3rd Edition: Chapter 3 - Simon Fraser University

3rd Edition: Chapter 3 - Simon Fraser University

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

Transport Layer 3-1 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 3-2 Transport services and protocols provide logical l ca gi lo network data link physical network data link physical d en den network data link physical a tr

network data link physical network data link physical rt po ns 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 applicatio n transport network data link physical

applicatio n transport network data link physical Transport Layer 3-3 Transport vs. network layer network layer: logical communication between hosts transport layer: logical communication between processes relies on, enhances, network layer services Household analogy: 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 Transport Layer 3-4 Internet transport-layer protocols reliable, in-order delivery (TCP) no-frills extension of best-effort IP services not available: delay guarantees bandwidth guarantees network data link physical network data link physical rt po ns

delivery: UDP network data link physical a tr unreliable, unordered network data link physical network data link physical d en den congestion control flow control connection setup l ca gi lo

applicatio n transport network data link physical applicatio n transport network data link physical Transport Layer 3-5 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 3-6 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 = process P 3 PP 11 application P 2 P 4 application transport transport transport network network network link

link link physical host 1 physical host 2 physical host 3 Transport Layer 3-7 How demultiplexing works host receives IP datagrams each datagram has source IP address, destination IP address each datagram carries 1 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 (message) TCP/UDP segment format Transport Layer 3-8 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 3-9 UDP: User Datagram Protocol 768] [RFC no frills, bare bones Internet transport protocol best effort service, UDP segments may be: lost delivered out of order to app connectionless: no handshaking between UDP sender, receiver each UDP segment

handled independently of others Why is there a UDP? no connection establishment (which can add delay) simple: no connection state at sender, receiver small segment header no congestion control: UDP can blast away as fast as desired Transport Layer 3-10 UDP: more often used for streaming multimedia apps loss tolerant rate sensitive Length, in bytes of UDP segment, including header other UDP uses DNS

SNMP reliable transfer over UDP: add reliability at application layer application-specific error recovery! 32 bits source port # dest port # length checksum Application data (message) UDP segment format Transport Layer 3-11 UDP checksum Goal: detect errors (e.g., flipped bits) in transmitted segment Sender: Receiver: treat segment contents

compute checksum of as sequence of 16-bit integers checksum: addition (1s complement sum) of segment contents sender puts checksum value into UDP checksum field received segment check if computed checksum equals checksum field value: NO - error detected YES - no error detected. But maybe errors nonetheless? More later . Transport Layer 3-12 Internet Checksum Example Note When adding numbers, a carryout from the most significant bit needs to be added to the result Example: add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0 1 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 wraparound 1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1 sum 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0 checksum 1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1 Transport Layer 3-13 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 3-14 Principles of Reliable data transfer important in app., transport, link layers characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt) Transport Layer 3-15 Principles of Reliable data transfer important in app., transport, link layers characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt) Transport Layer 3-16

Principles of Reliable data transfer important in app., transport, link layers characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt) Transport Layer 3-17 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 Transport Layer 3-18 Reliable data transfer: getting started Well: 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 state 1 event causing state transition actions taken on state transition event actions state 2

Transport Layer 3-19 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 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 Transport Layer 3-20 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 receiver feedback: control msgs (ACK,NAK) rcvr>sender Transport Layer 3-21 rdt2.0: FSM specification rdt_send(data)

sndpkt = make_pkt(data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && isNAK(rcvpkt) Wait for Wait for call from ACK or udt_send(sndpkt) above NAK rdt_rcv(rcvpkt) && isACK(rcvpkt) 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) Transport Layer 3-22 rdt2.0: operation with no errors

rdt_send(data) snkpkt = make_pkt(data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && isNAK(rcvpkt) Wait for Wait for call from ACK or udt_send(sndpkt) above NAK rdt_rcv(rcvpkt) && isACK(rcvpkt) 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) Transport Layer 3-23 rdt2.0: error scenario rdt_send(data) snkpkt = make_pkt(data, checksum)

