Transport Layer Our goals r understand principles behind

Transport Layer Our goals: r understand principles behind transport layer services: m m multiplexing/demultipl exing reliable data transfer flow control congestion control r learn about transport layer protocols in the Internet: m m m UDP: connectionless transport TCP: connection-oriented transport TCP congestion control 3 -1

Outline r Transport-layer services r Multiplexing and demultiplexing r Connectionless transport: UDP r Principles of reliable data transfer r Connection-oriented transport: TCP m m segment structure reliable data transfer flow control connection management r Principles of congestion control r TCP congestion control 3 -2

Transport services and protocols r provide logical communication network data link physical al ic g lo d en d- en network data link physical po s an tr rt between app processes running on different hosts r transport protocols run in end systems m send side: breaks app messages into segments, passes to network layer m rcv side: reassembles segments into messages, passes to app layer r more than one transport protocol available to apps m Internet: TCP and UDP application transport network data link physical 3 -3

Transport vs. network layer r network layer: logical communication between hosts r transport layer: logical communication between processes m relies on, enhances, network layer services Household analogy: 12 kids sending letters to 12 kids r processes = kids r app messages = letters in envelopes r hosts = houses r transport protocol = Ann and Bill r network-layer protocol = postal service 3 -4

Internet transport-layer protocols r reliable, in-order delivery (TCP) network data link physical po s an rt m delay guarantees bandwidth guarantees network data link physical tr m d en d- delivery: UDP r services not available: en r unreliable, unordered al m network data link physical ic m congestion control flow control connection setup g lo m application transport network data link physical 3 -5

Outline r Transport-layer services r Multiplexing and demultiplexing r Connectionless transport: UDP r Principles of reliable data transfer r Connection-oriented transport: TCP m m segment structure reliable data transfer flow control connection management r Principles of congestion control r TCP congestion control 3 -6

Multiplexing/demultiplexing Multiplexing at send host: gathering data from multiple sockets, enveloping data with header (later used for demultiplexing) Demultiplexing at rcv host: delivering received segments to correct socket = socket application transport network link = process P 3 P 1 application transport network P 2 P 4 application transport network link physical host 1 physical host 2 physical host 3 3 -7

How demultiplexing works r host receives IP datagrams each datagram has source IP address, destination IP address m each datagram carries 1 transport-layer segment m each segment has source, destination port number r host uses IP addresses & port numbers to direct segment to appropriate socket m 32 bits source port # dest port # other header fields application data (message) TCP/UDP segment format 3 -8

Connectionless demultiplexing r Create sockets with port numbers: Datagram. Socket my. Socket 1 = new Datagram. Socket(99111); Datagram. Socket my. Socket 2 = new Datagram. Socket(99222); r UDP socket identified by two-tuple: (dest IP address, dest port number) r When host receives UDP segment: m m checks destination port number in segment directs UDP segment to socket with that port number r IP datagrams with different source IP addresses and/or source port numbers directed to same socket 3 -9

Connectionless demux (cont) Datagram. Socket server. Socket = new Datagram. Socket(6428); P 2 SP: 6428 DP: 9157 SP: 6428 DP: 5775 SP: 9157 client IP: A P 1 P 3 DP: 6428 server IP: C SP: 5775 DP: 6428 Client IP: B SP provides “return address” 3 -10

Connection-oriented demux r TCP socket identified by 4 -tuple: m m source IP address source port number dest IP address dest port number r recv host uses all four values to direct segment to appropriate socket r Server host may support many simultaneous TCP sockets: m each socket identified by its own 4 -tuple r Web servers have different sockets for each connecting client m non-persistent HTTP will have different socket for each request 3 -11

Connection-oriented demux (cont) P 1 P 4 P 5 P 2 P 6 P 1 P 3 SP: 5775 DP: 80 S-IP: B D-IP: C client IP: A SP: 9157 DP: 80 S-IP: A D-IP: C SP: 9157 server IP: C DP: 80 S-IP: B D-IP: C Client IP: B 3 -12

Connection-oriented demux: Threaded Web Server P 1 P 2 P 4 P 1 P 3 SP: 5775 DP: 80 S-IP: B D-IP: C client IP: A SP: 9157 DP: 80 S-IP: A D-IP: C SP: 9157 server IP: C DP: 80 S-IP: B D-IP: C Client IP: B 3 -13

