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Chapter 3Transport Layer
Transport Layer 3-1
Transport Layer
Central piece of the layered network architecture
Provides communications services to applications
process
socket
host orserver
process
socket
host orserver
applicationapplicationmsg
msg
Transport Layer 3-2
helliprelying on the service of the network layer
TCP withbuffersvariables
TCP withbuffersvariables
Internet
transportnetworkdata linkphysical
transportnetworkdata linkphysical
Transport Layer
Central piece of the layered network architecture
Provides communications services to applications
process
socket
host orserver
process
socket
host orserver
application application
Transport Layer 3-3
helliprelying on the service of the network layer
socket
Internetnetworkdata linkphysical
networkdata linkphysical
transport transportmsgTH
msg TH
Best effort routing of datagrams from sender to destination host
Transport Layer
Transport entities run in end systems
send side breaks app messages into segments passes to network layer
rcv side reassembles
process
socket
host orserver
process
socket
host orserver
application applicationmsg
msg
msg
Transport Layer 3-4
rcv side reassembles segments into messages passes to app layer
Extends host to host datagram delivery to app to app message transfer
socket
Internetnetworkdata linkphysical
networkdata linkphysical
transport transport
Best effort routing of datagrams from sender to destination host
mTHgTH sTHmTH sTHgTH
msgmsg
Internet transport-layer protocols
reliable in-order delivery (TCP)
congestion control
flow control
connection setup
applicationtransportnetworkdata linkphysical
networkdata link
networkdata linkphysical
networkdata linkphysicalnetwork
data linkphysical
Transport Layer 3-5
connection setup
unreliable unordered delivery UDP
no-frills extension of ldquobest-effortrdquo IP
services not available delay guarantees
bandwidth guarantees
applicationtransportnetworkdata linkphysical
networkdata linkphysical
data linkphysical
Multiplexingdemultiplexing
= process= socket
delivering received segmentsto correct socket
Demultiplexing at rcv hostgathering data from multiplesockets enveloping data with header (later used for demultiplexing)
Multiplexing at send host
Transport Layer 3-6
application
transport
network
link
physical
P1 application
transport
network
link
physical
application
transport
network
link
physical
P2P3 P4P1
host 1 host 2 host 3
transport
How demultiplexing works
host receives IP datagrams
each datagram has source IP address destination IP address (in the network layer header)
each datagram carries 1
Source IP addr
Dest IP addr
Source port
Dest portTransport Layer
Network Layer
Header
Segm
ent
Datagram ndash Network Layer PDU
Transport Layer 3-7
each datagram carries 1 transport-layer segment
each segment has source destination port number (in the transport layer header)
host uses IP addresses amp port numbers to direct segment to appropriate socket
message
Transport Layer
Header
Application Layer
Message
Segm
ent ndash
Tran
sport L
ayer PD
U
Connectionless demultiplexing
Create sockets with port numbers
DatagramSocket mySocket1 = new DatagramSocket(9911)
DatagramSocket mySocket2 = new DatagramSocket(9922)
When host receives UDP segment
checks destination port number in segment
directs UDP segment to socket with that port
Transport Layer 3-8
DatagramSocket(9922)
UDP socket identified by two-tuple
(IP address port number)
socket with that port number
IP datagrams with different source IP addresses andor source port but same destination port are directed to same socket
Connectionless demux (cont)
P3 P1P1P3 P4A2145 C6428 C7527 B9825
A C
B C
Transport Layer 3-9
ClientIPB
clientIP A
serverIP C
msg
2145 6428
msg
A C
2145 7527
msg
B C
9825 7527
Connection-oriented demux
TCP socket identified by 4-tuple
source IP address
source port number
dest IP address
Server host may support many simultaneous TCP sockets
each socket identified by its own 4-tuple
Transport Layer 3-10
dest IP address
dest port number
recv host uses all four values to direct segment to appropriate socket
Web servers have different sockets for each connecting client
non-persistent HTTP will have different socket for each request
Connection-oriented demux (cont)
P3 P1P1P3C80
B9825
A C
B C
A2145C80
C80A2145
B9825C80
Transport Layer 3-11
ClientIPB
clientIP A
serverIP C
msg
2145 80
msg
B C
9825 80
UDP User Datagram Protocol [RFC 768]
ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol
ldquobest effortrdquo service UDP segments may be
lost
delivered out of order
Why is there a UDP no connection
establishment (which can add delay)
simple no connection state
Transport Layer 3-12
delivered out of order
to bare IP service UDP adds Muxdemux
checksum
connectionless
no handshaking between UDP sender receiver
each UDP segment handled independently of others
simple no connection state at sender receiver
small segment header
no congestion control UDP can blast away as fast as desired
UDP more
often used for streaming multimedia apps
loss tolerant
rate sensitive
other UDP uses
source port dest port
32 bits
length checksumLength in
bytes of UDPsegmentincludingheader
Transport Layer 3-13
DNS
SNMP
reliable transfer over UDP add reliability at application layer
application-specific error recovery
Applicationdata
(message)
UDP segment format
header
UDP checksum
Sender treat segment contents
as sequence of 16-bit
Receiver compute checksum of
received segment
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Transport Layer 3-14
as sequence of 16-bit integers
checksum addition (1rsquos complement sum) of segment contents
sender puts checksum value into UDP checksum field
received segment
check if computed checksum equals checksum field value
NO - error detected
YES - no error detected But maybe errors nonetheless More later hellip
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
1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1wraparound
Transport Layer 3-15
1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1
1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0
1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
wraparound
sum
checksum
Note when adding numbers a carryout from the most significant bit needs to be added to the result
Principles of Reliable data transfer
important in app transport link layers top-10 list of important networking topics
Transport Layer 3-16
characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Reliable data transfer getting started
send receive
rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer
deliver_data() called by rdt to deliver data to upper
Transport Layer 3-17
sendside
receiveside
udt_send() called by rdtto transfer packet over
unreliable channel to receiver
rdt_rcv() called when packet arrives on rcv-side of channel
Reliable data transfer getting started
Wersquoll
incrementally develop sender receiver sides of reliable data transfer protocol (rdt)
consider only unidirectional data transfer but control info will flow on both directions
Transport Layer 3-18
but control info will flow on both directions
use finite state machines (FSM) to specify sender receiver
state1
state2
event causing state transitionactions taken on state transition
state when in this ldquostaterdquo next state
uniquely determined by next event
eventactions
Rdt10 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
Transport Layer 3-19
sender sends data into underlying channel
receiver read data from underlying channel
Wait for call from above packet = make_pkt(data)
udt_send(packet)
rdt_send(data)
extract (packetdata)deliver_data(data)
Wait for call from
below
rdt_rcv(packet)
sender receiver
Rdt20 channel with bit errors
underlying channel may flip bits in packet recall UDP checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells sender
that pkt received OK
Transport Layer 3-20
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 rdt20 (beyond rdt10) error detection
receiver feedback control msgs (ACKNAK) rcvr-gtsender
rdt20 FSM specification
Wait for call from above
sndpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
receiverrdt_send(data)
Transport Layer 3-21
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowsender
Λ
rdt20 operation with no errors
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-22
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 error scenario
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-23
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
What to do
Handling duplicates sender adds sequence
number to each pkt
sender retransmits current pkt if ACKNAK garbled
receiver discards (doesnrsquot
Transport Layer 3-24
What to do sender ACKsNAKs
receiverrsquos ACKNAK What if sender ACKNAK lost
retransmit but this might cause retransmission of correctly received pkt
receiver discards (doesnrsquot deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21 sender handles garbled ACKNAKs
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
Wait for ACK or NAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
rdt_rcv(rcvpkt) rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
Transport Layer 3-25
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
Wait forcall 1 from
above
Wait for ACK or NAK 1
ΛΛ
rdt21 receiver handles garbled ACKNAKs
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
Transport Layer 3-26
Wait for 0 from below
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq0(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
Wait for 1 from below
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq1(rcvpkt)
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt21 discussion
Sender
seq added to pkt
two seq rsquos (01) will suffice Why
must check if received
Receiver
must check if received packet is duplicate
state indicates whether 0 or 1 is expected pkt
Transport Layer 3-27
must check if received ACKNAK corrupted
twice as many states state must ldquorememberrdquo
whether ldquocurrentrdquo pkt has 0 or 1 seq
0 or 1 is expected pkt seq
note receiver can notknow if its last ACKNAK received OK at sender
rdt22 a NAK-free protocol
same functionality as rdt21 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
Transport Layer 3-28
duplicate ACK at sender results in same action as NAK retransmit current pkt
rdt22 sender receiver fragments
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||
isACK(rcvpkt1) )
rdt_rcv(rcvpkt)
Wait for ACK
0
sender FSMfragment
Transport Layer 3-29
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
fragment
Wait for 0 from below
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSMfragment
Λ
rdt30 channels with errors and loss
New assumptionunderlying channel can also lose packets (data or ACKs)
checksum seq ACKs
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
Transport Layer 3-30
checksum seq ACKs retransmissions will be of help but not enough
Q how to deal with loss sender waits until
certain data or ACK lost then retransmits
yuck drawbacks
if pkt (or ACK) just delayed (not lost)
retransmission will be duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
rdt30 sendersndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
Wait for
ACK0
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait for call 0from
above
ΛΛ
Transport Layer 3-31
Wait for call 1 from
above
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )
ampamp isACK(rcvpkt1)
stop_timerstop_timer
udt_send(sndpkt)start_timer
timeoutWait for
ACK1
Λrdt_rcv(rcvpkt)
Λ
rdt30 in action
Transport Layer 3-32
rdt30 in action
Transport Layer 3-33
Performance of rdt30
rdt30 works but performance stinks
example 1 Gbps link 15 ms e-e prop delay 1KB packet
Ttransmit =
8kbpkt109 bsec
= 8 microsecL (packet length in bits)R (transmission rate bps)
=
Transport Layer 3-34
109 bsec
U sender utilization ndash fraction of time sender busy sending
1KB pkt every 30 msec -gt 267kbps thruput over 1 Gbps link
network protocol limits use of physical resources
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
R (transmission rate bps)
rdt30 stop-and-wait operation
first packet bit transmitted t = 0
sender receiver
RTT
last packet bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACK
Transport Layer 3-35
ACK arrives send next packet t = RTT + L R
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
Pipelined protocols
Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts
range of sequence numbers must be increased
buffering at sender andor receiver
Transport Layer 3-36
Two generic forms of pipelined protocols go-Back-N selective repeat
Pipelining increased utilization
first packet bit transmitted t = 0
sender receiver
RTT
last bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACKlast bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK
Transport Layer 3-37
ACK arrives send next packet t = RTT + L R
last bit of 3rd packet arrives send ACK
U sender
= 024
30024 = 00008
microsecon
3 L R
RTT +3 L =
Increase utilizationby a factor of 3
Pipelining Protocols
Go-back-N big picture Sender can have up to
N unacked packets in pipeline
Rcvr only sends cumulative acks
Selective Repeat big pic Sender can have up to
N unacked packets in pipeline
Rcvr acks individual packets
Transport Layer 3-38
Rcvr only sends cumulative acks
Doesnrsquot ack packet if therersquos a gap
Sender has timer for oldest unacked packet
If timer expires retransmit all unacked packets
Rcvr acks individual packets
Sender maintains timer for each unacked packet
When timer expires retransmit only unack packet
Go-Back-N
Trasmit multiple packets (up to N) without waiting for ACK
Sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
Transport Layer 3-39
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo
may receive duplicate ACKs (see receiver)
timeout(n) retransmit pkt n and all higher seq pkts in window
GBN sender extended FSM (1 timer)rdt_send(data)
if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum)udt_send(sndpkt[nextseqnum])if (base == nextseqnum)
start_timernextseqnum++
elserefuse_data(data)
base=1
Λ
Transport Layer 3-40
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])
timeout
base = getacknum(rcvpkt)+1If (base == nextseqnum)
stop_timerelsestart_timer
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Λ
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Internet transport-layer protocols
reliable in-order delivery (TCP)
congestion control
flow control
connection setup
applicationtransportnetworkdata linkphysical
networkdata link
networkdata linkphysical
networkdata linkphysicalnetwork
data linkphysical
Transport Layer 3-5
connection setup
unreliable unordered delivery UDP
no-frills extension of ldquobest-effortrdquo IP
services not available delay guarantees
bandwidth guarantees
applicationtransportnetworkdata linkphysical
networkdata linkphysical
data linkphysical
Multiplexingdemultiplexing
= process= socket
delivering received segmentsto correct socket
Demultiplexing at rcv hostgathering data from multiplesockets enveloping data with header (later used for demultiplexing)
Multiplexing at send host
Transport Layer 3-6
application
transport
network
link
physical
P1 application
transport
network
link
physical
application
transport
network
link
physical
P2P3 P4P1
host 1 host 2 host 3
transport
How demultiplexing works
host receives IP datagrams
each datagram has source IP address destination IP address (in the network layer header)
each datagram carries 1
Source IP addr
Dest IP addr
Source port
Dest portTransport Layer
Network Layer
Header
Segm
ent
Datagram ndash Network Layer PDU
Transport Layer 3-7
each datagram carries 1 transport-layer segment
each segment has source destination port number (in the transport layer header)
host uses IP addresses amp port numbers to direct segment to appropriate socket
message
Transport Layer
Header
Application Layer
Message
Segm
ent ndash
Tran
sport L
ayer PD
U
Connectionless demultiplexing
Create sockets with port numbers
DatagramSocket mySocket1 = new DatagramSocket(9911)
DatagramSocket mySocket2 = new DatagramSocket(9922)
When host receives UDP segment
checks destination port number in segment
directs UDP segment to socket with that port
Transport Layer 3-8
DatagramSocket(9922)
UDP socket identified by two-tuple
(IP address port number)
socket with that port number
IP datagrams with different source IP addresses andor source port but same destination port are directed to same socket
Connectionless demux (cont)
P3 P1P1P3 P4A2145 C6428 C7527 B9825
A C
B C
Transport Layer 3-9
ClientIPB
clientIP A
serverIP C
msg
2145 6428
msg
A C
2145 7527
msg
B C
9825 7527
Connection-oriented demux
TCP socket identified by 4-tuple
source IP address
source port number
dest IP address
Server host may support many simultaneous TCP sockets
each socket identified by its own 4-tuple
Transport Layer 3-10
dest IP address
dest port number
recv host uses all four values to direct segment to appropriate socket
Web servers have different sockets for each connecting client
non-persistent HTTP will have different socket for each request
Connection-oriented demux (cont)
P3 P1P1P3C80
B9825
A C
B C
A2145C80
C80A2145
B9825C80
Transport Layer 3-11
ClientIPB
clientIP A
serverIP C
msg
2145 80
msg
B C
9825 80
UDP User Datagram Protocol [RFC 768]
ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol
ldquobest effortrdquo service UDP segments may be
lost
delivered out of order
Why is there a UDP no connection
establishment (which can add delay)
simple no connection state
Transport Layer 3-12
delivered out of order
to bare IP service UDP adds Muxdemux
checksum
connectionless
no handshaking between UDP sender receiver
each UDP segment handled independently of others
simple no connection state at sender receiver
small segment header
no congestion control UDP can blast away as fast as desired
UDP more
often used for streaming multimedia apps
loss tolerant
rate sensitive
other UDP uses
source port dest port
32 bits
length checksumLength in
bytes of UDPsegmentincludingheader
Transport Layer 3-13
DNS
SNMP
reliable transfer over UDP add reliability at application layer
application-specific error recovery
Applicationdata
(message)
UDP segment format
header
UDP checksum
Sender treat segment contents
as sequence of 16-bit
Receiver compute checksum of
received segment
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Transport Layer 3-14
as sequence of 16-bit integers
checksum addition (1rsquos complement sum) of segment contents
sender puts checksum value into UDP checksum field
received segment
check if computed checksum equals checksum field value
NO - error detected
YES - no error detected But maybe errors nonetheless More later hellip
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
1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1wraparound
Transport Layer 3-15
1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1
1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0
1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
wraparound
sum
checksum
Note when adding numbers a carryout from the most significant bit needs to be added to the result
Principles of Reliable data transfer
important in app transport link layers top-10 list of important networking topics
Transport Layer 3-16
characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Reliable data transfer getting started
send receive
rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer
deliver_data() called by rdt to deliver data to upper
Transport Layer 3-17
sendside
receiveside
udt_send() called by rdtto transfer packet over
unreliable channel to receiver
rdt_rcv() called when packet arrives on rcv-side of channel
Reliable data transfer getting started
Wersquoll
incrementally develop sender receiver sides of reliable data transfer protocol (rdt)
consider only unidirectional data transfer but control info will flow on both directions
Transport Layer 3-18
but control info will flow on both directions
use finite state machines (FSM) to specify sender receiver
state1
state2
event causing state transitionactions taken on state transition
state when in this ldquostaterdquo next state
uniquely determined by next event
eventactions
Rdt10 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
Transport Layer 3-19
sender sends data into underlying channel
receiver read data from underlying channel
Wait for call from above packet = make_pkt(data)
udt_send(packet)
rdt_send(data)
extract (packetdata)deliver_data(data)
Wait for call from
below
rdt_rcv(packet)
sender receiver
Rdt20 channel with bit errors
underlying channel may flip bits in packet recall UDP checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells sender
that pkt received OK
Transport Layer 3-20
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 rdt20 (beyond rdt10) error detection
receiver feedback control msgs (ACKNAK) rcvr-gtsender
rdt20 FSM specification
Wait for call from above
sndpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
receiverrdt_send(data)
Transport Layer 3-21
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowsender
Λ
rdt20 operation with no errors
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-22
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 error scenario
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-23
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
What to do
Handling duplicates sender adds sequence
number to each pkt
sender retransmits current pkt if ACKNAK garbled
receiver discards (doesnrsquot
Transport Layer 3-24
What to do sender ACKsNAKs
receiverrsquos ACKNAK What if sender ACKNAK lost
retransmit but this might cause retransmission of correctly received pkt
receiver discards (doesnrsquot deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21 sender handles garbled ACKNAKs
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
Wait for ACK or NAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
rdt_rcv(rcvpkt) rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
Transport Layer 3-25
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
Wait forcall 1 from
above
Wait for ACK or NAK 1
ΛΛ
rdt21 receiver handles garbled ACKNAKs
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
Transport Layer 3-26
Wait for 0 from below
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq0(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
Wait for 1 from below
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq1(rcvpkt)
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt21 discussion
Sender
seq added to pkt
two seq rsquos (01) will suffice Why
must check if received
Receiver
must check if received packet is duplicate
state indicates whether 0 or 1 is expected pkt
Transport Layer 3-27
must check if received ACKNAK corrupted
twice as many states state must ldquorememberrdquo
whether ldquocurrentrdquo pkt has 0 or 1 seq
0 or 1 is expected pkt seq
note receiver can notknow if its last ACKNAK received OK at sender
rdt22 a NAK-free protocol
same functionality as rdt21 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
Transport Layer 3-28
duplicate ACK at sender results in same action as NAK retransmit current pkt
rdt22 sender receiver fragments
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||
isACK(rcvpkt1) )
rdt_rcv(rcvpkt)
Wait for ACK
0
sender FSMfragment
Transport Layer 3-29
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
fragment
Wait for 0 from below
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSMfragment
Λ
rdt30 channels with errors and loss
New assumptionunderlying channel can also lose packets (data or ACKs)
checksum seq ACKs
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
Transport Layer 3-30
checksum seq ACKs retransmissions will be of help but not enough
Q how to deal with loss sender waits until
certain data or ACK lost then retransmits
yuck drawbacks
if pkt (or ACK) just delayed (not lost)
retransmission will be duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
rdt30 sendersndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
Wait for
ACK0
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait for call 0from
above
ΛΛ
Transport Layer 3-31
Wait for call 1 from
above
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )
ampamp isACK(rcvpkt1)
stop_timerstop_timer
udt_send(sndpkt)start_timer
timeoutWait for
ACK1
Λrdt_rcv(rcvpkt)
Λ
rdt30 in action
Transport Layer 3-32
rdt30 in action
Transport Layer 3-33
Performance of rdt30
rdt30 works but performance stinks
example 1 Gbps link 15 ms e-e prop delay 1KB packet
Ttransmit =
8kbpkt109 bsec
= 8 microsecL (packet length in bits)R (transmission rate bps)
=
Transport Layer 3-34
109 bsec
U sender utilization ndash fraction of time sender busy sending
1KB pkt every 30 msec -gt 267kbps thruput over 1 Gbps link