udt_send(sndpkt) rdt_rcv(rcvpkt) && isNAK(rcvpkt) Wait for Wait for call from ACK or udt_send(sndpkt) above NAK rdt_rcv(rcvpkt) && isACK(rcvpkt) 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) Transport Layer 3-24 rdt2.0 has a fatal flaw! What happens if ACK/NAK corrupted? Handling duplicates:

sender doesnt know pkt if ACK/NAK garbled sender adds sequence number to each pkt receiver discards (doesnt deliver up) duplicate pkt what happened at receiver! cant just retransmit: possible duplicate sender retransmits current stop and wait Sender sends one packet, then waits for receiver response Transport Layer 3-25 rdt2.1: sender, handles garbled ACK/NAKs rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)

&& isACK(rcvpkt) Wait for call 0 from above ( corrupt(rcvpkt) || isNAK(rcvpkt) ) udt_send(sndpkt) Wait for ACK or NAK 0 rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) ) udt_send(sndpkt) Wait for ACK or NAK 1 Wait for call 1 from above

rdt_send(data) sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) Transport Layer 3-26 rdt2.1: receiver, handles garbled ACK/NAKs rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq0(rcvpkt) 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) 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) Wait for 0 from

below Wait for 1 from below rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt) rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq0(rcvpkt) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) Transport Layer 3-27 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 pkt has 0 or 1 seq. # 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 3-28 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 3-29 rdt2.2: 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_seq1(rcvpkt)) udt_send(sndpkt) Wait for 0 from below ( corrupt(rcvpkt) || isACK(rcvpkt,1) ) udt_send(sndpkt) Wait for ACK 0

sender FSM fragment rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) 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 3-30 rdt3.0: channels with errors and loss New assumption: underlying channel can also lose packets (data or ACKs) checksum, seq. #, ACKs, retransmissions will be of help, but not enough

Approach: sender waits 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 use of seq. #s already handles this receiver must specify seq # of pkt being ACKed requires countdown timer Transport Layer 3-31 rdt3.0 sender rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,1)

rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,0) ) timeout udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) stop_timer stop_timer timeout udt_send(sndpkt) start_timer Wait for ACK0 Wait for call 0from above rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) ) Wait

for ACK1 Wait for call 1 from above rdt_send(data) rdt_rcv(rcvpkt) sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) start_timer Transport Layer 3-32 rdt3.0 in action Transport Layer 3-33 rdt3.0 in action Transport Layer 3-34 rdt3.0: stop-and-wait operation sender

receiver first packet bit transmitted, t = 0 last packet bit transmitted, t = L / R first packet bit arrives last packet bit arrives, send ACK RTT ACK arrives, send next packet, t = RTT + L / R U sender = L/ R RTT +L / R = .008 30.008 = 0.00027 microsec onds Transport Layer

3-35 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 3-36 Pipelining: increased utilization sender receiver first packet bit transmitted, t = 0 last bit transmitted, t = 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 RTT ACK arrives, send next

packet, t = RTT + L / R Increase utilization by a factor of 3! U sender = 3*L/ R RTT +L / R = .024 30.008 = 0.0008 microsecon ds Transport Layer 3-37 Go-Back-N Sender: k-bit seq # in pkt header window of up to N, consecutive unacked pkts allowed ACK(n): ACKs all pkts up to, including seq # n - cumulative

ACK may receive duplicate ACKs (see receiver) timer for each in-flight pkt timeout(n): retransmit pkt n and all higher seq # pkts in window Transport Layer 3-38 GBN: sender extended FSM rdt_send(data) base=1 nextseqnum=1 rdt_rcv(rcvpkt) && corrupt(rcvpkt) 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) timeout start_timer Wait udt_send(sndpkt[base]) udt_send(sndpkt[base+1]) udt_send(sndpkt[nextseqnum1]) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) base = getacknum(rcvpkt)+1 If (base == nextseqnum) stop_timer else start_timer Transport Layer 3-39 GBN: receiver extended FSM default udt_send(sndpkt) 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 # may generate duplicate ACKs need only remember expectedseqnum out-of-order pkt: discard (dont buffer) -> no receiver buffering! Re-ACK pkt with highest in-order seq # Transport Layer 3-40 GBN in action Transport Layer 3-41 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 again limits seq #s of sent, unACKed pkts Transport Layer 3-42 Selective repeat: sender, receiver windows Transport Layer 3-43 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+N1] 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 3-44 Selective repeat in action Transport Layer

