Chapter_3_V6.01.pdf

Chapter 3 Transport Layer A note on the use of these ppt slides: We’re making these slides freely available to all (faculty, students, readers). They’re in PowerPoint form so you see the animations; and can add, modify, and delete slides (including this one) and slide content to suit your needs. The...

Chapter 3 Transport Layer A note on the use of these ppt slides: We’re making these slides freely available to all (faculty, students, readers). They’re in PowerPoint form so you see the animations; and can add, modify, and delete slides (including this one) and slide content to suit your needs. They obviously represent a lot of work on our part. In return for use, we only ask the following: v If you use these slides (e.g., in a class) that you mention their source (after all, we’d like people to use our book!) v If you post any slides on a www site, that you note that they are adapted from (or perhaps identical to) our slides, and note our copyright of this material. Computer Networking: A Top Down Approach 6th edition Jim Kurose, Keith Ross Addison-Wesley March 2012 Thanks and enjoy! JFK/KWR All material copyright 1996-2013 J.F Kurose and K.W. Ross, All Rights Reserved Transport Layer 3-1 Chapter 3: Transport Layer our goals: v understand principles behind transport layer services: § multiplexing, demultiplexing § reliable data transfer § flow control § congestion control v 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 v v v 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 v transport layer: logical communication between processes v § relies on, enhances, network layer services household analogy: 12 kids in Ann’s house sending letters to 12 kids in Bill’s house: v hosts = houses v processes = kids v app messages = letters in envelopes v transport protocol = Ann and Bill who demux to inhouse siblings v network-layer protocol = postal service Transport Layer 3-5 Internet transport-layer protocols v reliable, in-order delivery (TCP) § congestion control § flow control § connection setup v unreliable, unordered delivery: UDP § no-frills extension of “best-effort” IP v services not available: application transport network data link physical network data link physical network data link physical network data link physical network data link physical network data link physical network data link physical network data link physical application transport network data link physical § delay guarantees § bandwidth guarantees Transport Layer 3-6 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 3-7 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 application application P1 P2 application P3 transport P4 transport network transport network link network physical link link physical socket process physical Transport Layer 3-8 How demultiplexing works v 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 v 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 3-9 Connectionless demultiplexing v recall: created socket has host-local port #: v DatagramSocket mySocket1 = new DatagramSocket(12534); v when host receives UDP segment: § checks destination port # in segment § directs UDP segment to socket with that port # recall: when creating 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 Transport Layer 3-10 Connectionless demux: example DatagramSocket mySocket2 = new DatagramSocket (9157); DatagramSocket serverSocket = new DatagramSocket (6428); application application DatagramSocket mySocket1 = new DatagramSocket (5775); application P1 P3 P4 transport transport transport network network link network link physical link physical physical source port: 6428 dest port: 9157 source port: 9157 dest port: 6428 source port: ? dest port: ? source port: ? dest port: ? Transport Layer 3-11 Connection-oriented demux v TCP socket identified by 4-tuple: § § § § v source IP address source port number dest IP address dest port number demux: receiver uses all four values to direct segment to appropriate socket v server host may support many simultaneous TCP sockets: § each socket identified by its own 4-tuple v web servers have different sockets for each connecting client § non-persistent HTTP will have different socket for each request Transport Layer 3-12 Connection-oriented demux: example application application P4 P3 P5 application P6 P3 P2 transport network network link network link physical link physical host: IP address A transport transport server: IP address B source IP,port: B,80 dest IP,port: A,9157 source IP,port: A,9157 dest IP, port: B,80 three segments, all destined to IP address: B, dest port: 80 are demultiplexed to different sockets physical source IP,port: C,5775 dest IP,port: B,80 host: IP address C source IP,port: C,9157 dest IP,port: B,80 Transport Layer 3-13 Connection-oriented