Outline r Transport-layer services r Multiplexing and demultiplexing r Connectionless transport: UDP r Principles of reliable data transfer r Connection-oriented transport: TCP m m segment structure reliable data transfer flow control connection management r Principles of congestion control r TCP congestion control 3 -14

UDP: User Datagram Protocol r “no frills, ” “bare bones” Internet transport protocol r “best effort” service, UDP segments may be: m lost m delivered out of order to app r connectionless: m no handshaking between UDP sender, receiver m each UDP segment handled independently of others [RFC 768] Why is there a UDP? r no connection establishment (which can add delay) r simple: no connection state at sender, receiver r small segment header: 8 bytes m TCP header: 20 bytes r no congestion control: UDP can blast away as fast as desired 3 -15

UDP: more r often used for streaming multimedia apps m loss tolerant m rate sensitive Length, in bytes of UDP segment, including header r other UDP uses m DNS m SNMP r reliable transfer over UDP: add reliability at application layer m application-specific error recovery! 32 bits source port # dest port # length checksum Application data (message) UDP segment format 3 -16

UDP checksum Goal: detect “errors” (e. g. , flipped bits) in transmitted segment Sender: Receiver: r treat segment contents as r compute checksum of sequence of 16 -bit integers r checksum: addition (1’s complement sum) of segment contents r sender puts checksum value into UDP checksum field received segment r check if computed checksum equals checksum field value: m NO - error detected m YES - no error detected. But maybe errors nonetheless? More later …. 3 -17

Internet Checksum Example r Note m When adding numbers, a carryout from the most significant bit needs to be added to the result r Example: add two 16 -bit integers 1 1 0 0 1 1 1 0 1 0 1 wraparound 1 1 0 1 1 sum 1 1 0 1 1 0 0 checksum 1 0 0 0 0 1 1 3 -18

Outline r Transport-layer services r Multiplexing and demultiplexing r Connectionless transport: UDP r Principles of reliable data transfer r Connection-oriented transport: TCP m m segment structure reliable data transfer flow control connection management r Principles of congestion control r TCP congestion control 3 -19

Principles of Reliable data transfer r important in app. , transport, link layers r top-10 list of important networking topics! r characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt) 3 -20

Reliable data transfer: getting started rdt_send(): called from above, (e. g. , by app. ). Passed data to deliver to receiver upper layer send side udt_send(): called by rdt, to transfer packet over unreliable channel to receiver deliver_data(): called by rdt to deliver data to upper receive side rdt_rcv(): called when packet arrives on rcv-side of channel 3 -21

Reliable data transfer: getting started We’ll: r incrementally develop sender, receiver sides of reliable data transfer protocol (rdt) r consider only unidirectional data transfer m but control info will flow on both directions! r use finite state machines (FSM) to specify sender, receiver state: when in this “state” next state uniquely determined by next event state 1 event causing state transition actions taken on state transition event actions state 2 3 -22

Reliable data transfer Ideal r Rdt 1. 0: reliable channel, too good to be real r Rdt 2. 0: unreliable channel with bit errors m Stop and Wait m Rdt 2. 1: ACK-NAK m Rdt 2. 2: NAK-free r Rdt 3. 0: unreliable channel with bit errors and lost Real world 3 -23

Rdt 1. 0: reliable transfer over a reliable channel r underlying channel perfectly reliable m no bit errors m no loss of packets r separate FSMs for sender, receiver: m sender sends data into underlying channel m receiver read 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 3 -24

Rdt 2. 0: channel with bit errors r underlying channel may flip bits in packet m checksum to detect bit errors r the question: how to recover from errors: m acknowledgements (ACKs): receiver explicitly tells sender that pkt received OK m negative acknowledgements (NAKs): receiver explicitly tells sender that pkt had errors m sender retransmits pkt on receipt of NAK r new mechanisms in rdt 2. 0 (beyond rdt 1. 0): m m error detection receiver feedback: control msgs (ACK, NAK) rcvr->sender 3 -25

