Post on 30-Sep-2020
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© From Computer Networking, by Kurose&Ross Transport Layer 3-1
Introduction toComputer Networking
Guy Leduc
Chapter 3Transport Layer Computer Networking:
A Top Down Approach,5th edition.Jim Kurose, Keith RossAddison-Wesley, April2009.
© From Computer Networking, by Kurose&Ross Transport Layer 3-2
Chapter 3: Transport LayerOur goals:❒ understand principles
behind transportlayer services:❍ multiplexing/demultipl
exing❍ reliable data transfer❍ flow control❍ congestion control
❒ learn about transportlayer protocols in theInternet:❍ UDP: connectionless
transport❍ TCP: connection-oriented
transport❍ TCP congestion control
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© From Computer Networking, by Kurose&Ross Transport Layer 3-3
Chapter 3 outline
❒ 3.1 Transport-layerservices
❒ 3.2 Multiplexing anddemultiplexing
❒ 3.3 Connectionlesstransport: UDP
❒ 3.4 Principles ofreliable data transfer
❒ 3.5 Connection-orientedtransport: TCP❍ segment structure❍ reliable data transfer❍ flow control❍ connection management
❒ 3.6 Principles ofcongestion control
❒ 3.7 TCP congestioncontrol
© From Computer Networking, by Kurose&Ross Transport Layer 3-4
Transport services and protocols❒ provide logical communication
between app processesrunning on different hosts
❒ transport protocols run in endsystems❍ send side: breaks app
messages into segments,passes to network layer
❍ rcv side: reassemblessegments into messages,passes to app layer
❒ more than one transportprotocol available to apps❍ Internet: TCP and UDP
applicationtransportnetworkdata linkphysical
applicationtransportnetworkdata linkphysical
logical end-end transport
3
© From Computer Networking, by Kurose&Ross Transport Layer 3-5
Transport vs. network layer
❒ network layer: logicalcommunicationbetween hosts
❒ transport layer: logicalcommunicationbetween processes❍ relies on, enhances,
network layer services
Household analogy:(see example in reference
book)12 kids sending letters to 12
kids❒ processes = kids❒ app messages = letters in
envelopes❒ hosts = houses❒ transport protocol = Ann
and Bill❒ network-layer protocol =
postal service
© From Computer Networking, by Kurose&Ross Transport Layer 3-6
Internet transport-layer protocols
❒ reliable, in-orderdelivery (TCP)❍ congestion control❍ flow control❍ connection setup
❒ unreliable, unordereddelivery: UDP❍ no-frills extension of
“best-effort” IP❒ services not available:
❍ delay guarantees❍ bandwidth guarantees
applicationtransportnetworkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
applicationtransportnetworkdata linkphysical
logical end-end transport
4
© From Computer Networking, by Kurose&Ross Transport Layer 3-7
Chapter 3 outline
❒ 3.1 Transport-layerservices
❒ 3.2 Multiplexing anddemultiplexing
❒ 3.3 Connectionlesstransport: UDP
❒ 3.4 Principles ofreliable data transfer
❒ 3.5 Connection-orientedtransport: TCP❍ segment structure❍ reliable data transfer❍ flow control❍ connection management
❒ 3.6 Principles ofcongestion control
❒ 3.7 TCP congestioncontrol
© From Computer Networking, by Kurose&Ross Transport Layer 3-8
Multiplexing/demultiplexing
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 host:gathering data from multiplesockets, enveloping data with header (later used for demultiplexing)
Multiplexing at send host:
5
© From Computer Networking, by Kurose&Ross Transport Layer 3-9
How demultiplexing works❒ host receives IP datagrams
❍ each datagram has sourceIP address, destination IPaddress
❍ each datagram carries 1transport-layer segment
❍ each segment has source,destination port number
❒ host uses IP addresses & portnumbers to direct segment toappropriate socket
source port # dest port #
32 bits
applicationdata
(message)
other header fields
TCP/UDP segment format
© From Computer Networking, by Kurose&Ross Transport Layer 3-10
Connectionless demultiplexing
❒ Create sockets with portnumbers:
DatagramSocket mySocket1 = newDatagramSocket(12534);
DatagramSocket mySocket2 = newDatagramSocket(12535);
❒ UDP socket identified bytwo-tuple:
(dest IP address, dest port number)
❒ When host receives UDPsegment:❍ checks destination port
number in segment❍ directs UDP segment (+
source IP) to socket withthat port number
❒ IP datagrams withdifferent source IPaddresses and/or sourceport numbers directedto same socket
6
© From Computer Networking, by Kurose&Ross Transport Layer 3-11
Connectionless demux (cont)
DatagramSocket serverSocket = new DatagramSocket(6428);
ClientIP:B
P2
client IP: A
P1P1P3
serverIP: C
SP: 6428DP: 9157
SP: 9157DP: 6428
SP: 6428DP: 5775
SP: 5775DP: 6428
SP (together with Source IP) provides “return address” to app process
© From Computer Networking, by Kurose&Ross Transport Layer 3-12
Connection-oriented demux
❒ TCP socket identifiedby 4-tuple:❍ source IP address❍ source port number❍ dest IP address❍ dest port number
❒ recv host uses all fourvalues to directsegment to appropriatesocket
❒ Server host may supportmany simultaneous TCPsockets:❍ each socket identified by
its own 4-tuple❒ Web servers have
different sockets foreach connecting client❍ non-persistent HTTP will
have different socket foreach request
7
© From Computer Networking, by Kurose&Ross Transport Layer 3-13
Connection-oriented demux(cont)
ClientIP: B
P1
client IP: A
P1P2P4
serverIP: C
SP: 9157DP: 80
SP: 9157DP: 80
P5 P6 P3
D-IP:CS-IP: AD-IP:C
S-IP: B
SP: 5775DP: 80
D-IP:CS-IP: B
© From Computer Networking, by Kurose&Ross Transport Layer 3-14
Connection-oriented demux:Threaded Web Server
ClientIP: B
P1
client IP: A
P1P2
serverIP: C
SP: 9157DP: 80
SP: 9157DP: 80
P4 P3
D-IP:CS-IP: AD-IP:C
S-IP: B
SP: 5775DP: 80
D-IP:CS-IP: B
8
© From Computer Networking, by Kurose&Ross Transport Layer 3-15
Chapter 3 outline
❒ 3.1 Transport-layerservices
❒ 3.2 Multiplexing anddemultiplexing
❒ 3.3 Connectionlesstransport: UDP
❒ 3.4 Principles ofreliable data transfer
❒ 3.5 Connection-orientedtransport: TCP❍ segment structure❍ reliable data transfer❍ flow control❍ connection management
❒ 3.6 Principles ofcongestion control
❒ 3.7 TCP congestioncontrol
© From Computer Networking, by Kurose&Ross Transport Layer 3-16
UDP: User Datagram Protocol [RFC 768]
❒ “no frills,” “bare bones”Internet transportprotocol
❒ “best effort” service, UDPsegments may be:❍ lost❍ delivered out of order
to app❒ connectionless:
❍ no handshaking betweenUDP sender, receiver
❍ each UDP segmenthandled independentlyof others
Why is there a UDP?❒ no connection
establishment (which canadd delay)
❒ simple: no connection stateat sender, receiver
❒ small segment header❒ no congestion control: UDP
can blast away as fast asdesired
9
© From Computer Networking, by Kurose&Ross Transport Layer 3-17
UDP: more❒ often used for streaming
multimedia apps❍ loss tolerant❍ rate sensitive
❒ other UDP uses❍ DNS❍ SNMP
❒ reliable transfer over UDP:add reliability atapplication layer❍ application-specific
error recovery!