demux: example threaded server application application P3 application P4 P3 P2 transport network network link network link physical link physical host: IP address A transport transport server: IP address B source IP,port: B,80 dest IP,port: A,9157 source IP,port: A,9157 dest IP, port: B,80 physical source IP,port: C,5775 dest IP,port: B,80 host: IP address C source IP,port: C,9157 dest IP,port: B,80 Transport Layer 3-14 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 3-15 UDP: User Datagram Protocol [RFC 768] v v v “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 v UDP use: § streaming multimedia apps (loss tolerant, rate sensitive) § DNS § SNMP v reliable transfer over UDP: § add reliability at application layer § application-specific error recovery! Transport Layer 3-16 UDP: segment header 32 bits source port # dest port # length checksum application data (payload) length, in bytes of UDP segment, including header why is there a UDP? v v v UDP segment format v no connection establishment (which can add delay) simple: no connection state at sender, receiver small header size no congestion control: UDP can blast away as fast as desired Transport Layer 3-17 UDP checksum Goal: detect “errors” (e.g., flipped bits) in transmitted segment sender: receiver: v v v v 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 v 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 3-18 Internet checksum: example 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 Note: when adding numbers, a carryout from the most significant bit needs to be added to the result Transport Layer 3-19 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 3-20 Principles of reliable data transfer v important in application, transport, link layers § top-10 list of important networking topics! v characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt) Transport Layer 3-21 Principles of reliable data transfer v important in application, transport, link layers § top-10 list of important networking topics! v characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt) Transport Layer 3-22 Principles of reliable data transfer v important in application, transport, link layers § top-10 list of important networking topics! v characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt) Transport Layer 3-23 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 3-24 Reliable data transfer: getting started we’ll: v incrementally develop sender, receiver sides of reliable data transfer protocol (rdt) v consider only unidirectional data transfer § but control info will flow on both directions! v use finite state machines (FSM) to specify sender, receiver event causing state transition actions taken on state transition state: when in this “state” next state uniquely determined by next event state 1 event actions state 2 Transport Layer 3-25 rdt1.0: reliable transfer over a reliable channel v underlying channel perfectly reliable § no bit errors § no loss of packets v 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 3-26 rdt2.0: channel with bit errors v underlying channel may flip bits in packet § checksum to detect bit errors v v 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 pkt on receipt from of NAK Howretransmits do humans recover “errors” new mechanisms in rdt2.0 (beyond rdt1.0): during conversation? § error detection § receiver feedback: control msgs (ACK,NAK) rcvr>sender Transport Layer 3-27 rdt2.0: channel with bit errors v underlying channel may flip bits in packet § checksum to detect bit errors v v 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 3-28 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) Λ 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 3-29 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) Λ 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 3-30 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) Λ 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 3-31 rdt2.0 has a fatal flaw! what happens if ACK/ NAK corrupted? v v sender doesn’t know what happened at receiver! can’t just retransmit: possible duplicate handling duplicates: v v v 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 3-32 rdt2.1: sender, handles garbled ACK/NAKs rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && Wait for call 0 from above rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt) ( corrupt(rcvpkt) || isNAK(rcvpkt) ) udt_send(sndpkt) Wait for ACK or NAK 0 rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt) Λ rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) ) udt_send(sndpkt) Λ 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 3-33 rdt2.1: receiver, handles garbled ACK/NAKs rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq0(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_seq1(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_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 3-34 rdt2.1: discussion sender: v seq # added to pkt v two seq. #’s (0,1) will suffice. Why? v must check if received ACK/NAK corrupted v twice as many states § state must “remember” whether “expected” pkt should have seq # of 0 or 1 receiver: v must check if received packet is duplicate § state indicates whether 0 or 1 is expected pkt seq # v note: receiver can not know if its last ACK/ NAK received OK at sender Transport Layer 3-35 rdt2.2: a NAK-free protocol v v 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 v duplicate ACK at sender results in same action as NAK: retransmit current pkt Transport Layer 3-36 rdt2.2: sender, receiver fragments rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || Wait for Wait for isACK(rcvpkt,1) ) ACK call 0 from 0 udt_send(sndpkt) above sender FSM fragment rdt_rcv(rcvpkt) && (corrupt(rcvpkt) || has_seq1(rcvpkt)) udt_send(sndpkt) Wait for 0 from below rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) receiver FSM fragment Λ rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK1, chksum) udt_send(sndpkt) Transport Layer 3-37 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 v v v 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 3-38 rdt3.0 sender rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) Λ rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) ) Λ Wait for ACK0 Wait for call 0from above rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,1) timeout udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) stop_timer stop_timer timeout udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,0) ) Λ Wait for ACK1 Wait for call 1 from above rdt_send(data) rdt_rcv(rcvpkt) Λ sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) start_timer Transport Layer 3-39 rdt3.0 in action receiver sender send pkt0 rcv ack0 send pkt1 rcv ack1 send pkt0 pkt0 ack0 pkt1 ack1 pkt0 ack0 (a) no loss send pkt0 rcv pkt0 send ack0 rcv pkt1 send ack1 rcv pkt0 send ack0 receiver sender rcv ack0 send pkt1 pkt0 ack0 rcv pkt0 send ack0 pkt1 X loss timeout resend pkt1 rcv ack1 send pkt0 pkt1 ack1 pkt0 ack0 rcv pkt1 send ack1 rcv pkt0 send ack0 (b) packet loss Transport Layer 3-40 rdt3.0 in action receiver sender send pkt0 pkt0 rcv ack0 send pkt1 ack0 pkt1 ack1 X rcv pkt0 send ack0 timeout resend pkt1 pkt1 rcv ack1 send pkt0 ack1 pkt0 ack0 (c) ACK loss send pkt0 rcv ack0 send pkt1 rcv pkt1 send ack1 loss rcv pkt1 (detect duplicate) send ack1 rcv pkt0 send ack0 receiver sender pkt0 rcv pkt0 send ack0 ack0 pkt1 rcv pkt1 send ack1 ack1 timeout resend pkt1 rcv ack1 send pkt0 rcv ack1 send pkt0 pkt1 rcv pkt1 pkt0 ack1 ack0 pkt0 (detect duplicate) ack0 (detect duplicate) send ack1 rcv pkt0 send ack0 rcv pkt0 send ack0 (d) premature timeout/ delayed ACK Transport Layer 3-41 Performance of rdt3.0 v v rdt3.0 is correct, but performance stinks e.g.: 1 Gbps link, 15 ms prop. delay, 8000 bit packet: 8000 bits L Dtrans = R = 109 bits/sec = 8 microsecs § U sender: utilization – fraction of time sender busy sending U sender = L/R RTT + L / R =.008 30.008 = 0.00027 § if RTT=30 msec, 1KB pkt every 30 msec: 33kB/sec thruput over 1 Gbps link v network protocol limits use of physical resources! Transport Layer 3-42 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 Transport Layer 3-43 Pipelined protocols pipelining: sender allows multiple, “in-flight”, yetto-be-acknowledged pkts § range of sequence numbers must be increased § buffering at sender and/or receiver v two generic forms of pipelined protocols: go-Back-N, selective repeat Transport Layer 3-44 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 3-packet pipelining increases utilization by a factor of 3! U sender = 3L / R RTT + L / R =.0024 30.008 = 0.00081 Transport Layer 3-45 Pipelined protocols: overview Go-back-N: v sender can have up to N unacked packets in pipeline v receiver only sends cumulative ack Selective Repeat: v sender can have up to N unack’ed packets in pipeline v rcvr sends individual ack for each packet § doesn’t ack packet if there’s a gap v sender has timer for oldest unacked packet § when timer expires, retransmit all unacked packets v sender maintains timer for each unacked packet § when timer expires, retransmit only that unacked packet Transport Layer 3-46 Go-Back-N: sender v v v v v 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 3-47 GBN: sender extended FSM rdt_send(data) Λ 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[nextseqnum-1] ) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) base = getacknum(rcvpkt)+1 If (base == nextseqnum) stop_timer else start_timer Transport Layer 3-48 GBN: receiver extended FSM default udt_send(sndpkt) Λ 