rdt 2. 0: FSM specification rdt_send(data) snkpkt = make_pkt(data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && is. NAK(rcvpkt) Wait for call from ACK or udt_send(sndpkt) above NAK rdt_rcv(rcvpkt) && is. ACK(rcvpkt) L sender 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) 3 -26

rdt 2. 0: operation with no errors rdt_send(data) snkpkt = make_pkt(data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && is. NAK(rcvpkt) Wait for call from ACK or udt_send(sndpkt) above NAK rdt_rcv(rcvpkt) && is. ACK(rcvpkt) L 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) 3 -27

rdt 2. 0: error scenario rdt_send(data) snkpkt = make_pkt(data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && is. NAK(rcvpkt) Wait for call from ACK or udt_send(sndpkt) above NAK rdt_rcv(rcvpkt) && is. ACK(rcvpkt) L 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) 3 -28

rdt 2. 0 has a fatal flaw! What happens if ACK/NAK corrupted? r sender doesn’t know what happened at receiver! r can’t just retransmit: possible duplicate !!! Handling duplicates: r sender retransmits current pkt if ACK/NAK garbled r sender adds sequence number to each pkt: restransmission or new packet? r receiver discards (doesn’t deliver up) duplicate pkt stop and wait Sender sends one packet, then waits for receiver response 3 -29

rdt 2. 1: sender, handles garbled ACK/NAKs rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && is. ACK(rcvpkt) Wait for call 0 from above rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && is. ACK(rcvpkt) L rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || is. NAK(rcvpkt) ) udt_send(sndpkt) Wait for ACK or NAK 0 L Wait for ACK or NAK 1 Wait for call 1 from above rdt_send(data) sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) 3 -30

rdt 2. 1: receiver, handles garbled ACK/NAKs rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq 0(rcvpkt) rdt_rcv(rcvpkt) && (corrupt(rcvpkt) extract(rcvpkt, data) deliver_data(data) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && (corrupt(rcvpkt) sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq 1(rcvpkt) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt) Wait for 0 from below Wait for 1 from below rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq 1(rcvpkt) rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq 0(rcvpkt) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) extract(rcvpkt, data) deliver_data(data) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) 3 -31

rdt 2. 1: discussion Resending old packet or sending new packet? Sender: r seq # added to pkt r two seq. #’s (0, 1) will suffice. Why? r must check if received ACK/NAK corrupted r twice as many states m state must “remember” whether “current” pkt has 0 or 1 seq. # Receiver: r must check if received packet is duplicate m state indicates whether 0 or 1 is expected pkt seq # r note: receiver can not know if its last ACK/NAK received OK at sender 3 -32

rdt 2. 2: a NAK-free protocol r same functionality as rdt 2. 1, using ACKs only r instead of NAK, receiver sends ACK for last pkt received OK m receiver must explicitly include seq # of pkt being ACKed r duplicate ACK at sender results in same action as NAK: retransmit current pkt 3 -33

rdt 2. 2: sender, receiver fragments rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && Wait for call 0 from above rdt_rcv(rcvpkt) && (corrupt(rcvpkt) || has_seq 1(rcvpkt)) udt_send(sndpkt) Wait for 0 from below ( corrupt(rcvpkt) || is. ACK(rcvpkt, 1) ) udt_send(sndpkt) Wait for ACK 0 sender FSM fragment rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && is. ACK(rcvpkt, 0) receiver FSM fragment L rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq 1(rcvpkt) extract(rcvpkt, data) deliver_data(data) sndpkt = make_pkt(ACK 1, chksum) udt_send(sndpkt) 3 -34

rdt 3. 0: channels with errors and loss New assumption: underlying channel can also lose packets (data or ACKs) m checksum, seq. #, ACKs, retransmissions will be of help, but not enough Approach: sender waits “reasonable” amount of time for ACK r retransmits if no ACK received in this time r if pkt (or ACK) just delayed (not lost): m retransmission will be duplicate, but use of seq. #’s already handles this m receiver must specify seq # of pkt being ACKed r requires countdown timer 3 -35

rdt 3. 0 sender rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) L rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && is. ACK(rcvpkt, 1) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || is. ACK(rcvpkt, 0) ) timeout udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && is. ACK(rcvpkt, 0) stop_timer timeout udt_send(sndpkt) start_timer L Wait for ACK 0 Wait for call 0 from above L rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || is. ACK(rcvpkt, 1) ) Wait for ACK 1 Wait for call 1 from above rdt_send(data) rdt_rcv(rcvpkt) L sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) start_timer 3 -36

rdt 3. 0 in action 3 -37

rdt 3. 0 in action 3 -38

Performance of rdt 3. 0 r example: 1 Gbps link, 15 ms end-end prop. delay, 1 KB packet: r Performance, in term of U sender: utilization – fraction of time sender busy sending? 3 -39

