chapter 3: transport layer our goals: r understand principles behind transport layer services: m...
TRANSCRIPT
Chapter 3 Transport Layer
Our goals understand principles
behind transport layer services multiplexing
demultiplexing reliable data transfer flow control congestion control
learn about transport layer protocols in the Internet UDP connectionless
transport TCP connection-oriented
transport TCP congestion control
Transport services and protocols provide logical
communication between app processes running on different hosts
transport protocols run in end systems
reliable in-order delivery (TCP)
unreliable unordered delivery UDP
application
transportnetworkdata linkphysical
application
transportnetworkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysicalnetwork
data linkphysical
logical end-end transport
Multiplexingdemultiplexing
application
transport
network
link
physical
P1 application
transport
network
link
physical
application
transport
network
link
physical
P2P3 P4P1
host 1 host 2 host 3
= 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
How demultiplexing works host receives IP datagrams host uses IP addresses amp port
numbers to direct segment to appropriate socket
UDP socket identified by two-tuple (dest IP address dest port number)
TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number
source port dest port
32 bits
applicationdata
(message)
other header fields
TCPUDP segment format
Connectionless demux (cont)
DatagramSocket serverSocket = new DatagramSocket(6428)
ClientIPB
P2
client IP A
P1P1P3
serverIP C
SP 6428
DP 9157
SP 9157
DP 6428
SP 6428
DP 5775
SP 5775
DP 6428
SP provides ldquoreturn addressrdquo
Connection-oriented demux (cont)
ClientIPB
P1
client IP A
P1P2P4
serverIP C
SP 9157
DP 80
SP 9157
DP 80
P5 P6 P3
D-IPCS-IP A
D-IPC
S-IP B
SP 5775
DP 80
D-IPCS-IP B
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 to
app connectionless
no handshaking between UDP sender receiver
each UDP segment handled independently of others
Why is there a UDP no connection establishment
(which can add delay) 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 DNS SNMP
reliable transfer over UDP add reliability at application layer application-specific error
recovery
source port dest port
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength in
bytes of UDPsegmentincluding
header
UDP checksum
Sender treat segment contents as
sequence of 16-bit integers checksum addition (1rsquos
complement sum) of segment contents
sender puts checksum value into UDP checksum field
Receiver compute checksum of received
segment check if computed checksum
equals checksum field value NO - error detected YES - no error detected
But maybe errors nonetheless More later hellip
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Principles of Reliable data transfer important in app transport link layers characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Transport services and protocols provide logical
communication between app processes running on different hosts
transport protocols run in end systems
reliable in-order delivery (TCP)
unreliable unordered delivery UDP
application
transportnetworkdata linkphysical
application
transportnetworkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysicalnetwork
data linkphysical
logical end-end transport
Multiplexingdemultiplexing
application
transport
network
link
physical
P1 application
transport
network
link
physical
application
transport
network
link
physical
P2P3 P4P1
host 1 host 2 host 3
= 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
How demultiplexing works host receives IP datagrams host uses IP addresses amp port
numbers to direct segment to appropriate socket
UDP socket identified by two-tuple (dest IP address dest port number)
TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number
source port dest port
32 bits
applicationdata
(message)
other header fields
TCPUDP segment format
Connectionless demux (cont)
DatagramSocket serverSocket = new DatagramSocket(6428)
ClientIPB
P2
client IP A
P1P1P3
serverIP C
SP 6428
DP 9157
SP 9157
DP 6428
SP 6428
DP 5775
SP 5775
DP 6428
SP provides ldquoreturn addressrdquo
Connection-oriented demux (cont)
ClientIPB
P1
client IP A
P1P2P4
serverIP C
SP 9157
DP 80
SP 9157
DP 80
P5 P6 P3
D-IPCS-IP A
D-IPC
S-IP B
SP 5775
DP 80
D-IPCS-IP B
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 to
app connectionless
no handshaking between UDP sender receiver
each UDP segment handled independently of others
Why is there a UDP no connection establishment
(which can add delay) 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 DNS SNMP
reliable transfer over UDP add reliability at application layer application-specific error
recovery
source port dest port
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength in