network protocol limits use of physical resources
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
R (transmission rate bps)
rdt30 stop-and-wait operation
first packet bit transmitted t = 0
sender receiver
RTT
last packet bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACK
Transport Layer 3-35
ACK arrives send next packet t = RTT + L R
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
Pipelined protocols
Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts
range of sequence numbers must be increased
buffering at sender andor receiver
Transport Layer 3-36
Two generic forms of pipelined protocols go-Back-N selective repeat
Pipelining increased utilization
first packet bit transmitted t = 0
sender receiver
RTT
last bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACKlast bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK
Transport Layer 3-37
ACK arrives send next packet t = RTT + L R
last bit of 3rd packet arrives send ACK
U sender
= 024
30024 = 00008
microsecon
3 L R
RTT +3 L =
Increase utilizationby a factor of 3
Pipelining Protocols
Go-back-N big picture Sender can have up to
N unacked packets in pipeline
Rcvr only sends cumulative acks
Selective Repeat big pic Sender can have up to
N unacked packets in pipeline
Rcvr acks individual packets
Transport Layer 3-38
Rcvr only sends cumulative acks
Doesnrsquot ack packet if therersquos a gap
Sender has timer for oldest unacked packet
If timer expires retransmit all unacked packets
Rcvr acks individual packets
Sender maintains timer for each unacked packet
When timer expires retransmit only unack packet
Go-Back-N
Trasmit multiple packets (up to N) without waiting for ACK
Sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
Transport Layer 3-39
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo
may receive duplicate ACKs (see receiver)
timeout(n) retransmit pkt n and all higher seq pkts in window
GBN sender extended FSM (1 timer)rdt_send(data)
if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum)udt_send(sndpkt[nextseqnum])if (base == nextseqnum)
start_timernextseqnum++
elserefuse_data(data)
base=1
Λ
Transport Layer 3-40
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])
timeout
base = getacknum(rcvpkt)+1If (base == nextseqnum)
stop_timerelsestart_timer
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Λ
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Connectionless demux (cont)
P3 P1P1P3 P4A2145 C6428 C7527 B9825
A C
B C
Transport Layer 3-9
ClientIPB
clientIP A
serverIP C
msg
2145 6428
msg
A C
2145 7527
msg
B C
9825 7527
Connection-oriented demux
TCP socket identified by 4-tuple
source IP address
source port number
dest IP address
Server host may support many simultaneous TCP sockets
each socket identified by its own 4-tuple
Transport Layer 3-10
dest IP address
dest port number
recv host uses all four values to direct segment to appropriate socket
Web servers have different sockets for each connecting client
non-persistent HTTP will have different socket for each request
Connection-oriented demux (cont)
P3 P1P1P3C80
B9825
A C
B C
A2145C80
C80A2145
B9825C80
Transport Layer 3-11
ClientIPB
clientIP A
serverIP C
msg
2145 80
msg
B C
9825 80
UDP User Datagram Protocol [RFC 768]
ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol
ldquobest effortrdquo service UDP segments may be
lost
delivered out of order
Why is there a UDP no connection
establishment (which can add delay)
simple no connection state
Transport Layer 3-12
delivered out of order
to bare IP service UDP adds Muxdemux
checksum
connectionless
no handshaking between UDP sender receiver
each UDP segment handled independently of others
simple no connection state at sender receiver
small segment header
no congestion control UDP can blast away as fast as desired
UDP more
often used for streaming multimedia apps
loss tolerant
rate sensitive
other UDP uses
source port dest port
32 bits
length checksumLength in
bytes of UDPsegmentincludingheader
Transport Layer 3-13
DNS
SNMP
reliable transfer over UDP add reliability at application layer
application-specific error recovery
Applicationdata
(message)
UDP segment format
header
UDP checksum
Sender treat segment contents
as sequence of 16-bit
Receiver compute checksum of
received segment
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Transport Layer 3-14
as sequence of 16-bit integers
checksum addition (1rsquos complement sum) of segment contents
sender puts checksum value into UDP checksum field
received segment
check if computed checksum equals checksum field value
NO - error detected
YES - no error detected But maybe errors nonetheless More later hellip
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
1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1wraparound
Transport Layer 3-15
1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1
1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0
1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
wraparound
sum
checksum
Note when adding numbers a carryout from the most significant bit needs to be added to the result
Principles of Reliable data transfer
important in app transport link layers top-10 list of important networking topics
Transport Layer 3-16
characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Reliable data transfer getting started
send receive
rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer
deliver_data() called by rdt to deliver data to upper
Transport Layer 3-17
sendside
receiveside
udt_send() called by rdtto transfer packet over
unreliable channel to receiver
rdt_rcv() called when packet arrives on rcv-side of channel
Reliable data transfer getting started
Wersquoll
incrementally develop sender receiver sides of reliable data transfer protocol (rdt)
consider only unidirectional data transfer but control info will flow on both directions
Transport Layer 3-18
but control info will flow on both directions
use finite state machines (FSM) to specify sender receiver
state1
state2
event causing state transitionactions taken on state transition
state when in this ldquostaterdquo next state
uniquely determined by next event
eventactions
Rdt10 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
Transport Layer 3-19
sender sends data into underlying channel
receiver read data from underlying channel
Wait for call from above packet = make_pkt(data)
udt_send(packet)
rdt_send(data)
extract (packetdata)deliver_data(data)
Wait for call from
below
rdt_rcv(packet)
sender receiver
Rdt20 channel with bit errors
underlying channel may flip bits in packet recall UDP checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells sender
that pkt received OK
Transport Layer 3-20
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 rdt20 (beyond rdt10) error detection
receiver feedback control msgs (ACKNAK) rcvr-gtsender
rdt20 FSM specification
Wait for call from above
sndpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
receiverrdt_send(data)
Transport Layer 3-21
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowsender
Λ
rdt20 operation with no errors
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-22
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 error scenario
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-23
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
What to do
Handling duplicates sender adds sequence
number to each pkt
sender retransmits current pkt if ACKNAK garbled
receiver discards (doesnrsquot
Transport Layer 3-24
What to do sender ACKsNAKs
receiverrsquos ACKNAK What if sender ACKNAK lost
retransmit but this might cause retransmission of correctly received pkt
receiver discards (doesnrsquot deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21 sender handles garbled ACKNAKs
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
Wait for ACK or NAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
rdt_rcv(rcvpkt) rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
Transport Layer 3-25
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
Wait forcall 1 from
above
Wait for ACK or NAK 1
ΛΛ
rdt21 receiver handles garbled ACKNAKs
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
Transport Layer 3-26
Wait for 0 from below
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq0(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
Wait for 1 from below
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq1(rcvpkt)
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt21 discussion
Sender
seq added to pkt
two seq rsquos (01) will suffice Why
must check if received
Receiver
must check if received packet is duplicate
state indicates whether 0 or 1 is expected pkt
Transport Layer 3-27
must check if received ACKNAK corrupted
twice as many states state must ldquorememberrdquo
whether ldquocurrentrdquo pkt has 0 or 1 seq
0 or 1 is expected pkt seq
note receiver can notknow if its last ACKNAK received OK at sender
rdt22 a NAK-free protocol
same functionality as rdt21 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
Transport Layer 3-28
duplicate ACK at sender results in same action as NAK retransmit current pkt
rdt22 sender receiver fragments
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||
isACK(rcvpkt1) )
rdt_rcv(rcvpkt)
Wait for ACK
0
sender FSMfragment
Transport Layer 3-29
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
fragment
Wait for 0 from below
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSMfragment
Λ
rdt30 channels with errors and loss
New assumptionunderlying channel can also lose packets (data or ACKs)
checksum seq ACKs
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
Transport Layer 3-30
checksum seq ACKs retransmissions will be of help but not enough
Q how to deal with loss sender waits until
certain data or ACK lost then retransmits
yuck drawbacks
if pkt (or ACK) just delayed (not lost)
retransmission will be duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
rdt30 sendersndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
Wait for
ACK0
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait for call 0from
above
ΛΛ
Transport Layer 3-31
Wait for call 1 from
above
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )
ampamp isACK(rcvpkt1)
stop_timerstop_timer
udt_send(sndpkt)start_timer
timeoutWait for
ACK1
Λrdt_rcv(rcvpkt)
Λ
rdt30 in action
Transport Layer 3-32
rdt30 in action
Transport Layer 3-33
Performance of rdt30
rdt30 works but performance stinks
example 1 Gbps link 15 ms e-e prop delay 1KB packet
Ttransmit =
8kbpkt109 bsec
= 8 microsecL (packet length in bits)R (transmission rate bps)
=
Transport Layer 3-34
109 bsec
U sender utilization ndash fraction of time sender busy sending
1KB pkt every 30 msec -gt 267kbps thruput over 1 Gbps link
network protocol limits use of physical resources
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
R (transmission rate bps)
rdt30 stop-and-wait operation
first packet bit transmitted t = 0
sender receiver
RTT
last packet bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACK
Transport Layer 3-35
ACK arrives send next packet t = RTT + L R
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
Pipelined protocols
Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts
range of sequence numbers must be increased
buffering at sender andor receiver
Transport Layer 3-36
Two generic forms of pipelined protocols go-Back-N selective repeat
Pipelining increased utilization
first packet bit transmitted t = 0
sender receiver
RTT
last bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACKlast bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK
Transport Layer 3-37
ACK arrives send next packet t = RTT + L R
last bit of 3rd packet arrives send ACK
U sender
= 024
30024 = 00008
microsecon
3 L R
RTT +3 L =
Increase utilizationby a factor of 3
Pipelining Protocols
Go-back-N big picture Sender can have up to
N unacked packets in pipeline
Rcvr only sends cumulative acks
Selective Repeat big pic Sender can have up to
N unacked packets in pipeline
Rcvr acks individual packets
Transport Layer 3-38
Rcvr only sends cumulative acks
Doesnrsquot ack packet if therersquos a gap
Sender has timer for oldest unacked packet
If timer expires retransmit all unacked packets
Rcvr acks individual packets
Sender maintains timer for each unacked packet
When timer expires retransmit only unack packet
Go-Back-N
Trasmit multiple packets (up to N) without waiting for ACK
Sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
Transport Layer 3-39
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo
may receive duplicate ACKs (see receiver)
timeout(n) retransmit pkt n and all higher seq pkts in window
GBN sender extended FSM (1 timer)rdt_send(data)
if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum)udt_send(sndpkt[nextseqnum])if (base == nextseqnum)
start_timernextseqnum++
elserefuse_data(data)
base=1
Λ
Transport Layer 3-40
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])
timeout
base = getacknum(rcvpkt)+1If (base == nextseqnum)
stop_timerelsestart_timer
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Λ
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
UDP more
often used for streaming multimedia apps
loss tolerant
rate sensitive
other UDP uses
source port dest port
32 bits
length checksumLength in
bytes of UDPsegmentincludingheader
Transport Layer 3-13
DNS
SNMP
reliable transfer over UDP add reliability at application layer
application-specific error recovery
Applicationdata
(message)
UDP segment format
header
UDP checksum
Sender treat segment contents
as sequence of 16-bit
Receiver compute checksum of
received segment
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Transport Layer 3-14
as sequence of 16-bit integers
checksum addition (1rsquos complement sum) of segment contents
sender puts checksum value into UDP checksum field
received segment
check if computed checksum equals checksum field value
NO - error detected
YES - no error detected But maybe errors nonetheless More later hellip
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
1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1wraparound
Transport Layer 3-15
1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1
1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0
1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
wraparound
sum
checksum
Note when adding numbers a carryout from the most significant bit needs to be added to the result
Principles of Reliable data transfer
important in app transport link layers top-10 list of important networking topics
Transport Layer 3-16
characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Reliable data transfer getting started
send receive
rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer
deliver_data() called by rdt to deliver data to upper
Transport Layer 3-17
sendside
receiveside
udt_send() called by rdtto transfer packet over
unreliable channel to receiver
rdt_rcv() called when packet arrives on rcv-side of channel
Reliable data transfer getting started
Wersquoll
incrementally develop sender receiver sides of reliable data transfer protocol (rdt)
consider only unidirectional data transfer but control info will flow on both directions
Transport Layer 3-18
but control info will flow on both directions
use finite state machines (FSM) to specify sender receiver
state1
state2
event causing state transitionactions taken on state transition
state when in this ldquostaterdquo next state
uniquely determined by next event
eventactions
Rdt10 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
Transport Layer 3-19
sender sends data into underlying channel
receiver read data from underlying channel
Wait for call from above packet = make_pkt(data)
udt_send(packet)
rdt_send(data)
extract (packetdata)deliver_data(data)
Wait for call from
below
rdt_rcv(packet)
sender receiver
Rdt20 channel with bit errors
underlying channel may flip bits in packet recall UDP checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells sender
that pkt received OK
Transport Layer 3-20
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 rdt20 (beyond rdt10) error detection
receiver feedback control msgs (ACKNAK) rcvr-gtsender
rdt20 FSM specification
Wait for call from above
sndpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
receiverrdt_send(data)
Transport Layer 3-21
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowsender
Λ
rdt20 operation with no errors
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-22
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 error scenario
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-23
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
What to do
Handling duplicates sender adds sequence
number to each pkt
sender retransmits current pkt if ACKNAK garbled
receiver discards (doesnrsquot
Transport Layer 3-24
What to do sender ACKsNAKs
receiverrsquos ACKNAK What if sender ACKNAK lost
retransmit but this might cause retransmission of correctly received pkt
receiver discards (doesnrsquot deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21 sender handles garbled ACKNAKs
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
Wait for ACK or NAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
rdt_rcv(rcvpkt) rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
Transport Layer 3-25
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
Wait forcall 1 from
above
Wait for ACK or NAK 1
ΛΛ
rdt21 receiver handles garbled ACKNAKs
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
Transport Layer 3-26
Wait for 0 from below
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq0(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
Wait for 1 from below
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq1(rcvpkt)
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt21 discussion
Sender
seq added to pkt
two seq rsquos (01) will suffice Why
must check if received
Receiver
must check if received packet is duplicate
state indicates whether 0 or 1 is expected pkt
Transport Layer 3-27
must check if received ACKNAK corrupted
twice as many states state must ldquorememberrdquo
whether ldquocurrentrdquo pkt has 0 or 1 seq
0 or 1 is expected pkt seq
note receiver can notknow if its last ACKNAK received OK at sender
rdt22 a NAK-free protocol
same functionality as rdt21 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
Transport Layer 3-28
duplicate ACK at sender results in same action as NAK retransmit current pkt
rdt22 sender receiver fragments
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||
isACK(rcvpkt1) )
rdt_rcv(rcvpkt)
Wait for ACK
0
sender FSMfragment
Transport Layer 3-29
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
fragment
Wait for 0 from below
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSMfragment
Λ
rdt30 channels with errors and loss
New assumptionunderlying channel can also lose packets (data or ACKs)
checksum seq ACKs
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
Transport Layer 3-30
checksum seq ACKs retransmissions will be of help but not enough
Q how to deal with loss sender waits until
certain data or ACK lost then retransmits
yuck drawbacks
if pkt (or ACK) just delayed (not lost)
retransmission will be duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
rdt30 sendersndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
Wait for
ACK0
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait for call 0from
above
ΛΛ
Transport Layer 3-31
Wait for call 1 from
above
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )
ampamp isACK(rcvpkt1)
stop_timerstop_timer
udt_send(sndpkt)start_timer
timeoutWait for
ACK1
Λrdt_rcv(rcvpkt)
Λ
rdt30 in action
Transport Layer 3-32
rdt30 in action
Transport Layer 3-33
Performance of rdt30
rdt30 works but performance stinks
example 1 Gbps link 15 ms e-e prop delay 1KB packet
Ttransmit =
8kbpkt109 bsec
= 8 microsecL (packet length in bits)R (transmission rate bps)
=
Transport Layer 3-34
109 bsec
U sender utilization ndash fraction of time sender busy sending
1KB pkt every 30 msec -gt 267kbps thruput over 1 Gbps link
network protocol limits use of physical resources
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
R (transmission rate bps)
rdt30 stop-and-wait operation
first packet bit transmitted t = 0
sender receiver
RTT
last packet bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACK
Transport Layer 3-35
ACK arrives send next packet t = RTT + L R
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
Pipelined protocols
Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts
range of sequence numbers must be increased
buffering at sender andor receiver
Transport Layer 3-36
Two generic forms of pipelined protocols go-Back-N selective repeat
Pipelining increased utilization
first packet bit transmitted t = 0
sender receiver
RTT
last bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACKlast bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK
Transport Layer 3-37
ACK arrives send next packet t = RTT + L R
last bit of 3rd packet arrives send ACK
U sender
= 024
30024 = 00008
microsecon
3 L R
RTT +3 L =
Increase utilizationby a factor of 3
Pipelining Protocols
Go-back-N big picture Sender can have up to
N unacked packets in pipeline
Rcvr only sends cumulative acks
Selective Repeat big pic Sender can have up to
N unacked packets in pipeline
Rcvr acks individual packets
Transport Layer 3-38
Rcvr only sends cumulative acks
Doesnrsquot ack packet if therersquos a gap
Sender has timer for oldest unacked packet
If timer expires retransmit all unacked packets
Rcvr acks individual packets
Sender maintains timer for each unacked packet
When timer expires retransmit only unack packet
Go-Back-N
Trasmit multiple packets (up to N) without waiting for ACK
Sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
Transport Layer 3-39
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo
may receive duplicate ACKs (see receiver)
timeout(n) retransmit pkt n and all higher seq pkts in window
GBN sender extended FSM (1 timer)rdt_send(data)
if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum)udt_send(sndpkt[nextseqnum])if (base == nextseqnum)
start_timernextseqnum++
elserefuse_data(data)
base=1
Λ
Transport Layer 3-40
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])
timeout
base = getacknum(rcvpkt)+1If (base == nextseqnum)
stop_timerelsestart_timer
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Λ
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Reliable data transfer getting started
send receive
rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer
deliver_data() called by rdt to deliver data to upper
Transport Layer 3-17
sendside
receiveside
udt_send() called by rdtto transfer packet over
unreliable channel to receiver
rdt_rcv() called when packet arrives on rcv-side of channel
Reliable data transfer getting started
Wersquoll
incrementally develop sender receiver sides of reliable data transfer protocol (rdt)
consider only unidirectional data transfer but control info will flow on both directions
Transport Layer 3-18
but control info will flow on both directions
use finite state machines (FSM) to specify sender receiver
state1
state2
event causing state transitionactions taken on state transition
state when in this ldquostaterdquo next state
uniquely determined by next event
eventactions
Rdt10 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
Transport Layer 3-19
sender sends data into underlying channel
receiver read data from underlying channel
Wait for call from above packet = make_pkt(data)
udt_send(packet)
rdt_send(data)
extract (packetdata)deliver_data(data)
Wait for call from
below
rdt_rcv(packet)
sender receiver
Rdt20 channel with bit errors
underlying channel may flip bits in packet recall UDP checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells sender
that pkt received OK
Transport Layer 3-20
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 rdt20 (beyond rdt10) error detection
receiver feedback control msgs (ACKNAK) rcvr-gtsender
rdt20 FSM specification
Wait for call from above
sndpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
receiverrdt_send(data)
Transport Layer 3-21
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowsender
Λ
rdt20 operation with no errors
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-22
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 error scenario
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-23
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
What to do
Handling duplicates sender adds sequence
number to each pkt
sender retransmits current pkt if ACKNAK garbled
receiver discards (doesnrsquot
Transport Layer 3-24
What to do sender ACKsNAKs
receiverrsquos ACKNAK What if sender ACKNAK lost
retransmit but this might cause retransmission of correctly received pkt
