Pipelined protocols: overview

Transcription

1 Pipelined protocols: overview Go-back-N: sender can have up to N unacked packets in pipeline receiver only sends cumulative ack doesn t ack packet if there s a gap sender has timer for oldest unacked packet when timer expires, retransmit all unacked packets Selective Repeat: sender can have up to N unack ed packets in pipeline rcvr sends individual ack for each packet sender maintains timer for each unacked packet when timer expires, retransmit only that unacked packet Transport Layer 3-1

2 Go-Back-N: sender k-bit seq # in pkt header window of up to N, consecutive unack ed pkts allowed ACK(n): ACKs all pkts up to, including seq # n - cumulative ACK may receive duplicate ACKs (see receiver) timer for oldest in-flight pkt timeout(n): retransmit packet n and all higher seq # pkts in window Transport Layer 3-2

3 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) Wait timeout start_timer udt_send(sndpkt[base]) udt_send(sndpkt[base+1]) udt_send(sndpkt[nextseqnum-1]) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) base = getacknum(rcvpkt)+1 If (base == nextseqnum) stop_timer else start_timer Transport Layer 3-3

4 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 (don t buffer): no receiver buffering! re-ack pkt with highest in-order seq # Transport Layer 3-4

5 Selective repeat receiver individually acknowledges all correctly received pkts buffers pkts, as needed, for eventual in-order delivery to upper layer sender only resends pkts for which ACK not received sender timer for each unacked pkt sender window N consecutive seq # s limits seq #s of sent, unacked pkts Transport Layer 3-5

6 Selective repeat: sender, receiver windows Transport Layer 3-6

7 Selective repeat sender data from above: if next available seq # in window, send pkt timeout(n): resend pkt n, restart timer ACK(n) in [sendbase,sendbase+n]: mark pkt n as received if n smallest unacked pkt, advance window base to next unacked seq # receiver pkt n in [rcvbase, rcvbase+n-1] send ACK(n) out-of-order: buffer in-order: deliver (also deliver buffered, in-order pkts), advance window to next not-yet-received pkt pkt n in [rcvbase-n,rcvbase-1] ACK(n) otherwise: ignore Transport Layer 3-7

8 TCP segment structure 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) 32 bits source port # dest port # head len sequence number acknowledgement number not used UAP R S F checksum receive window application data (variable length) Urg data pointer options (variable length) counting by bytes of data (not segments!) # bytes rcvr willing to accept Transport Layer 3-8

9 TCP seq. numbers, ACKs sequence numbers: byte stream number of first byte in segment s data acknowledgements: seq # of next byte expected from other side cumulative ACK Q: how receiver handles out-of-order segments A: TCP spec doesn t say, outgoing segment from sender source port # dest port # sequence number acknowledgement number rwnd checksum urg pointer sent ACKed window size N sender sequence number space sent, notyet ACKed ( inflight ) usable but not yet sent incoming segment to sender - up to implementor source port # dest port # sequence number not usable acknowledgement number A rwnd checksum urg pointer Transport Layer 3-9

10 TCP round trip time, timeout EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT exponential weighted moving average influence of past sample decreases exponentially fast typical value: = RTT: gaia.cs.umass.edu to fantasia.eurecom.fr 350 RTT: gaia.cs.umass.edu to fantasia.eurecom.fr RTT (milliseconds) RTT (milliseconds) samplertt EstimatedRTT time (seconnds) time (seconds) SampleRTT Estimated RTT Transport Layer 3-10

11 TCP round trip time, timeout timeout interval: EstimatedRTT plus safety margin large variation in EstimatedRTT -> larger safety margin estimate SampleRTT deviation from EstimatedRTT: DevRTT = (1-)*DevRTT + * SampleRTT-EstimatedRTT (typically, = 0.25) TimeoutInterval = EstimatedRTT + 4*DevRTT estimated RTT safety margin Transport Layer 3-11

12 TCP sender events: data rcvd from app: create segment with seq # seq # is byte-stream number of first data byte in segment start timer if not already running think of timer as for oldest unacked segment expiration interval: TimeOutInterval timeout: retransmit segment that caused timeout restart timer ack rcvd: if ack acknowledges previously unacked segments update what is known to be ACKed start timer if there are still unacked segments Transport Layer 3-12