bytes of UDPsegmentincluding
header
UDP checksum
Sender treat segment contents as
sequence of 16-bit integers checksum addition (1rsquos
complement sum) of segment contents
sender puts checksum value into UDP checksum field
Receiver compute checksum of received
segment check if computed checksum
equals checksum field value NO - error detected YES - no error detected
But maybe errors nonetheless More later hellip
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Principles of Reliable data transfer important in app transport link layers characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Multiplexingdemultiplexing
application
transport
network
link
physical
P1 application
transport
network
link
physical
application
transport
network
link
physical
P2P3 P4P1
host 1 host 2 host 3
= 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
How demultiplexing works host receives IP datagrams host uses IP addresses amp port
numbers to direct segment to appropriate socket
UDP socket identified by two-tuple (dest IP address dest port number)
TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number
source port dest port
32 bits
applicationdata
(message)
other header fields
TCPUDP segment format
Connectionless demux (cont)
DatagramSocket serverSocket = new DatagramSocket(6428)
ClientIPB
P2
client IP A
P1P1P3
serverIP C
SP 6428
DP 9157
SP 9157
DP 6428
SP 6428
DP 5775
SP 5775
DP 6428
SP provides ldquoreturn addressrdquo
Connection-oriented demux (cont)
ClientIPB
P1
client IP A
P1P2P4
serverIP C
SP 9157
DP 80
SP 9157
DP 80
P5 P6 P3
D-IPCS-IP A
D-IPC
S-IP B
SP 5775
DP 80
D-IPCS-IP B
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 to
app connectionless
no handshaking between UDP sender receiver
each UDP segment handled independently of others
Why is there a UDP no connection establishment
(which can add delay) 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 DNS SNMP
reliable transfer over UDP add reliability at application layer application-specific error
recovery
source port dest port
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength in
bytes of UDPsegmentincluding
header
UDP checksum
Sender treat segment contents as
sequence of 16-bit integers checksum addition (1rsquos
complement sum) of segment contents
sender puts checksum value into UDP checksum field
Receiver compute checksum of received
segment check if computed checksum
equals checksum field value NO - error detected YES - no error detected
But maybe errors nonetheless More later hellip
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Principles of Reliable data transfer important in app transport link layers characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
How demultiplexing works host receives IP datagrams host uses IP addresses amp port
numbers to direct segment to appropriate socket
UDP socket identified by two-tuple (dest IP address dest port number)
TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number
source port dest port
32 bits
applicationdata
(message)
other header fields
TCPUDP segment format
Connectionless demux (cont)
DatagramSocket serverSocket = new DatagramSocket(6428)
ClientIPB
P2
client IP A
P1P1P3
serverIP C
SP 6428
DP 9157
SP 9157
DP 6428
SP 6428
DP 5775
SP 5775
DP 6428
SP provides ldquoreturn addressrdquo
Connection-oriented demux (cont)
ClientIPB
P1
client IP A
P1P2P4
serverIP C
SP 9157
DP 80
SP 9157
DP 80
P5 P6 P3
D-IPCS-IP A
D-IPC
S-IP B
SP 5775
DP 80
D-IPCS-IP B
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 to
app connectionless
no handshaking between UDP sender receiver
each UDP segment handled independently of others
Why is there a UDP no connection establishment
(which can add delay) 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 DNS SNMP
reliable transfer over UDP add reliability at application layer application-specific error
recovery
source port dest port
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength in
bytes of UDPsegmentincluding
header
UDP checksum
Sender treat segment contents as
sequence of 16-bit integers checksum addition (1rsquos
complement sum) of segment contents
sender puts checksum value into UDP checksum field
Receiver compute checksum of received
segment check if computed checksum
equals checksum field value NO - error detected YES - no error detected
But maybe errors nonetheless More later hellip
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Principles of Reliable data transfer important in app transport link layers characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Connectionless demux (cont)
DatagramSocket serverSocket = new DatagramSocket(6428)
ClientIPB
P2
client IP A
P1P1P3
serverIP C
SP 6428
DP 9157
SP 9157
DP 6428
SP 6428
DP 5775
SP 5775
DP 6428
SP provides ldquoreturn addressrdquo
Connection-oriented demux (cont)
