程序代写代做代考 dns case study ER Chapter 3 Transport Layer

Chapter 3 Transport Layer
Transport Layer 3-1

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

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

Transport services and protocols
 provide logical communication between app processes running on different hosts
 transport protocols run in end systems
 send side: breaks app messages into segments, passes to network layer
 rcv side: reassembles segments into messages, passes to app layer
 more than one transport protocol available to apps
 Internet: TCP and UDP
application
transport
network
data link
physical
application
transport
network
data link physical
Transport Layer 3-4

Transport vs. network layer
 network layer: logical communication between hosts
household analogy:
12 kids in Ann’s house sending letters to 12 kids in Bill’s house:
 hosts = houses
 processes = kids
 app messages = letters in envelopes
 transport protocol = Ann and Bill who demux to in- house siblings
 network-layer protocol = postal service
 transport layer: logical
communication between processes
relies on, enhances, network layer services
Transport Layer 3-5

Internet transport-layer protocols
 reliable, in-order delivery (TCP)
 congestion control  flow control
 connection setup
 unreliable, unordered delivery: UDP
 no-frills extension of “best-effort” IP
 services not available:  delay guarantees
 bandwidth guarantees
application
transport
network data link physical
application
transport
network
data link
physical
network
network
data link
data link physical
physical
network
data link
physical
network
data link
physical
network
data link physical
network
data link
physical
network data link physical
Transport Layer 3-6

IP Service
 The IP service model is a best-effort delivery service. This means that IP makes its “best effort” to deliver segments between communicating hosts, but it makes no guarantees.
 It does not guarantee segment delivery, it does not guarantee orderly delivery of segments, and it does not guarantee the integrity of the data in the segments. For these reasons, IP is said to be an unreliable service.
Transport Layer 3-7

What do UDP and TCP do?
 The most fundamental responsibility of UDP and TCP is to extend IP’s delivery service between two end systems to a delivery service between two processes running on the end systems.
 Extending host-to-host delivery to process-to- process delivery is called transport-layer multiplexing and demultiplexing.
Transport Layer 3-9

Multiplexing/demultiplexing
multiplexing at sender:
handle data from multiple sockets, add transport header (later used for demultiplexing)
demultiplexing at receiver:
use header info to deliver received segments to correct socket
socket
process
application
P1 P2
transport
network
link
physical
application
P4
transport
network
link
physical
application
P3
transport
network
link
physical
Transport Layer 3-10

How demultiplexing works
 host receives IP datagrams
 each datagram has source IP address, destination IP address
 each datagram carries one transport-layer segment
 each segment has source, destination port number
 host uses IP addresses & port numbers to direct segment to appropriate socket
32 bits
source port #
dest port #
other header fields
application data
(payload)
TCP/UDP segment format
Transport Layer 3-11

Connectionless demultiplexing
 recall: created socket has  recall: when creating
host-local port #:
DatagramSocket mySocket1
= new DatagramSocket(12534);
datagram to send into UDP socket, must specify
 destination IP address  destination port #
IP datagrams with same dest. port #, but different source IP addresses and/or source port numbers will be directed to same socket at dest
 when host receives UDP segment:
 checks destination port # in segment
 directs UDP segment to socket with that port #
Transport Layer 3-12

Connectionless demux: example
DatagramSocket
mySocket2 = new
DatagramSocket
(9157);
DatagramSocket
serverSocket = new
DatagramSocket
(6428);
DatagramSocket
mySocket1 = new
DatagramSocket
(5775);
application
P1
application
P4
transport
network
link
transport
network
link
physical
application
P3
transport
network
link
physical
physical
source port: 6428 dest port: 9157
source port: ? dest port: ?
source port: ? dest port: ?
source port: 9157 dest port: 6428
Transport Layer 3-13

Connection-oriented demux
 TCP socket identified by 4-tuple:
 source IP address
 source port number  dest IP address
 dest port number
 demux: receiver uses all four values to direct segment to appropriate socket
 server host may support many simultaneous TCP sockets:
 each socket identified by its own 4-tuple
 web servers have different sockets for each connecting client
non-persistent HTTP will have different socket for each request
Transport Layer 3-14

Connection-oriented demux: example
application
P4 P5 P6
transport
network
link
physical
application
P2 P3
transport
network
link
physical
application
P3
transport
network
link
physical
host: IP address A
source IP,port: B,80 dest IP,port: A,9157
source IP,port: A,9157 dest IP, port: B,80
server: IP address B
source IP,port: C,5775 dest IP,port: B,80
source IP,port: C,9157 dest IP,port: B,80
host: IP address C
three segments, all destined to IP address: B, dest port: 80 are demultiplexed to different sockets
Transport Layer 3-15

Connection-oriented demux: example
threaded server
application
P4
transport
network
link
physical
application
P2 P3
transport
network
link
physical
application
P3
transport
network
link
physical
host: IP address A
source IP,port: B,80 dest IP,port: A,9157
source IP,port: A,9157 dest IP, port: B,80
server: IP address B
source IP,port: C,5775 dest IP,port: B,80
source IP,port: C,9157 dest IP,port: B,80
host: IP address C
Transport Layer 3-16