Performance of rdt 3. 0 r rdt 3. 0 works, but performance stinks r example: 1 Gbps link, 15 ms e-e prop. delay, 1 KB packet: Ttransmit = m m m L (packet length in bits) 8 kb/pkt = = 8 microsec R (transmission rate, bps) 10**9 b/sec U sender: utilization – fraction of time sender busy sending 1 KB pkt every 30 msec -> 33 k. B/sec thruput over 1 Gbps link network protocol limits use of physical resources! 3 -40

rdt 3. 0: stop-and-wait operation sender receiver first packet bit transmitted, t = 0 last packet bit transmitted, t = L / R RTT first packet bit arrives last packet bit arrives, send ACK arrives, send next packet, t = RTT + L / R 3 -41

Pipelined protocols Pipelining: sender allows multiple, “in-flight”, yet-tobe-acknowledged pkts m m range of sequence numbers must be increased buffering at sender and/or receiver r Two generic forms of pipelined protocols: go-Back-N, selective repeat 3 -42

Pipelining: increased utilization sender receiver first packet bit transmitted, t = 0 last bit transmitted, t = L / R RTT first packet bit arrives last packet bit arrives, send ACK last bit of 2 nd packet arrives, send ACK last bit of 3 rd packet arrives, send ACK arrives, send next packet, t = RTT + L / R Increase utilization by a factor of 3! 3 -43

Go-Back-N Sender: r k-bit seq # in pkt header r “window” of up to N, consecutive unack’ed pkts allowed r ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK” may receive duplicate ACKs (see receiver) r timer for the oldest transmitted not yet acked packet! r timeout(n): retransmit pkt n and all higher seq # pkts in window m 3 -44

GBN: sender extended FSM rdt_send(data) L base=1 nextseqnum=1 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) Wait rdt_rcv(rcvpkt) && corrupt(rcvpkt) L 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) //non outstanding packet stop_timer else start_timer 3 -45

GBN: receiver extended FSM default udt_send(sndpkt) L Wait expectedseqnum=1 sndpkt = make_pkt(expectedseqnum, ACK, chksum) rdt_rcv(rcvpkt) && notcurrupt(rcvpkt) && hasseqnum(rcvpkt, expectedseqnum) extract(rcvpkt, data) deliver_data(data) sndpkt = make_pkt(expectedseqnum, ACK, chksum) udt_send(sndpkt) expectedseqnum++ ACK-only: always send ACK for correctly-received pkt with highest in-order seq # m m may generate duplicate ACKs need only remember expectedseqnum r out-of-order pkt: m discard (don’t buffer) -> no receiver buffering! m Re-ACK pkt with highest in-order seq # 3 -46

GBN in action 3 -47

GBN: Discussions r A single packet error cause GBN to transmit a large number of packtes r In-order, but out-of-order in real world r Unnecessary packet re-transmission!! performance!!! 3 -48

Selective Repeat r receiver individually acknowledges all correctly received pkts, could be out-of-order!!! m buffers pkts, as needed, for eventual in-order delivery to upper layer r sender only resends pkts for which ACK not received m m sender timer for each un. ACKed pkt Avoid unnecessary re-transmission in GBN!!! r sender window m N consecutive seq #’s m again limits seq #s of sent, un. ACKed pkts 3 -49

Selective repeat: sender, receiver windows 3 -50

Selective repeat sender data from above : receiver pkt n in [rcvbase, rcvbase+N-1] r if next available seq # in r send ACK(n) timeout(n): r in-order: deliver (also window, send pkt r resend pkt n, restart timer ACK(n) in [sendbase, sendbase+N]: r mark pkt n as received r if n is smallest un. ACKed pkt, advance window base to next un. ACKed seq # r out-of-order: buffer deliver buffered, in-order pkts), advance window to next not-yet-received pkt n in [rcvbase-N, rcvbase-1] r ACK(n) otherwise: r ignore 3 -51