receiver discards (doesnrsquot deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21 sender handles garbled ACKNAKs
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
Wait for ACK or NAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
rdt_rcv(rcvpkt) rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
Transport Layer 3-25
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
Wait forcall 1 from
above
Wait for ACK or NAK 1
ΛΛ
rdt21 receiver handles garbled ACKNAKs
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
Transport Layer 3-26
Wait for 0 from below
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq0(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
Wait for 1 from below
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq1(rcvpkt)
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt21 discussion
Sender
seq added to pkt
two seq rsquos (01) will suffice Why
must check if received
Receiver
must check if received packet is duplicate
state indicates whether 0 or 1 is expected pkt
Transport Layer 3-27
must check if received ACKNAK corrupted
twice as many states state must ldquorememberrdquo
whether ldquocurrentrdquo pkt has 0 or 1 seq
0 or 1 is expected pkt seq
note receiver can notknow if its last ACKNAK received OK at sender
rdt22 a NAK-free protocol
same functionality as rdt21 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
Transport Layer 3-28
duplicate ACK at sender results in same action as NAK retransmit current pkt
rdt22 sender receiver fragments
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||
isACK(rcvpkt1) )
rdt_rcv(rcvpkt)
Wait for ACK
0
sender FSMfragment
Transport Layer 3-29
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
fragment
Wait for 0 from below
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSMfragment
Λ
rdt30 channels with errors and loss
New assumptionunderlying channel can also lose packets (data or ACKs)
checksum seq ACKs
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
Transport Layer 3-30
checksum seq ACKs retransmissions will be of help but not enough
Q how to deal with loss sender waits until
certain data or ACK lost then retransmits
yuck drawbacks
if pkt (or ACK) just delayed (not lost)
retransmission will be duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
rdt30 sendersndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
Wait for
ACK0
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait for call 0from
above
ΛΛ
Transport Layer 3-31
Wait for call 1 from
above
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )
ampamp isACK(rcvpkt1)
stop_timerstop_timer
udt_send(sndpkt)start_timer
timeoutWait for
ACK1
Λrdt_rcv(rcvpkt)
Λ
rdt30 in action
Transport Layer 3-32
rdt30 in action
Transport Layer 3-33
Performance of rdt30
rdt30 works but performance stinks
example 1 Gbps link 15 ms e-e prop delay 1KB packet
Ttransmit =
8kbpkt109 bsec
= 8 microsecL (packet length in bits)R (transmission rate bps)
=
Transport Layer 3-34
109 bsec
U sender utilization ndash fraction of time sender busy sending
1KB pkt every 30 msec -gt 267kbps thruput over 1 Gbps link
network protocol limits use of physical resources
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
R (transmission rate bps)
rdt30 stop-and-wait operation
first packet bit transmitted t = 0
sender receiver
RTT
last packet bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACK
Transport Layer 3-35
ACK arrives send next packet t = RTT + L R
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
Pipelined protocols
Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts
range of sequence numbers must be increased
buffering at sender andor receiver
Transport Layer 3-36
Two generic forms of pipelined protocols go-Back-N selective repeat
Pipelining increased utilization
first packet bit transmitted t = 0
sender receiver
RTT
last bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACKlast bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK
Transport Layer 3-37
ACK arrives send next packet t = RTT + L R
last bit of 3rd packet arrives send ACK
U sender
= 024
30024 = 00008
microsecon
3 L R
RTT +3 L =
Increase utilizationby a factor of 3
Pipelining Protocols
Go-back-N big picture Sender can have up to
N unacked packets in pipeline
Rcvr only sends cumulative acks
Selective Repeat big pic Sender can have up to
N unacked packets in pipeline
Rcvr acks individual packets
Transport Layer 3-38
Rcvr only sends cumulative acks
Doesnrsquot ack packet if therersquos a gap
Sender has timer for oldest unacked packet
If timer expires retransmit all unacked packets
Rcvr acks individual packets
Sender maintains timer for each unacked packet
When timer expires retransmit only unack packet
Go-Back-N
Trasmit multiple packets (up to N) without waiting for ACK
Sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
Transport Layer 3-39
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo
may receive duplicate ACKs (see receiver)
timeout(n) retransmit pkt n and all higher seq pkts in window
GBN sender extended FSM (1 timer)rdt_send(data)
if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum)udt_send(sndpkt[nextseqnum])if (base == nextseqnum)
start_timernextseqnum++
elserefuse_data(data)
base=1
Λ
Transport Layer 3-40
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])
timeout
base = getacknum(rcvpkt)+1If (base == nextseqnum)
stop_timerelsestart_timer
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Λ
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
rdt20 FSM specification
Wait for call from above
sndpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
receiverrdt_send(data)
Transport Layer 3-21
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowsender
Λ
rdt20 operation with no errors
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-22
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 error scenario
Wait for call from above
snkpkt = make_pkt(data checksum)udt_send(sndpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampampisNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Wait for ACK or
NAK
rdt_send(data)
Transport Layer 3-23
extract(rcvpktdata)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)Wait for call from
belowΛ
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
What to do
Handling duplicates sender adds sequence
number to each pkt
sender retransmits current pkt if ACKNAK garbled
receiver discards (doesnrsquot
Transport Layer 3-24
What to do sender ACKsNAKs
receiverrsquos ACKNAK What if sender ACKNAK lost
retransmit but this might cause retransmission of correctly received pkt
receiver discards (doesnrsquot deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21 sender handles garbled ACKNAKs
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
Wait for ACK or NAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
rdt_rcv(rcvpkt) rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
Transport Layer 3-25
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
Wait forcall 1 from
above
Wait for ACK or NAK 1
ΛΛ
rdt21 receiver handles garbled ACKNAKs
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
Transport Layer 3-26
Wait for 0 from below
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq0(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
Wait for 1 from below
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq1(rcvpkt)
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt21 discussion
Sender
seq added to pkt
two seq rsquos (01) will suffice Why
must check if received
Receiver
must check if received packet is duplicate
state indicates whether 0 or 1 is expected pkt
Transport Layer 3-27
must check if received ACKNAK corrupted
twice as many states state must ldquorememberrdquo
whether ldquocurrentrdquo pkt has 0 or 1 seq
0 or 1 is expected pkt seq
note receiver can notknow if its last ACKNAK received OK at sender
rdt22 a NAK-free protocol
same functionality as rdt21 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
Transport Layer 3-28
duplicate ACK at sender results in same action as NAK retransmit current pkt
rdt22 sender receiver fragments
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||
isACK(rcvpkt1) )
rdt_rcv(rcvpkt)
Wait for ACK
0
sender FSMfragment
Transport Layer 3-29
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
fragment
Wait for 0 from below
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSMfragment
Λ
rdt30 channels with errors and loss
New assumptionunderlying channel can also lose packets (data or ACKs)
checksum seq ACKs
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
Transport Layer 3-30
checksum seq ACKs retransmissions will be of help but not enough
Q how to deal with loss sender waits until
certain data or ACK lost then retransmits
yuck drawbacks
if pkt (or ACK) just delayed (not lost)
retransmission will be duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
rdt30 sendersndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
Wait for
ACK0
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait for call 0from
above
ΛΛ
Transport Layer 3-31
Wait for call 1 from
above
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )
ampamp isACK(rcvpkt1)
stop_timerstop_timer
udt_send(sndpkt)start_timer
timeoutWait for
ACK1
Λrdt_rcv(rcvpkt)
Λ
rdt30 in action
Transport Layer 3-32
rdt30 in action
Transport Layer 3-33
Performance of rdt30
rdt30 works but performance stinks
example 1 Gbps link 15 ms e-e prop delay 1KB packet
Ttransmit =
8kbpkt109 bsec
= 8 microsecL (packet length in bits)R (transmission rate bps)
=
Transport Layer 3-34
109 bsec
U sender utilization ndash fraction of time sender busy sending
1KB pkt every 30 msec -gt 267kbps thruput over 1 Gbps link
network protocol limits use of physical resources
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
R (transmission rate bps)
rdt30 stop-and-wait operation
first packet bit transmitted t = 0
sender receiver
RTT
last packet bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACK
Transport Layer 3-35
ACK arrives send next packet t = RTT + L R
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
Pipelined protocols
Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts
range of sequence numbers must be increased
buffering at sender andor receiver
Transport Layer 3-36
Two generic forms of pipelined protocols go-Back-N selective repeat
Pipelining increased utilization
first packet bit transmitted t = 0
sender receiver
RTT
last bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACKlast bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK
Transport Layer 3-37
ACK arrives send next packet t = RTT + L R
last bit of 3rd packet arrives send ACK
U sender
= 024
30024 = 00008
microsecon
3 L R
RTT +3 L =
Increase utilizationby a factor of 3
Pipelining Protocols
Go-back-N big picture Sender can have up to
N unacked packets in pipeline
Rcvr only sends cumulative acks
Selective Repeat big pic Sender can have up to
N unacked packets in pipeline
Rcvr acks individual packets
Transport Layer 3-38
Rcvr only sends cumulative acks
Doesnrsquot ack packet if therersquos a gap
Sender has timer for oldest unacked packet
If timer expires retransmit all unacked packets
Rcvr acks individual packets
Sender maintains timer for each unacked packet
When timer expires retransmit only unack packet
Go-Back-N
Trasmit multiple packets (up to N) without waiting for ACK
Sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
Transport Layer 3-39
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo
may receive duplicate ACKs (see receiver)
timeout(n) retransmit pkt n and all higher seq pkts in window
GBN sender extended FSM (1 timer)rdt_send(data)
if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum)udt_send(sndpkt[nextseqnum])if (base == nextseqnum)
start_timernextseqnum++
elserefuse_data(data)
base=1
Λ
Transport Layer 3-40
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])
timeout
base = getacknum(rcvpkt)+1If (base == nextseqnum)
stop_timerelsestart_timer
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Λ
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
rdt21 sender handles garbled ACKNAKs
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
Wait for ACK or NAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
rdt_rcv(rcvpkt) rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
Transport Layer 3-25
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)