TCP vs UDP Sockets
 Important
In contrast with UDP, two arriving TCP segments with different source IP addresses or source port numbers will (with the exception of a TCP segment carrying the original connection- establishment request) be directed to two different sockets.
Transport Layer 3-17

Why Use UDP? TCP Seems So much Easier?
 Finer application-level control over
what data is sent, and when
TCP – Congestion Control, Resending Segments UDP – ‘Closer’ to the Network Layer
 No connection establishment TCP – 3 Way Handshake, UDP – None
 No connection state
Less overhead for server, server can handle more UDP connections compared to TCP
 Small packet header overhead TCP – 20 Bytes, UDP 8 Bytes
Transport Layer 3-18

UDP: User Datagram Protocol [RFC 768]
 “no frills,” “bare bones” Internet transport protocol
 UDP use:
 streaming multimedia apps (loss tolerant, rate
 “best effort” service, sensitive)
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
 DNS  SNMP
 reliable transfer over UDP:
 add reliability at application layer
 application-specific error recovery!
Transport Layer 3-20

UDP: segment header
32 bits
length, in bytes of UDP segment, including header
why is there a UDP?
source port #
dest port #
length
application data
(payload)
checksum
 no connection establishment (which can add delay) ie DNS
 simple: no connection state at sender, receiver
 small header size
 no congestion control: UDP can blast away as fast as desired
UDP segment format
Transport Layer 3-21

UDP checksum
Goal: detect “errors” (e.g., flipped bits) in transmitted segment
sender:
 treat segment contents, including header fields, as sequence of 16-bit integers
 checksum: addition (one’s 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 ….
Transport Layer 3-22

Internet checksum: example
example: add two 16-bit integers
11110011001100110 11101010101010101
wraparound
sum checksum
11011101110111011 11011101110111100
10100010001000011 Note: when adding numbers, a carryout from the most
significant bit needs to be added to the result
Transport Layer 3-23

Principles of reliable data transfer
 important in application, transport, link layers  top-10 list of important networking topics!
 characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Transport Layer 3-25

Reliable data transfer: getting started
rdt_send(): called from above, (e.g., by app.). Passed data to deliver to receiver upper layer
deliver_data(): called by rdt to deliver data to upper
send receive side side
udt_send(): called by rdt, to transfer packet over unreliable channel to receiver
rdt_rcv(): called when packet arrives on rcv-side of channel
Transport Layer 3-28

Reliable data transfer: getting started
we’ll:
 incrementally develop sender, receiver sides of
reliable data transfer protocol (rdt)
 consider only unidirectional data transfer
 but control info will flow on both directions!
 use finite state machines (FSM) to specify sender,
receiver
state: when in this “state” next state uniquely determined by next event
event causing state transition actions taken on state transition
state 1
event actions
state 2
Transport Layer 3-29

rdt1.0: reliable transfer over a reliable channel
 underlying channel perfectly reliable  no bit errors
 no loss of packets
 separate FSMs for sender, receiver:
 sender sends data into underlying channel
 receiver reads data from underlying channel
Wait for call from above
rdt_send(data)
packet = make_pkt(data) udt_send(packet)
sender
Wait for call from below
rdt_rcv(packet)
extract (packet,data) deliver_data(data)
receiver
Transport Layer 3-30

rdt2.0: channel with bit errors
 underlying channel may flip bits in packet  checksum to detect bit errors
 the question: how to recover from errors:
 acknowledgements (ACKs): receiver explicitly tells sender that pkt received OK
 negative acknowledgements (NAKs): receiver explicitly tells sender that pkt had errors
sender retransmits pkt on receipt of NAK
How do humans recover from “errors”
 new mechanisms in rdt2.0 (beyond rdt1.0): during conversation?
 error detection
 receiver feedback: control msgs (ACK,NAK) rcvr- >sender
Transport Layer 3-31

rdt2.0: channel with bit errors
 underlying channel may flip bits in packet  checksum to detect bit errors
 the question: how to recover from errors:
 acknowledgements (ACKs): receiver explicitly tells sender
that pkt received OK
 negative acknowledgements (NAKs): receiver explicitly tells sender that pkt had errors
sender retransmits pkt on receipt of NAK
 new mechanisms in rdt2.0 (beyond rdt1.0):
 error detection
 feedback: control msgs (ACK,NAK) from receiver to sender
Transport Layer 3-32

rdt2.0: FSM specification
rdt_send(data)
sndpkt = make_pkt(data, checksum)
receiver
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
udt_send(NAK)
Wait for call from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(ACK)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(sndpkt) rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
sender
Wait for call from above
Wait for ACK or NAK
Transport Layer 3-33

rdt2.0: operation with no errors
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(sndpkt)
Wait for call from above
Wait for ACK or NAK
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
udt_send(NAK)
Wait for call from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(ACK)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
Transport Layer 3-34

rdt2.0: error scenario
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(sndpkt)
Wait for call from above
Wait for ACK or NAK
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
udt_send(NAK)
Wait for call from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(ACK)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
Transport Layer 3-35

rdt2.0 has a fatal flaw!
what happens if ACK/NAK corrupted?
 sender doesn’t know what happened at receiver!
 can’t just retransmit: possible duplicate
handling duplicates:
 sender retransmits current pkt if ACK/NAK corrupted
 sender adds sequence number to each pkt
 receiver discards (doesn’t deliver up) duplicate pkt
stop and wait
sender sends one packet, then waits for receiver response
Transport Layer 3-36

