Chapter3 Dsse

Chapter 3
Transport Layer
A note on the use of these ppt slides:

We’re making these slides freely available to all (faculty, students, readers). Computer
They’re in PowerPoint form so you see the animations; and can add, modify,
and delete slides (including this one) and slide content to suit your needs. Networking: A Top
They obviously represent a lot of work on our part. In return for use, we only
ask the following: Down Approach
 If you use these slides (e.g., in a class) that you mention their source
(after all, we’d like people to use our book!)
6th edition
 If you post any slides on a www site, that you note that they are adapted Jim Kurose, Keith Ross
from (or perhaps identical to) our slides, and note our copyright of this
material.
Addison-Wesley
March 2012
Thanks and enjoy! JFK/KWR
All material copyright 1996-2013

J.F Kurose and K.W. Ross, All Rights Reserved
Transport Layer 3-1

Chapter 3: Transport Layer
our goals:
 understand  learn about Internet
principles behind transport layer protocols:
transport layer  UDP: connectionless
services: transport
 multiplexing,  TCP: connection-oriented
demultiplexing reliable transport
 reliable data transfer  TCP congestion control
 flow control
 congestion control
Transport Layer 3-2

Chapter 3 outline
3.1 transport-layer 3.5 connection-oriented
services transport: TCP
3.2 multiplexing and  segment structure
demultiplexing  reliable data transfer
3.3 connectionless  flow control
transport: UDP  connection management
3.4 principles of reliable 3.6 principles of congestion
data transfer control
3.7 TCP congestion control
Transport Layer 3-3

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

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

protocol available to apps
 Internet: TCP and UDP
Transport Layer 3-4
Transport vs. network layer
 network layer: logical household analogy:
communication
between hosts 12 kids in Ann’s house sending
letters to 12 kids in Bill’s
 transport layer: house:
logical  hosts = houses
communication  processes = kids
between processes  app messages = letters in

envelopes
 relies on, enhances,  transport protocol = Ann
network layer and Bill who demux to in-
services house siblings
 network-layer protocol =
postal service
Transport Layer 3-5

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

physical
network
 congestion control network

data link
data link
physical
physical
 flow control network
data link
 connection setup physical
network
 unreliable, unordered data link

physical
delivery: UDP network

data link
physical
 no-frills extension of network
data link application
“best-effort” IP physical
network
data link
transport
network
data link
 services not available: physical
physical
 delay guarantees
 bandwidth guarantees
Transport Layer 3-6

Chapter 3 outline
Transport Layer 3-7

Multiplexing/demultiplexing
multiplexing at sender:
handle data from multiple demultiplexing at receiver:
sockets, add transport header use header info to deliver
(later used for demultiplexing) received segments to correct
socket
application
application P1 P2 application socket

P3 transport P4
process
transport network transport
network link network
link physical link
physical physical
Transport Layer 3-8

How demultiplexing works
 host receives IP datagrams 32 bits
 each datagram has source IP source port # dest port #
address, destination IP
address
other header fields
 each datagram carries one
transport-layer segment
 each segment has source, application
destination port number data
 host uses IP addresses & (payload)
port numbers to direct
segment to appropriate
TCP/UDP segment format
socket
Transport Layer 3-9

Connectionless demultiplexing
 when host receives UDP IP datagrams with same

segment: dest. port #, but different
 checks destination port # source IP addresses
in segment and/or source port
numbers will be directed
 directs UDP segment to to same socket at dest
socket with that port #
Transport Layer 3-10

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

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

dest port: 6428 dest port: ?
Connection-oriented demux
 TCP socket identified  server host may support
by 4-tuple: many simultaneous TCP
 source IP address sockets:
 source port number  each socket identified by
 dest IP address its own 4-tuple
 dest port number  web servers have
 demux: receiver uses different sockets for
all four values to direct each connecting client
segment to appropriate  non-persistent HTTP will
socket have different socket for
each request