Selective repeat in action 3 -52

Selective repeat: dilemma Example: r seq #’s: 0, 1, 2, 3 r window size=3 r receiver sees no difference in two scenarios! r incorrectly passes duplicate data as new in (a) Q: what relationship between seq # size and window size? 3 -53

Outline r Transport-layer services r Multiplexing and demultiplexing r Connectionless transport: UDP r Principles of reliable data transfer r Connection-oriented transport: TCP m m segment structure reliable data transfer flow control connection management r Principles of congestion control r TCP congestion control 3 -54

TCP: Overview r point-to-point: m one sender, one receiver r reliable, in-order byte steam: m no “message boundaries” r pipelined: m TCP congestion and flow control set window size r send & receive buffers RFCs: 793, 1122, 1323, 2018, 2581 r full duplex data: m bi-directional data flow in same connection m MSS: maximum segment size r connection-oriented: m handshaking (exchange of control msgs) init’s sender, receiver state before data exchange r flow controlled: m sender will not overwhelm receiver 3 -55

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) source port # dest port # sequence number acknowledgement number head not UA P R S F len used checksum Receive window Urg data pnter Options (variable length) counting by bytes of data (not segments!) # bytes rcvr willing to accept application data (variable length) 3 -56

TCP seq. #’s and ACKs Seq. #’s: m byte stream “number” of first byte in segment’s data ACKs: m seq # of next byte expected from other side m cumulative ACK Q: how receiver handles out-of-order segments m A: TCP spec doesn’t say, - up to implementor Host B Host A User types ‘C’ Seq=4 2, ACK = 79, da ta ata = d , 3 4 K= , AC q=79 Se host ACKs receipt of echoed ‘C’ = ‘C’ host ACKs receipt of ‘C’, echoes back ‘C’ Seq=4 3, ACK =80 simple telnet scenario time 3 -57

TCP Round Trip Time and Timeout Q: how to set TCP timeout value? r longer than RTT m but RTT varies r too short: premature timeout m unnecessary retransmissions r too long: slow reaction to segment loss Q: how to estimate RTT? r Sample. RTT: measured time from segment transmission until ACK receipt m ignore retransmissions r Sample. RTT will vary, want estimated RTT “smoother” m average several recent measurements, not just current Sample. RTT 3 -58

TCP Round Trip Time and Timeout Estimated. RTT = (1 - )*Estimated. RTT + *Sample. RTT r Exponential weighted moving average r influence of past sample decreases exponentially fast r typical value: = 0. 125 3 -59

Example RTT estimation: 3 -60

TCP Round Trip Time and Timeout Setting the timeout r Estimted. RTT plus “safety margin” m large variation in Estimated. RTT -> larger safety margin r first estimate of how much Sample. RTT deviates from Estimated. RTT: Dev. RTT = (1 - )*Dev. RTT + *|Sample. RTT-Estimated. RTT| (typically, = 0. 25) Then set timeout interval: Timeout. Interval = Estimated. RTT + 4*Dev. RTT 3 -61

Outline r Transport-layer services r Multiplexing and demultiplexing r Connectionless transport: UDP r Principles of reliable data transfer r Connection-oriented transport: TCP m m segment structure reliable data transfer flow control connection management r Principles of congestion control r TCP congestion control 3 -62

TCP reliable data transfer r TCP creates rdt service on top of IP’s unreliable service r Pipelined segments r Cumulative acks r TCP uses single retransmission timer r Retransmissions are triggered by: m m timeout events duplicate acks r Initially consider simplified TCP sender: m m ignore duplicate acks ignore flow control, congestion control 3 -63

TCP sender events: data rcvd from app: r Create segment with seq # r seq # is byte-stream number of first data byte in segment r start timer if not already running (think of timer as for oldest unacked segment) r expiration interval: Time. Out. Interval timeout: r retransmit segment that caused timeout r restart timer Ack rcvd: r If acknowledges previously unacked segments m m update what is known to be acked start timer if there are outstanding segments 3 -64