rdt2.1: sender, handles garbled ACK/NAKs
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) )
udt_send(sndpkt)
Wait for call 0 from above
Wait for ACK or NAK 0
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt)
L
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) )
udt_send(sndpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt)
L
Wait for ACK or NAK 1
rdt_send(data)
sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt)
Wait for call 1 from above
Transport Layer 3-37

rdt2.1: receiver, handles garbled ACK/NAKs
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) && has_seq1(rcvpkt)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq0(rcvpkt)
extract(rcvpkt,data) deliver_data(data)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
Wait for 0 from below
Wait for 1 from below
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) && has_seq0(rcvpkt)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt)
extract(rcvpkt,data) deliver_data(data)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
Transport Layer 3-38

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 “expected” pkt should have seq # of 0 or 1
receiver:
 must check if received packet is duplicate
 state indicates whether 0 or 1 is expected pkt seq #
 note: receiver can not know if its last ACK/NAK received OK at sender
Transport Layer 3-39

rdt2.2: a NAK-free protocol
 same functionality as rdt2.1, using ACKs only
 instead of NAK, receiver sends ACK for last pkt
received OK
 receiver must explicitly include seq # of pkt being ACKed
 duplicate ACK at sender results in same action as NAK: retransmit current pkt
Transport Layer 3-40

rdt2.2: sender, receiver fragments
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
Wait for call 0 from above
Wait for ACK 0
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) ||
isACK(rcvpkt,1) ) udt_send(sndpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt,0)
L
rdt_rcv(rcvpkt) && (corrupt(rcvpkt) ||
has_seq1(rcvpkt)) udt_send(sndpkt)
Wait for 0 from below
sender FSM fragment
receiver FSM fragment
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt)
extract(rcvpkt,data) deliver_data(data)
sndpkt = make_pkt(ACK1, chksum) udt_send(sndpkt)
Transport Layer 3-41

rdt3.0: channels with errors and loss
new assumption:
underlying channel can also lose packets (data, ACKs)
checksum, seq. #, ACKs, retransmissions will be of help … but not enough
approach: sender waits “reasonable” 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 seq. #’s already handles this
 receiver must specify seq # of pkt being ACKed
 requires countdown timer Transport Layer 3-42

rdt3.0 sender
rdt_send(data)
sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt)
start_timer L
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
rdt_rcv(rcvpkt)
L
Wait for call 0from above
Wait for ACK0
timeout udt_send(sndpkt) start_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt,0)
stop_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt,1)
stop_timer
timeout
udt_send(sndpkt) start_timer
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,0) )
L
Wait for ACK1
Wait for call 1 from above
rdt_rcv(rcvpkt)
L
rdt_send(data)
sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt)
start_timer
Transport Layer 3-43

rdt3.0 in action
sender
send pkt0
rcv ack0 send pkt1
rcv ack1 send pkt0
receiver sender
receiver
rcv pkt0 send ack0
pkt0
rcv pkt0 ack0 send ack0
pkt1
rcv pkt1 ack1 send ack1
pkt0 rcv pkt0 ack0 send ack0
(a) no loss
send pkt0 pkt0 rcv ack0 ack0
send pkt1 pkt1 X
timeout
loss
pkt1
resend pkt1
rcv pkt1 ack1 send ack1
pkt0
rcv pkt0 ack0 send ack0
rcv ack1 send pkt0
(b) packet loss
Transport Layer 3-44

rdt3.0 in action
sender
send pkt0
rcv ack0 send pkt1
receiver
rcv pkt0 send ack0
rcv pkt1 send ack1
sender
send pkt0
rcv ack0 send pkt1
receiver
pkt0 rcv pkt0 ack0 send ack0
pkt1 rcv pkt1 ack1 send ack1
pkt0
ack0
pkt1
ack1
pkt1 pkt0
ack1 ack0
pkt0
ack0
timeout
rcv ack1 send pkt0
rcv ack1 send pkt0
rcv pkt1 (detect duplicate)
resend pkt1
pkt1
X
timeout
loss
resend pkt1
rcv pkt1
ack1 (detect duplicate)
pkt0
rcv pkt0 ack0 send ack0
send ack1
rcv ack1 send pkt0
send ack1
rcv pkt0 send ack0
rcv pkt0 (detect duplicate)
(c) ACK loss
send ack0 (d) premature timeout/ delayed ACK
Transport Layer 3-45

Performance of rdt3.0
 rdt3.0 is correct, but performance stinks
 e.g.: 1 Gbps link, 15 ms prop. delay, 8000 bit packet:
D =L=8000bits = trans R 109 bits/sec
 U sender: utilization – fraction of time sender busy sending Usender= L/R = .008 = 0.00027
8 microsecs
RTT + L / R 30.008
if RTT=30 msec, 1KB pkt every 30 msec: 33kB/sec thruput over 1 Gbps link
 network protocol limits use of physical resources! Transport Layer 3-46

rdt3.0: stop-and-wait operation
sender first packet bit transmitted, t = 0
last packet bit transmitted, t = L / R RTT
ACK arrives, send next packet, t = RTT + L / R
Usender= L/R RTT + L / R
receiver
first packet bit arrives
last packet bit arrives, send ACK
= .008 30.008
= 0.00027
Transport Layer 3-47