source port # dest port #
32 bits
Applicationdata
(message)
UDP segment format
length checksumLength, in
bytes of UDPsegment,including
header
© From Computer Networking, by Kurose&Ross Transport Layer 3-18
UDP checksum
Sender:❒ treat segment contents as
sequence of 16-bitintegers
❒ checksum: addition (1’scomplement sum) ofsegment contents
❒ sender puts checksumvalue into UDP checksumfield
Receiver:❒ compute checksum of
received segment❒ check if computed checksum
equals checksum field value:❍ NO - error detected❍ YES - no error detected.
But maybe errorsnonetheless? More later….
Goal: detect “errors” (e.g., flipped bits) intransmitted segment
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© From Computer Networking, by Kurose&Ross Transport Layer 3-19
Internet Checksum Example❒ Note
❍ When adding numbers, a carryout from themost significant bit needs to be added to theresult
❒ Example: add two 16-bit integers
1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1
1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1
1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
wraparound
sumchecksum
© From Computer Networking, by Kurose&Ross Transport Layer 3-20
Chapter 3 outline
❒ 3.1 Transport-layerservices
❒ 3.2 Multiplexing anddemultiplexing
❒ 3.3 Connectionlesstransport: UDP
❒ 3.4 Principles ofreliable data transfer
❒ 3.5 Connection-orientedtransport: TCP❍ segment structure❍ reliable data transfer❍ flow control❍ connection management
❒ 3.6 Principles ofcongestion control
❒ 3.7 TCP congestioncontrol
11
© From Computer Networking, by Kurose&Ross Transport Layer 3-21
Principles of Reliable data transfer❒ important in app., transport, link layers❒ top-10 list of important networking topics!
❒ characteristics of unreliable channel will determinecomplexity of reliable data transfer protocol (rdt)
© From Computer Networking, by Kurose&Ross Transport Layer 3-22
Principles of Reliable data transfer❒ important in app., transport, link layers❒ top-10 list of important networking topics!
❒ characteristics of unreliable channel will determinecomplexity of reliable data transfer protocol (rdt)
12
© From Computer Networking, by Kurose&Ross Transport Layer 3-23
Principles of Reliable data transfer❒ important in app., transport, link layers❒ top-10 list of important networking topics!
❒ characteristics of unreliable channel will determinecomplexity of reliable data transfer protocol (rdt)
© From Computer Networking, by Kurose&Ross Transport Layer 3-24
Reliable data transfer: getting started
sendside
receiveside
rdt_send(): called from above,(e.g., by app.). Passed data to
deliver to receiver upper layer
udt_send(): called by rdt,to transfer packet over
unreliable channel to receiver
rdt_rcv(): called when packetarrives on rcv-side of channel
deliver_data(): called byrdt to deliver data to upper
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© From Computer Networking, by Kurose&Ross Transport Layer 3-25
Reliable data transfer: getting startedWe’ll:❒ incrementally develop sender, receiver sides of
reliable data transfer protocol (rdt)❒ consider only unidirectional data transfer
❍ but control info will flow on both directions!❒ use finite state machines (FSM) to specify
sender, receiver
state1
state2
event causing state transitionactions taken on state transition
state: when in this“state” next state
uniquely determinedby next event
eventactions
© From Computer Networking, by Kurose&Ross Transport Layer 3-26
Rdt1.0: reliable transfer over a reliable channel
❒ underlying channel perfectly reliable❍ no bit errors❍ no loss of packets
❒ separate FSMs for sender, receiver:❍ sender sends data into underlying channel❍ receiver reads data from underlying channel
Wait forcall fromabove packet = make_pkt(data)
udt_send(packet)
rdt_send(data)data = extract (packet)deliver_data(data)
Wait forcall from
below
rdt_rcv(packet)
sender receiver
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© From Computer Networking, by Kurose&Ross Transport Layer 3-27
Rdt2.0: channel with bit errors❒ underlying channel may flip bits in packet
❍ checksum to detect bit errors❒ the question: how to recover from errors:
❍ acknowledgements (ACKs): receiver explicitlytells sender that pkt received OK
❍ negative acknowledgements (NAKs): receiverexplicitly tells sender that pkt had errors
❍ sender retransmits pkt on receipt of NAK❒ new mechanisms in rdt2.0 (beyond
rdt1.0):❍ error detection❍ receiver feedback: control msgs (ACK,NAK)
rcvr->sender
pkt A
pkt B
pkt B
ACK
NAK
ACK OK
Error
OK
© From Computer Networking, by Kurose&Ross Transport Layer 3-28
rdt2.0: FSM specification
Wait forcall fromabove
sndpkt = make_pkt(data, checksum)udt_send(sndpkt)
data = extract(rcvpkt)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Wait forACK or
NAK
Wait forcall from
belowsender
receiverrdt_send(data)
Λ
15
© From Computer Networking, by Kurose&Ross Transport Layer 3-29
rdt2.0: operation with no errors
Wait forcall fromabove
snkpkt = make_pkt(data, checksum)udt_send(sndpkt)
extract(rcvpkt,data)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Wait forACK or
NAK
Wait forcall from
below
rdt_send(data)
Λ
© From Computer Networking, by Kurose&Ross Transport Layer 3-30
rdt2.0: error scenario
Wait forcall fromabove
snkpkt = make_pkt(data, checksum)udt_send(sndpkt)
extract(rcvpkt,data)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Wait forACK or
NAK
Wait forcall from
below
rdt_send(data)
Λ
16
© From Computer Networking, by Kurose&Ross Transport Layer 3-31
rdt2.0 has a fatal flaw!What happens if ACK/NAK
corrupted?❒ garbled ACK/NAK detected by
checksum too❒ garbled ACK/NAK discarded❒ sender doesn’t know what
happened at receiver!❒ can’t just retransmit: possible
duplicate
Handling duplicates:❒ sender retransmits current
pkt if ACK/NAK garbled❒ sender adds sequence
number to each pkt❒ receiver discards (doesn’t
deliver up) duplicate pkt
Sender sends one packet, then waits for receiver response
stop and waitpkt
same pkt
garbled ACK
Duplicate!
