6 Transport Layer

Transport Layer
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
 Connectionless transport: UDP
 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
log
ica
le
 transport protocols actions in end
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 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 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 Actions
Sender:
application  is passed an application- app. msg
application
layer message
transport
 determines segment TTh htransport
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)  demultiplexes message up network (IP)
link to application via socket link
physical physical
Th app. msg

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
Chapter 3: roadmap
 Multiplexing and demultiplexing
 Principles of reliable data transfer
functionality
Chapter 3: roadmap
functionality

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


UDP sender actions:
application  is passed an application- SNMP msg
application
layer message
transport  determines UDP segment UDPhtransport
UDP h SNMP msg
(UDP) header fields values (UDP)
network (IP)
 creates UDP segment network (IP)
link
 passes segment to IP link
physical physical
Simple Network Management Protocol (SNMP)


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

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
Internet checksum: an example
example: add two 16-bit integers
1110011001100110
1101010101010101
wraparound 11011101110111011
sum 1011101110111100
checksum 0100010001000011
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/
Internet checksum: weak protection!
example: add two 16-bit integers
01
1110011001100110 10
1101010101010101
wraparound 11011101110111011 Even though
numbers have
sum 1011101110111100 changed (bit
flips), no change
checksum 0100010001000011 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?)

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 data transfer protocol (rdt): interfaces
rdt_send(): called from above, deliver_data(): called by rdt to
(e.g., by app.). Passed data 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
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

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)

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
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
send pkt2 receive pkt0, send ack0
012345678
send pkt3 Xloss receive pkt1, send ack1
012345678
(wait)
receive pkt3, discard,
012345678 rcv ack0, send pkt4 (re)send ack1
012345678 rcv ack1, send pkt5 receive pkt4, discard,
(re)send ack1
ignore duplicate ACK receive pkt5, discard,
(re)send ack1
pkt 2 timeout
012345678 send pkt2
012345678 send pkt3
012345678 send pkt4 rcv pkt2, deliver, send ack2
012345678 send pkt5 rcv pkt3, deliver, send ack3
rcv pkt4, deliver, send ack4
rcv pkt5, deliver, send ack5

Selective repeat
 receiver individually acknowledges all correctly received 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

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]: 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

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?

receiver window
Selective repeat:
sender window
(after receipt) (after receipt)
pkt0
a dilemma!
0123012
0123012 pkt1 0123012
0123012 pkt2 0123012
0123012
example: 0123012 pkt3
X
0123012
 seq #s: 0, 1, 2, 3 (base 4 counting) 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!
receiver window
Selective repeat:
sender window
(after receipt) (after receipt)
pkt0
a dilemma!
0123012
0123012 pkt1 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 with seq number 0
 window size=3 (a) no problem
 receiver
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!
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
C, E: congestion notification length)
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
yet sent
Q: how receiver handles out-of- (“in-flight”)
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: 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

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
Seq=92
ACKs for same data (“triple Seq=1
, 8 bytes
of data
duplicate ACKs”), resend unACKed 0 0, 20 b
ytes o
f data
segment with smallest seq # X
 likely that unACKed segment lost,
=100
so don’t wait for timeout ACK
=100
timeout
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!

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

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

TCP flow control
application
process

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

TCP flow control
application
process

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

TCP flow control
application
process

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

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

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 autoadjust
RcvBuffer
 sender limits amount of unACKed
(“in-flight”) data to received rwnd
 guarantees receive buffer will not
overflow
TCP segment format

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();
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

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!

choose x
req_conn(x)
ESTAB
retransmit acc_conn(x)
req_conn(x)
ESTAB
req_conn(x)
connection
client x completes server
terminates forgets x
ESTAB
acc_conn(x)
Problem: half open
connection! (no client)
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
terminates forgets x
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

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

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
arrival rate lin
delay
throughput:
approaches R/2?
lin R/2 lin R/2
maximum per-connection large delays as arrival rate
throughput: R/2 lin approaches capacity
 one router, finite buffers
 sender retransmits lost, timed-out packet
• application-layer input = application-layer output: lin = lout
• transport-layer input includes retransmissions : l’in lin
Host A lin : original data

