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

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?

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
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
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

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

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?

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

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

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
rcv pkt1 ack1
ack1 send ack1
X timeout
loss resend pkt1
pkt1 rcv pkt1
timeout
resend pkt1 pkt1
rcv pkt1 rcv ack1 (detect duplicate)
(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

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

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

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
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
Go-Back-N 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, discard,
012345678 rcv ack0, send pkt4 (re)send ack1
012345678 rcv ack1, send pkt5 receive pkt4, discard,
(re)send ack1
ignore duplicate ACK receive pkt5, discard,
(re)send ack1
pkt 2 timeout
012345678 send pkt2
012345678 send pkt3
012345678 send pkt4 rcv pkt2, deliver, send ack2
012345678 send pkt5 rcv pkt3, deliver, send ack3
rcv pkt4, deliver, send ack4
rcv pkt5, deliver, send ack5

Selective repeat: the approach
§pipelining: multiple packets in flight
§receiver individually ACKs all correctly received packets
• buffers packets, as needed, for in-order delivery to upper layer
§sender:
• maintains (conceptually) a timer for each unACKed pkt
• timeout: retransmits single unACKed packet associated with timeout
• maintains (conceptually) “window” over N consecutive seq #s
• limits pipelined, “in flight” packets to be within this window

Selective repeat: sender, receiver windows

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-1]: received packet
§ mark packet n as received packet n in [rcvbase-N,rcvbase-1]
§ if n smallest unACKed packet, § ACK(n)
advance window base to next otherwise:
unACKed seq # § ignore

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
record ack3 arrived send ack4
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?

Selective repeat:
sender window receiver window
(after receipt) (after receipt)
a dilemma!
0123012 pkt0
0123012 pkt1 0123012
0123012 pkt2 0123012
example:
0123012
0123012 pkt3
X
§ seq #s: 0, 1, 2, 3 (base 4 counting) 0123012
pkt0 will accept packet
with seq number 0
§ window size=3 (a) no problem
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!
Selective repeat:
sender window receiver window
(after receipt) (after receipt)
a dilemma!
0123012 pkt0
0123012 pkt1 0123012
0123012 pkt2 0123012
example:
0123012
0123012 pkt3
X
§ seq #s: 0, 1, 2, 3 (base 4 counting) § 0receiver
1 2 3 0 1 2 can’t
pkt0 will accept packet
see sender side with seq number 0
§ window size=3 (a) no problem
§ receiver
behavior
identical in both
cases!
§0something’s
123012 pkt0
Q: what relationship is needed (very)
0123012
wrong!pkt1 0123012
between sequence # size and 0123012 pkt2 X

X
0123012
window size to avoid problem

0123012
X
in scenario (b)? timeout
retransmit pkt0
0123012 pkt0
will accept packet
with seq number 0
(b) oops!
Chapter 3: roadmap
• segment structure
• reliable data transfer
• flow control
• connection management
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
steam: 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

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 length)

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

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
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

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

EstimatedRTT = (1- a)*EstimatedRTT + a*SampleRTT
§ exponential weighted moving average (EWMA)
§ influence of past sample decreases exponentially fast
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
§ typical value: a = 0.125 350
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr

(milliseconds)
300
250
RTT (milliseconds)
RTT
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)
SampleRTT Estimated RTT
§ timeout interval: EstimatedRTT plus “safety margin”
• large variation in EstimatedRTT: want a larger safety margin
TimeoutInterval = EstimatedRTT + 4*DevRTT
estimated RTT “safety margin”
§ DevRTT: EWMA of SampleRTT deviation from EstimatedRTT:

DevRTT = (1-b)*DevRTT + b*|SampleRTT-EstimatedRTT|
(typically, b = 0.25)
* Check out the online interactive exercises for more examples: http://gaia.cs.umass.edu/kurose_ross/interactive/
TCP Sender (simplified)
event: data received from event: timeout
application § retransmit segment that
caused timeout
§ create segment with seq #
§ restart timer
§ seq # is byte-stream number
of first data byte in segment
event: ACK received
§ start timer if not already
running § if ACK acknowledges
previously unACKed segments
• think of timer as for oldest
unACKed segment • update what is known to be
ACKed
• expiration interval:
TimeOutInterval • start timer if there are still
unACKed segments
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