Pipelined protocols
pipelining: sender allows multiple, “in-flight”, yet- to-be-acknowledged pkts
 range of sequence numbers must be increased  buffering at sender and/or receiver
 two generic forms of pipelined protocols: go-Back-N, selective repeat
Transport Layer 3-48

Pipelining: increased utilization
sender
receiver
first packet bit transmitted, t = 0 last bit transmitted, t = L / R
RTT
ACK arrives, send next packet, t = RTT + L / R
first packet bit arrives
last packet bit arrives, send ACK
last bit of 2nd packet arrives, send ACK last bit of 3rd packet arrives, send ACK
3-packet pipelining increases utilization by a factor of 3!
.0024 = 0.00081 30.008
U sender = 3L / R RTT + L / R
=
Transport Layer 3-49

Pipelined protocols: overview
Go-back-N:
 sender can have up to N unacked packets in pipeline
 receiver only sends cumulative ack
 doesn’t ack packet if there’s a gap
 sender has timer for oldest unacked packet
 when timer expires, retransmit all unacked packets
Selective Repeat:
 sender can have up to N unack’ed packets in pipeline
 rcvr sends individual ack for each packet
 sender maintains timer for each unacked packet
 when timer expires, retransmit only that unacked packet
Transport Layer 3-50

Go-Back-N: sender
 
k-bit seq # in pkt header
“window” of up to N, consecutive unack’ed pkts allowed

 
ACK(n): ACKs all pkts up to, including seq # n – “cumulative ACK”
 may receive duplicate ACKs (see receiver) timer for oldest in-flight pkt
timeout(n): retransmit packet n and all higher seq # pkts in window
Transport Layer 3-51