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 v out-of-order pkt: § discard (don’t buffer): no receiver buffering! § re-ACK pkt with highest in-order seq # Transport Layer 3-49 GBN in action sender window (N=4) 012345678 012345678 012345678 012345678 012345678 012345678 sender send pkt0 send pkt1 send pkt2 send pkt3 (wait) rcv ack0, send pkt4 rcv ack1, send pkt5 ignore duplicate ACK 012345678 012345678 012345678 012345678 pkt 2 timeout send pkt2 send pkt3 send pkt4 send pkt5 receiver Xloss 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 rcv rcv rcv pkt2, pkt3, pkt4, pkt5, deliver, deliver, deliver, deliver, send send send send ack2 ack3 ack4 ack5 Transport Layer 3-50 Selective repeat v receiver individually acknowledges all correctly received pkts § buffers pkts, as needed, for eventual in-order delivery to upper layer v sender only resends pkts for which ACK not received § sender timer for each unACKed pkt v sender window § N consecutive seq #’s § limits seq #s of sent, unACKed pkts Transport Layer 3-51 Selective repeat: sender, receiver windows Transport Layer 3-52 Selective repeat sender data from above: v if next available seq # in window, send pkt timeout(n): resend pkt n, restart timer ACK(n) in [sendbase,sendbase+N]: v mark pkt n as received v if n smallest unACKed pkt, advance window base to next unACKed seq # receiver pkt n in [rcvbase, rcvbase+N-1] v v v v 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] v ACK(n) otherwise: v ignore Transport Layer 3-53 Selective repeat in action sender window (N=4) 012345678 012345678 012345678 012345678 012345678 012345678 sender send pkt0 send pkt1 send pkt2 send pkt3 (wait) receiver Xloss rcv ack0, send pkt4 rcv ack1, send pkt5 record ack3 arrived 012345678 012345678 012345678 012345678 pkt 2 timeout send pkt2 record ack4 arrived record ack5 arrived 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 Q: what happens when ack2 arrives? Transport Layer 3-54 Selective repeat: dilemma example: v v v v 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)? receiver window (after receipt) sender window (after receipt) 0123012 pkt0 0123012 pkt1 0123012 0123012 pkt2 0123012 0123012 pkt3 0123012 pkt0 (a) no problem 0123012 X will accept packet with seq number 0 receiver can’t see sender side. receiver behavior identical in both cases! something’s (very) wrong! 0123012 pkt0 0123012 pkt1 0123012 0123012 pkt2 0123012 X X timeout retransmit pkt0 X 0123012 0123012 (b) oops! pkt0 will accept packet with seq number 0 Transport Layer 3-55 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 3-56 TCP: Overview v RFCs: 793,1122,1323, 2018, 2581 point-to-point: v § one sender, one receiver v v § bi-directional data flow in same connection § MSS: maximum segment size reliable, in-order byte steam: § no “message boundaries” full duplex data: v connection-oriented: § handshaking (exchange of control msgs) inits sender, receiver state before data exchange pipelined: § TCP congestion and flow control set window size v flow controlled: § sender will not overwhelm receiver Transport Layer 3-57 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 len used U A P R S F checksum receive window Urg data pointer options (variable length) counting by bytes of data (not segments!) # bytes rcvr willing to accept application data (variable length) Transport Layer 3-58 TCP seq. numbers, ACKs outgoing segment from sender 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 source port # dest port # sequence number acknowledgement number rwnd checksum urg pointer window size N sender sequence number space sent ACKed sent, notyet ACKed (“inflight”) usable not but not usable yet sent incoming segment to sender source port # dest port # sequence number acknowledgement number rwnd A checksum urg pointer Transport Layer 3-59 TCP seq. numbers, ACKs Host B Host A User types ‘C’ host ACKs receipt of echoed ‘C’ Seq=42, ACK=79, data = ‘C’ Seq=79, ACK=43, data = ‘C’ host ACKs receipt of ‘C’, echoes back ‘C’ Seq=43, ACK=80 simple telnet scenario Transport Layer 3-60 TCP round trip time, timeout Q: how to set TCP timeout value? v Q: how to estimate RTT? v longer than RTT § but RTT varies v v too short: premature timeout, unnecessary retransmissions too long: slow reaction to segment loss v 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 3-61 TCP round trip time, timeout EstimatedRTT = (1- α)*EstimatedRTT + α*SampleRTT v v exponential weighted moving average influence of past sample decreases exponentially fast typical value: α = 0.125 RTT: gaia.cs.umass.edu to fantasia.eurecom.fr 350 RTT: gaia.cs.umass.edu to fantasia.eurecom.fr RTT (milliseconds) RTT (milliseconds) v 300 250 200 sampleRTT 150 EstimatedRTT 100 1 8 15 22 29 36 43 50 57 64 71 time (seconnds) time (seconds) SampleRTT