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

link buffers
Idealization: perfect knowledge R/2
lout
 sender sends only when router buffers available
throughput:
Host A lin : original data lin
copy l'in: original data, plus lout R/2
retransmitted data
free buffer space!
R R

link buffers
Idealization: some perfect knowledge
 packets can be lost (dropped at router) due to
full buffers
 sender knows when packet has been dropped:
only resends if packet known to be lost

copy l'in: original data, plus
retransmitted data
no buffer space!
R R

link buffers
Idealization: some perfect knowledge R/2
“wasted” capacity due
lout
 packets can be lost (dropped at router) due to to retransmissions
full buffers
throughput:
when sending at
 sender knows when packet has been dropped: R/2, some packets
only resends if packet known to be lost are needed
retransmissions
Host A lin : original data lin R/2

l'in: original data, plus
retransmitted data
free buffer space!
R R

link buffers
Realistic scenario: un-needed duplicates R/2
 packets can be lost, dropped at router due to
lout
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
Host A lin : original data lin
and un-needed
timeout R/2 duplicates, that are
copy l'in: original data, plus delivered!
retransmitted data
free buffer space!
R R

link buffers
Realistic scenario: un-needed duplicates R/2
 packets can be lost, dropped at router due to
lout
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

 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 B
l'in: original data, plus
retransmitted data
finite shared
output link buffers
Host D
lout
Host C

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!

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
Approaches towards congestion control
End-end congestion control:

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

Approaches towards congestion control
Network-assisted congestion
control: explicit congestion info
 routers provide direct feedback
to sending/receiving hosts with data data
ACKs
flows passing through congested ACKs
router
 may indicate congestion level or
explicitly set sending rate
 TCP ECN, ATM, DECbit protocols
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: 3-77

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 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

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
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
cwnd
SS : Bắt đầu chậm (Slow start)
AI : Additive increase (Tăng theo cấp số cộng)
26
24 MD : Multiplicative decrease (Giảm theo cấp số nhân)
22 X
20
18
16
14
12
AI
MD
10 SS
08
MD
06 SS AI
04 AI
02
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
Round
Summary: TCP congestion control
New
New ACK!
ACK! new ACK
duplicate ACK
dupACKcount++ new ACK .
cwnd = cwnd + MSS (MSS/cwnd)
dupACKcount = 0
cwnd = cwnd+MSS transmit new segment(s), as allowed
dupACKcount = 0
L transmit new segment(s), as allowed
cwnd = 1 MSS
ssthresh = 64 KB cwnd > ssthresh
dupACKcount = 0
slow L congestion
start timeout avoidance
ssthresh = cwnd/2
cwnd = 1 MSS duplicate ACK
timeout dupACKcount = 0 dupACKcount++
ssthresh = cwnd/2 retransmit missing segment
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
timeout
New
ACK!
ssthresh = cwnd/2
cwnd = 1 New ACK
dupACKcount = 0
cwnd = ssthresh dupACKcount == 3
dupACKcount == 3 retransmit missing segment dupACKcount = 0
ssthresh= cwnd/2 ssthresh= cwnd/2
cwnd = ssthresh + 3 cwnd = ssthresh + 3
fast
recovery
duplicate ACK
cwnd = cwnd + MSS
transmit new segment(s), as allowed

6 Transport Layer

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6 Transport Layer

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Transport Layer

Transport Layer: 3-1

Transport Layer: 3-2

Transport Layer: 3-3

 provide logical communication mobile

between application processes national or global ISP

running on different hosts

 two transport protocols available to data link

Internet applications enterprise

 network layer: logical household analogy:

Transport Layer: 3-7

link to application via socket link

Transport Layer: 3-8

 TCP: Transmission Control Protocol mobile

 services not available: data link

• delay guarantees enterprise

Transport Layer: 3-11

Transport Layer: 3-13

SNMP client SNMP server

network (IP) network (IP)

Transport Layer: 3-14

SNMP client SNMP server

(UDP) header fields values (UDP)

Simple Network Management Protocol (SNMP)

SNMP client SNMP server

Transport Layer: 3-16

application length, in bytes of

Transport Layer: 3-17

Transport Layer: 3-18

Transport Layer: 3-21

reliable service abstraction

Transport Layer: 3-24

sending receiving sending receiving

reliable service implementation

Transport Layer: 3-25

Transport Layer: 3-26

Transport Layer: 3-27

sender-side data receiver-side

udt_send() Header data Header data rdt_rcv()

Transport Layer: 3-29

 separate FSMs for sender, receiver:

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

Transport Layer: 3-30

How do humans recover from “errors” during conversation?

Transport Layer: 3-31

stop and wait

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

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

Transport Layer: 3-35

Transport Layer: 3-36

Transport Layer: 3-37

Transport Layer: 3-38

Q: what happens when ack2 arrives?

Transport Layer: 3-39

Transport Layer: 3-43

source port # dest port # segment seq #: counting

Transport Layer: 3-44

first byte in segment’s data checksum

• seq # of next byte expected

• cumulative ACK sent sent, not- usable not

order segments outgoing segment from receiver

• A: TCP spec doesn’t say, - up

to implementor acknowledgement number

simple telnet scenario