GBN: sender extended FSM
rdt_send(data)
if (nextseqnum < base+N) { sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) L base=1 nextseqnum=1 rdt_rcv(rcvpkt) && corrupt(rcvpkt) start_timer nextseqnum++ } else refuse_data(data) Wait timeout start_timer udt_send(sndpkt[base]) udt_send(sndpkt[base+1]) ... udt_send(sndpkt[nextseqnum-1]) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) base = getacknum(rcvpkt)+1 If (base == nextseqnum) stop_timer else start_timer Transport Layer 3-52 GBN: receiver extended FSM L expectedseqnum=1 sndpkt = default udt_send(sndpkt) Wait rdt_rcv(rcvpkt) && notcurrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpkt) expectedseqnum++ make_pkt(expectedseqnum,ACK,chksum) ACK-only: always send ACK for correctly-received pkt with highest in-order seq #  may generate duplicate ACKs  need only remember expectedseqnum  out-of-order pkt:  discard (don’t buffer): no receiver buffering! re-ACK pkt with highest in-order seq # Transport Layer 3-53 GBN in action sender window (N=4) 012345678 012345678 45678 45678 012345678 012345678 sender receiver receive pkt0, send ack0 receive pkt1, send ack1 receive pkt3, discard, (re)send ack1 receive pkt4, discard, (re)send ack1 receive pkt5, discard, (re)send ack1 rcv pkt2, deliver, send ack2 rcv pkt3, deliver, send ack3 rcv pkt4, deliver, send ack4 rcv pkt5, deliver, send ack5 send pkt0 send pkt1 send pkt2 send pkt3 0123 0123 012345678 012345678 01 678 01 678 rcv ack0, send pkt4 rcv ack1, send pkt5 ignore duplicate ACK pkt 2 timeout send pkt2 send pkt3 send pkt4 send pkt5 Xloss (wait) 2345 2345 Transport Layer 3-54 Selective repeat  receiver individually acknowledges all correctly received pkts  buffers pkts, as needed, for eventual in-order delivery to upper layer  sender only resends pkts for which ACK not received  sender timer for each unACKed pkt  sender window  N consecutive seq #’s  limits seq #s of sent, unACKed pkts Transport Layer 3-55 Selective repeat: sender, receiver windows Transport Layer 3-56 Selective repeat sender data from above:  if next available seq # in window, send pkt timeout(n):  resend pkt n, restart timer ACK(n) in [sendbase,sendbase+N]:  mark pkt n as received  if n smallest unACKed pkt, advance window base to next unACKed seq # receiver pkt 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-N,rcvbase-1]  ACK(n) otherwise:  ignore Transport Layer 3-57 Selective repeat in action sender window (N=4) 012345678 012345678 45678 45678 012345678 012345678 sender receiver receive pkt0, send ack0 receive pkt1, send ack1 receive pkt3, buffer, send ack3 receive pkt4, buffer, send ack4 receive pkt5, buffer, send ack5 rcv pkt2; deliver pkt2, pkt3, pkt4, pkt5; send ack2 send pkt0 send pkt1 send pkt2 send pkt3 0123 0123 2345 2345 2345 2345 01 678 01 678 01 678 01 678 rcv ack0, send pkt4 rcv ack1, send pkt5 record ack3 arrived pkt 2 timeout send pkt2 record ack4 arrived record ack5 arrived Q: what happens when ack2 arrives? Xloss (wait) Transport Layer 3-58 Selective repeat: dilemma example:  seq #’s: 0, 1, 2, 3  window size=3  receiver sees no difference in two scenarios!  duplicate data accepted as new in (b) Q: what relationship between seq # size and window size to avoid problem in (b)? sender window (after receipt) 0123012 0123012 0123012 0123012 0 1 2 3 0 1 2 receiver window (after receipt) pkt0 pkt1 0123012 pkt2 0123012 pkt3 0123012 X pkt0 will accept packet (a) no problem with seq number 0 receiver can’t see sender side. receiver behavior identical in both cases! something’s (very) wrong! 0 1 2 3 0 1 2 pkt0 0123012 pkt1 0123012 0123012 pkt2 0123012 X 0123012 timeout X retransmit pkt0 X 0 1 2 3 0 1 2 pkt0 will accept packet (b) oops! with seq number 0 Transport Layer 3-59 Chapter 3 outline 3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 principles of reliable data transfer 3.5 connection-oriented transport: TCP  segment structure  reliable data transfer  flow control  connection management 3.6 principles of congestion control 3.7 TCP congestion control Transport Layer 3-60 TCP: Overview RFCs: 793,1122,1323, 2018, 2581  point-to-point:  one sender, one receiver  reliable, in-order byte steam:  no “message boundaries”  pipelined:  TCP congestion and flow control set window size  full duplex data:  bi-directional data flow in same connection  MSS: maximum segment size  connection-oriented:  handshaking (exchange of control msgs) inits sender, receiver state before data exchange  flow controlled:  sender will not overwhelm receiver Transport Layer 3-61 TCP segment structure 32 bits URG: urgent data (generally not used) ACK: ACK # valid PSH: push data now (generally not used) RST, SYN, FIN: connection estab (setup, teardown commands) Internet checksum (as in UDP) counting by bytes of data (not segments!) # bytes rcvr willing to accept source port # sequence number acknowledgement number dest port # head len not used checksum U A P R S F options (variable length) application data (variable length) receive window Urg data pointer Transport Layer 3-62 TCP seq. numbers, ACKs sequence numbers: byte stream “number” of first byte in segment’s data acknowledgements: seq # of next byte expected from other side cumulative ACK Q: how receiver handles out-of-order segments A: TCP spec doesn’t say, - up to implementor outgoing segment from sender source port # dest port # sequence number acknowledgement number rwnd checksum urg pointer window size N sender sequence number space sent ACKed sent, not- yet ACKed (“in- flight”) usable not but not usable yet sent incoming segment to sender source port # dest port # sequence number acknowledgement number A rwnd checksum urg pointer Transport Layer 3-63 TCP seq. numbers, ACKs Host A Host B User types ‘C’ host ACKs receipt of echoed ‘C’ Seq=42, ACK=79, data = ‘C’ host ACKs Seq=79, ACK=43, data = ‘C’ receipt of ‘C’, echoes back ‘C’ Seq=43, ACK=80 simple telnet scenario Transport Layer 3-64 TCP round trip time, timeout Q: 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 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 Transport Layer 3-65 TCP round trip time, timeout EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT  exponential weighted moving average  influence of past sample decreases exponentially fast  typical value:  = 0.125 RTT: gaia.cs.umass.edu to fantasia.eurecom.fr 350 300 250 200 150 100 1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106 RTT: gaia.cs.umass.edu to fantasia.eurecom.fr sampleRTT EstimatedRTT time (seconnds) time (seconds) Transport Layer 3-66 SampleRTT Estimated RTT RTT (milliseconds) RTT (milliseconds) TCP round trip time, timeout  timeout interval: EstimatedRTT plus “safety margin”  large variation in EstimatedRTT -> larger safety margin  estimate SampleRTT deviation from EstimatedRTT:
DevRTT = (1-)*DevRTT + *|SampleRTT-EstimatedRTT|
(typically,  = 0.25) TimeoutInterval = EstimatedRTT + 4*DevRTT
estimated RTT
“safety margin” Transport Layer 3-67

TCP reliable data transfer
 TCP creates rdt service on top of IP’s unreliable service
 pipelined segments
 cumulative acks
 single retransmission timer
 retransmissions triggered by:
 timeout events  duplicate acks
let’s initially consider simplified TCP sender:
 ignore duplicate acks  ignore flow control,
congestion control
Transport Layer 3-69

TCP sender events:
data 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 ack acknowledges previously unacked segments
 update what is known to be ACKed
 start timer if there are still unacked segments
Transport Layer 3-70

TCP sender (simplified)
L
NextSeqNum = InitialSeqNum SendBase = InitialSeqNum
event
ACK received, with ACK field value y
data received from application above
create segment, seq. #: NextSeqNum
pass segment to IP (i.e., “send”) NextSeqNum = NextSeqNum + length(data) if (timer currently not running)
start timer
timeout
retransmit not-yet-acked segment with smallest seq. #
start timer
wait for
if (y > SendBase) {
SendBase = y
/* SendBase–1: last cumulatively ACKed byte */ if (there are currently not-yet-acked segments)
start timer else stop timer
}
Transport Layer 3-71