ClientIPB
P1
client IP A
P1P2P4
serverIP C
SP 9157
DP 80
SP 9157
DP 80
P5 P6 P3
D-IPCS-IP A
D-IPC
S-IP B
SP 5775
DP 80
D-IPCS-IP B
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 to
app connectionless
no handshaking between UDP sender receiver
each UDP segment handled independently of others
Why is there a UDP no connection establishment
(which can add delay) 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 DNS SNMP
reliable transfer over UDP add reliability at application layer application-specific error
recovery
source port dest port
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength in
bytes of UDPsegmentincluding
header
UDP checksum
Sender treat segment contents as
sequence of 16-bit integers checksum addition (1rsquos
complement sum) of segment contents
sender puts checksum value into UDP checksum field
Receiver compute checksum of received
segment check if computed checksum
equals checksum field value NO - error detected YES - no error detected
But maybe errors nonetheless More later hellip
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Principles of Reliable data transfer important in app transport link layers characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Connection-oriented demux (cont)
ClientIPB
P1
client IP A
P1P2P4
serverIP C
SP 9157
DP 80
SP 9157
DP 80
P5 P6 P3
D-IPCS-IP A
D-IPC
S-IP B
SP 5775
DP 80
D-IPCS-IP B
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 to
app connectionless
no handshaking between UDP sender receiver
each UDP segment handled independently of others
Why is there a UDP no connection establishment
(which can add delay) 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 DNS SNMP
reliable transfer over UDP add reliability at application layer application-specific error
recovery
source port dest port
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength in
bytes of UDPsegmentincluding
header
UDP checksum
Sender treat segment contents as
sequence of 16-bit integers checksum addition (1rsquos
complement sum) of segment contents
sender puts checksum value into UDP checksum field
Receiver compute checksum of received
segment check if computed checksum
equals checksum field value NO - error detected YES - no error detected
But maybe errors nonetheless More later hellip
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Principles of Reliable data transfer important in app transport link layers characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
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 to
app connectionless
no handshaking between UDP sender receiver
each UDP segment handled independently of others
Why is there a UDP no connection establishment
(which can add delay) 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 DNS SNMP
reliable transfer over UDP add reliability at application layer application-specific error
recovery
source port dest port
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength in
bytes of UDPsegmentincluding
header
UDP checksum
Sender treat segment contents as
sequence of 16-bit integers checksum addition (1rsquos
complement sum) of segment contents
sender puts checksum value into UDP checksum field
Receiver compute checksum of received
segment check if computed checksum
equals checksum field value NO - error detected YES - no error detected
But maybe errors nonetheless More later hellip
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Principles of Reliable data transfer important in app transport link layers characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
UDP more
often used for streaming multimedia apps loss tolerant rate sensitive
other UDP uses DNS SNMP
reliable transfer over UDP add reliability at application layer application-specific error
recovery
source port dest port
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength in
bytes of UDPsegmentincluding
header
UDP checksum
Sender treat segment contents as
sequence of 16-bit integers checksum addition (1rsquos
complement sum) of segment contents
sender puts checksum value into UDP checksum field
Receiver compute checksum of received
segment check if computed checksum
equals checksum field value NO - error detected YES - no error detected
But maybe errors nonetheless More later hellip
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Principles of Reliable data transfer important in app transport link layers characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
UDP checksum
Sender treat segment contents as
sequence of 16-bit integers checksum addition (1rsquos
complement sum) of segment contents
sender puts checksum value into UDP checksum field
Receiver compute checksum of received
segment check if computed checksum
equals checksum field value NO - error detected YES - no error detected