13 TCP ACK generation [RFC 1122, RFC 2581] event at receiver arrival of in-order segment with expected seq #. All data up to expected seq # already ACKed arrival of in-order segment with expected seq #. One other segment has ACK pending arrival of out-of-order segment higher-than-expect seq. #. Gap detected arrival of segment that partially or completely fills gap TCP receiver action delayed ACK. Wait up to 500ms for next segment. If no next segment, send ACK immediately send single cumulative ACK, ACKing both in-order segments immediately send duplicate ACK, indicating seq. # of next expected byte immediate send ACK, provided that segment starts at lower end of gap Transport Layer 3-13

14 TCP fast retransmit time-out period often relatively long: long delay before resending lost packet detect lost segments via duplicate ACKs. sender often sends many segments 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 duplicate ACKs ), resend unacked segment with smallest seq # likely that unacked segment lost, so don t wait for timeout Transport Layer 3-14

15 TCP flow control application may remove data from TCP socket buffers. slower than TCP receiver is delivering (sender is sending) application process TCP socket receiver buffers TCP code application OS flow control receiver controls sender, so sender won t overflow receiver s buffer by transmitting too much, too fast from sender IP code receiver protocol stack Transport Layer 3-15

16 TCP flow control receiver advertises free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via socket options (typical default is 4096 bytes) many operating systems autoadjust RcvBuffer sender limits amount of unacked ( in-flight ) data to receiver s rwnd value guarantees receive buffer will not overflow RcvBuffer rwnd to application process buffered data free buffer space TCP segment payloads receiver-side buffering Transport Layer 3-16

17 Connection Management before exchanging data, sender/receiver handshake : agree to establish connection (each knowing the other willing to establish connection) agree on connection parameters application connection state: ESTAB connection variables: seq # client-to-server server-to-client rcvbuffer size at server,client network application connection state: ESTAB connection Variables: seq # client-to-server server-to-client rcvbuffer size at server,client network Socket clientsocket = newsocket("hostname","port number"); Socket connectionsocket = welcomesocket.accept(); Transport Layer 3-17

18 TCP 3-way handshake client state CLOSED SYNSENT ESTAB choose init seq num, x send TCP SYN msg received SYNACK(x) indicates server is live; send ACK for SYNACK; this segment may contain client-to-server data SYNbit=1, Seq=x SYNbit=1, Seq=y ACKbit=1; ACKnum=x+1 ACKbit=1, ACKnum=y+1 choose init seq num, y send TCP SYNACK msg, acking SYN received ACK(y) indicates client is live server state LISTEN SYN RCVD ESTAB Transport Layer 3-18

19 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 wait for server close FINbit=1, seq=x ACKbit=1; ACKnum=x+1 can still send data CLOSE_WAIT TIMED_WAIT timed wait for 2*max segment lifetime FINbit=1, seq=y ACKbit=1; ACKnum=y+1 can no longer send data LAST_ACK CLOSED CLOSED Transport Layer 3-19

20 Causes/costs of congestion: scenario 1 two senders, two receivers one router, infinite buffers output link capacity: R no retransmission original data: in Host A unlimited shared output link buffers throughput: out Host B R/2 out delay in R/2 in R/2 maximum per-connection throughput: R/2 large delays as arrival rate, in, approaches capacity Transport Layer 3-20

21 Causes/costs of congestion: scenario 2 one router, finite buffers sender retransmission of timed-out packet application-layer input = application-layer output: in = out transport-layer input includes retransmissions : in in in : original data ' in : original data, plus retransmitted data out Host A Host B finite shared output link buffers Transport Layer 3-21

22 Causes/costs of congestion: scenario 2 idealization: perfect knowledge sender sends only when router buffers available R/2 out in R/2 copy in : original data ' in : original data, plus retransmitted data out A free buffer space! Host B finite shared output link buffers Transport Layer 3-22

23 Causes/costs of congestion: scenario 2 Idealization: known loss packets can be lost, dropped at router due to full buffers sender only resends if packet known to be lost copy in : original data ' in : original data, plus retransmitted data out A no buffer space! Host B Transport Layer 3-23

24 Causes/costs of congestion: scenario 2 Idealization: known loss packets can be lost, dropped at router due to full buffers sender only resends if packet known to be lost R/2 out in R/2 when sending at R/2, some packets are retransmissions but asymptotic goodput is still R/2 in : original data ' in : original data, plus retransmitted data out A free buffer space! Host B Transport Layer 3-24