Next. Seq. Num = Initial. Seq. Num Send. Base = Initial. Seq. Num loop (forever) { switch(event) event: data received from application above create TCP segment with sequence number Next. Seq. Num if (timer currently not running) start timer pass segment to IP Next. Seq. Num = Next. Seq. Num + 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 > Send. Base) { Send. Base = y if (there are currently not-yet-acknowledged segments) start timer } } /* end of loop forever */ TCP sender (simplified) Comment: • Send. Base-1: last cumulatively ack’ed byte Example: • Send. Base-1 = 71; y= 73, so the rcvr wants 73+ ; y > Send. Base, so that new data is acked 3 -65

TCP: retransmission scenarios Host A 2, 8 by tes da t Seq=92 timeout a 100 X = ACK loss Seq=9 2, 8 by tes da ta 100 Sendbase = 100 Send. Base = 120 = ACK Send. Base = 100 time Host B Seq=9 Send. Base = 120 lost ACK scenario 2, 8 by tes da ta Seq= 100, 2 0 byte s data 0 10 = K 120 = C K A AC Seq=9 2, 8 by Seq=92 timeout Seq=9 timeout Host A Host B time tes da t a 20 K=1 AC premature timeout 3 -66

TCP retransmission scenarios (more) Host A Host B Seq=9 timeout 2, 8 by Send. Base = 120 Seq=1 tes da t a =100 K C A 00, 20 bytes data X loss 120 = ACK time Cumulative ACK scenario 3 -67

TCP ACK generation [RFC 1122, RFC 2581] Event at Receiver TCP Receiver action Arrival of in-order segment with expected seq #. All data up to expected seq # already ACKed Delayed ACK. Wait up to 500 ms for next segment. If no next segment, send ACK Arrival of in-order segment with expected seq #. One other segment has ACK pending Immediately send single cumulative ACK, ACKing both in-order segments Arrival of out-of-order segment higher-than-expect seq. #. Gap detected Immediately send duplicate ACK, indicating seq. # of next expected byte Arrival of segment that partially or completely fills gap Immediately send ACK, provided that segment starts at lower end of gap 3 -68

Fast Retransmit r Time-out period often relatively long: m long delay before resending lost packet r Detect lost segments via duplicate ACKs. m m Sender often sends many segments back-toback If segment is lost, there will likely be many duplicate ACKs. r If sender receives 3 ACKs for the same data, it supposes that segment after ACKed data was lost: m fast retransmit: resend segment before timer expires 3 -69

Fast retransmit algorithm: event: ACK received, with ACK field value of y if (y > Send. Base) { Send. Base = 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 } a duplicate ACK for already ACKed segment fast retransmit 3 -70

Outline r Transport-layer services r Multiplexing and demultiplexing r Connectionless transport: UDP r Principles of reliable data transfer r Connection-oriented transport: TCP m m segment structure reliable data transfer flow control connection management r Principles of congestion control r TCP congestion control 3 -71

TCP Flow Control r receive side of TCP connection has a receive buffer: flow control sender won’t overflow receiver’s buffer by transmitting too much, too fast r speed-matching r app process may be service: matching the send rate to the receiving app’s drain rate slow at reading from buffer 3 -72

TCP Flow control: how it works r Rcvr advertises spare (Suppose TCP receiver discards out-of-order segments) r spare room in buffer room by including value of Rcv. Window in segments r Sender limits un. ACKed data to Rcv. Window m guarantees receive buffer doesn’t overflow = Rcv. Window = Rcv. Buffer-[Last. Byte. Rcvd Last. Byte. Read] 3 -73

TCP Flow control: how it works r If the sender gets notified of Rcv. Window = 0 from the receiver? m Problem: the sender never able to send data to the receiver m Why: the receiver will not send segments to the sender unless having data/ack to send m Solution: the sender continues to send a segment with 1 data byte when Rcv. Window = 0; then the receiver will update it. 3 -74

Outline r Transport-layer services r Multiplexing and demultiplexing r Connectionless transport: UDP r Principles of reliable data transfer r Connection-oriented transport: TCP m m segment structure reliable data transfer flow control connection management r Principles of congestion control r TCP congestion control 3 -75