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 send cumulative
SendBase=120 ACK for 120
ACK=100
ACK=120
SendBase=120
lost ACK scenario premature timeout

TCP: retransmission scenarios
Host A Host B

ACK=100
X
ACK=120
cumulative ACK covers

for earlier lost ACK

TCP fast retransmit
Host A Host B
TCP fast retransmit
if sender receives 3 additional
ACKs for same data (“triple Seq=92
Seq=1
, 8 byte
s o f da
ta
duplicate ACKs”), resend unACKed 00, 20
bytes
of data
segment with smallest seq # X
§ likely that unACKed segment lost,
=100
so don’t wait for timeout A CK
=100
timeout
A CK
=100
A CK
=100
Receipt of three duplicate ACKs A CK
indicates 3 segments received Seq=100, 20 bytes of data
after a missing segment – lost

segment is likely. So retransmit!

Chapter 3: roadmap
• segment structure
• reliable data transfer
• flow control
• connection management
TCP flow control
application
Q: What happens if network Application removing
process
layer delivers data faster than data from TCP socket
application layer removes buffers
TCP socket
data from socket buffers? receiver buffers
TCP
code
Network layer
delivering IP datagram
payload into TCP
socket buffers IP
code
from sender
receiver protocol stack

TCP flow control
application
process
TCP socket
TCP
code
Network layer
delivering IP datagram
payload into TCP
socket buffers IP
code
from sender

TCP flow control
application
process
TCP socket
TCP
code
receive window
flow control: # bytes
receiver willing to accept IP
code
from sender

TCP flow control
application
process
TCP socket
TCP
flow control code
receiver controls sender, so

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

from sender

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 auto-adjust
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

TCP flow control
flow control: # bytes receiver willing to accept
§ TCP receiver “advertises” free buffer

space in rwnd field in TCP header
• RcvBuffer size set via socket
receive window
options (typical default is 4096 bytes)
• many operating systems auto-adjust
RcvBuffer
§ sender limits amount of unACKed
(“in-flight”) data to received rwnd
§ guarantees receive buffer will not
overflow
TCP segment format

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

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

Chapter 3: roadmap
functionality
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: 3-86

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

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
cwnd
ACKed sent, but not- available but
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)

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Chapter 3

Transport Layer: 3-2

§ provide logical communication mobile

running on different hosts

§ two transport protocols available to data link

Internet applications enterprise

Transport Layer: 3-5

Transport Layer: 3-6

multiplexing as sender: demultiplexing as receiver:

application P1 P2 application socket

Transport Layer: 3-8

Transport Layer: 3-9

transport network transport

Transport Layer: 3-10

numbers to direct segment to

Transport Layer: 3-19

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

Transport Layer: 3-24

Transport Layer: 3-27

Transport Layer: 3-28

SNMP client SNMP server

network (IP) network (IP)

Transport Layer: 3-29

SNMP client SNMP server

network (IP) § creates UDP segment network (IP)

Transport Layer: 3-30

SNMP client SNMP server

Transport Layer: 3-31

application length, in bytes of

Transport Layer: 3-32

Transport Layer: 3-33

Transport Layer: 3-36

reliable service abstraction

Transport Layer: 3-39

sending receiving sending receiving

reliable service implementation

Transport Layer: 3-40

Transport Layer: 3-41

message reliable service implementation

Transport Layer: 3-42

How do humans recover from “errors” during conversation?

Transport Layer: 3-43

stop and wait

stop and wait

Transport Layer: 3-46

§ same functionality as rdt2.1, using ACKs only

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

Transport Layer: 3-47

Q: How do humans handle lost sender-to-

Transport Layer: 3-48

Transport Layer: 3-49

(b) packet loss

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

Transport Layer: 3-52

first packet bit arrives

Transport Layer: 3-53

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

Receiver view of sequence number space:

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

Transport Layer: 3-56

Transport Layer: 3-57

Transport Layer: 3-58

Transport Layer: 3-59

Q: what happens when ack2 arrives?