Wait forcall 1 from
above
Wait for ACK or NAK 1
ΛΛ
rdt21 receiver handles garbled ACKNAKs
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)
sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)
Transport Layer 3-26
Wait for 0 from below
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq0(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
Wait for 1 from below
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamphas_seq1(rcvpkt)
sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)
rdt21 discussion
Sender
seq added to pkt
two seq rsquos (01) will suffice Why
must check if received
Receiver
must check if received packet is duplicate
state indicates whether 0 or 1 is expected pkt
Transport Layer 3-27
must check if received ACKNAK corrupted
twice as many states state must ldquorememberrdquo
whether ldquocurrentrdquo pkt has 0 or 1 seq
0 or 1 is expected pkt seq
note receiver can notknow if its last ACKNAK received OK at sender
rdt22 a NAK-free protocol
same functionality as rdt21 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
Transport Layer 3-28
duplicate ACK at sender results in same action as NAK retransmit current pkt
rdt22 sender receiver fragments
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||
isACK(rcvpkt1) )
rdt_rcv(rcvpkt)
Wait for ACK
0
sender FSMfragment
Transport Layer 3-29
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
fragment
Wait for 0 from below
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSMfragment
Λ
rdt30 channels with errors and loss
New assumptionunderlying channel can also lose packets (data or ACKs)
checksum seq ACKs
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
Transport Layer 3-30
checksum seq ACKs retransmissions will be of help but not enough
Q how to deal with loss sender waits until
certain data or ACK lost then retransmits
yuck drawbacks
if pkt (or ACK) just delayed (not lost)
retransmission will be duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
rdt30 sendersndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
Wait for
ACK0
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait for call 0from
above
ΛΛ
Transport Layer 3-31
Wait for call 1 from
above
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )
ampamp isACK(rcvpkt1)
stop_timerstop_timer
udt_send(sndpkt)start_timer
timeoutWait for
ACK1
Λrdt_rcv(rcvpkt)
Λ
rdt30 in action
Transport Layer 3-32
rdt30 in action
Transport Layer 3-33
Performance of rdt30
rdt30 works but performance stinks
example 1 Gbps link 15 ms e-e prop delay 1KB packet
Ttransmit =
8kbpkt109 bsec
= 8 microsecL (packet length in bits)R (transmission rate bps)
=
Transport Layer 3-34
109 bsec
U sender utilization ndash fraction of time sender busy sending
1KB pkt every 30 msec -gt 267kbps thruput over 1 Gbps link
network protocol limits use of physical resources
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
R (transmission rate bps)
rdt30 stop-and-wait operation
first packet bit transmitted t = 0
sender receiver
RTT
last packet bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACK
Transport Layer 3-35
ACK arrives send next packet t = RTT + L R
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
Pipelined protocols
Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts
range of sequence numbers must be increased
buffering at sender andor receiver
Transport Layer 3-36
Two generic forms of pipelined protocols go-Back-N selective repeat
Pipelining increased utilization
first packet bit transmitted t = 0
sender receiver
RTT
last bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACKlast bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK
Transport Layer 3-37
ACK arrives send next packet t = RTT + L R
last bit of 3rd packet arrives send ACK
U sender
= 024
30024 = 00008
microsecon
3 L R
RTT +3 L =
Increase utilizationby a factor of 3
Pipelining Protocols
Go-back-N big picture Sender can have up to
N unacked packets in pipeline
Rcvr only sends cumulative acks
Selective Repeat big pic Sender can have up to
N unacked packets in pipeline
Rcvr acks individual packets
Transport Layer 3-38
Rcvr only sends cumulative acks
Doesnrsquot ack packet if therersquos a gap
Sender has timer for oldest unacked packet
If timer expires retransmit all unacked packets
Rcvr acks individual packets
Sender maintains timer for each unacked packet
When timer expires retransmit only unack packet
Go-Back-N
Trasmit multiple packets (up to N) without waiting for ACK
Sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
Transport Layer 3-39
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo
may receive duplicate ACKs (see receiver)
timeout(n) retransmit pkt n and all higher seq pkts in window
GBN sender extended FSM (1 timer)rdt_send(data)
if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum)udt_send(sndpkt[nextseqnum])if (base == nextseqnum)
start_timernextseqnum++
elserefuse_data(data)
base=1
Λ
Transport Layer 3-40
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])
timeout
base = getacknum(rcvpkt)+1If (base == nextseqnum)
stop_timerelsestart_timer
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Λ
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
rdt22 sender receiver fragments
Wait for call 0 from
above
sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||
isACK(rcvpkt1) )
rdt_rcv(rcvpkt)
Wait for ACK
0
sender FSMfragment
Transport Layer 3-29
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
fragment
Wait for 0 from below
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK1 chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
receiver FSMfragment
Λ
rdt30 channels with errors and loss
New assumptionunderlying channel can also lose packets (data or ACKs)
checksum seq ACKs
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
Transport Layer 3-30
checksum seq ACKs retransmissions will be of help but not enough
Q how to deal with loss sender waits until
certain data or ACK lost then retransmits
yuck drawbacks
if pkt (or ACK) just delayed (not lost)
retransmission will be duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
rdt30 sendersndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
Wait for
ACK0
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait for call 0from
above
ΛΛ
Transport Layer 3-31
Wait for call 1 from
above
sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer
rdt_send(data)
ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)
rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )
ampamp isACK(rcvpkt1)
stop_timerstop_timer
udt_send(sndpkt)start_timer
timeoutWait for
ACK1
Λrdt_rcv(rcvpkt)
Λ
rdt30 in action
Transport Layer 3-32
rdt30 in action
Transport Layer 3-33
Performance of rdt30
rdt30 works but performance stinks
example 1 Gbps link 15 ms e-e prop delay 1KB packet
Ttransmit =
8kbpkt109 bsec
= 8 microsecL (packet length in bits)R (transmission rate bps)
=
Transport Layer 3-34
109 bsec
U sender utilization ndash fraction of time sender busy sending
1KB pkt every 30 msec -gt 267kbps thruput over 1 Gbps link
network protocol limits use of physical resources
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
R (transmission rate bps)
rdt30 stop-and-wait operation
first packet bit transmitted t = 0
sender receiver
RTT
last packet bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACK
Transport Layer 3-35
ACK arrives send next packet t = RTT + L R
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
Pipelined protocols
Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts
range of sequence numbers must be increased
buffering at sender andor receiver
Transport Layer 3-36
Two generic forms of pipelined protocols go-Back-N selective repeat
Pipelining increased utilization
first packet bit transmitted t = 0
sender receiver
RTT
last bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACKlast bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK
Transport Layer 3-37
ACK arrives send next packet t = RTT + L R
last bit of 3rd packet arrives send ACK
U sender
= 024
30024 = 00008
microsecon
3 L R
RTT +3 L =
Increase utilizationby a factor of 3
Pipelining Protocols
Go-back-N big picture Sender can have up to
N unacked packets in pipeline
Rcvr only sends cumulative acks
Selective Repeat big pic Sender can have up to
N unacked packets in pipeline
Rcvr acks individual packets
Transport Layer 3-38
Rcvr only sends cumulative acks
Doesnrsquot ack packet if therersquos a gap
Sender has timer for oldest unacked packet
If timer expires retransmit all unacked packets
Rcvr acks individual packets
Sender maintains timer for each unacked packet
When timer expires retransmit only unack packet
Go-Back-N
Trasmit multiple packets (up to N) without waiting for ACK
Sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
Transport Layer 3-39
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo
may receive duplicate ACKs (see receiver)
timeout(n) retransmit pkt n and all higher seq pkts in window
GBN sender extended FSM (1 timer)rdt_send(data)
if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum)udt_send(sndpkt[nextseqnum])if (base == nextseqnum)
start_timernextseqnum++
elserefuse_data(data)
base=1
Λ
Transport Layer 3-40
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])
timeout
base = getacknum(rcvpkt)+1If (base == nextseqnum)
stop_timerelsestart_timer
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Λ
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
rdt30 in action
Transport Layer 3-33
Performance of rdt30
rdt30 works but performance stinks
example 1 Gbps link 15 ms e-e prop delay 1KB packet
Ttransmit =
8kbpkt109 bsec
= 8 microsecL (packet length in bits)R (transmission rate bps)
=
Transport Layer 3-34
109 bsec
U sender utilization ndash fraction of time sender busy sending
1KB pkt every 30 msec -gt 267kbps thruput over 1 Gbps link
network protocol limits use of physical resources
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
R (transmission rate bps)
rdt30 stop-and-wait operation
first packet bit transmitted t = 0
sender receiver
RTT
last packet bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACK
Transport Layer 3-35
ACK arrives send next packet t = RTT + L R
U sender
= 008
30008 = 000027
microsec
L R
RTT + L R =
Pipelined protocols
Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts
range of sequence numbers must be increased
buffering at sender andor receiver
Transport Layer 3-36
Two generic forms of pipelined protocols go-Back-N selective repeat
Pipelining increased utilization
first packet bit transmitted t = 0
sender receiver
RTT
last bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACKlast bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK
Transport Layer 3-37
ACK arrives send next packet t = RTT + L R
last bit of 3rd packet arrives send ACK
U sender
= 024
30024 = 00008
microsecon
3 L R
RTT +3 L =
Increase utilizationby a factor of 3
Pipelining Protocols
Go-back-N big picture Sender can have up to
N unacked packets in pipeline
Rcvr only sends cumulative acks
Selective Repeat big pic Sender can have up to
N unacked packets in pipeline
Rcvr acks individual packets
Transport Layer 3-38
Rcvr only sends cumulative acks
Doesnrsquot ack packet if therersquos a gap
Sender has timer for oldest unacked packet
If timer expires retransmit all unacked packets
Rcvr acks individual packets
Sender maintains timer for each unacked packet
When timer expires retransmit only unack packet
Go-Back-N
Trasmit multiple packets (up to N) without waiting for ACK
Sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
Transport Layer 3-39
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo
may receive duplicate ACKs (see receiver)
timeout(n) retransmit pkt n and all higher seq pkts in window
GBN sender extended FSM (1 timer)rdt_send(data)
if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum)udt_send(sndpkt[nextseqnum])if (base == nextseqnum)
start_timernextseqnum++
elserefuse_data(data)
base=1
Λ
Transport Layer 3-40
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])
timeout
base = getacknum(rcvpkt)+1If (base == nextseqnum)
stop_timerelsestart_timer
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Λ
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Pipelining increased utilization
first packet bit transmitted t = 0
sender receiver
RTT
last bit transmitted t = L R
first packet bit arriveslast packet bit arrives send ACKlast bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK
Transport Layer 3-37
ACK arrives send next packet t = RTT + L R
last bit of 3rd packet arrives send ACK
U sender
= 024
30024 = 00008
microsecon
3 L R
RTT +3 L =
Increase utilizationby a factor of 3
Pipelining Protocols
Go-back-N big picture Sender can have up to
N unacked packets in pipeline
Rcvr only sends cumulative acks
Selective Repeat big pic Sender can have up to
N unacked packets in pipeline
Rcvr acks individual packets
Transport Layer 3-38
Rcvr only sends cumulative acks
Doesnrsquot ack packet if therersquos a gap
Sender has timer for oldest unacked packet
If timer expires retransmit all unacked packets
Rcvr acks individual packets
Sender maintains timer for each unacked packet
When timer expires retransmit only unack packet
Go-Back-N
Trasmit multiple packets (up to N) without waiting for ACK
Sender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
Transport Layer 3-39
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo
may receive duplicate ACKs (see receiver)
timeout(n) retransmit pkt n and all higher seq pkts in window
GBN sender extended FSM (1 timer)rdt_send(data)
if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum)udt_send(sndpkt[nextseqnum])if (base == nextseqnum)
start_timernextseqnum++
elserefuse_data(data)
base=1
Λ
Transport Layer 3-40
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])
timeout
base = getacknum(rcvpkt)+1If (base == nextseqnum)
stop_timerelsestart_timer
rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)
Λ
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
GBN receiver extended FSM
Wait
udt_send(sndpkt)
default
rdt_rcv(rcvpkt)ampamp notcurrupt(rcvpkt)ampamp hasseqnum(rcvpktexpectedseqnum)
extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0ACKchksum)
Λ
Transport Layer 3-41
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 (donrsquot buffer) -gt no receiver buffering
Re-ACK pkt with highest in-order seq
GBN inaction
Transport Layer 3-42
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
Transport Layer 3-43
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window N consecutive seq rsquos
again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Transport Layer 3-44
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) resend pkt n restart timer
senderpkt n in [rcvbase rcvbase+N-1] send ACK(n)
out-of-order buffer
in-order deliver (also deliver buffered in-order pkts) advance window to
receiver
Transport Layer 3-45
resend pkt n restart timer
ACK(n) in [sendbasesendbase+N] mark pkt n as received
if n smallest unACKed pkt advance window base to next unACKed seq
pkts) advance window to next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1] ACK(n)
otherwise ignore
Selective repeat in action
Transport Layer 3-46
Selective repeatdilemma
Example seq rsquos 0 1 2 3
window size=3
receiver sees no
Transport Layer 3-47
receiver sees no difference in two scenarios
incorrectly passes duplicate data as new in (a)
Q what relationship between seq size and window size
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data bi-directional data flow
in same connection
MSS maximum segment size
connection-oriented
point-to-point one sender one receiver
reliable in-order byte steam
no ldquomessage boundariesrdquo
Transport Layer 3-48
connection-oriented handshaking (exchange
of control msgs) initrsquos sender receiver state before data exchange
flow controlled sender will not
overwhelm receiver
no ldquomessage boundariesrdquo
pipelined TCP congestion and flow
control set window size
send amp receive buffers
socketdoor
TCPsend buffer
TCPreceive buffer
socketdoor
segment
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
TCP segment structure
source port dest port
32 bits
sequence number
acknowledgement number
Receive window
Urg data pnterchecksum
FSRPAUheadlen
notused
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used) bytes
rcvr willing
countingby bytes of data(not segments)
Transport Layer 3-49
applicationdata
(variable length)
Urg data pnterchecksum
Options (variable length)RST SYN FINconnection estab(setup teardown
commands)
rcvr willingto accept
Internetchecksum
(as in UDP)
Sequence and Acknowledgement Numbers
TCP views data as unstructured but ordered data
In a segment
Sequence number is the byte-stream number of the first byte in the segment
Initial sequence number is randomly chosen
Transport Layer 3-50
Initial sequence number is randomly chosen
Ack number is the number of the next byte expected from the other side
TCP uses cumulative acknowledgements
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
TCP seq numbers ACKs
sequence numbers
byte stream ldquonumberrdquoof first byte in segmentrsquos data
acknowledgements
seq of next byte
window size
N
source port dest port
sequence number
acknowledgement number
checksum
rwnd
urg pointer
outgoing segment from sender
Transport Layer 3-51
seq of next byte expected from other side
cumulative ACK
Q how receiver handles out-of-order segments
A TCP spec doesnrsquot say - up to implementor
source port dest port
sequence number
acknowledgement number
checksum
rwndurg pointer
incoming segment to sender
A
sent
ACKed
sent not-yet ACKed
(ldquoin-flightrdquo)
usable
but not
yet sent
not
usable
sender sequence number space
TCP seq rsquos and ACKs
Host A Host B
UsertypeslsquoCrsquo
host ACKsreceipt oflsquoCrsquo echoesback lsquoCrsquo
Transport Layer 3-52
host ACKsreceipt of echoed
lsquoCrsquo
back lsquoCrsquo
timesimple telnet scenario
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
TCP reliable data transfer
TCP creates rdt service on top of IPrsquos unreliable service
Pipelined segments
Cumulative acks
Retransmissions are triggered by
timeout events
duplicate acks
Initially consider
Transport Layer 3-53
Cumulative acks
TCP should use a single retransmission timer
Initially consider simplified TCP sender
ignore duplicate acks
ignore flow control congestion control
TCP sender events
data rcvd from app
Create segment with seq
seq is byte-stream number of first data
timeout
retransmit segment that caused timeout
restart timer
Ack rcvd
Transport Layer 3-54
number of first data byte in segment
start timer for that segment
expiration interval TimeOutInterval
Ack rcvd
If acknowledges previously unacked segments
update what is known to be acked
TCP sender(simplified)
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
loop (forever) switch(event)
event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running)
start timerpass segment to IP NextSeqNum = NextSeqNum + length(data) break
Transport Layer 3-55
breakevent timer timeout
retransmit not-yet acked segment with smallest sequence number
start timer break
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timerbreak
end of loop forever
TCP sender (simplified)
wait
for NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
Λ
create segment seq NextSeqNum
pass segment to IP (ie ldquosendrdquo)
NextSeqNum = NextSeqNum + length(data)
if (timer currently not running)
start timer
data received from application above
Transport Layer 3-56
eventSendBase = InitialSeqNum
retransmit not-yet-acked segment with smallest seq
start timer
timeout
if (y gt SendBase)
SendBase = y
SendBasendash1 last cumulatively ACKed byte
if (there are currently not-yet-acked segments)
start timer
else stop timer
ACK received with ACK field value y
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
TCP retransmission scenariosHost A Host B
Seq=9
2 t
imeout
Host A
loss
tim
eout
Host B
X
Seq=1
00
tim
eout
Transport Layer 3-57
timepremature timeout
Seq=9
2 t
imeout
loss
lost ACK scenariotime
Seq=1
00
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
loss
tim
eout
Host B
X
Transport Layer 3-58
loss
Cumulative ACK scenariotime
SendBase= 120
TCP ACK generation [RFC 1122 RFC 2581]
Event at Receiver
Arrival of in-order segment withexpected seq All data up toexpected seq already ACKed
Arrival of in-order segment with
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
Immediately send single cumulative
Transport Layer 3-59
Arrival of in-order segment withexpected seq One other segment has ACK pending
Arrival of out-of-order segmenthigher-than-expect seq Gap detected
Arrival of segment that partially or completely fills gap
Immediately send single cumulative ACK ACKing both in-order segments
Immediately send duplicate ACK indicating seq of next expected byte
Immediate send ACK provided thatsegment starts at lower end of gap
Fast Retransmit
Time-out period often relatively long
long delay before resending lost packet
Detect lost segments
If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost
Transport Layer 3-60
Detect lost segments via duplicate ACKs
Sender often sends many segments back-to-back
If segment is lost there will likely be many duplicate ACKs
data was lost fast retransmit resend
segment before timer expires
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
event ACK received with ACK field value of y if (y gt SendBase)
Sendbase=yif( there are currently any not yet acked segment)
start timer
Fast retransmit algorithm
Transport Layer 3-61
else
increment count of dup ACKs received for yif (count of dup ACKs received for y = 3)
resend segment with sequence number y
break
a duplicate ACK for already ACKed segment
fast retransmit
X
Host BHost A
Seq=92 8 bytes of data
TCP fast retransmit
Seq=100 20 bytes of data
Transport Layer 3-62
fast retransmit after sender
receipt of triple duplicate ACK
ACK=100
timeout
ACK=100
ACK=100
ACK=100
Seq=100 20 bytes of data
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT but RTT varies
too short premature timeout
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
ignore retransmitted segments
Transport Layer 3-63
too short premature timeout
unnecessary retransmissions
too long slow reaction to segment loss
segments
SampleRTT will vary want estimated RTT ldquosmootherrdquo
average several recent measurements not just current SampleRTT
TCP Round Trip Time and Timeout
EstimatedRTT = (1- αααα)EstimatedRTT + ααααSampleRTT
Exponential weighted moving average
influence of past sample decreases exponentially fast
typical value αααα = 0125
Transport Layer 3-64
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
250
300
350
RT
T (
mil
lisec
on
ds)
Transport Layer 3-65
100
150
200
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lisec
on
ds)
SampleRTT Estimated RTT
TCP Round Trip Time and Timeout
Setting the timeout EstimtedRTT plus ldquosafety marginrdquo
large variation in EstimatedRTT -gt larger safety margin
first estimate of how much SampleRTT deviates from EstimatedRTT
Transport Layer 3-66
EstimatedRTT
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-ββββ)DevRTT +ββββ|SampleRTT-EstimatedRTT|
(typically ββββ = 025)
Then set timeout interval
TCP flow controlapplication
process
TCP socket
receiver buffers
TCP
application
OS
application may
remove data from
TCP socket buffers hellip
hellip slower than TCP
receiver is delivering
(sender is sending)
Transport Layer 3-67
code
IP
code
receiver protocol stack
from sender
receiver controls sender so
sender wonrsquot overflow receiverrsquos buffer by transmitting too much
too fast
flow control
TCP flow control
buffered data
free buffer space
RcvBuffer
to application process receiver ldquoadvertisesrdquo
free buffer space by including rwnd value in TCP header of receiver-to-sender segments
Transport Layer 3-68
free buffer spacerwnd
TCP segment payloads
to-sender segments RcvBuffer size set via
socket options (typical default is 4096 bytes)
many operating systems autoadjust RcvBuffer
sender limits amount of unacked (ldquoin-flightrdquo) data to receiverrsquos rwnd value
guarantees receive
receiver-side buffering
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
TCP Connection Management
Recall TCP sender receiver establish ldquoconnectionrdquo before exchanging data segments
initialize TCP variables
seq s
buffers flow control info (eg RcvWindow)
Transport Layer 3-69
buffers flow control info (eg RcvWindow)
client connection initiatorSocket clientSocket = new Socket(hostnameport
number)
server contacted by clientSocket connectionSocket = welcomeSocketaccept()
Connection Managementbefore exchanging data senderreceiver ldquohandshakerdquo agree to establish connection (each knowing the other willing to
establish connection)
agree on connection parameters
application application
Transport Layer 3-70