3-45 Selective repeat: dilemma Example: seq #s: 0, 1, 2, 3 window size=3 receiver sees no difference in two scenarios! incorrectly passes duplicate data as new in (a) Q: what relationship between seq # size and window size? Transport Layer 3-46 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 3-47 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 send & receive buffers socket door a p p lic a t io n w r it e s d a t a a p p lic a t io n re a d s d a ta TC P s e n d b u ffe r full duplex data: bi-directional data flow in same connection MSS: maximum segment size connection-oriented: TC P r e c e iv e b u f f e r

handshaking (exchange of control msgs) inits sender, receiver state before data exchange flow controlled: sender will not socket door overwhelm receiver segm ent Transport Layer 3-48 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 not U A Pnumber R S F Receive window head len used checksum Urg data pnter Options (variable length) counting by bytes of data (not segments!) # bytes rcvr willing to accept application data (variable length) Transport Layer 3-49

TCP seq. #s and ACKs Seq. #s: byte stream number of first byte in segments data ACKs: seq # of next byte expected from other side cumulative ACK Q: how receiver handles out-of-order segments A: TCP spec doesnt say, - up to implementor Host B Host A User types C host ACKs receipt of echoed C Seq=4 2, AC K =79, d

ata = C host ACKs receipt of C C, echoes ta = a d , 3 back C CK=4 A , 9 7 = Seq Seq=4 3, ACK =80 time simple telnet scenario Transport Layer 3-50

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

Transport Layer 3-51 TCP Round Trip Time and Timeout EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT Exponential weighted moving average influence of past sample decreases exponentially fast typical value: = 0.125 Transport Layer 3-52 Example RTT estimation: RTT: gaia.cs.umass.edu to fantasia.eurecom.fr 350 RTT (milliseconds) 300 250 200 150 100 1

8 15 22 29 36 43 50 57 64 71 78 85 92 99 106 time (seconnds) SampleRTT Estimated RTT

Transport Layer 3-53 TCP Round Trip Time and Timeout Setting the timeout EstimtedRTT plus safety margin large variation in EstimatedRTT -> larger safety margin first estimate of how much SampleRTT deviates from EstimatedRTT: DevRTT = (1-)*DevRTT + *|SampleRTT-EstimatedRTT|SampleRTT-EstimatedRTT|SampleRTT-EstimatedRTT| (typically, = 0.25)) Then set timeout interval: TimeoutInterval = EstimatedRTT + 4*DevRTT Transport Layer 3-54 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 3-55 TCP reliable data transfer TCP creates rdt service on top of IPs unreliable service Pipelined segments Cumulative acks

TCP uses single retransmission timer Retransmissions are triggered by: timeout events duplicate acks Initially consider simplified TCP sender: ignore duplicate acks ignore flow control, congestion control Transport Layer 3-56 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 acknowledges previously unacked segments update what is known to be acked start timer if there are outstanding segments Transport Layer 3-57 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 */ TCP sender (simplified ) Comment: SendBase-1: last cumulatively acked byte Example: SendBase-1 = 71; y= 73, so the rcvr wants 73+ ; y > SendBase, so that new data is acked

Transport Layer 3-58 TCP: retransmission scenarios Host A Seq=92 timeout X tes da ta =100 K C A loss Seq=9 2, 8 byte s data =100 K C A SendBase = 100

Sendbase = 100 SendBase = 120 SendBase = 120 time lost ACK scenario Host B Seq=9 2, 8 b ytes d ata Seq= 100, 20 by t es da ta 0 10 = K 120 = C K A AC Seq=9 2,

Seq=92 timeout timeout Seq=9 2, 8 b y time Host A Host B 8 byte s data 20 1 = K AC premature timeout Transport Layer 3-59 TCP retransmission scenarios (more) Host A Host B

timeout Seq=9 2, 8 b y SendBase = 120 tes da ta =100 Seq=1 K C A 00, 20 b y te s data X loss =120 K C A time Cumulative ACK scenario Transport Layer