TCP: retransmission scenarios
Host A Host B
Seq=92, 8 bytes of data
Host A Host B SendBase=92
ACK=100
X
Seq=92, 8 bytes of data
ACK=100
lost ACK scenario
SendBase=100 SendBase=120
SendBase=120
Seq=92, 8 bytes of data Seq=100, 20 bytes of data
ACK=100 ACK=120
Seq=92, 8 bytes of data
ACK=120
premature timeout
Transport Layer 3-72
timeout
timeout

TCP: retransmission scenarios
Host A
Host B
Seq=92, 8 bytes of data
Seq=100, 20 bytes of data
X ACK=100 ACK=120
Seq=120, 15 bytes of data
cumulative ACK
Transport Layer 3-73
timeout

TCP ACK generation [RFC 1122, RFC 2581]
event at receiver
arrival of in-order segment with expected seq #. All data up to expected seq # already ACKed
arrival of in-order segment with expected seq #. One other segment has ACK pending
arrival of out-of-order segment higher-than-expect seq. # . Gap detected
arrival of segment that partially or completely fills gap
TCP receiver action
delayed ACK. Wait up to 500ms
for 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 that segment starts at lower end of gap
Transport Layer 3-74

TCP fast retransmit
 time-out period often relatively 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 many duplicate ACKs.
TCP fast retransmit
if sender receives 3 ACKs for same data
(“triple duplicate ACKs”), (“triple duplicate ACKs”),
resend unacked segment with smallest seq #
 likely that unacked segment lost, so don’t wait for timeout
Transport Layer 3-75

TCP fast retransmit
Host A
Host B
Seq=92, 8 bytes of data
Seq=100, 20 bytes of data
X
ACK=100
ACK=100
ACK=100 ACK=100
Seq=100, 20 bytes of data
fast retransmit after sender receipt of triple duplicate ACK
Transport Layer 3-76
timeout

TCP flow control
application process
TCP code
TCP socket receiver buffers
IP code
application may remove data from TCP socket buffers ….
… slower than TCP receiver is delivering (sender is sending)
application OS
flow control
receiver controls sender, so sender won’t overflow receiver’s buffer by transmitting too much, too fast
from sender
receiver protocol stack
Transport Layer 3-78

TCP flow control
 receiver “advertises” free buffer space by including rwnd value in TCP header of receiver-to-sender segments
 RcvBuffer size set via socket options (typical default is 4096 bytes)
 many operating systems autoadjust RcvBuffer
 sender limits amount of unacked (“in-flight”) data to receiver’srwnd value
 guarantees receive buffer will not overflow
to application process
buffered data
free buffer space
RcvBuffer
rwnd
TCP segment payloads receiver-side buffering
Transport Layer 3-79

Connection Management
before exchanging data, sender/receiver “handshake”:  agree to establish connection (each knowing the other willing
to establish connection)
 agree on connection parameters
application
connection state: ESTAB connection variables:
seq # client-to-server server-to-client
rcvBuffer size at server,client
network
application
connection state: ESTAB connection Variables:
seq # client-to-server server-to-client
rcvBuffer size at server,client
network
Socket clientSocket = newSocket(“hostname”,”port
Socket connectionSocket = welcomeSocket.accept();
number”);
Transport Layer 3-81

Agreeing to establish a connection
2-way handshake:
Q: will 2-way handshake always work in network?
 variable delays
 retransmitted messages (e.g. req_conn(x)) due to message loss
 message reordering
 can’t “see” other side
Let’s talk OK
ESTAB
ESTAB
choose x
ESTAB
req_conn(x) acc_conn(x)
ESTAB
Transport Layer 3-82

Agreeing to establish a connection
2-way handshake failure scenarios:
choose x
retransmit req_conn(x)
ESTAB
client terminates
req_conn(x)
choose x
retransmit req_conn(x)
ESTAB
retransmit data(x+1)
client terminates
req_conn(x)
ESTAB
accept data(x+1)
server forgets x
ESTAB
accept data(x+1)
ESTAB
acc_conn(x)
acc_conn(x) data(x+1)
connection x completes
req_conn(x) data(x+1)
req_conn(x)
connection x completes
server forgets x
ESTAB
half open connection! (no client!)
Transport Layer 3-83

TCP 3-way handshake
client state
LISTEN
SYNSENT
choose init seq num, x send TCP SYN msg
server state
LISTEN
SYN RCVD
SYNbit=1, Seq=x
SYNbit=1, Seq=y ACKbit=1; ACKnum=x+1
ACKbit=1, ACKnum=y+1
choose init seq num, y send TCP SYNACK msg, acking SYN
ESTAB
received SYNACK(x) indicates server is live; send ACK for SYNACK;
this segment may contain client-to-server data
received ACK(y) indicates client is live
ESTAB
Transport Layer 3-84

TCP 3-way handshake: FSM
closed
Socket connectionSocket = welcomeSocket.accept();
SYN(x)
L
listen
Socket clientSocket = newSocket(“hostname”,”port
number”);
SYN(seq=x)
SYN sent
SYNACK(seq=y,ACKnum=x+1) ACK(ACKnum=y+1)
SYNACK(seq=y,ACKnum=x+1) create new socket for
communication back to client
SYN rcvd
ACK(ACKnum=y+1)
L
ESTAB
Transport Layer 3-85