OK
© From Computer Networking, by Kurose&Ross Transport Layer 3-32
rdt2.1: sender, handles garbled ACK/NAKs
Wait forcall 0 from
above
sndpkt = make_pkt(0, data, checksum)udt_send(sndpkt)
rdt_send(data)
Wait forACK orNAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) &&( corrupt(rcvpkt) ||isNAK(rcvpkt) )
sndpkt = make_pkt(1, data, checksum)udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt)&& not corrupt(rcvpkt)&& isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&( corrupt(rcvpkt) ||isNAK(rcvpkt) )
rdt_rcv(rcvpkt)&& not corrupt(rcvpkt)&& isACK(rcvpkt)
Wait for call 1 from
above
Wait forACK orNAK 1
ΛΛ
17
© From Computer Networking, by Kurose&Ross Transport Layer 3-33
rdt2.1: receiver, handles garbled ACK/NAKs
Wait for0 frombelow
sndpkt = make_pkt(NAK, chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq0(rcvpkt)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq1(rcvpkt)
data = extract(rcvpkt)deliver_data(data)sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
Wait for1 frombelow
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq0(rcvpkt)
data = extract(rcvpkt)deliver_data(data)sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq1(rcvpkt)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
sndpkt = make_pkt(NAK, chksum)udt_send(sndpkt)
Note: receiver sends ACK in this case.
Why?
© From Computer Networking, by Kurose&Ross Transport Layer 3-34
rdt2.1: discussion
Sender:❒ seq # added to pkt❒ two seq. #’s (0,1) will
suffice. Why?❒ must check if received
ACK/NAK corrupted❒ twice as many states
❍ state must “remember”whether “current” pkthas 0 or 1 seq. #
Receiver:❒ must check if received
packet is duplicate❍ state indicates whether
0 or 1 is expected pktseq #
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© From Computer Networking, by Kurose&Ross Transport Layer 3-35
rdt3.0: channels with errors and loss
New assumption:underlying channel canalso lose packets (dataor ACKs)❍ checksum, seq. #, ACKs,
retransmissions will beof help, but not enough
Approach: sender waits“reasonable” amount oftime for ACK
❒ retransmits if no ACKreceived in this time
❒ requires countdown timer
pkt(0)
pkt(0)ACK
Timeout
pkt(0)
pkt(0)ACK
ACKTimeout Discard but ACK!
No duplicate
X X
© From Computer Networking, by Kurose&Ross Transport Layer 3-36
Rdt3.0: actually no need for NAKs!❒ Up to now:
❍ Timer for packet loss❍ NAK for packet errors
❒ Simpler:❍ Timer for both packet loss
and errors!❒ NAK would improve
recovery time, but it’s notour concern here!
pkt(0)
pkt(1)
pkt(1)
ACK
NAK
ACK
With ACKs & NAKs
pkt(0)
pkt(1)
pkt(1)
ACK
ACK
With ACKs only
Timeout
OK
OK
Error
OK
OK
Error
19
© From Computer Networking, by Kurose&Ross Transport Layer 3-37
Flaw: Delayed packets or ACKs❒ If pkt (or ACK) just delayed (not lost):
❍ retransmission will be duplicate, butuse of seq. #’s already handles this
pkt(0)
pkt(0)
ACK
pkt(0)
pkt(1)ACK
ACKDiscard (no dupl)Resend ACK
❒ However, raceconditions arepossible between thereceived ACK and theretransmitted packet!
Discard again,Still expects #1!
B ACKedbut lost!
C ACKedbut lost!
X
Timeout
A
B
C
A received
B not recovered
C not recovered
Note: such race conditions are only possible over full-duplex channels
© From Computer Networking, by Kurose&Ross Transport Layer 3-38
Fixing rdt3.0
❒ receiver must specify seq # of pkt being ACKed
pkt(0)
pkt(0)
ACK
pkt(1)
pkt(1)ACK(0)
ACK(0)Discard (no dupl)Resend ACK
B notACKed
X
Timeout
A
B
A received
B recoveredTimeout
20
© From Computer Networking, by Kurose&Ross Transport Layer 3-39
rdt3.0 sendersndpkt = make_pkt(0, data, chksum)udt_send(sndpkt)start_timer
rdt_send(data)
Waitfor
ACK0
rdt_rcv(rcvpkt) &&( corrupt(rcvpkt) ||isACK(rcvpkt,1) )
Wait forcall 1 from
above
sndpkt = make_pkt(1, data, chksum)udt_send(sndpkt)start_timer
rdt_send(data)
rdt_rcv(rcvpkt)&& not corrupt(rcvpkt)&& isACK(rcvpkt,0)
rdt_rcv(rcvpkt) &&( corrupt(rcvpkt) ||isACK(rcvpkt,0) )
rdt_rcv(rcvpkt)&& not corrupt(rcvpkt)&& isACK(rcvpkt,1)
stop_timerstop_timer
udt_send(sndpkt)start_timer
timeout
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait forcall 0from
above
Waitfor
ACK1
Λrdt_rcv(rcvpkt)
ΛΛ
Λ
© From Computer Networking, by Kurose&Ross Transport Layer 3-40
rdt3.0 = Alternating-bit protocol(1969)
21
© From Computer Networking, by Kurose&Ross Transport Layer 3-41
rdt3.0 in action (2)
© From Computer Networking, by Kurose&Ross Transport Layer 3-42
rdt3.0 still incorrect if pkt or ACKreordering is possible
pkt(0)
pkt(0)
ACK(0)
pkt(1)
ACK(0)
ACK(1)
Timeouton pkt(0)
Duplicate!Consideredas new pkt(0)!pkt(0)pkt(0)
ACKedbut lost!
pkt reordering
Solution: Choose timeout so large that when a pkt is retransmitted the sender is surethat the previous copy of this pkt and its ACK have disappeared from the network.Better solution: Use a much larger seq# space (see later).
X
Can be arbitrarily small
22
© From Computer Networking, by Kurose&Ross Transport Layer 3-43
Performance of rdt3.0:stop-and-wait operation
first packet bit transmitted, t = 0
sender receiver
RTT
last packet bit transmitted, t = L / R
first packet bit arriveslast packet bit arrives, sendACK
ACK arrives, send nextpacket, t = RTT + L / R
!
Usender
=
LR
RTT + LR
=1
1+R " RTT
L
R . RTT/2 = Bandwidth-delay product= “in-flight” bits
U sender: utilization = fraction of time sender busy sending
© From Computer Networking, by Kurose&Ross Transport Layer 3-44
Performance of rdt3.0❒ Was OK over local low-speed networks, but…❒ Example: 1 Gbps link, 15 ms end-to-end prop. delay, 1KB packet:
Ttransmit = 8kb10**9 bps = 8 µsec
U sender
= .008
30.008 = 0.00027
microsec
onds
L / R
RTT + L / R =
L (packet length in bits)R (transmission rate, bps) =
❍ 1KB pkt every 30 msec -> 33kB/sec throughput over 1 Gbps link❍ network protocol limits use of physical resources!