Connection-oriented demux: example
application
application P4 P5 P6 application
P3 P2 P3
transport
transport transport
network
link physical link
physical server: IP physical
address B
host: IP source IP,port: B,80 host: IP

address A dest IP,port: A,9157 source IP,port: C,5775 address C
dest IP,port: B,80
source IP,port: A,9157
dest IP, port: B,80
source IP,port: C,9157
dest IP,port: B,80
three segments, all destined to IP address: B,
dest port: 80 are demultiplexed to different sockets Transport Layer 3-13
Connection-oriented demux: example
threaded server
application
P4
P3 P2 P3
transport
transport transport
network
link physical link
physical server: IP physical
address B
host: IP source IP,port: B,80 host: IP

address A dest IP,port: A,9157 source IP,port: C,5775 address C
dest IP,port: B,80
source IP,port: A,9157
dest IP, port: B,80
source IP,port: C,9157
dest IP,port: B,80

Chapter 3 outline

UDP: User Datagram Protocol [RFC 768]
 “no frills,” “bare bones”  UDP use:
Internet transport  streaming multimedia
protocol apps (loss tolerant, rate
 “best effort” service, sensitive)
UDP segments may be:  DNS
 lost  SNMP
 delivered out-of-order  reliable transfer over
to app
UDP:
 connectionless:
 add reliability at
 no handshaking application layer
between UDP sender,
receiver  application-specific error
recovery!
 each UDP segment
handled independently
of others
UDP: segment header
length, in bytes of
32 bits UDP segment,
source port # dest port # including header
length checksum
why is there a UDP?
 no connection
application establishment (which can
data add delay)
(payload)  simple: no connection
state at sender, receiver
 small header size
UDP segment format  no congestion control:
UDP can blast away as
fast as desired

UDP checksum
Goal: detect “errors” (e.g., flipped bits) in transmitted
segment
sender: receiver:
 treat segment contents,  compute checksum of
including header fields, received segment
as sequence of 16-bit  check if computed
integers
checksum equals checksum
 checksum: addition field value:
(one’s complement
sum) of segment  NO - error detected
contents  YES - no error detected.
 sender puts checksum But maybe errors
value into UDP nonetheless? More later
checksum field ….
Internet checksum: example
example: add two 16-bit integers
1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0
1 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1
wraparound 1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1
sum 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0
checksum 1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
Note: when adding numbers, a carryout from the most

significant bit needs to be added to the result

Chapter 3 outline

Principles of reliable data transfer
 important in application, transport, link layers
 top-10 list of important networking topics!
 characteristics of unreliable channel will determine

complexity of reliable data transfer protocol (rdt)


Reliable data transfer: getting started
we’ll:
 incrementally develop sender, receiver sides of
reliable data transfer protocol (rdt)
 consider only unidirectional data transfer
 but control info will flow on both directions!

rdt1.0: reliable transfer over a reliable channel
 underlying channel perfectly reliable
 no bit errors
 no loss of packets

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
Howretransmits
do humanspkt on receipt from
recover of NAK“errors”
 new mechanisms in rdt2.0 (beyond rdt1.0):
 error detection
during conversation?
 receiver feedback: control msgs (ACK,NAK) rcvr-
>sender

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
 new mechanisms in rdt2.0 (beyond rdt1.0):
 error detection
 feedback: control msgs (ACK,NAK) from receiver to
sender

rdt2.0 has a fatal flaw!
what happens if handling duplicates:
ACK/NAK corrupted?  sender retransmits
 sender doesn’t know current pkt if ACK/NAK
what happened at corrupted
receiver!
 sender adds sequence
 can’t just retransmit: number to each pkt
possible 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
 two seq. #’s (0,1) will packet is duplicate
suffice. Why?  note: receiver can not
 must check if received know if its last
ACK/NAK corrupted ACK/NAK received
OK at sender