3-60 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 500ms 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 Immediate send ACK, provided that segment startsat lower end of gap Transport Layer 3-61 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 backto-back If segment is lost, there will likely be many duplicate ACKs. If sender receives 3 ACKs for the same

data, it supposes that segment after ACKed data was lost: fast retransmit: resend segment before timer expires Transport Layer 3-62 Fast retransmit algorithm: 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 } a duplicate ACK for already ACKed segment fast retransmit Transport Layer 3-63

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 3-64

TCP Flow Control flow control receive side of TCP connection has a receive buffer: sender wont overflow receivers buffer by transmitting too much, too fast speed-matching service: matching the send rate to the receiving apps drain rate app process may be slow at reading from buffer Transport Layer 3-65 TCP Flow control: how it works Rcvr advertises spare

(Suppose TCP receiver discards out-of-order segments) spare room in buffer room by including value of RcvWindow in segments Sender limits unACKed data to RcvWindow guarantees receive buffer doesnt overflow = RcvWindow = RcvBuffer-[LastByteRcvd LastByteRead] Transport Layer 3-66 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 3-67 TCP Connection Management Recall: TCP sender, receiver establish connection before exchanging 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(); Three way handshake: Step 1: client host sends TCP SYN segment to server specifies initial seq # no data Step 2: server host receives SYN, replies with SYNACK segment server allocates buffers specifies server initial seq. # Step 3: client receives SYNACK, replies with ACK segment, which may contain data Transport Layer 3-68 TCP Connection Management (cont.) Closing a connection: client closes socket: clientSocket.close(); client

close Step 1: client end system close FIN timed wait replies with ACK. Closes connection, sends FIN. FIN ACK sends TCP FIN control segment to server Step 2: server receives FIN, server ACK closed Transport Layer 3-69 TCP Connection Management (cont.)

Step 3: client receives FIN, replies with ACK. Enters timed wait - will respond with ACK to received FINs client closing closing FIN timed wait Connection closed. can handle simultaneous FINs. FIN ACK Step 4: server, receives ACK. Note: with small modification, server ACK closed

closed Transport Layer 3-70 TCP Connection Management (cont) TCP server lifecycle TCP client lifecycle Transport Layer 3-71 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 3-72 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 3-73

Causes/costs of congestion: scenario 1 Host A two senders, two receivers one router, infinite buffers no retransmission Host B out in : original data unlimited shared output link buffers large delays when congested maximum achievable throughput Transport Layer 3-74 Causes/costs of congestion: scenario 2 one router, finite buffers

sender retransmission of lost packet Host A in : original data out 'in : original data, plus retransmitted data Host B finite shared output link buffers Transport Layer 3-75 Causes/costs of congestion: scenario 2 always: (goodput) = out in perfect retransmission only when loss: > out in retransmission of delayed (not lost) packet makes in

larger (than perfect case) forsame out R/2 R/2 R/2 in a. R/2 out out out R/3 in b. R/2 R/4 in

R/2 c. costs of congestion: more work (retrans) for given goodput unneeded retransmissions: link carries multiple copies of pkt Transport Layer 3-76 Causes/costs of congestion: scenario 3 four senders multihop paths timeout/retransmit Host A Q: what happens as in and in increase ? in : original data out 'in : original data, plus retransmitted data finite shared output

link buffers Host B Transport Layer 3-77 Causes/costs of congestion: scenario 3 H o s t A o u t H o s t B Another cost of congestion: when packet dropped, any upstream transmission capacity used for that packet was wasted! Transport Layer 3-78

Approaches towards congestion control Two broad approaches towards congestion control: Network-assisted End-end congestion congestion control: control: no explicit feedback from routers provide feedback network congestion inferred from end-system observed loss, delay approach taken by TCP to end systems single bit indicating congestion (SNA, DECbit, TCP/IP ECN, ATM) explicit rate sender should send at Transport Layer 3-79 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 3-80 TCP congestion control: additive increase, multiplicative decrease