Estimated RTT 78 85 92 99 106 Transport Layer 3-62 TCP round trip time, timeout v timeout interval: EstimatedRTT plus “safety margin” § large variation in EstimatedRTT -> larger safety margin v estimate SampleRTT deviation from EstimatedRTT: DevRTT = (1-β)*DevRTT + β*|SampleRTT-EstimatedRTT| (typically, β = 0.25) TimeoutInterval = EstimatedRTT + 4*DevRTT estimated RTT “safety margin” 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 reliable data transfer v TCP creates rdt service on top of IP’s unreliable service § pipelined segments § cumulative acks § single retransmission timer v retransmissions triggered by: let’s initially consider simplified TCP sender: § ignore duplicate acks § ignore flow control, congestion control § timeout events § duplicate acks Transport Layer 3-65 TCP sender events: data rcvd from app: v create segment with seq # v seq # is byte-stream number of first data byte in segment v start timer if not already running § think of timer as for oldest unacked segment § expiration interval: TimeOutInterval timeout: v retransmit segment that caused timeout v restart timer ack rcvd: v if ack acknowledges previously unacked segments § update what is known to be ACKed § start timer if there are still unacked segments Transport Layer 3-66 TCP sender (simplified) data received from application above Λ NextSeqNum = InitialSeqNum SendBase = InitialSeqNum wait for event 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 ACK received, with ACK field value y if (y > SendBase) { SendBase = y if (there are currently not-yet-acked segments) start timer else stop timer } Transport Layer 3-67 TCP: retransmission scenarios Host B Host A Host B Host A SendBase=92 X ACK=100 Seq=92, 8 bytes of data timeout timeout Seq=92, 8 bytes of data Seq=100, 20 bytes of data ACK=100 ACK=120 Seq=92, 8 bytes of data SendBase=100 ACK=100 Seq=92, 8 bytes of data SendBase=120 ACK=120 SendBase=120 lost ACK scenario premature timeout Transport Layer 3-68 TCP: retransmission scenarios Host B Host A Seq=92, 8 bytes of data timeout Seq=100, 20 bytes of data X ACK=100 ACK=120 Seq=120, 15 bytes of data cumulative ACK Transport Layer 3-69 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 starts at lower end of gap Transport Layer 3-70 TCP fast retransmit v time-out period often relatively long: § long delay before resending lost packet v detect lost segments via duplicate ACKs. § sender often sends many segments backto-back § if segment is lost, there will likely be many duplicate ACKs. TCP fast retransmit if sender receives 3 ACKs for same data (“triple (“triple duplicate duplicate ACKs”), ACKs”), resend unacked segment with smallest seq # § likely that unacked segment lost, so don’t wait for timeout Transport Layer 3-71 TCP fast retransmit Host B Host A Seq=92, 8 bytes of data Seq=100, 20 bytes of data X timeout 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 3-72 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 3-73 TCP flow control application may remove data from TCP socket buffers …. … slower than TCP receiver is delivering (sender is sending) application process application TCP code IP code flow control receiver controls sender, so sender won’t overflow receiver’s buffer by transmitting too much, too fast OS TCP socket receiver buffers from sender receiver protocol stack Transport Layer 3-74 TCP flow control v 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 v v sender limits amount of unacked (“in-flight”) data to receiver’s rwnd value guarantees receive buffer will not overflow to application process RcvBuffer rwnd buffered data free buffer space TCP segment payloads receiver-side buffering Transport Layer 3-75 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 3-76 Connection Management before exchanging data, sender/receiver “handshake”: v v 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 Socket clientSocket = newSocket("hostname","port number"); application connection state: ESTAB connection Variables: seq # client-to-server server-to-client rcvBuffer size at server,client network Socket connectionSocket = welcomeSocket.accept(); Transport Layer 3-77 Agreeing to establish a connection 2-way handshake: Q: will 2-way handshake always work in network? Let’s talk ESTAB OK ESTAB v v v choose x ESTAB v req_conn(x) acc_conn(x) variable delays retransmitted messages (e.g. req_conn(x)) due to message loss message reordering can’t “see” other side ESTAB Transport Layer 3-78 Agreeing to establish a connection 2-way handshake failure scenarios: choose x choose x req_conn(x) req_conn(x) ESTAB ESTAB retransmit req_conn(x) retransmit req_conn(x) acc_conn(x) ESTAB ESTAB req_conn(x) client terminates connection x completes acc_conn(x) data(x+1) accept data(x+1) retransmit data(x+1) server forgets x ESTAB half