23
© From Computer Networking, by Kurose&Ross Transport Layer 3-45
Pipelined protocolsPipelining: sender allows multiple, “in-flight”, yet-to-
be-acknowledged pkts❍ range of sequence numbers must be increased❍ buffering at sender and/or receiver
❒ Two generic forms of pipelined protocols:go-Back-N, selective repeat
© From Computer Networking, by Kurose&Ross Transport Layer 3-46
Pipelining: increased utilization
first packet bit transmitted, t = 0
sender receiver
RTT
last bit transmitted, t = L / R
first packet bit arriveslast packet bit arrives, send ACK
ACK arrives, send nextpacket, t = RTT + L / R
last bit of 2nd packet arrives, send ACKlast bit of 3rd packet arrives, send ACK
U sender
= .024
30.008 = 0.0008
microsecon
ds
3 * L / R
RTT + L / R =
Increase utilizationby a factor of 3!
24
© From Computer Networking, by Kurose&Ross Transport Layer 3-47
Pipelining Protocols
Go-back-N: overview❒ sender: up to N
unACKed pkts inpipeline
❒ receiver: only sendscumulative ACKs❍ doesn’t ACK pkt if
there’s a gap❒ sender: has timer for
oldest unACKed pkt❍ if timer expires:
retransmit all unACKedpackets
Selective Repeat: overview❒ sender: up to N unACKed
packets in pipeline❒ receiver: ACKs individual
pkts❒ sender: maintains timer
for each unACKed pkt❍ if timer expires: retransmit
only unACKed packet
© From Computer Networking, by Kurose&Ross Transport Layer 3-48
Go-Back-N (GBN)Sender:❒ k-bit seq # in pkt header❒ “window” of up to N, consecutive unack’ed pkts allowed
❒ ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK”❍ may receive duplicate ACKs (see receiver)
❒ single timer: conceptually on the oldest transmitted and unack’ed pkt❒ timeout: retransmit all pkts in window
25
© From Computer Networking, by Kurose&Ross Transport Layer 3-49
GBN: sender extended FSM
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])…udt_send(sndpkt[nextseqnum-1])
timeout
rdt_send(data)
if (nextseqnum < base+N) { sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ }else refuse_data(data)
base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Λ
Λ
© From Computer Networking, by Kurose&Ross Transport Layer 3-50
GBN: receiver extended FSM
ACK-only: always send ACK for correctly-received pktwith highest in-order seq #❍ may generate duplicate ACKs❍ need only remember expectedseqnum
❒ out-of-order pkt:❍ discard (don’t buffer) -> no receiver buffering!❍ Re-ACK pkt with highest in-order seq #
Wait
udt_send(sndpkt)default
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum)
data = extract(rcvpkt)deliver_data(data)sndpkt = make_pkt(expectedseqnum,ACK,chksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0,ACK,chksum)
Λ
26
© From Computer Networking, by Kurose&Ross Transport Layer 3-51
GBN inaction
Window size (N) = 4
© From Computer Networking, by Kurose&Ross Transport Layer 3-52
Maximum window size with GBN❒ If seq# size = K (“The modulo”)❒ Q: What is the maximum window size N?
01234567012
3
ACK2
DiscardACK2
01234567012
3
ACK2
ACK2
Suppose K = 8, N =8No pkt loss
W=[3..2] W=[3..2]
With pkt loss:For sender: same view
For receiver: different view!
XX
XXXXX
27
© From Computer Networking, by Kurose&Ross Transport Layer 3-53
GBN: maximum window size❒ Constraint:
❍ window size (N) ≤ seq# size (K) - 1
0123456701
resend 3resend 4…
ACK2
DiscardACK2
W=[3..1]
XXXXX
X
© From Computer Networking, by Kurose&Ross Transport Layer 3-54
Selective Repeat
❒ receiver individually acknowledges all correctlyreceived pkts❍ buffers pkts, as needed, for eventual in-order delivery
to upper layer❒ sender only resends pkts for which ACK not
received❍ sender timer for each unACKed pkt
❒ sender window❍ N consecutive seq #’s❍ again limits seq #s of sent, unACKed pkts
28
© From Computer Networking, by Kurose&Ross Transport Layer 3-55
Selective repeat: sender, receiver windows
© From Computer Networking, by Kurose&Ross Transport Layer 3-56
Selective repeat
data from above :❒ if next available seq # in
window, send pkttimeout(n):❒ resend pkt n, restart timerACK(n) in [sendbase,sendbase+N-1]:
❒ mark pkt n as received❒ if n smallest unACKed pkt,
advance window base to nextunACKed seq #
senderpkt n in [rcvbase, rcvbase+N-1]
❒ send ACK(n)❒ out-of-order: buffer❒ in-order: deliver (also
deliver buffered, in-orderpkts), advance window tonext not-yet-received pkt
pkt n in [rcvbase-N,rcvbase-1]
❒ ACK(n)otherwise:❒ ignore
receiver
29
© From Computer Networking, by Kurose&Ross Transport Layer 3-57
Selective repeat in action
© From Computer Networking, by Kurose&Ross Transport Layer 3-58
Selective repeat: dilemmaExample:❒ seq #’s: 0, 1, 2, 3❒ window size=3
❒ receiver sees nodifference in twoscenarios!
❒ incorrectly passesduplicate data as newin (a)
Q: what relationshipbetween seq # sizeand window size?
30
© From Computer Networking, by Kurose&Ross Transport Layer 3-59
Maximum window size withselective repeat❒ If seq# size = K (“The modulo”)❒ Q: What is the maximum window size N?❒ A: N ≤ K/2
(a) Initial situation with a window of size N=3. (b) After 3 frames have been sent and received but not ACK’ed.
The upper side of the rec window falls in the sending window(c) Initial situation with a window of size N=2. (d) After 2 frames sent and received but not ACK’ed.
The upper side of the rec window does not fall in the sending window
Sender 0 1 2 3 0 1 2 3 0 1 2 3 0 1 2 3
Receiver 0 1 2 3 0 1 2 3 0 1 2 3 0 1 2 3
(a) (b) (c) (d)
Suppose K = 4, N =3 K = 4, but N =2
Retransmission of received pkts fall
outside rec window
© From Computer Networking, by Kurose&Ross Transport Layer 3-60
Maximum window size:general principle❒ Notations:
❍ seq# space = [0..K-1], i.e. modulo K❍ Maximum sender window size = Ns❍ Maximum receiver window size = Nr
❒ Requirement:❍ Ns + Nr ≤ K
❒ Particular cases:❍ GBN: Nr = 1, Ns ≤ K - 1❍ Selective repeat: Ns = Nr ≤ K/2❍ Alternating-bit: K = 2, Ns = Nr = 1
• Special case of both GBN and Selective Repeat
31
© From Computer Networking, by Kurose&Ross Transport Layer 3-61
Chapter 3 outline
❒ 3.1 Transport-layerservices
❒ 3.2 Multiplexing anddemultiplexing
❒ 3.3 Connectionlesstransport: UDP
❒ 3.4 Principles ofreliable data transfer
❒ 3.5 Connection-orientedtransport: TCP❍ segment structure❍ reliable data transfer❍ flow control❍ connection management
❒ 3.6 Principles ofcongestion control
❒ 3.7 TCP congestioncontrol
© From Computer Networking, by Kurose&Ross Transport Layer 3-62
TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581
❒ full duplex data:❍ bi-directional data flow
in same connection❍ MSS: maximum segment
size❒ connection-oriented:
❍ handshaking (exchangeof control msgs) init’ssender, receiver statebefore data exchange
❒ flow controlled:❍ sender will not
overwhelm receiver
❒ point-to-point:❍ one sender, one receiver
❒ reliable, in-order bytestream:❍ no “message boundaries”
❒ pipelined:❍ TCP congestion and flow
control set window size❒ send & receive buffers
socket
door
TCP
send buffer
TCP
receive buffer
socket
door
segment
application
writes data
application
reads data
32
© From Computer Networking, by Kurose&Ross Transport Layer 3-63
TCP segment structure
source port # dest port #
32 bits
applicationdata
(variable length)
sequence numberacknowledgement number
Receive windowUrg 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, FIN:connection estab(setup, teardown
commands)
# bytes rcvr willingto accept
countingby bytes of data(not segments!)