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

rdt3.0: channels with errors and loss
new assumption: approach: sender waits

underlying channel can “reasonable” amount of
also lose packets time for ACK
(data, ACKs)  retransmits if no ACK
 checksum, seq. #, received in this time
ACKs, retransmissions  if pkt (or ACK) just delayed
(not lost):
will be of help … but
not enough  retransmission will be
duplicate, but seq. #’s
already handles this
 receiver must specify seq
# of pkt being ACKed
 requires countdown timer

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
ack1 send ack1 loss
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
send pkt0 pkt0 rcv pkt0
send ack0
rcv pkt0 ack0
send ack0 rcv ack0
ack0 send pkt1 pkt1
rcv ack0 rcv pkt1
send pkt1 pkt1
send ack1
rcv pkt1 ack1
ack1 send ack1
X
loss timeout
resend pkt1 pkt1
rcv pkt1
timeout
resend pkt1 pkt1 rcv ack1 (detect duplicate)
rcv pkt1 send pkt0
pkt0
send ack1
(detect duplicate) ack1
ack1 send ack1 rcv ack1 rcv pkt0
rcv ack1 send pkt0
ack0 send ack0
send pkt0 pkt0 pkt0
rcv pkt0
rcv pkt0 ack0 (detect duplicate)
ack0 send ack0 send ack0
(c) ACK loss (d) premature timeout/ delayed ACK

Performance of rdt3.0
 rdt3.0 is correct, but performance stinks
 e.g.: 1 Gbps link, 15 ms prop. delay, 8000 bit packet:
L 8000 bits
Dtrans = R = = 8 microsecs
109 bits/sec
 U sender: utilization – fraction of time sender busy sending
U L/R .008
sender = = = 0.00027
RTT + L / R 30.008
 if RTT=30 msec, 1KB pkt every 30 msec: 33kB/sec thruput

over 1 Gbps link
 network protocol limits use of physical resources!
rdt3.0: stop-and-wait operation
sender receiver
first packet bit transmitted, t = 0
last packet bit transmitted, t = L / R
first packet bit arrives

RTT last packet bit arrives, send ACK
ACK arrives, send next

packet, t = RTT + L / R
U L/R .008
sender = = = 0.00027
RTT + L / R 30.008

Pipelined protocols
pipelining: sender allows multiple, “in-flight”, yet-
to-be-acknowledged pkts
 range of sequence numbers must be increased
 buffering at sender and/or receiver
 two generic forms of pipelined protocols: go-Back-N,

selective repeat
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

Pipelined protocols: overview
Go-back-N: Selective Repeat:
 sender can have up to  sender can have up to N
N unacked packets in unack’ed packets in
pipeline pipeline
 receiver only sends  rcvr sends individual ack
cumulative ack for each packet
 doesn’t ack packet if
there’s a gap
 sender has timer for  sender maintains timer
oldest unacked packet for each unacked packet
 when timer expires,  when timer expires,
retransmit all unacked retransmit only that
packets unacked packet

Go-Back-N: sender
 k-bit seq # in pkt header
 “window” of up to N, consecutive unack’ed pkts allowed
 ACK(n): ACKs all pkts up to, including seq # n - “cumulative

ACK”
 may receive duplicate ACKs (see receiver)
 timer for oldest in-flight pkt
 timeout(n): retransmit packet n and all higher seq # pkts in
window
GBN: receiver
ACK-only: always send ACK for correctly-received

pkt with highest in-order seq #
 may generate duplicate ACKs
 need only remember expectedseqnum
 out-of-order pkt:
 discard (don’t buffer): no receiver buffering!
 re-ACK pkt with highest in-order seq #

GBN 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
 receiver individually acknowledges all correctly
received pkts
 buffers pkts, as needed, for eventual in-order delivery
to upper layer
 sender only resends pkts for which ACK not
received
 sender timer for each unACKed pkt
 sender window
 N consecutive seq #’s
 limits seq #s of sent, unACKed pkts