Approach: increase transmission rate (window Saw tooth behavior: probing for bandwidth congestion window size size), probing for usable bandwidth, until loss occurs additive increase: increase CongWin by 1 MSS every RTT until loss detected multiplicative decrease: cut CongWin in half after loss c o n g e s t io n w in d o w 2 4 K b y te s 1 6 K b y te s 8 K b y te s time tim e Transport Layer 3-81 TCP Congestion Control: details sender limits transmission: LastByteSent-LastByteAcked

CongWin Roughly, How does sender perceive congestion? loss event = timeout or 3 duplicate acks TCP sender reduces CongWin rate = Bytes/sec rate (CongWin) after RTT loss event CongWin is dynamic, function three mechanisms: of perceived network AIMD congestion slow start conservative after timeout events Transport Layer 3-82 TCP Slow Start When connection begins, CongWin = 1 MSS

Example: MSS = 500 bytes & RTT = 200 msec initial rate = 20 kbps When connection begins, increase rate exponentially fast until first loss event available bandwidth may be >> MSS/RTT desirable to quickly ramp up to respectable rate Transport Layer 3-83 TCP Slow Start (more) When connection

double CongWin every RTT done by incrementing CongWin for every ACK received RTT begins, increase rate exponentially until first loss event: Host A Host B one segm en t two segm ents four segm ents Summary: initial rate is slow but ramps up exponentially fast time

Transport Layer 3-84 Refinement Q: When should the exponential increase switch to linear? A: When CongWin gets to 1/2 of its value before timeout. Implementation: Variable Threshold At loss event, Threshold is set to 1/2 of CongWin just before loss event Transport Layer 3-85 Refinement: inferring loss After 3 dup ACKs: CongWin is cut in half window then grows linearly But after timeout event: CongWin instead set to 1 MSS; window then grows exponentially to a threshold, then grows linearly

Philosophy: 3 dup ACKs indicates network capable of delivering some segments timeout indicates a more alarming congestion scenario Transport Layer 3-86 Summary: TCP Congestion Control When CongWin is below Threshold, sender in slow-start phase, window grows exponentially. When CongWin is above Threshold, sender is in congestion-avoidance phase, window grows linearly. When a triple duplicate ACK occurs, Threshold set to CongWin/2 and CongWin set to Threshold. When timeout occurs, Threshold set to CongWin/2 and CongWin is set to 1 MSS.

Transport Layer 3-87 TCP sender congestion control State Event TCP Sender Action Commentary Slow Start (SS) ACK receipt for previously unacked data CongWin = CongWin + MSS, If (CongWin > Threshold) set state to Congestion Avoidance Resulting in a doubling of CongWin every RTT Congestion Avoidance (CA)

ACK receipt for previously unacked data CongWin = CongWin+MSS * (MSS/CongWin) Additive increase, resulting in increase of CongWin by 1 MSS every RTT SS or CA Loss event detected by triple duplicate ACK Threshold = CongWin/2, CongWin = Threshold, Set state to Congestion Avoidance Fast recovery, implementing multiplicative decrease. CongWin will not drop below 1 MSS. SS or CA Timeout

Threshold = CongWin/2, CongWin = 1 MSS, Set state to Slow Start Enter slow start SS or CA Duplicate ACK Increment duplicate ACK count for segment being acked CongWin and Threshold not changed Transport Layer 3-88 TCP throughput Whats the average throughout of TCP as a function of window size and RTT? Ignore slow start Let W be the window size when loss occurs. When window is W, throughput is W/RTT Just after loss, window drops to W/2,

throughput to W/2RTT. Average throughout: .75 W/RTT Transport Layer 3-89 TCP Futures 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: 1. 22 MSS RTT L L = 210-10 Wow New versions of TCP for high-speed needed! Transport Layer 3-90 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 3-91 Chapter 3: Summary principles behind transport layer services: multiplexing, demultiplexing reliable data transfer flow control congestion control instantiation and implementation in the Internet UDP TCP Next: leaving the network edge (application, transport layers) into the network core Transport Layer 3-92

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