Internetchecksum
(as in UDP)
© From Computer Networking, by Kurose&Ross Transport Layer 3-64
TCP seq. #’s and ACKsSeq. #’s:
❍ byte stream“number” of firstbyte in segment’sdata
ACKs:❍ seq # of next byte
expected fromother side
❍ cumulative ACKQ: how receiver handles
out-of-order segments❍ A: TCP spec doesn’t
say, - up toimplementer
Host A Host B
Seq=42, ACK=79, data = ‘C’
Seq=79, ACK=43, data = ‘C’
Seq=43, ACK=80
Usertypes
‘C’
host ACKsreceipt
of echoed‘C’
host ACKsreceipt of‘C’, echoes
back ‘C’
timesimple telnet scenario
33
© From Computer Networking, by Kurose&Ross Transport Layer 3-65
TCP Round Trip Time and TimeoutQ: how to set TCP timeout value?❒ longer than RTT
❍ but RTT varies❒ too short: premature timeout
❍ unnecessary retransmissions❒ too long: slow reaction to segment loss
0 1 2 3 4 5RTT (msec)
T
0
0.1
0.2
0.3
Prob.density
0
0.1
0.2
0.3
Prob.density
0 10 20 30 40 50RTT (msec)
T1 T2
Over a link
Over a network
From Computer Networks, by Tanenbaum © Prentice Hall
© From Computer Networking, by Kurose&Ross Transport Layer 3-66
TCP Round Trip Time and Timeout
EstimatedRTT = (1-α)*EstimatedRTT + α*SampleRTT
❒ Exponential weighted moving average❒ Influence of past sample decreases exponentially fast❒ Typical value: α = 0.125
Q: how to estimate RTT?❒ SampleRTT: measured time from segment transmission until ACK receipt
❍ ignore retransmissions❒ SampleRTT will vary, want estimated RTT “smoother”
❍ average several recent measurements, not just current SampleRTT
34
© From Computer Networking, by Kurose&Ross Transport Layer 3-67
Example RTT estimation:RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
100
150
200
250
300
350
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
RT
T (
mil
lise
con
ds)
SampleRTT Estimated RTT
© From Computer Networking, by Kurose&Ross Transport Layer 3-68
TCP Round Trip Time and TimeoutSetting the timeout❒ EstimatedRTT plus “safety margin”
❍ large variation in EstimatedRTT -> larger safety margin❒ first estimate of how much SampleRTT deviates from
EstimatedRTT:
TimeoutInterval = EstimatedRTT + 4*DevRTT
DevRTT = (1-β)*DevRTT + β*|SampleRTT-EstimatedRTT|
(typically, β = 0.25)
Then set timeout interval:
35
© From Computer Networking, by Kurose&Ross Transport Layer 3-69
Chapter 3 outline
❒ 3.1 Transport-layerservices
❒ 3.2 Multiplexing anddemultiplexing
❒ 3.3 Connectionlesstransport: UDP
❒ 3.4 Principles ofreliable data transfer
❒ 3.5 Connection-orientedtransport: TCP❍ segment structure❍ reliable data transfer❍ flow control❍ connection management
❒ 3.6 Principles ofcongestion control
❒ 3.7 TCP congestioncontrol
© From Computer Networking, by Kurose&Ross Transport Layer 3-70
TCP reliable data transfer
❒ TCP creates rdtservice on top of IP’sunreliable service
❒ Pipelined segments❒ Cumulative acks❒ TCP uses single
retransmission timer
❒ Retransmissions aretriggered by:❍ timeout events❍ duplicate acks
❒ Initially considersimplified TCP sender:❍ ignore duplicate acks❍ ignore flow control,
congestion control
36
© From Computer Networking, by Kurose&Ross Transport Layer 3-71
TCP sender events:data rcvd from app:❒ Create segment with
seq #❒ seq # is byte-stream
number of first databyte in segment
❒ start timer if notalready running (thinkof timer as for oldestunacked segment)
❒ expiration interval:TimeOutInterval
timeout:❒ retransmit segment
that caused timeout❒ restart timer Ack rcvd:❒ If acknowledges
previously unackedsegments❍ update what is known to
be acked❍ start timer if there are
outstanding segments
© From Computer Networking, by Kurose&Ross Transport Layer 3-72
TCPsender(simplified)
NextSeqNum = InitialSeqNum SendBase = InitialSeqNum
loop (forever) { switch(event)
event: data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)
event: timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer
event: ACK received, with ACK field value of y if (y > SendBase) { SendBase = y if (there are currently not-yet-acknowledged segments) start timer } } /* end of loop forever */
Comment:• SendBase-1: last cumulatively ack’ed byteExample:• SendBase-1 = 71;y= 73, so the rcvrwants 73+ ;y > SendBase, sothat new data is acked
37
© From Computer Networking, by Kurose&Ross Transport Layer 3-73
TCP: retransmission scenariosHost 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 t
imeo
ut
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
SendBase= 100
SendBase = 120
Sendbase = 100Timer start
Sendbase = 92
Timer start
© From Computer Networking, by Kurose&Ross Transport Layer 3-74
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
38
© From Computer Networking, by Kurose&Ross Transport Layer 3-75
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 segment,send 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 starts at lower end of gap
© From Computer Networking, by Kurose&Ross Transport Layer 3-76
Fast Retransmit
❒ Time-out period oftenrelatively long:❍ long delay before
resending lost packet❒ Detect lost segments
via duplicate ACKs:❍ Sender often sends
many segments back-to-back
❍ If segment is lost,there will likely be manyduplicate ACKs for thatsegment.