Selective repeat: sender, receiver windows

Selective repeat
sender receiver
data from above: pkt n in [rcvbase, rcvbase+N-1]
 if next available seq # in  send ACK(n)
window, send pkt  out-of-order: buffer
timeout(n):  in-order: deliver (also
 resend pkt n, restart deliver buffered, in-order
timer pkts), advance window to
next not-yet-received pkt
ACK(n) in [sendbase,sendbase+N]:
 mark pkt n as received
pkt n in [rcvbase-N,rcvbase-1]
 ACK(n)
 if n smallest unACKed
pkt, advance window base otherwise:
to next 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 receive pkt4, buffer,
send ack4
record ack3 arrived receive pkt5, buffer,
send ack5
pkt 2 timeout
012345678 send pkt2
012345678 record ack4 arrived
012345678 rcv pkt2; deliver pkt2,
record ack5 arrived
012345678 pkt3, pkt4, pkt5; send ack2
Q: what happens when ack2 arrives?

sender window receiver window
Selective repeat: (after receipt) (after receipt)
dilemma 0123012 pkt0

pkt1
0123012 0123012
pkt2 0123012
example:
0123012
0123012
pkt3
seq #’s: 0, 1, 2, 3
0123012
 X
0123012
 window size=3 pkt0 will accept packet
with seq number 0
(a) no problem
 receiver sees no
difference in two receiver can’t see sender side.
scenarios! receiver behavior identical in both cases!
something’s (very) wrong!
 duplicate data
accepted as new in 0123012 pkt0
(b) 0123012 pkt1 0123012
0123012 pkt2 0123012
X 0123012
Q: what relationship X
between seq # size timeout
retransmit pkt0 X
and window size to 0123012 pkt0
will accept packet
avoid problem in (b)? with seq number 0
(b) oops!
Chapter 3 outline

TCP: Overview RFCs: 793,1122,1323, 2018, 2581
 point-to-point:  full duplex data:

 one sender, one receiver  bi-directional data flow
 reliable, in-order byte in same connection
steam:  MSS: maximum segment
size
 no “message
boundaries”  connection-oriented:
 pipelined:  handshaking (exchange
of control msgs) inits
 TCP congestion and sender, receiver state
flow control set window before data exchange
size
 flow controlled:
 sender will not
overwhelm receiver
TCP segment structure
32 bits
URG: urgent data counting
(generally not used) source port # dest port #
by bytes
sequence number of data
ACK: ACK #
valid acknowledgement number (not segments!)
head not
PSH: push data now len used
UAP R S F receive window
(generally not used) # bytes
checksum Urg data pointer
rcvr willing
RST, SYN, FIN: to accept
options (variable length)
connection estab
(setup, teardown
commands)
application
Internet data
checksum (variable length)
(as in UDP)

TCP seq. numbers, ACKs
sequence numbers:
byte stream “number” of
first byte in segment’s
data
acknowledgements:
seq # of next byte
expected from other side
cumulative ACK
Q: how receiver handles
out-of-order segments
A: TCP spec doesn’t say,
- up to implementor

TCP seq. numbers, ACKs
Host A Host B
User
types
‘C’ Seq=42, ACK=79, data = ‘C’
host ACKs
receipt of
‘C’, echoes
Seq=79, ACK=43, data = ‘C’ back ‘C’
host ACKs
receipt
of echoed
‘C’ Seq=43, ACK=80
simple telnet scenario

TCP round trip time, timeout
Q: how to set TCP Q: how to estimate RTT?
timeout value?  SampleRTT: measured
time from segment
 longer than RTT transmission until ACK
 but RTT varies receipt
 too short: premature  ignore retransmissions
timeout, unnecessary  SampleRTT will vary, want
retransmissions estimated RTT “smoother”
 average several recent
 too long: slow reaction measurements, not just
to segment loss current SampleRTT