25 Causes/costs of congestion: scenario 2 Realistic: duplicates packets can be lost, dropped at router due to full buffers sender times out prematurely, sending two copies, both of which are delivered R/2 out in R/2 when sending at R/2, some packets are retransmissions including duplicated that are delivered! copy timeout in ' in out A free buffer space! Host B Transport Layer 3-25

26 Causes/costs of congestion: scenario 3 four senders multihop paths timeout/retransmit Host A Q: what happens as in and in increase? A: as red in increases, all arriving blue pkts at upper queue are dropped, blue throughput 0 in : original data ' in : original data, plus retransmitted data finite shared output link buffers out Host B Host D Host C Transport Layer 3-26

27 Approaches towards congestion control two broad approaches towards congestion control: end-end congestion control: no explicit feedback from network congestion inferred from end-system observed loss, delay approach taken by TCP network-assisted congestion control: routers provide feedback to end systems single bit indicating congestion (SNA, DECbit, TCP/IP ECN, ATM) explicit rate for sender to send at Transport Layer 3-27

28 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 additively increase window size. until loss occurs (then cut window in half) time Transport Layer 3-28

29 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 summary: initial rate is slow but ramps up exponentially fast Host A RTT Host B time Transport Layer 3-29

30 TCP: detecting, reacting to loss loss indicated by timeout: cwnd set to 1 MSS; window then grows exponentially (as in slow start) to threshold, then grows linearly loss indicated by 3 duplicate ACKs: TCP RENO dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks) Transport Layer 3-30

31 TCP: switching from slow start to CA Q: when should the exponential increase switch to linear? A: when cwnd gets to 1/2 of its value before timeout. Implementation: variable ssthresh on loss event, ssthresh is set to 1/2 of cwnd just before loss event Transport Layer 3-31

32 Network layer transport segment from sending to receiving host on sending side encapsulates segments into datagrams on receiving side, delivers segments to transport layer network layer protocols in every host, router router examines header fields in all IP datagrams passing through it 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 network data link physical network data link physical network data link physical network data link physical application transport network data link physical Network Layer 4-32

33 Two key network-layer functions forwarding: move packets from router s input to appropriate router output routing: determine route taken by packets from source to dest. routing algorithms analogy: routing: process of planning trip from source to dest forwarding: process of getting through single interchange Network Layer 4-33

34 Network service model Q: What service model for channel transporting datagrams from sender to receiver? example services for individual datagrams: guaranteed delivery guaranteed delivery with less than 40 msec delay example services for a flow of datagrams: in-order datagram delivery guaranteed minimum bandwidth to flow restrictions on changes in inter-packet spacing Network Layer 4-34

35 Virtual circuits source-to-dest path behaves much like telephone circuit performance-wise network actions along source-to-dest path call setup, teardown for each call before data can flow each packet carries VC identifier (not destination host address) every router on source-dest path maintains state for each passing connection link, router resources (bandwidth, buffers) may be allocated to VC (dedicated resources = predictable service) Network Layer 4-35

36 Datagram networks no call setup at network layer routers: no state about end-to-end connections no network-level concept of connection packets forwarded using destination host address application transport network data link physical 1. send datagrams 2. receive datagrams application transport network data link physical Network Layer 4-36

37 Datagram forwarding table routing algorithm local forwarding table dest address output link address-range 1 address-range 2 address-range 3 address-range billion IP addresses, so rather than list individual destination address list range of addresses (aggregate table entries) IP destination address in arriving packet s header Network Layer 4-37

38 Longest prefix matching longest prefix matching when looking for forwarding table entry for given destination address, use longest address prefix that matches destination address. Destination Address Range *** ********* ********* *** ********* otherwise examples: Link interface DA: DA: which interface? which interface? Network Layer 4-38

39 Datagram or VC network: why? Internet (datagram) data exchange among computers elastic service, no strict timing req. many link types different characteristics uniform service difficult smart end systems (computers) can adapt, perform control, error recovery simple inside network, complexity at edge ATM (VC) evolved from telephony human conversation: strict timing, reliability requirements need for guaranteed service dumb end systems telephones complexity inside network Network Layer 4-39