But maybe errors nonetheless More later hellip
Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment
Principles of Reliable data transfer important in app transport link layers characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Principles of Reliable data transfer important in app transport link layers characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Rdt10 reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors no loss of packets
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Rdt20 channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question how to recover from errors acknowledgements (ACKs) negative acknowledgements (NAKs) sender retransmits pkt on receipt of NAK
new mechanisms in rdt20 error detection receiver feedback
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
rdt20 has a fatal flaw
What happens if ACKNAK corrupted
sender doesnrsquot know what happened at receiver
canrsquot just retransmit possible duplicate
Handling duplicates sender adds sequence
number to each pkt sender retransmits current
pkt if ACKNAK garbled receiver discards (doesnrsquot
deliver up) duplicate pkt
Sender sends one packet then waits for receiver response
stop and wait
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
rdt21
Sender seq added to pkt two seq rsquos (01) will
suffice must check if received
ACKNAK corrupted
Receiver must check if received
packet is duplicate note receiver can not
know 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 duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
rdt22 a NAK-free protocol
same functionality as rdt21 using ACKs only instead of NAK receiver sends ACK for last pkt
received OK duplicate ACK at sender results in same action as NAK
retransmit current pkt
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
rdt30 channels with errors and loss
New assumption underlying channel can also lose packets (data or ACKs) checksum seq ACKs
retransmissions will be of help but not enough
Approach sender waits ldquoreasonablerdquo amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost) retransmission will be
duplicate but use of seq rsquos already handles this
receiver must specify seq of pkt being ACKed
requires countdown timer
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
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
Two generic forms of pipelined protocols go-Back-N selective repeat
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed
ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Receiver ACK-only always send ACK for correctly-received
pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum
out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering Re-ACK pkt with highest in-order seq
Go-Back-N
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Selective Repeat
receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order delivery to
upper layer
sender only resends pkts for which ACK not received sender timer for each unACKed pkt
sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Selective repeat sender receiver windows
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Selective repeat
data from above if next available seq in
window send pkt
timeout(n) 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
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 next not-yet-received pkt
pkt n in [rcvbase-Nrcvbase-1]
ACK(n)
otherwise ignore
receiver
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP Overview RFCs 793 1122 1323 2018 2581
full duplex data connection-oriented flow controlled
point-to-point reliable in-order byte
steam pipelined send amp receive buffers
socketdoor
T C Psend buffer
T C Preceive buffer
socketdoor
segm ent
applicationwrites data
applicationreads data
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP segment structure
source port dest port
32 bits
applicationdata
(variable length)
sequence number
acknowledgement numberReceive window
Urg data pnterchecksum
FSRPAUheadlen
notused
Options (variable length)
URG urgent data (generally not used)
ACK ACK valid
PSH push data now(generally not used)
RST SYN FINconnection estab(setup teardown
commands)
bytes rcvr willingto accept
countingby bytes of data(not segments)
Internetchecksum
(as in UDP)
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP seq rsquos and ACKsSeq rsquos
byte stream ldquonumberrdquo of first byte in segmentrsquos data
ACKs seq of next byte
expected from other side
Host A Host B
Seq=42 ACK=79 data = lsquoCrsquo
Seq=79 ACK=43 data = lsquoCrsquo
Seq=43 ACK=80
Usertypes
lsquoCrsquo
host ACKsreceipt
of echoedlsquoCrsquo
host ACKsreceipt of
lsquoCrsquo echoesback lsquoCrsquo
timesimple telnet scenario
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP Round Trip Time and Timeout
Q how to set TCP timeout value