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

RTT (milliseconds)
300
250
RTT (milliseconds)
200
sampleRTT
150
EstimatedRTT
100
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
time (seconds) Transport Layer 3-53
SampleRTT Estimated RTT
 timeout interval: EstimatedRTT plus “safety margin”
 large variation in EstimatedRTT -> larger safety margin
 estimate SampleRTT deviation from EstimatedRTT:
DevRTT = (1-)*DevRTT +
*|SampleRTT-EstimatedRTT|
(typically,  = 0.25)
TimeoutInterval = EstimatedRTT + 4*DevRTT
estimated RTT “safety margin”

Chapter 3 outline

TCP reliable data transfer
 TCP creates rdt service
on top of IP’s unreliable
service
 pipelined segments
 cumulative acks let’s initially consider
 single retransmission simplified TCP sender:
timer  ignore duplicate acks
 retransmissions  ignore flow control,
triggered by: congestion control
 timeout events
 duplicate acks

TCP sender events:
data rcvd from app: timeout:
 create segment with  retransmit segment
seq # that caused timeout
 seq # is byte-stream  restart timer
number of first data ack rcvd:
byte in segment  if ack acknowledges
 start timer if not previously unacked
already running segments
 think of timer as for  update what is known
oldest unacked to be ACKed
segment
 start timer if there are
 expiration interval: still unacked segments
TimeOutInterval

TCP: retransmission scenarios
Host A Host B Host A Host B
SendBase=92
Seq=92, 8 bytes of data Seq=92, 8 bytes of data
Seq=100, 20 bytes of data

timeout
timeout
ACK=100
X
ACK=100
ACK=120
Seq=92, 8 bytes of data Seq=92, 8

SendBase=100 bytes of data
SendBase=120
ACK=100
ACK=120
SendBase=120
lost ACK scenario premature timeout

TCP: retransmission scenarios
Host A Host B

timeout
ACK=100
X
ACK=120
cumulative ACK
TCP ACK generation [RFC 1122, RFC 2581]
event at receiver TCP receiver action

arrival of in-order segment with delayed ACK. Wait up to 500ms
expected seq #. All data up to for next segment. If no next segment,
expected seq # already ACKed send ACK
arrival of in-order segment with immediately send single cumulative

expected seq #. One other ACK, ACKing both in-order segments
segment has ACK pending
arrival of out-of-order segment immediately send duplicate ACK,

higher-than-expect seq. # . indicating seq. # of next expected byte
Gap detected
arrival of segment that immediate send ACK, provided that

partially or completely fills gap segment starts at lower end of gap

TCP fast retransmit
 time-out period often
relatively long: TCP fast retransmit
 long delay before if sender receives 3
resending lost packet ACKs for same data
 detect lost segments (“triple
(“triple duplicate
duplicate ACKs”),
ACKs”),
via duplicate ACKs. resend unacked
 sender often sends segment with smallest
many segments back- seq #
to-back
 likely that unacked
 if segment is lost, there segment lost, so don’t
will likely be many wait for timeout
duplicate ACKs.

TCP fast retransmit
Host A Host B

X
ACK=100
timeout
ACK=100
ACK=100
ACK=100
fast retransmit after sender

receipt of triple duplicate ACK
Chapter 3 outline

TCP flow control
application
application may process
remove data from application
TCP socket buffers ….
TCP socket OS
receiver buffers
… slower than TCP
receiver is delivering
(sender is sending) TCP
code
IP
flow control code
receiver controls sender, so
sender won’t overflow
receiver’s buffer by transmitting from sender
too much, too fast
receiver protocol stack

TCP flow control
 receiver “advertises” free
buffer space by including to application process
rwnd value in TCP header
of receiver-to-sender
segments RcvBuffer buffered data
 RcvBuffer size set via
socket options (typical default rwnd free buffer space
is 4096 bytes)
 many operating systems
autoadjust RcvBuffer TCP segment payloads
 sender limits amount of
unacked (“in-flight”) data to receiver-side buffering
receiver’s rwnd value
 guarantees receive buffer
will not overflow
Chapter 3 outline