TCP: closing a connection
 client, server each close their side of connection  send TCP segment with FIN bit = 1
 respond to received FIN with ACK
 on receiving FIN, ACK can be combined with own FIN
 simultaneous FIN exchanges can be handled
Transport Layer 3-86

TCP: closing a connection
client state
ESTAB
server state
ESTAB CLOSE_WAIT
LAST_ACK
CLOSED
clientSocket.close()
FIN_WAIT_1 FIN_WAIT_2
TIMED_WAIT
can no longer send but can receive data
wait for server close
FINbit=1, seq=x ACKbit=1; ACKnum=x+1
FINbit=1, seq=y ACKbit=1; ACKnum=y+1
can still send data
CLOSED
timed wait for 2*max
segment lifetime
can no longer send data
Transport Layer 3-87

Principles of congestion control
congestion:
 informally: “too many sources sending too much
data too fast for network to handle”  different from flow control!
 manifestations:
lost packets (buffer overflow at routers)
long delays (queueing in router buffers)  a top-10 problem!
Transport Layer 3-89

Causes/costs of congestion: scenario 1
 two senders, two receivers
 one router, infinite buffers
 output link capacity: R  no retransmission
original data: lin Host A
throughput: lout unlimited shared
output link buffers
Host B
R/2
lin R/2
lin R/2
 maximum per-connection  large delays as arrival rate, lin, throughput: R/2 approaches capacity
Transport Layer 3-90
lout
delay

Causes/costs of congestion: scenario 2
 one router, finite buffers
 sender retransmission of timed-out packet
 application-layer input = application-layer output: lin =
lout
 transport-layer input includes retransmissions : l
l
‘
in
in
lin : original data
l’in: original data, plus
retransmitted data
lout
Host A
Host B
finite shared output link buffers
Transport Layer 3-91

Causes/costs of congestion: scenario 2
R/2
idealization: perfect knowledge
 sender sends only when router buffers available
copy
A
lin R/2 lin : original data
lout
l’in: original data, plus retransmitted data
free buffer space!
Host B
finite shared output link buffers
Transport Layer 3-92
lout

Causes/costs of congestion: scenario 2
Idealization: known loss packets can be lost, dropped at router due to full buffers
 sender only resends if packet known to be lost
copy
A
lin : original data
l’in: original data, plus
retransmitted data
no buffer space!
lout
Host B
Transport Layer 3-93

Causes/costs of congestion: scenario 2
Idealization: known loss packets can be lost, dropped at router due to full buffers
 sender only resends if packet known to be lost
R/2
lin lin : original data
when sending at R/2, some packets are retransmissions but asymptotic goodput is still R/2 (why?)
R/2
lout
A
l’in: original data, plus retransmitted data
free buffer space!
Host B
Transport Layer 3-94
lout

Causes/costs of congestion: scenario 2
Realistic: duplicates
 packets can be lost, dropped
at router due to full buffers
 sender times out prematurely, sending two copies, both of which are delivered
lin l’in
R/2
timeout
lin
l
R/2
when sending at R/2, some packets are retransmissions including duplicated that are delivered!
copy
out
A
free buffer space!
Host B
Transport Layer 3-95
lout

Causes/costs of congestion: scenario 2
Realistic: duplicates
 packets can be lost, dropped
at router due to full buffers
R/2
 sender times out prematurely, sending two copies, both of which are delivered
lin
when sending at R/2, some packets are retransmissions including duplicated that are delivered!
R/2
“costs” of congestion:
 more work (retrans) for given “goodput”
 unneeded retransmissions: link carries multiple copies of pkt
 decreasing goodput
Transport Layer 3-96
lout

Causes/costs of congestion: scenario 3
 four senders
 multihop paths
 timeout/retransmit
Q: what happens as lin and lin’ increase ?
A: as red lin’ increases, all arriving blue pkts at upper queue are dropped, blue throughput g 0
Host A
lin : original data
l’in: original data, plus
retransmitted data
finite shared output link buffers
lout Host B
Host D
Host C
Transport Layer 3-97

Causes/costs of congestion: scenario 3
C/2
lin’
another “cost” of congestion:
 when packet dropped, any “upstream transmission capacity used for that packet was wasted!
C/2
Transport Layer 3-98
lout

Approaches towards congestion control
two broad approaches towards congestion control:
end-end congestion network-assisted
control:
 no explicit feedback from network
 congestion inferred from end-system observed loss, delay
 approach taken by TCP
congestion control:
 routers provide feedback to end systems
single bit indicating congestion (SNA, DECbit, TCP/IP ECN, ATM)
explicit rate for sender to send at
Transport Layer 3-99

Case study: ATM ABR congestion control
ABR: available bit rate:
 “elastic service”  if sender’s path
“underloaded”:
 sender should use
available bandwidth
 if sender’s path congested:
 sender throttled to minimum guaranteed rate
RM (resource management) cells:
 sent by sender, interspersed with data cells
 bits in RM cell set by switches (“network-assisted”)
 NI bit: no increase in rate (mild congestion)
 CI bit: congestion indication
 RM cells returned to sender by receiver, with bits intact
Transport Layer 3-100

Case study: ATM ABR congestion control
RM cell data cell
 two-byte ER (explicit rate) field in RM cell
 congested switch may lower ER value in cell
 senders’ send rate thus max supportable rate on path
 EFCI bit in data cells: set to 1 in congested switch
 if data cell preceding RM cell has EFCI set, receiver sets CI bit in returned RM cell
Transport Layer 3-101