TCP Connection Management Recall: TCP sender, receiver establish “connection” before exchanging data segments r initialize TCP variables: m seq. #s m buffers, flow control info (e. g. Rcv. Window) r client: connection initiator Socket client. Socket = new Socket("hostname", "port number"); r server: contacted by client Socket connection. Socket = welcome. Socket. accept(); Three way handshake: Step 1: client host sends TCP SYN segment to server m specifies initial seq # m no data Step 2: server host receives SYN, replies with SYNACK segment server allocates buffers m specifies server initial seq. # Step 3: client receives SYNACK, replies with ACK segment, which may contain data, allocate buffers and variables 3 -76 m

TCP Connection Management (cont. ) Closing a connection: client closes socket: client. Socket. close(); client close Step 1: client end system close FIN timed wait FIN, replies with ACK, sends FIN ACK sends TCP FIN control segment to server Step 2: server receives server ACK closed 3 -77

TCP Connection Management (cont. ) Step 3: client receives FIN, replies with ACK. m client closing Enters “timed wait” will respond with ACK to received FINs server FIN ACK Step 4: server, receives closing FIN Note: with small modification, can handle simultaneous FINs. timed wait ACK. Connection closed. ACK closed 3 -78

TCP Connection Management (cont) TCP server lifecycle TCP client lifecycle 3 -79

Popular SYN flood attack Denial of service attack 3 -80

Outline r Transport-layer services r Multiplexing and demultiplexing r Connectionless transport: UDP r Principles of reliable data transfer r Connection-oriented transport: TCP m m segment structure reliable data transfer flow control connection management r Principles of congestion control r TCP congestion control 3 -81

Principles of Congestion Control Congestion: r informally: “too many sources sending too much data too fast for network to handle” r different from flow control! r manifestations: m lost packets (buffer overflow at routers) m long delays (queueing in router buffers) r a top-10 problem! 3 -82

Causes/costs of congestion: scenario 1 Host A r two senders, two receivers r one router, infinite buffers r no retransmission Host B lout lin : original data unlimited shared output link buffers r large delays when congested r maximum achievable throughput 3 -83

Causes/costs of congestion: scenario 2 r one router, finite buffers r sender retransmission of lost packet Host A lin : original data lout l'in : original data, plus retransmitted data Host B finite shared output link buffers 3 -84

Causes/costs of congestion: scenario 2 (goodput) = l out in r “perfect” retransmission only when loss: r always: l l > lout in r retransmission of delayed (not lost) packet makes (than perfect case) for same R/2 l lout R/2 in larger R/2 lin a. R/2 lout R/3 lin b. R/2 R/4 lin R/2 c. “costs” of congestion: r more work (retrans) for given “goodput” r unneeded retransmissions: link carries multiple copies of pkt 3 -85

Causes/costs of congestion: scenario 3 r four senders Q: what happens as l in and l increase ? r multihop paths in r timeout/retransmit Host A lin : original data lout l'in : original data, plus retransmitted data finite shared output link buffers Host B 3 -86

Causes/costs of congestion: scenario 3 H o st A l o u t H o st B Another “cost” of congestion: r when packet dropped, any “upstream transmission capacity used for that packet wasted! 3 -87

Approaches towards congestion control Two broad approaches towards congestion control: End-end congestion control: r no explicit feedback from network r congestion inferred from end-system observed loss, delay r approach taken by TCP Network-assisted congestion control: r routers provide feedback to end systems m single bit indicating congestion (SNA, DECbit, TCP/IP ECN, ATM) m explicit rate sender should send at 3 -88

Outline r Transport-layer services r Multiplexing and demultiplexing r Connectionless transport: UDP r Principles of reliable data transfer r Connection-oriented transport: TCP m m segment structure reliable data transfer flow control connection management r Principles of congestion control r TCP congestion control 3 -89

TCP Congestion Control r end-end control (no network assistance) r sender limits transmission: Last. Byte. Sent-Last. Byte. Acked Cong. Win r Roughly, rate = Cong. Win Bytes/sec RTT r Cong. Win is dynamic, function of perceived network congestion How does sender perceive congestion? r loss event = timeout or 3 duplicate acks r TCP sender reduces rate (Cong. Win) after loss event three mechanisms: m m m AIMD slow start conservative after timeout events 3 -90