connection state ESTAB
connection variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
connection state ESTAB
connection Variables
seq client-to-server
server-to-client
rcvBuffer size
at serverclient network
Q will 2-way handshake always work in network
variable delays
2-way handshake
Letrsquos talk
OKESTAB
ESTAB
Agreeing to establish a connection
Transport Layer 3-71
variable delays
retransmitted messages (eg req_conn(x)) due to message loss
message reordering
canrsquot ldquoseerdquo other side
ESTAB
choose xreq_conn(x)
ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
2-way handshake failure scenarios
retransmit
req_conn(x)
retransmit
req_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
choose xreq_conn(x)
ESTAB
acc_conn(x)
Transport Layer 3-72
req_conn(x)
ESTAB
req_conn(x)
half open connection
(no client)
client terminates
server
forgets x
connection
x completes
req_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmit
data(x+1)
accept
data(x+1)
ESTAB
client terminates
ESTABdata(x+1) accept
data(x+1)
connection
x completes server
forgets x
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
TCP 3-way handshake
SYNbit=1 Seq=x
choose init seq num x
send TCP SYN msg
choose init seq num y
send TCP SYNACK
SYNSENT
SYN RCVD
client state
LISTEN
server state
LISTEN
Transport Layer 3-73
ESTAB
SYNbit=1 Seq=y
ACKbit=1 ACKnum=x+1
send TCP SYNACK
msg acking SYN
ACKbit=1 ACKnum=y+1
received SYNACK(x)
indicates server is live
send ACK for SYNACK
this segment may contain
client-to-server data
received ACK(y)
indicates client is live
ESTAB
SYN RCVD
TCP 3-way handshake FSM
closed
ΛSocket clientSocket =
newSocket(hostnameport number)
SYN(seq=x)
Socket connectionSocket = welcomeSocketaccept()
SYN(x)
SYNACK(seq=yACKnum=x+1)
Transport Layer 3-74
listen
SYN
rcvd
SYN
sent
ESTAB
SYN(seq=x)SYNACK(seq=yACKnum=x+1)
create new socket for
communication back to client
SYNACK(seq=yACKnum=x+1)
ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
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
Transport Layer 3-75
on receiving FIN ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
FIN_WAIT_2
CLOSE_WAIT
ACKbit=1 ACKnum=x+1
wait for servercan still
TCP closing a connection
FIN_WAIT_1 FINbit=1 seq=xcan no longer
send but can
receive data
clientSocketclose()
client state server state
ESTABESTAB
Transport Layer 3-76
FIN_WAIT_2
FINbit=1 seq=y
ACKbit=1 ACKnum=y+1
wait for server
closesend data
can no longer
send data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait
for 2max
segment lifetime
CLOSED
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much
data too fast for network to handlerdquo
different from flow control
manifestations
Transport Layer 3-77
manifestations
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)
a top-10 problem
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
unlimited shared output link buffers
Host Aλin original data
Host B
λout
Transport Layer 3-78
no retransmission
large delays when congested
maximum achievable throughput
Causescosts of congestion scenario 2
one router finite buffers
sender retransmission of lost packet
Host Aλin original data
λout
λ original data plus
Transport Layer 3-79
finite shared output link buffersHost B
λin original data plus retransmitted data
idealization perfect knowledge
sender sends only when router buffers available
R2
R2
λ out
λin
Causescosts of congestion scenario 2
Transport Layer 3-80
finite shared output link buffers
λin original data
λoutλin original data plusretransmitted data
copy
free buffer space
Host B
A
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
Causescosts of congestion scenario 2
Transport Layer 3-81
λin original data
λoutλin original data plusretransmitted data
copy
no buffer spaceA
Host B
Causescosts of congestion scenario 2
Idealization known
loss packets can be lost dropped at router due to full buffers
sender only resends if packet known to be lost
R2
R2λin
λ out
when sending at R2 some packets are retransmissions but asymptotic goodput is still R2 (why)
Transport Layer 3-82
λin original data
λoutλin original data plusretransmitted data
free buffer spaceA
Host B
R2
R2λin
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
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
Causescosts of congestion scenario 2
Transport Layer 3-83
A
λin λoutλincopy
free buffer space
timeout
Host B
R2
λ out
when sending at R2 some packets are retransmissions including duplicated that are delivered
R2λin
Causescosts of congestion scenario 2
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
Transport Layer 3-84
ldquocostsrdquo of congestion more work (retrans) for given ldquogoodputrdquo unneeded retransmissions link carries multiple
copies of pktdecreasing goodput
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Causescosts of congestion scenario 3
four senders
multihop paths
timeoutretransmit
λin
Q what happens as and increase λ
in
Host Aλin original data λout
λin original data plus retransmitted data
Transport Layer 3-85
finite shared output link buffers
Host B
Causescosts of congestion scenario 3
Host A
Host B
λou
t
Transport Layer 3-86
Another ldquocostrdquo of congestion
when packet dropped any ldquoupstream transmission capacity used for that packet was wasted
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
Transport Layer 3-87
network
congestion inferred from end-system observed loss delay
approach taken by TCP
to end systems
single bit indicating congestion (SNA DECbit TCPIP ECN ATM)
explicit rate sender should send at
Case study ATM ABR congestion control
ABR available bit rate ldquoelastic servicerdquo
if senderrsquos path ldquounderloadedrdquo
sender should use available bandwidth
RM (resource management) cells
sent by sender interspersed with data cells
bits in RM cell set by switches (ldquonetwork-assistedrdquo)
Transport Layer 3-88
available bandwidth
if senderrsquos path congested
sender throttled to minimum guaranteed rate
(ldquonetwork-assistedrdquo)
NI bit no increase in rate (mild congestion)
CI bit congestion indication
RM cells returned to sender by receiver with bits intact
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Case study ATM ABR congestion control
Transport Layer 3-89
two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell
senderrsquo send rate thus maximum supportable rate on path
EFCI bit in data cells set to 1 in congested switch if data cell preceding RM cell has EFCI set sender sets CI bit in returned RM cell
TCP Congestion Control
end-end control (no network
assistance)
How does sender perceive congestion
loss event = timeout or3 duplicate acks
TCP sender reduces
last byte
ACKedsent not-yet ACKed
(ldquoin-flightrdquo)
last byte sent
cwndsender sequence number space
Transport Layer 3-90
assistance)
sender limits transmissionLastByteSent-LastByteAcked
lelelele minCongWin RcvWindow
Roughly
CongWin is dynamic function of perceived network congestion
TCP sender reduces rate (CongWin) after loss event
three mechanisms slow start
AIMD
conservative after timeout events
rate =CongWinRTT
Bytessec
TCP Slow Start
When connection begins CongWin = 1 MSS
Example MSS = 500 bytes amp RTT = 200 msec
initial rate = 20 kbps
available bandwidth may
When connection begins increase rate exponentially fast until first loss event
Transport Layer 3-91
available bandwidth may be gtgt MSSRTT
desirable to quickly ramp up to respectable rate
TCP Slow Start (more)
When connection begins increase rate exponentially until first loss event
double CongWin every
Host A
RTT
Host B
Transport Layer 3-92
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast time
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
TCP AIMD
24 Kbytes
congestionwindow
multiplicative decreasecut CongWin in half after loss event
additive increaseincrease CongWin by 1 MSS every RTT in the absence of loss events probing
Transport Layer 3-93
8 Kbytes
16 Kbytes
24 Kbytes
time
events probing
Long-lived TCP connection
Refinement inferring loss
After 3 dup ACKs
CongWin is cut in half
window then grows linearly
But after timeout event
bull 3 dup ACKs indicates network capable of delivering some segmentsbull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Philosophy
Transport Layer 3-94
But after timeout event
CongWin instead set to 1 MSS
window then grows exponentially
to a threshold then grows linearly
bull timeout before 3 dup ACKs is ldquomore alarmingrdquo
Refinement (more)
Q When should the exponential increase switch to linear
A When CongWingets to 12 of its 4
6
8
10
12
14
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
threshold
Transport Layer 3-95
gets to 12 of its value before timeout
Implementation Variable Threshold
At loss event Threshold is set to 12 of CongWin just before loss event
0
2
4
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Transmission round
con
ges
tio
n w
ind
ow
siz
e (s
egm
ents
)
Series1 Series2
threshold
TCPTahoe
TCPReno
Summary TCP Congestion Control
When CongWin is below Threshold sender in slow-start phase window grows exponentially
When CongWin is above Threshold sender is in congestion-avoidance phase window grows linearly
Transport Layer 3-96
When a triple duplicate ACK occurs Thresholdset to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS)
ACK receipt for previously unacked data
CongWin = CongWin + MSS If (CongWin gt Threshold)
set state to ldquoCongestion Avoidancerdquo
Resulting in a doubling of CongWin every RTT
CongestionAvoidance (CA)
ACK receipt for previously unacked data
CongWin = CongWin+MSS (MSSCongWin)
Additive increase resulting in increase of CongWin by 1 MSS every RTT
Transport Layer 3-97
data
SS or CA Loss event detected by triple duplicate ACK
Threshold = CongWin2 CongWin = ThresholdSet state to ldquoCongestion Avoidancerdquo
Fast recovery implementing multiplicative decrease CongWin will not drop below 1 MSS
SS or CA Timeout Threshold = CongWin2 CongWin = 1 MSSSet state to ldquoSlow Startrdquo
Enter slow start
SS or CA Duplicate ACK
Increment duplicate ACK count for segment being acked
CongWin and Threshold not changed
Summary TCP Congestion Control
timeoutssthresh = cwnd2
cwnd = 1 MSS
Λcwnd gt ssthresh
congestion
avoidance
cwnd = cwnd + MSS (MSScwnd)
dupACKcount = 0
transmit new segment(s) as allowed
new ACK
dupACKcount++
duplicate ACK
slow
start
timeout
cwnd = cwnd+MSS
dupACKcount = 0
transmit new segment(s) as allowed
new ACKdupACKcount++
duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB
dupACKcount = 0
New
ACK
New
ACK
Transport Layer 3-98
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
dupACKcount++
fast
recovery
cwnd = cwnd + MSS
transmit new segment(s) as allowed
duplicate ACK
ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd2
cwnd = 1
dupACKcount = 0
retransmit missing segment ssthresh= cwnd2
cwnd = ssthresh + 3
retransmit missing segment
dupACKcount == 3cwnd = ssthresh
dupACKcount = 0
New ACK
timeoutssthresh = cwnd2
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
New
ACK
TCP throughput
avg TCP thruput as function of window size RTT ignore slow start assume always data to send
W window size (measured in bytes) where loss occurs avg window size ( in-flight bytes) is frac34 W
avg thruput is 34W per RTT
Transport Layer 3-99
W
W2
avg TCP thruput = 34
WRTT
bytessec
TCP Futures TCP over ldquolong fat pipesrdquo
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments
Throughput in terms of loss rate
Transport Layer 3-100
Throughput in terms of loss rate
L = 210-10 Wow
New versions of TCP for high-speed
LRTT
MSSsdot221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
TCP Fairness
Transport Layer 3-101
bottleneckrouter
capacity R
TCP connection 2
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughout increases
multiplicative decrease decreases throughput proportionally
R equal bandwidth share
Transport Layer 3-102
RConnection 1 throughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Fairness (more)
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion
Fairness and parallel TCP connections
nothing prevents app from opening parallel cnctions between 2 hosts
Transport Layer 3-103
throttled by congestion control
Instead use UDP pump audiovideo at constant rate tolerate packet loss
Research area TCP friendly
between 2 hosts
Web browsers do this
Example link of rate R supporting 9 cnctions
new app asks for 1 TCP gets rate R10
new app asks for 11 TCPs gets R2