TCP congestion control: additive increase multiplicative decrease
 approach: sender increases transmission rate (window size), probing for usable bandwidth, until loss occurs
additive increase: increase cwnd by 1 MSS every RTT until loss detected
 multiplicative decrease: cut cwnd in half after loss additively increase window size …
…. until loss occurs (then cut window in half)
AIMD saw tooth behavior: probing for bandwidth
time
Transport Layer 3-103
cwnd: TCP sender congestion window size

TCP Congestion Control: details
sender sequence number space
TCP sending rate:
 roughly: send cwnd bytes, wait RTT for ACKS, then send more bytes
cwnd
last byte ACKed
sent, not- yet ACKed (“in- flight”)
last byte sent
 sender limits transmission:
 cwnd is dynamic, function of perceived network congestion
rate
~ cwnd bytes/sec
~
RTT
LastByteSent- < LastByteAcked cwnd Transport Layer 3-104 TCP Slow Start  when connection begins, increase rate exponentially until first loss event:  initially cwnd = 1 MSS doublecwndeveryRTT  done by incrementing cwnd for every ACK received  summary: initial rate is slow but ramps up exponentially fast Host A Host B time Transport Layer 3-105 RTT TCP: detecting, reacting to loss  loss indicated by timeout: cwnd set to 1 MSS; window then grows exponentially (as in slow start) to threshold, then grows linearly  loss indicated by 3 duplicate ACKs: TCP RENO dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly  TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks) Transport Layer 3-106 TCP: switching from slow start to CA Q: when should the exponential increase switch to linear? A: when cwnd gets to 1/2 of its value before timeout. Implementation:  variable ssthresh  on loss event, ssthresh is set to 1/2 of cwnd just before loss event Transport Layer 3-107 Summary: TCP Congestion Control New ACK! new ACK cwnd = cwnd+MSS dupACKcount = 0 transmit new segment(s), as allowed cwnd > ssthresh
L
timeout ssthresh = cwnd/2
cwnd = 1 MSS dupACKcount = 0 retransmit missing segment
New ACK!
transmit new segment(s), as allowed
duplicate ACK dupACKcount++
slow start
new ACK
.
cwnd = cwnd + MSS dupACKcount = 0
(MSS/cwnd)
L
cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0
timeout
ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0
retransmit missing segment
dupACKcount == 3
ssthresh= cwnd/2 cwnd = ssthresh + 3 retransmit missing segment
congestion avoidance
duplicate ACK dupACKcount++
New ACK!
New ACK
cwnd = ssthresh dupACKcount = 0
fast recovery
timeout
ssthresh = cwnd/2
cwnd = 1
dupACKcount = 0 retransmit missing segment
dupACKcount == 3
ssthresh= cwnd/2
cwnd = ssthresh + 3 retransmit missing segment
duplicate ACK
cwnd = cwnd + MSS
transmit new segment(s), as allowed
Transport Layer 3-108

TCP throughput
 avg. TCP thruput as function of window size, RTT?  ignore slow start, assume always data to send
 W: window size (measured in bytes) where loss occurs  avg. window size (# in-flight bytes) is 3⁄4 W
avg. thruput is 3/4W per RTT
avg TCP thruput = 3 W bytes/sec 4 RTT
W W/2
Transport Layer 3-109

TCP Futures: TCP over “long, fat pipes”  example: 1500 byte segments, 100ms RTT, want
10 Gbps throughput
 requires W = 83,333 in-flight segments
 throughput in terms of segment loss probability, L [Mathis 1997]:
TCP throughput = 1.22 . MSS RTT L
➜ to achieve 10 Gbps throughput, need a loss rate of L = 2·10-10 – a very small loss rate!
 new versions of TCP for high-speed
Transport Layer 3-110

TCP Fairness
fairness goal: if K TCP sessions share same bottleneck link of bandwidth R, each should have average rate of R/K
TCP connection 1
TCP connection 2
bottleneck router capacity R
Transport Layer 3-111

Why is TCP fair?
two competing sessions:
 additive increase gives slope of 1, as throughout increases
 multiplicative decrease decreases throughput proportionally
R
equal bandwidth share
loss: decrease window by factor of 2 congestion avoidance: additive increase
loss: decrease window by factor of 2 congestion avoidance: additive increase
Connection 1 throughput
R
Transport Layer 3-112

Fairness (more)
Fairness and UDP
 multimedia apps often do not use TCP
 do not want rate throttled by congestion control
 instead use UDP:
 send audio/video at constant rate, tolerate packet loss
Fairness, parallel TCP connections
 application can open multiple parallel connections between two hosts
 web browsers do this
 e.g., link of rate R with 9
existing connections:
 new app asks for 1 TCP, gets rate R/10
 new app asks for 11 TCPs, gets R/2 Transport Layer 3-113

Chapter 3: summary
 principles behind transport layer services:
 multiplexing, demultiplexing
reliable data transfer flow control congestion control
 instantiation, implementation in the Internet
 UDP  TCP
next:
 leaving the network “edge” (application, transport layers)
 into the network “core”
Transport Layer 3-114

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