TCP AIMD multiplicative decrease: cut Cong. Win in half after loss event additive increase: increase Cong. Win by MSS*(MSS/Cong. Win) bytes every RTT in the absence of loss events: probing Long-lived TCP connection 3 -91

TCP Slow Start r When connection begins, Cong. Win = 1 MSS m m Example: MSS = 500 bytes & RTT = 200 msec initial rate = 20 kbps r When connection begins, increase rate exponentially fast until first loss event r available bandwidth may be >> MSS/RTT m desirable to quickly ramp up to respectable rate 3 -92

TCP Slow Start (more) r When connection m m double Cong. Win every RTT done by incrementing Cong. Win for every ACK received RTT begins, increase rate exponentially until first loss event: Host A Host B one segme nt two segme nts four segme nts r Summary: initial rate is slow but ramps up exponentially fast time 3 -93

Refinement Philosophy: r After 3 dup ACKs: m Cong. Win m window is cut in half then grows linearly r But after timeout event: m Cong. Win instead set to 1 MSS; m window then grows exponentially m to a threshold (half of Cong. Win before loss event), then grows linearly • 3 dup ACKs indicates network capable of delivering some segments • timeout before 3 dup ACKs is “more alarming” 3 -94

Refinement (more) Q: When should the exponential increase switch to linear? A: When Cong. Win gets to 1/2 of its value before timeout. Implementation: r Variable Threshold r At loss event, Threshold is set to 1/2 of Cong. Win just before loss event 3 -95

Summary: TCP Congestion Control r When Cong. Win is below Threshold, sender in slow-start phase, window grows exponentially. r When Cong. Win is above Threshold, sender is in congestion-avoidance phase, window grows linearly. r When a triple duplicate ACK occurs, Threshold set to Cong. Win/2 and Cong. Win set to Threshold. r When timeout occurs, Threshold set to Cong. Win/2 and Cong. Win is set to 1 MSS. 3 -96

TCP sender congestion control State Event TCP Sender Action Commentary Slow Start (SS) ACK receipt Cong. Win = Cong. Win + MSS, for previously If (Cong. Win > Threshold) unacked data set state to “Congestion Avoidance” Resulting in a doubling of Cong. Win every RTT Congestion Avoidance (CA) ACK receipt Cong. Win = Cong. Win+MSS * for previously (MSS/Cong. Win) unacked data Additive increase, resulting in increase of Cong. Win by 1 MSS every RTT SS or CA Loss event detected by triple duplicate ACK Threshold = Cong. Win/2, Cong. Win = Threshold, Set state to “Congestion Avoidance” Fast recovery, implementing multiplicative decrease. Cong. Win will not drop below 1 MSS. SS or CA Timeout Threshold = Cong. Win/2, Cong. Win = 1 MSS, Set state to “Slow Start” Enter slow start SS or CA Duplicate ACK Increment duplicate ACK count for segment being acked Cong. Win and Threshold not changed 3 -97

TCP throughput r What’s the average throughout of TCP as a function of window size and RTT? m Ignore slow start r Let W be the window size when loss occurs. r When window is W, throughput is W/RTT r Just after loss, window drops to W/2, throughput to W/2 RTT. r Average throughout: . 75 W/RTT 3 -98

TCP 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 3 -99

Why is TCP fair? Two competing sessions: r Additive increase gives slope of 1, as throughout increases r multiplicative decreases throughput proportionally equal bandwidth share Connection 2 throughput R loss: decrease window by factor of 2 congestion avoidance: additive increase Connection 1 throughput R 3 -100

Fairness (more) Fairness and UDP r Multimedia apps often do not use TCP m do not want rate throttled by congestion control r Instead use UDP: m pump audio/video at constant rate, tolerate packet loss r Research area: TCP friendly Fairness and parallel TCP connections r nothing prevents app from opening parallel cnctions between 2 hosts. r Web browsers do this r Example: link of rate R supporting 9 connections; m m new app asks for 1 TCP, gets rate R/10 new app asks for 11 TCPs, gets R/2 ! 3 -101

Summary r principles behind transport layer services: m multiplexing, demultiplexing m reliable data transfer m flow control m congestion control r instantiation and implementation in the Internet m UDP m TCP Next: r leaving the network “edge” (application, transport layers) r into the network “core” 3 -102