❒ If sender receives 3ACKs for the samedata, it supposes thatsegment after ACKeddata was lost:❍ fast retransmit: resend
segment before timerexpires
39
© From Computer Networking, by Kurose&Ross Transport Layer 3-77
Host A
tim
eout
Host B
time
X
resend seq # x2
seq # x1seq # x2seq # x3seq # x4seq # x5
ACK x2
ACK x2ACK x2ACK x2
tripleduplicate
ACKs
= Waiting for byte x2
© From Computer Networking, by Kurose&Ross Transport Layer 3-78
event: ACK received, with ACK field value of y if (y > 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 retransmit algorithm:
a duplicate ACK for already ACKed segment
fast retransmit
40
© From Computer Networking, by Kurose&Ross Transport Layer 3-79
Chapter 3 outline
❒ 3.1 Transport-layerservices
❒ 3.2 Multiplexing anddemultiplexing
❒ 3.3 Connectionlesstransport: UDP
❒ 3.4 Principles ofreliable data transfer
❒ 3.5 Connection-orientedtransport: TCP❍ segment structure❍ reliable data transfer❍ flow control❍ connection management
❒ 3.6 Principles ofcongestion control
❒ 3.7 TCP congestioncontrol
© From Computer Networking, by Kurose&Ross Transport Layer 3-80
TCP Flow Control
❒ receive side of TCPconnection has areceive buffer:
❒ speed-matchingservice: matchingsend rate to receivingapplication’s drain rate
❒ app process may beslow at reading frombuffer
sender won’t overflowreceiver’s buffer by
transmitting too much, too fast
flow control
IPdatagrams
TCP data(in buffer)
(currently)unused buffer
space
applicationprocess
41
© From Computer Networking, by Kurose&Ross Transport Layer 3-81
TCP Flow control: how it works
(suppose TCP receiverdiscards out-of-ordersegments)
❒ unused buffer space:= rwnd= RcvBuffer-[LastByteRcvd -
LastByteRead]
❒ receiver: advertisesunused buffer space byincluding rwnd value insegment header
❒ sender: limits # ofunACKed bytes to rwnd❍ guarantees receiver’s
buffer doesn’t overflow
IPdatagrams
TCP data(in buffer)
(currently)unused buffer
space
applicationprocess
rwndRcvBuffer
© From Computer Networking, by Kurose&Ross Transport Layer 3-82
Nagle algorithm❒ When data come in from socket one byte at the
time❍ send first byte and buffer all the rest until the
outstanding byte is acknowledged❍ sends other segments as one per RTT
❒ Useful for Telnet❍ Otherwise:❍ 41 bytes segments containing 1 byte of data❍ 40 bytes of TCP/IP header overhead
42
© From Computer Networking, by Kurose&Ross Transport Layer 3-83
Silly window syndrome
Receiverʼs buffer is full
Application reads 1 byte
Window update segment sent
New byte arrives
Receiverʼs buffer is full
Header
Header
Room for one more byte
1 Byte
Still a problemwith applicationsreading one byteat a time
Clark's solution: receiver sends a windowupdate only if the buffer is half empty, or if a full segment can be received
From Computer Networks, by Tanenbaum © Prentice Hall
© From Computer Networking, by Kurose&Ross Transport Layer 3-84
Chapter 3 outline
❒ 3.1 Transport-layerservices
❒ 3.2 Multiplexing anddemultiplexing
❒ 3.3 Connectionlesstransport: UDP
❒ 3.4 Principles ofreliable data transfer
❒ 3.5 Connection-orientedtransport: TCP❍ segment structure❍ reliable data transfer❍ flow control❍ connection management
❒ 3.6 Principles ofcongestion control
❒ 3.7 TCP congestioncontrol
43
© From Computer Networking, by Kurose&Ross Transport Layer 3-85
TCP Connection ManagementRecall:
TCP sender, receiver establish “connection” beforeexchanging data segments
❒ initialize TCP variables:❍ seq. #s❍ buffers, flow control info (e.g. RcvWindow)
❒ client: connection initiator Socket clientSocket = new Socket("hostname","port
number");
❒ server: contacted by client Socket connectionSocket = welcomeSocket.accept();
© From Computer Networking, by Kurose&Ross Transport Layer 3-86
Three-way handshakeStep 1: client host sends TCP SYN
segment to server❍ specifies initial seq #❍ no data❍ starts timer for SYN
retransmissionStep 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
client
SYN (seq=x)
server
SYNACK (seq=y, ack=x+1)
ACK (seq=x+1, ack=y+1)
connectlisten
+ accept
connected
connected
If server port not open, TCP server sends back a RST segment
44
© From Computer Networking, by Kurose&Ross Transport Layer 3-87
TCP Connection Management (cont.)Closing a connection:client closes socket:
clientSocket.close();
Step 1: client end-system sendsTCP FIN control segment toserver
Step 2: server receives FIN,replies with ACK. When alldata sent, closes connection,sends FIN
Note: two directions closedseparately
Symmetric/graceful release
client
FIN
server
ACK
ACK
FIN
close
close
closed
tim
ed w
ait
© From Computer Networking, by Kurose&Ross Transport Layer 3-88
TCP Connection Management (cont.)Step 3: client receives FIN,
replies with ACK.
❍ Enters “timed wait” - willrespond with ACK toreceived FINs
Step 4: server, receives ACK.Connection closed.
FINs have seq#, like datasegments
Can recover loss of lastdata bytes
Note: with small modification, canhandle simultaneous FINs.
client
FIN
server
ACK
ACK
FIN
closing
closing
closed
tim
ed w
ait
closed
45
© From Computer Networking, by Kurose&Ross Transport Layer 3-89
TCP Connection Management (cont)
TCP clientlifecycle
TCP serverlifecycle
© From Computer Networking, by Kurose&Ross Transport Layer 3-90
Choosing initial seq#❒ Consider choosing x❒ Constraint: x not used
recently in a formerconnection by this client (i.e.same port#) to send TCPsegments to this server(same port#)
❒ Recently = less than packetlifetime
❒ Otherwise a still aliveduplicate segment from theformer connection can beconfused with this SYN
client
SYN (seq=x)
server
SYNACK (seq=y, ack=x+1)
ACK (seq=x+1, ack=y+1)
connect listen+ accept
connected
connected
46
© From Computer Networking, by Kurose&Ross Transport Layer 3-91
Choosing initial seq# (2)
1. In practice1. If client can only reuse the same port# when a
maximum packet lifetime has elapsed after the lastpacket has been sent, and
2. If, when a host fails (= unknown duration), it waits atleast a maximum packet lifetime before opening thefirst connection,
3. Then, any initial seq# x and y are OK, they could berandomly chosen
2. In principle, could avoid these delays by usingclocks (see next slide)
© From Computer Networking, by Kurose&Ross Transport Layer 3-92
Using clocks❒ Let T be the maximum packet lifetime in the network❒ Each host has a "clock”
❍ that never stops running, even when the station is down❍ that need not be synchronized with other clocks
❒ The clock is merely a regular and reliable counter of say k bitssuch that 2k ticks > T
2k-1
t
47
© From Computer Networking, by Kurose&Ross Transport Layer 3-93
Using clocks (2)❒ Initial seq# x is the local
clock value❒ Don’t enter forbidden
region (neither from top,nor from bottom)❍ Throughput is bounded!
t
T
x
SYN xis sent
DT x+1is sentDT x+kis sent
y
SYN yis sent
New connectionbetween same portsof same hosts
z
At that time,an old duplicate DT z
anda new packet DT z
can be both present in the network
clock Numbers used byconnection
Forb
idden
regio
n
© From Computer Networking, by Kurose&Ross Transport Layer 3-94
Chapter 3 outline
❒ 3.1 Transport-layerservices
❒ 3.2 Multiplexing anddemultiplexing
❒ 3.3 Connectionlesstransport: UDP
❒ 3.4 Principles ofreliable data transfer
❒ 3.5 Connection-orientedtransport: TCP❍ segment structure❍ reliable data transfer❍ flow control❍ connection management
❒ 3.6 Principles ofcongestion control
❒ 3.7 TCP congestioncontrol
48
© From Computer Networking, by Kurose&Ross Transport Layer 3-95
Principles of Congestion Control
❒ informally: “too many sources sending too much data too fast fornetwork to handle”
❒ manifestations:❍ lost packets (buffer overflow at routers)❍ long delays (queuing in router buffers)
❒ a top-10 problem!