open connection! (no client!) client terminates connection x completes req_conn(x) data(x+1) server forgets x ESTAB accept data(x+1) Transport Layer 3-79 TCP 3-way handshake client state server state LISTEN LISTEN choose init seq num, x send TCP SYN msg SYNSENT received SYNACK(x) indicates server is live; ESTAB send ACK for SYNACK; this segment may contain client-to-server data SYNbit=1, Seq=x choose init seq num, y send TCP SYNACK SYN RCVD msg, acking SYN SYNbit=1, Seq=y ACKbit=1; ACKnum=x+1 ACKbit=1, ACKnum=y+1 received ACK(y) indicates client is live ESTAB Transport Layer 3-80 TCP 3-way handshake: FSM closed Socket connectionSocket = welcomeSocket.accept(); Λ SYN(x) SYNACK(seq=y,ACKnum=x+1) create new socket for communication back to client listen SYN(seq=x) SYN sent SYN rcvd ACK(ACKnum=y+1) Socket clientSocket = newSocket("hostname","port number"); ESTAB SYNACK(seq=y,ACKnum=x+1) ACK(ACKnum=y+1) Λ Transport Layer 3-81 TCP: closing a connection v client, server each close their side of connection § send TCP segment with FIN bit = 1 v respond to received FIN with ACK § on receiving FIN, ACK can be combined with own FIN v simultaneous FIN exchanges can be handled Transport Layer 3-82 TCP: closing a connection client state server state ESTAB ESTAB clientSocket.close() FIN_WAIT_1 FIN_WAIT_2 can no longer send but can receive data FINbit=1, seq=x CLOSE_WAIT ACKbit=1; ACKnum=x+1 wait for server close FINbit=1, seq=y TIMED_WAIT timed wait for 2*max segment lifetime can still send data LAST_ACK can no longer send data ACKbit=1; ACKnum=y+1 CLOSED CLOSED Transport Layer 3-83 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 3-84 Principles of congestion control congestion: v v v v 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 3-85 Causes/costs of congestion: scenario 1 v v λout Host A unlimited shared output link buffers Host B R/2 delay v two senders, two receivers one router, infinite buffers output link capacity: R no retransmission throughput: λout v original data: λin v λin R/2 maximum per-connection throughput: R/2 v λin R/2 large delays as arrival rate, λin, approaches capacity Transport Layer 3-86 Causes/costs of congestion: scenario 2 v v one router, finite buffers sender retransmission of timed-out packet § application-layer input = application-layer output: λin = λout § transport-layer input includes retransmissions : λ‘in λin λin : original data λ'in: original data, plus λout retransmitted data Host A Host B finite shared output link buffers Transport Layer 3-87 Causes/costs of congestion: scenario 2 λout idealization: perfect knowledge v sender sends only when router buffers available R/2 λin : original data λ'in: original data, plus copy λin R/2 λout retransmitted data A Host B free buffer space! finite shared output link buffers Transport Layer 3-88 Causes/costs of congestion: scenario 2 Idealization: known loss v packets can be lost, dropped at router due to full buffers sender only resends if packet known to be lost λin : original data λ'in: original data, plus copy λout retransmitted data A no buffer space! Host B Transport Layer 3-89 Causes/costs of congestion: scenario 2 v packets can be lost, dropped at router due to full buffers sender only resends if packet known to be lost R/2 when sending at R/2, some packets are retransmissions but asymptotic goodput is still R/2 (why?) λout Idealization: known loss λin : original data λ'in: original data, plus λin R/2 λout retransmitted data A free buffer space! Host B Transport Layer 3-90 Causes/costs of congestion: scenario 2 v v packets can be lost, dropped at router due to full buffers sender times out prematurely, sending two copies, both of which are delivered R/2 λin λ'in timeout copy A when sending at R/2, some packets are retransmissions including duplicated that are delivered! λout Realistic: duplicates λin R/2 λout free buffer space! Host B Transport Layer 3-91 Causes/costs of congestion: scenario 2 v v packets can be lost, dropped at router due to full buffers sender times out prematurely, sending two copies, both of which are delivered R/2 when sending at R/2, some packets are retransmissions including duplicated that are delivered! λout Realistic: duplicates λin R/2 “costs” of congestion: v v more work (retrans) for given “goodput” unneeded retransmissions: link carries multiple copies of pkt § decreasing goodput Transport Layer 3-92 Causes/costs of congestion: scenario 3 v v v four senders multihop paths timeout/retransmit Host A Q: what happens as λin and λin’ increase ? A: as red λin’ increases, all arriving blue pkts at upper queue are dropped, blue throughput g 0 λin : original data λ'in: original data, plus λout Host B retransmitted data finite shared output link buffers Host D Host C Transport Layer 3-93 Causes/costs of