longer than RTT too short premature
timeout too long slow reaction to
segment loss
Q how to estimate RTT SampleRTT measured time from
segment transmission until ACK receipt
SampleRTT will vary want estimated RTT ldquosmootherrdquo
EstimatedRTT = (1- )EstimatedRTT + SampleRTT
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RTT
(mill
isec
onds
)
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
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
TimeoutInterval = EstimatedRTT + 4DevRTT
DevRTT = (1-)DevRTT + |SampleRTT-EstimatedRTT|
(typically = 025)
Then set timeout interval
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP reliable data transfer
TCP creates reliable data transfer service on top of IPrsquos unreliable service
Pipelined segments Cumulative acks Retransmissions are triggered by
timeout events duplicate acks
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP sender eventsdata rcvd from app Create segment with
seq seq is byte-stream
number of first data byte in segment
start timer if not already running (think of timer as for oldest unacked segment)
expiration interval TimeOutInterval
timeout retransmit segment that
caused timeout restart timer
Ack rcvd If acknowledges
previously unacked segments update what is known to
be acked start timer if there are
outstanding segments
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP retransmission scenarios
Host A
Seq=100 20 bytes data
ACK=100
timepremature timeout
Host B
Seq=92 8 bytes data
ACK=120
Seq=92 8 bytes data
Seq=
92
tim
eout
ACK=120
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
lost ACK scenario
Host B
X
Seq=92 8 bytes data
ACK=100
time
Seq=
92
tim
eout
SendBase= 100
SendBase= 120
SendBase= 120
Sendbase= 100
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP retransmission scenarios (more)
Host A
Seq=92 8 bytes data
ACK=100
loss
tim
eout
Cumulative ACK scenario
Host B
X
Seq=100 20 bytes data
ACK=120
time
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
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 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
TCP Receiver action
Delayed ACK Wait up to 500msfor next segment If no next segmentsend ACK
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 startsat lower end of gap
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Fast Retransmit Time-out period often relatively long Detect lost segments via duplicate ACKs If sender receives 3 ACKs for the same data it supposes that
segment after ACKed data was lost
event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y
fast retransmita duplicate ACK for already ACKed segment
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP Flow Control
receiver
speed-matching service matching the send rate to the receiving apprsquos drain rate
sender wonrsquot overflow
receiverrsquos buffer bytransmitting too
much too fast
flow control
Rcvr advertises spare room by including value of RcvWindow in segments
Sender limits unACKed data to RcvWindow
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP Connection Management
Three way handshake
Step 1 client host sends TCP SYN segment to server specifies initial seq no data
Step 2 server host receives SYN replies with SYNACK segment
server allocates buffers specifies server initial seq
Step 3 client receives SYNACK replies with ACK segment which may contain data
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP Connection Management (cont)
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Principles of Congestion Control
Congestion informally ldquotoo many sources sending too much data
too fast for network to handlerdquo different from flow control manifestations
lost packets (buffer overflow at routers) long delays (queueing in router buffers)
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Causescosts of congestion scenario 1
two senders two receivers
one router infinite buffers
no retransmission
large delays when congested
maximum achievable throughput
unlimited shared output link buffers
Host Ain original data
Host B
out
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Causescosts of congestion scenario 2
one router finite buffers sender retransmission of
lost packet finite shared output link buffers
Host Ain original data
Host B
out
in original data plus retransmitted data
ldquocostsrdquo of congestion more work (retrans) for
given ldquogoodputrdquo unneeded retransmissions
link carries multiple copies of pkt
always (goodput)
ldquoperfectrdquo retransmission only when
loss
retransmission of delayed (not lost)
packet makes larger (than
perfect case) for same
in
out
=
in
out
gt
in
out
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Causescosts of congestion scenario 3
four senders multihop paths timeoutretransmit when packet dropped any