Transmissionrate adjustment
Transmissionnetwork
Small-capacityreceiver
Large-capacityreceiver
InternalcongestionNeed
flowcontrol
Needcongestion
control
From Computer Networks, by Tanenbaum © Prentice Hall
© From Computer Networking, by Kurose&Ross Transport Layer 3-96
Causes/costs of congestion: scenario 1
❒ two senders, tworeceivers
❒ one router,infinite buffers
❒ no retransmission
❒ large delayswhen congested
❒ maximumachievablethroughput
unlimited sharedoutput link buffers
Host Aλin : original data
Host B
λout
capacity = C
49
© From Computer Networking, by Kurose&Ross Transport Layer 3-97
Causes/costs of congestion: scenario 2
❒ one router, finite buffers❒ sender retransmission of lost packet
finite shared outputlink buffers
Host A λin : original data
Host B
λout
λ'in : original data, plusretransmitted data
capacity = C
© From Computer Networking, by Kurose&Ross Transport Layer 3-98
Causes/costs of congestion: scenario 2❒ always: λin = λout (goodput)
❒ “perfect” retransmission only when loss: λ’in > λout
❒ retransmission of delayed (not lost) packet makes λ’in larger (than perfect case)
for same λout
“costs” of congestion:❒ more work (retransmission) for given “goodput”❒ unneeded retransmissions: link carries multiple copies of packet
b) only lost packetsretransmitted
C/2
C/2λin
λ out
a) sender avoids loss c) all packetsretransmitted once
C/2
C/2λin
λ out
C/4
C/2
C/2λin
λ out
e.g. C/3
50
© From Computer Networking, by Kurose&Ross Transport Layer 3-99
Causes/costs of congestion: scenario 3❒ four senders❒ multihop paths❒ timeout/retransmit
λin
Q: what happens asand increase ?λ
in
finite shared outputlink buffers
Host Aλin : original data
Host B
λout
λ'in : original data, plusretransmitted data
© From Computer Networking, by Kurose&Ross Transport Layer 3-100
Causes/costs of congestion: scenario 3
Another “cost” of congestion:❒ when packet dropped, any “upstream” transmission
capacity used for that packet was wasted!
HostA
HostB
λout
51
© From Computer Networking, by Kurose&Ross Transport Layer 3-101
Approaches towards congestion control
End-end congestioncontrol:
❒ no explicit feedback fromnetwork
❒ congestion inferred fromend-system observed loss,delay
❒ approach taken by TCP
Network-assistedcongestion control:
❒ routers provide feedbackto end-systems❍ single bit indicating
congestion (SNA,DECbit, TCP/IP ECN,ATM)
❍ explicit rate sendershould send at (ATMABR)
❒ Not covered in this course
Two broad approaches towards congestion control:
© From Computer Networking, by Kurose&Ross Transport Layer 3-102
Chapter 3 outline
❒ 3.1 Transport-layerservices
❒ 3.2 Multiplexing anddemultiplexing
❒ 3.3 Connectionlesstransport: UDP
❒ 3.4 Principles ofreliable data transfer
❒ 3.5 Connection-orientedtransport: TCP❍ segment structure❍ reliable data transfer❍ flow control❍ connection management
❒ 3.6 Principles ofcongestion control
❒ 3.7 TCP congestioncontrol
52
© From Computer Networking, by Kurose&Ross Transport Layer 3-103
TCP congestion control:❒ goal: TCP sender should transmit as fast as possible,
but without congesting network❍ Q: how to find rate just below congestion level
❒ decentralized: each TCP sender sets its own rate,based on implicit feedback:❍ ACK: segment received (a good thing!), network not
congested, so increase sending rate❍ lost segment: assume loss due to congested
network, so decrease sending rate
© From Computer Networking, by Kurose&Ross Transport Layer 3-104
TCP congestion control: bandwidth probing
❒ “probing for bandwidth”: increase transmission rateon receipt of ACK, until eventually loss occurs, thendecrease transmission rate❍ continue to increase on ACK, decrease on loss (since available
bandwidth is changing, depending on other connections innetwork)
ACKs being received, so increase rate
X
X
XX
X loss, so decrease rate
send
ing
rate
time
❒ Q: how fast to increase/decrease?❍ details to follow
TCP’s“sawtooth”behavior
53
© From Computer Networking, by Kurose&Ross Transport Layer 3-105
TCP Congestion Control: details
❒ sender limits rate by limiting numberof unACKed bytes “in pipeline”:
❍ cwnd: differs from rwnd (how, why?)❍ sender limited by min(cwnd,rwnd)
❒ roughly,
❒ cwnd is dynamic, function ofperceived network congestion
rate = cwnd
RTT bytes/sec
LastByteSent-LastByteAcked ≤ cwnd
cwndbytes
RTT
ACK(s)
© From Computer Networking, by Kurose&Ross Transport Layer 3-106
TCP Congestion Control: more details
segment loss event:reducing cwnd
❒ timeout: no responsefrom receiver❍ cut cwnd to 1
❒ 3 duplicate ACKs: atleast some segmentsgetting through (recallfast retransmit)❍ cut cwnd in half, less
aggressively than ontimeout
ACK received: increasecwnd
❒ slowstart phase:❍ increase exponentially
fast (despite name) atconnection start, orfollowing timeout
❒ congestion avoidance:❍ increase linearly
54
© From Computer Networking, by Kurose&Ross Transport Layer 3-107
TCP Slow Start❒ when connection begins, cwnd = 1
MSS (Maximum Segment Size)❍ example: MSS = 500 bytes &
RTT = 200 msec❍ initial rate = 20 kbps
❒ available bandwidth may be >>MSS/RTT❍ desirable to quickly ramp up to
respectable rate❒ increase rate exponentially until
first loss event or when thresholdreached❍ double cwnd every RTT❍ done by incrementing cwnd by
1 for every ACK received
Host A
one segment
RTT
Host B
time
two segments
four segments
1
2
34
cwnd
© From Computer Networking, by Kurose&Ross Transport Layer 3-108
Transitioning into/out of slowstartssthresh: cwnd threshold maintained by TCP❒ on loss event: set ssthresh to cwnd/2
❍ remember (half of) TCP rate when congestion last occurred❒ when cwnd >= ssthresh: transition from slowstart to congestion
avoidance phase
slow start timeout
ssthresh = cwnd/2cwnd = 1 MSS
dupACKcount = 0retransmit missing segment timeout
ssthresh = cwnd/2 cwnd = 1 MSS
dupACKcount = 0retransmit missing segment
Λcwnd > ssthresh
cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s),as allowed
new ACKdupACKcount++duplicate ACK
Λ
cwnd = 1 MSSssthresh = 64 KBdupACKcount = 0 congestion
avoidance
55
© From Computer Networking, by Kurose&Ross Transport Layer 3-109
TCP: congestion avoidance
❒ when cwnd > ssthreshgrow cwnd linearly❍ increase cwnd by 1
MSS per RTT❍ approach possible
congestion slowerthan in slowstart
❍ implementation: cwnd= cwnd + MSS/cwndfor each ACK received
❒ ACKs: increase cwndby 1 MSS per RTT:additive increase
❒ loss: cut cwnd in half(non-timeout-detectedloss): multiplicativedecrease
AIMD
AIMD: Additive IncreaseMultiplicative Decrease
© From Computer Networking, by Kurose&Ross Transport Layer 3-110
TCP congestion control FSM: overview
slow start
congestionavoidance
fastrecovery
cwnd > ssthresh
loss:timeout
loss:timeout
new ACK loss:3dupACK
loss:3dupACK
loss:timeout
56
© From Computer Networking, by Kurose&Ross Transport Layer 3-111
TCP congestion control FSM: details
slow start
congestionavoidance
fastrecovery
timeoutssthresh = cwnd/2
cwnd = 1 MSSdupACKcount = 0
retransmit missing segment timeoutssthresh = cwnd/2
cwnd = 1 MSSdupACKcount = 0
retransmit missing segment
Λcwnd > ssthresh
cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s),as allowed
new ACKcwnd = cwnd + MSS (MSS/cwnd)
dupACKcount = 0transmit new segment(s),as allowed
new ACK.