congestion: scenario 3 λout C/2 λin’ C/2 another “cost” of congestion: v when packet dropped, any “upstream transmission capacity used for that packet was wasted! Transport Layer 3-94 Approaches towards congestion control two broad approaches towards congestion control: end-end congestion control: v v v no explicit feedback from network congestion inferred from end-system observed loss, delay approach taken by TCP network-assisted congestion control: v 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 3-95 Case study: ATM ABR congestion control ABR: available bit rate: v v v “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: v v v 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 3-96 Case study: ATM ABR congestion control RM cell v 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 v 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 3-97 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 3-98 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 AIMD saw tooth behavior: probing for bandwidth cwnd: TCP sender congestion window size v additively increase window size … …. until loss occurs (then cut window in half) time Transport Layer 3-99 TCP Congestion Control: details sender sequence number space cwnd last byte ACKed v sent, notyet ACKed (“inflight”) last byte sent sender limits transmission: TCP sending rate: v roughly: send cwnd bytes, wait RTT for ACKS, then send more bytes rate ~ ~ cwnd RTT bytes/sec LastByteSent< cwnd LastByteAcked v cwnd is dynamic, function of perceived network congestion Transport Layer 3-100 TCP Slow Start when connection begins, increase rate exponentially until first loss event: § initially cwnd = 1 MSS § double cwnd every RTT § done by incrementing cwnd for every ACK received v summary: initial rate is slow but ramps up exponentially fast RTT v Host B Host A one segm ent two segm ents four segm ents time Transport Layer 3-101 TCP: detecting, reacting to loss v v v 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 3-102 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: v v variable ssthresh on loss event, ssthresh is set to 1/2 of cwnd just before loss event Transport Layer 3-103 Summary: TCP Congestion Control duplicate ACK dupACKcount++ Λ cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0 slow start timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment dupACKcount == 3 ssthresh= cwnd/2 cwnd = ssthresh + 3 retransmit missing segment New ACK! new ACK cwnd = cwnd+MSS dupACKcount = 0 transmit new segment(s), as allowed cwnd > ssthresh Λ timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment timeout ssthresh = cwnd/2 cwnd = 1 dupACKcount = 0 retransmit missing segment. New ACK! new ACK cwnd = cwnd + MSS (MSS/cwnd) dupACKcount = 0 transmit new segment(s), as allowed congestion avoidance duplicate ACK dupACKcount++ New ACK! New ACK cwnd = ssthresh dupACKcount = 0 fast recovery dupACKcount == 3 ssthresh= cwnd/2 cwnd = ssthresh + 3 retransmit missing segment duplicate ACK cwnd = cwnd + MSS transmit new segment(s), as allowed Transport Layer 3-104 TCP throughput v avg. TCP thruput as function of window size, RTT? § ignore slow start, assume always data to send v W: window size (measured in bytes) where loss occurs § avg. window size (# in-flight bytes) is ¾ W § avg. thruput is 3/4W per RTT avg TCP thruput = 3 W bytes/sec 4 RTT W W/2 Transport Layer 3-105 TCP Futures: TCP over “long, fat pipes” v v v 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]:. MSS 1.22 TCP throughput = RTT L ➜ to achieve 10 Gbps throughput, need a loss rate of L = 2·10-10 – a very small loss rate! v new versions of TCP for high-speed Transport Layer 3-106 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 3-107 Why is TCP fair? two competing sessions: v additive increase gives slope of 1, as throughout increases multiplicative decrease decreases throughput proportionally R Connection 2 throughput v 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 3-108 Fairness (more) Fairness and UDP v multimedia apps often do not use TCP v Fairness, parallel TCP connections v application can open § do not want rate multiple parallel throttled by congestion connections between two control hosts instead use UDP: v web browsers do this § send audio/video at v e.g., link of rate R with 9 constant rate, tolerate packet loss existing connections: § new app asks for 1 TCP, gets rate R/10 § new app asks for 11 TCPs, gets R/2 Transport Layer 3-109 Chapter 3: summary v v principles behind transport layer services: § multiplexing, demultiplexing § reliable data transfer § flow control § congestion control instantiation, implementation in the Internet next: v leaving the network “edge” (application , transport layers) v into the network “core” § UDP § TCP Transport Layer 3-110