Connection Management
before exchanging data, sender/receiver “handshake”:
 agree to establish connection (each knowing the other willing
to establish connection)
 agree on connection parameters
connection state: ESTAB connection state: ESTAB

connection variables: connection Variables:
seq # client-to-server seq # client-to-server
server-to-client server-to-client
rcvBuffer size rcvBuffer size
at server,client at server,client
network network
Socket clientSocket = Socket connectionSocket =

newSocket("hostname","port welcomeSocket.accept();
number");

TCP 3-way handshake
client state server state

LISTEN LISTEN
choose init seq num, x
send TCP SYN msg
SYNSENT SYNbit=1, Seq=x
choose init seq num, y
send TCP SYNACK
msg, acking SYN SYN RCVD
SYNbit=1, Seq=y
ACKbit=1; ACKnum=x+1
received SYNACK(x)
ESTAB indicates server is live;
send ACK for SYNACK;
this segment may contain ACKbit=1, ACKnum=y+1
client-to-server data
received ACK(y)
indicates client is live
ESTAB

TCP: closing a connection
 client, server each close their side of connection
 send TCP segment with FIN bit = 1
 respond to received FIN with ACK
 on receiving FIN, ACK can be combined with own FIN
 simultaneous FIN exchanges can be handled

TCP: closing a connection
client state server state
ESTAB ESTAB
clientSocket.close()
FIN_WAIT_1 can no longer FINbit=1, seq=x
send but can
receive data CLOSE_WAIT
ACKbit=1; ACKnum=x+1
can still
FIN_WAIT_2 wait for server send data
close
LAST_ACK
FINbit=1, seq=y
TIMED_WAIT can no longer
send data
ACKbit=1; ACKnum=y+1
timed wait
for 2*max CLOSED
segment lifetime
CLOSED

Chapter 3 outline

Principles of congestion control
congestion:
 informally: “too many sources sending too much
data too fast for network to handle”
 different from flow control!
 manifestations:
 lost packets (buffer overflow at routers)
 long delays (queueing in router buffers)
 a top-10 problem!

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

buffers output link buffers
 output link capacity: R

 no retransmission
Host B

 one router, finite buffers
 sender retransmission of timed-out packet
 application-layer input = application-layer output: lin =
lout
 transport-layer input includes retransmissions : l‘in lin
lin : original data

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

Host B
link buffers
idealization: perfect
knowledge
 sender sends only when
router buffers available
lin : original data

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

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

lout
copy l'in: original data, plus
retransmitted data
A
no buffer space!
Host B
Idealization: known loss
packets can be lost,
dropped at router due
to full buffers
 sender only resends if
packet known to be lost
lin : original data

lout
l'in: original data, plus
retransmitted data
A
free buffer space!
Host B
Realistic: duplicates
 packets can be lost, dropped
at router due to full buffers
 sender times out prematurely,
sending two copies, both of
which are delivered
lin
timeout
copy l'in lout
A
free buffer space!
Host B
Realistic: duplicates
 packets can be lost, dropped
at router due to full buffers
 sender times out prematurely,
sending two copies, both of
which are delivered

Approaches towards congestion control
two broad approaches towards congestion control:
end-end congestion network-assisted

control: congestion control:
 no explicit feedback  routers provide
from network feedback to end systems
 congestion inferred  single bit indicating
from end-system congestion (SNA,
observed loss, delay DECbit, TCP/IP ECN,
 approach taken by ATM)
TCP  explicit rate for
sender to send at