dupACKcount++duplicate ACK
ssthresh= cwnd/2cwnd = ssthresh + 3 MSSretransmit missing segment
dupACKcount == 3
dupACKcount++duplicate ACK
ssthresh= cwnd/2cwnd = ssthresh + 3 MSS
retransmit missing segment
dupACKcount == 3
timeoutssthresh = cwnd/2cwnd = 1 dupACKcount = 0retransmit missing segment
cwnd = cwnd + MSStransmit new segment(s), as allowed
duplicate ACK
cwnd = ssthreshdupACKcount = 0
New ACK
Λ
cwnd = 1 MSSssthresh = 64 KBdupACKcount = 0
© From Computer Networking, by Kurose&Ross Transport Layer 3-112
Popular “flavors” of TCP
ssthresh
ssthresh
TCPTahoe
TCPReno
Transmissionround
cwnd windowsize(insegments)
57
© From Computer Networking, by Kurose&Ross Transport Layer 3-113
Summary: TCP Congestion Control
❒ when cwnd < ssthresh, sender in slow-start phase, windowgrows exponentially.
❒ when cwnd >= ssthresh, sender is in congestion-avoidancephase, window grows linearly.
❒ when triple duplicate ACK occurs, ssthresh set tocwnd/2, cwnd set to ~ ssthresh
❒ when timeout occurs, ssthresh set to cwnd/2, cwnd setto 1 MSS.
© From Computer Networking, by Kurose&Ross Transport Layer 3-114
TCP “goodput” in steady state❒ Goodput = not counting overhead and retransmitted bytes❒ What’s the average goodput of TCP as a function of window size and
RTT?❍ Ignoring slow start
❒ Let W be the window size (in MSS) when losses occur❒ When window is W, goodput is W/RTT❒ Just after loss, window drops to W/2, goodput to 0.5 W/RTT❒ Average goodput: 0.75 W/RTT
Win
dow
size
in M
SS W
W/2
W/2 . RTT
3W/4 . W/2MSS percycle
58
© From Computer Networking, by Kurose&Ross Transport Layer 3-115
TCP goodput in steady state (2)
❒ Average window size (in MSS) = 3W/4❒ Number of MSS per cycle = 3W/4 . W/2 = 3W2/8 = 1/p
❍ Where p is the packet loss ratio• One packet loss per cycle! (if p small enough)
❍ So
❒ Average goodput (in MSS/sec) = aver. window / RTT = 3W / 4RTT
❒ Average goodput (in bps)
!
W = 83p
= 32
1
RTT p
!
= 32
MSS
RTT p
© From Computer Networking, by Kurose&Ross Transport Layer 3-116
TCP Futures: TCP over “long, fat pipes”
❒ Example: 1500 byte segments, 100ms RTT,want 10 Gbps throughput
❒ Requires window size W = 83 333 in-flight segments❒ Throughput in terms of loss rate:
❒ Needs packet loss rate p = 2 . 10-10 Wow!❒ New versions of TCP for high-speed needed!
!
1.22 " MSS
RTT p
59
© From Computer Networking, by Kurose&Ross Transport Layer 3-117
Fairness goal: if K TCP sessions share samebottleneck link of bandwidth R, each should haveaverage rate of R/K
TCP connection 1
bottleneckrouter
capacity R
TCP connection 2
TCP Fairness
© From Computer Networking, by Kurose&Ross Transport Layer 3-118
Why is TCP fair?Two competing sessions having the same RTT:❒ Additive increase gives slope of 1, as throughout increases❒ Multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughput
Conn
ecti
on 2
thr
ough
put
1. congestion avoidance: additive increase
2. loss: decrease window by factor of 23. congestion avoidance: additive increase
4. loss: decrease window by factor of 2
60
© From Computer Networking, by Kurose&Ross Transport Layer 3-119
And when RTTs are different?❒ If RTT of connection 2 = 2 x RTT of connection 1❒ Connection 1 ramps up twice more quickly
R
R
bandwidth share inversely
proportional to RTT
Connection 1 throughput
Conn
ecti
on 2
thr
ough
put
© From Computer Networking, by Kurose&Ross Transport Layer 3-120
Fairness (more)Fairness and UDP❒ Multimedia apps often
do not use TCP❍ do not want rate
throttled by congestioncontrol
❒ Instead use UDP:❍ pump audio/video at
constant rate, toleratepacket loss
❒ Research area: TCPfriendly
Fairness and parallel TCPconnections
❒ nothing prevents app fromopening parallelconnections between 2hosts
❒ Web browsers do this❒ Example: link of rate R
supporting 9 connections;❍ new app asks for 1 TCP, gets
rate R/10❍ new app asks for 10 TCPs,
gets more than R/2 !
61
© From Computer Networking, by Kurose&Ross Transport Layer 3-121
Chapter 3: Summary❒ principles behind transport
layer services:❍ multiplexing,
demultiplexing❍ reliable data transfer❍ flow control❍ congestion control
❒ instantiation andimplementation in theInternet❍ UDP❍ TCP
Next:❒ leaving the network
“edge” (application,transport layers)
❒ into the network“core”