40 Router architecture overview two key router functions: run routing algorithms/protocol (RIP, OSPF, BGP) forwarding datagrams from incoming to outgoing link forwarding tables computed, pushed to input ports routing processor routing, management control plane (software) forwarding data plane (hardware) high-speed switching fabric router input ports router output ports Network Layer 4-40

41 Input port functions line termination link layer protocol (receive) lookup, forwarding queueing switch fabric physical layer: bit-level reception data link layer: e.g., Ethernet see chapter 5 decentralized switching: given datagram dest., lookup output port using forwarding table in input port memory ( match plus action ) goal: complete input port processing at line speed queuing: if datagrams arrive faster than forwarding rate into switch fabric Network Layer 4-41

42 Switching fabrics transfer packet from input buffer to appropriate output buffer switching rate: rate at which packets can be transfer from inputs to outputs often measured as multiple of input/output line rate N inputs: switching rate N times line rate desirable three types of switching fabrics memory memory bus crossbar Network Layer 4-42

43 Output ports switch fabric datagram buffer queueing link layer protocol (send) line termination buffering required when datagrams arrive from fabric faster than the transmission rate scheduling discipline chooses among queued datagrams for transmission Network Layer 4-43

44 Output port queueing switch fabric switch fabric at t, packets more from input to output one packet time later buffering when arrival rate via switch exceeds output line speed queueing (delay) and loss due to output port buffer overflow! Network Layer 4-44

45 Input port queuing fabric slower than input ports combined -> queueing may occur at input queues queueing delay and loss due to input buffer overflow! Head-of-the-Line (HOL) blocking: queued datagram at front of queue prevents others in queue from moving forward switch fabric switch fabric output port contention: only one red datagram can be transferred. lower red packet is blocked one packet time later: green packet experiences HOL blocking Network Layer 4-45

46 IP datagram format IP protocol version number header length (# of 32-bit words) type of data max number remaining hops (decremented at each router) upper layer protocol to deliver payload to how much overhead? 20 bytes of TCP 20 bytes of IP = 40 bytes + app layer overhead ver head. len 16-bit identifier time to live type of service upper layer 32 bits flgs length header checksum 32 bit source IP address data (variable length, typically a TCP or UDP segment) fragment offset 32 bit destination IP address options (if any) total datagra length (bytes for fragmentatio reassembly e.g. timestam record route taken, specify list of routers to visit. Network Layer 4-46

47 IP fragmentation, reassembly network links have MTU (max. transfer size) - largest possible link-level frame different link types, different MTUs large IP datagram divided ( fragmented ) within net one datagram becomes several datagrams reassembled only at final destination IP header bits used to identify, order related fragments reassembly fragmentation: in: one large datagram out: 3 smaller datagrams Network Layer 4-47

48 IP fragmentation, reassembly example: 4000 byte datagram MTU = 1500 bytes 1480 bytes in data field offset = 1480/8 lengthidfragflagoffset =4000=x =0 =0 one large datagram becomes several smaller datagrams lengthidfragflag =1500=x =1 lengthidfragflag =1500=x =1 lengthidfragflag =1040=x =0 offset =0 offset =185 offset =370 Network Layer 4-48

49 IP addressing: introduction IP address: 32-bit identifier for host, router interface interface: connection between host/router and physical link router s typically have multiple interfaces host typically has one or two interfaces (e.g., wired Ethernet, wireless ) IP addresses associated with each interface = Network Layer 4-49

50 Subnets IP address: subnet part - high order bits host part - low order bits what s a subnet? device interfaces with same subnet part of IP address can physically reach each other without intervening router subnet network consisting of 3 subnets Network Layer 4-50

51 IP addressing: CIDR CIDR: Classless InterDomain Routing subnet portion of address of arbitrary length address format: a.b.c.d/x, where x is # bits in subnet portion of address subnet host part part /23 Network Layer 4-51

52 DHCP: Dynamic Host Configuration Protocol goal: allow host to dynamically obtain its IP address from network server when it joins network can renew its lease on address in use allows reuse of addresses (only hold address while connected/ on ) support for mobile users who want to join network (more shortly) DHCP overview: host broadcasts DHCP discover msg [optional] DHCP server responds with DHCP offer msg [optional] host requests IP address: DHCP request msg DHCP server sends address: DHCP ack msg Network Layer 4-52