Chapter 3 outline

TCP congestion control: additive increase
multiplicative decrease
 approach: sender increases transmission rate (window
size), probing for usable bandwidth, until loss occurs
 additive increase: increase cwnd by 1 MSS every
RTT until loss detected
 multiplicative decrease: cut cwnd in half after loss
additively increase window size …
…. until loss occurs (then cut window in half)
congestion window size
cwnd: TCP sender
AIMD saw tooth

behavior: probing
for bandwidth
time
TCP Congestion Control: details
sender sequence number space
cwnd TCP sending rate:
 roughly: send cwnd
bytes, wait RTT for
last byte last byte
ACKS, then send
ACKed sent, not-
yet ACKed
sent more bytes
(“in-
flight”) cwnd
 sender limits transmission: rate ~
~
RTT
bytes/sec
LastByteSent- < cwnd

LastByteAcked
 cwnd is dynamic, function

of perceived network
congestion
TCP Slow Start
Host A Host B
 when connection begins,
increase rate
exponentially until first
loss event:
RTT
 initially cwnd = 1 MSS
 double cwnd every RTT
 done by incrementing
cwnd for every ACK
received
 summary: initial rate is
slow but ramps up
exponentially fast time

TCP: detecting, reacting to loss
 loss indicated by timeout:
 cwnd set to 1 MSS;
 window then grows exponentially (as in slow start)
to threshold, then grows linearly
 loss indicated by 3 duplicate ACKs: TCP RENO
 dup ACKs indicate network capable of delivering
some segments
 cwnd is cut in half window then grows linearly
 TCP Tahoe always sets cwnd to 1 (timeout or 3
duplicate acks)

TCP: switching from slow start to CA
Q: when should the
exponential
increase switch to
linear?
A: when cwnd gets
to 1/2 of its value
before timeout.
Implementation:
 variable ssthresh
 on loss event, ssthresh
is set to 1/2 of cwnd just
before loss event

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

Chapter3 Dsse

Загружено:

Сведения о документе

Оригинальное название

Авторское право

Доступные форматы

Поделиться этим документом

Поделиться или встроить документ

Параметры публикации

Этот документ был вам полезен?

Это неприемлемый материал?

Авторское право:

Доступные форматы

Chapter3 Dsse

Загружено:

Авторское право:

Доступные форматы

Chapter 3

A note on the use of these ppt slides:

All material copyright 1996-2013

Transport Layer 3-1

Transport Layer 3-2

Transport Layer 3-3

running on different hosts

 more than one transport

between processes  app messages = letters in

Transport Layer 3-5

delivery (TCP) data link

 congestion control network

 connection setup physical

 unreliable, unordered data link

delivery: UDP network

Transport Layer 3-6

Transport Layer 3-7

application P1 P2 application socket

Transport Layer 3-8

Transport Layer 3-9

 when host receives UDP IP datagrams with same

Transport Layer 3-10

source port: 6428 source port: ?

source port: 9157 source port: ?

Transport Layer 3-12

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

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

Transport Layer 3-14

Transport Layer 3-15

Transport Layer 3-17

Note: when adding numbers, a carryout from the most

Transport Layer 3-19

Transport Layer 3-20

 characteristics of unreliable channel will determine

 characteristics of unreliable channel will determine

 characteristics of unreliable channel will determine

Transport Layer 3-24

Transport Layer 3-25

Transport Layer 3-26

Transport Layer 3-27

Transport Layer 3-28

Transport Layer 3-29

Transport Layer 3-30

new assumption: approach: sender waits

Transport Layer 3-31

(b) packet loss

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

Transport Layer 3-33

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

 if RTT=30 msec, 1KB pkt every 30 msec: 33kB/sec thruput

first packet bit arrives

ACK arrives, send next

Transport Layer 3-35

 two generic forms of pipelined protocols: go-Back-N,

first packet bit arrives

Transport Layer 3-37

Transport Layer 3-38

 ACK(n): ACKs all pkts up to, including seq # n - “cumulative

ACK-only: always send ACK for correctly-received

Transport Layer 3-40

Transport Layer 3-41

Transport Layer 3-42

Transport Layer 3-43