ldquoupstream transmission capacity used for that packet was wasted
finite shared output link buffers
Host A in original data
Host B
out
in original data plus retransmitted data
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Approaches towards congestion control
End-end congestion control
no explicit feedback from network
congestion inferred from end-system observed loss delay
approach taken by TCP
Network-assisted congestion control
routers provide feedback to end systems
Two broad approaches towards congestion control
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP Congestion Control
end-end control (no network assistance)
sender limits transmission LastByteSent-LastByteAcked
CongWin Roughly
CongWin is dynamic function of perceived network congestion
How does sender perceive congestion
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms AIMD slow start conservative after
timeout events
rate = CongWin
RTT Bytessec
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP AIMD
8 Kbytes
16 Kbytes
24 Kbytes
time
congestionwindow
multiplicative decrease cut CongWin in half after loss event
additive increase increase CongWin by 1 MSS every RTT in the absence of loss events probing
Long-lived TCP connection
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP Slow Start
When connection begins increase rate exponentially until first loss event double CongWin every
RTT done by incrementing CongWin for every ACK received
Summary initial rate is slow but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Refinement
After 3 dup ACKs CongWin is cut in half window then grows linearly (congestion avoidance)
But after timeout event CongWin instead set to 1 MSS window then grows exponentially (slow start) to a threshold then grows linearly (congestion
avoidance)
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
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
When a triple duplicate ACK occurs Threshold set to CongWin2 and CongWin set to Threshold
When timeout occurs Threshold set to CongWin2 and CongWin is set to 1 MSS
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP throughput
Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start
Let W be the window size when loss occurs When window is W throughput is WRTT Just after loss window drops to W2
throughput to W2RTT Average throughout 75 WRTT
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
TCP Futures
Example 1500 byte segments 100ms RTT want 10 Gbps throughput
Requires window size W = 83333 in-flight segments Throughput in terms of loss rate
L = 210-10 Wow New versions of TCP for high-speed needed
LRTT
MSS221
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Why is TCP fair
Two competing sessions Additive increase gives slope of 1 as throughput increases multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughputConnect
ion 2
th
roughput
congestion avoidance additive increaseloss decrease window by factor of 2
congestion avoidance additive increaseloss decrease window by factor of 2
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-
Chapter 3 Summary principles behind transport
layer services multiplexing demultiplexing reliable data transfer flow control congestion control
instantiation and implementation in the Internet UDP TCP
Next leaving the network
ldquoedgerdquo (application transport layers)
into the network ldquocorerdquo
- Chapter 3 Transport Layer
- Transport services and protocols
- Multiplexingdemultiplexing
- How demultiplexing works
- Connectionless demux (cont)
- Connection-oriented demux (cont)
- UDP User Datagram Protocol [RFC 768]
- UDP more
- UDP checksum
- Principles of Reliable data transfer
- Rdt10 reliable transfer over a reliable channel
- Rdt20 channel with bit errors
- rdt20 has a fatal flaw
- rdt21
- rdt22 a NAK-free protocol
- rdt30 channels with errors and loss
- Pipelined protocols
- Go-Back-N
- Slide 19
- Selective Repeat
- Selective repeat sender receiver windows
- Selective repeat
- TCP Overview RFCs 793 1122 1323 2018 2581
- TCP segment structure
- TCP seq rsquos and ACKs
- TCP Round Trip Time and Timeout
- Example RTT estimation
- Slide 28
- TCP reliable data transfer
- TCP sender events
- TCP retransmission scenarios
- TCP retransmission scenarios (more)
- TCP ACK generation [RFC 1122 RFC 2581]
- Fast Retransmit
- TCP Flow Control
- TCP Connection Management
- TCP Connection Management (cont)
- TCP Connection Management (cont)
- Principles of Congestion Control
- Causescosts of congestion scenario 1
- Causescosts of congestion scenario 2
- Causescosts of congestion scenario 3
- Approaches towards congestion control
- TCP Congestion Control
- TCP AIMD
- TCP Slow Start
- Refinement
- Summary TCP Congestion Control
- TCP throughput
- TCP Futures
- TCP Fairness
- Why is TCP fair
- Chapter 3 Summary
-