1. Transport Layer 3-1 Chapter 3 Transport Layer Computer Networking: A Top Down Approach 4 th edition. Jim Kurose, Keith Ross Addison-Wesley, July 2007. 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 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:  If you use these slides (e.g., in a class) in substantially unaltered form, that you mention their source (after all, we’d like people to use our book!)  If you post any slides in substantially unaltered form 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. Thanks and enjoy! JFK/KWR All material copyright 1996-2007 J.F Kurose and K.W. Ross, All Rights Reserved

2. Transport Layer 3-2 Chapter 3: Transport Layer Our goals: r understand principles behind transport layer services: m multiplexing/demultipl exing m reliable data transfer m flow control m congestion control r learn about transport layer protocols in the Internet: m UDP: connectionless transport m TCP: connection-oriented transport m TCP congestion control

3. Transport Layer 3-3 Chapter 3 outline r 3.1 Transport-layer services r 3.2 Multiplexing and demultiplexing r 3.3 Connectionless transport: UDP r 3.4 Principles of reliable data transfer r 3.5 Connection-oriented transport: TCP m segment structure m reliable data transfer m flow control m connection management r 3.6 Principles of congestion control r 3.7 TCP congestion control

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

5. Transport Layer 3-5 Transport vs. network layer r network layer: logical communication between hosts r transport layer: logical communication between processes m relies on, enhances, network layer services Household analogy: 12 kids sending letters to 12 kids r processes = kids r app messages = letters in envelopes r hosts = houses r transport protocol = Ann and Bill r network-layer protocol = postal service

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

7. Transport Layer 3-7 Chapter 3 outline r 3.1 Transport-layer services r 3.2 Multiplexing and demultiplexing r 3.3 Connectionless transport: UDP r 3.4 Principles of reliable data transfer r 3.5 Connection-oriented transport: TCP m segment structure m reliable data transfer m flow control m connection management r 3.6 Principles of congestion control r 3.7 TCP congestion control

8. Transport Layer 3-8 Multiplexing/demultiplexing application transport network link physical P1 application transport network link physical application transport network link physical P2 P3 P4 P1 host 1 host 2 host 3 = process= socket delivering received segments to correct socket Demultiplexing at rcv host: gathering data from multiple sockets, enveloping data with header (later used for demultiplexing) Multiplexing at send host:

9. Transport Layer 3-9 How demultiplexing works r host receives IP datagrams m each datagram has source IP address, destination IP address m each datagram carries 1 transport-layer segment m each segment has source, destination port number r host uses IP addresses & port numbers to direct segment to appropriate socket source port #dest port # 32 bits application data (message) other header fields TCP/UDP segment format

10. Transport Layer 3-10 Connectionless demultiplexing r Create sockets with port numbers: DatagramSocket mySocket1 = new DatagramSocket(12534); DatagramSocket mySocket2 = new DatagramSocket(12535); r UDP socket identified by two-tuple: ( dest IP address, dest port number) r When host receives UDP segment: m checks destination port number in segment m directs UDP segment to socket with that port number r IP datagrams with different source IP addresses and/or source port numbers directed to same socket

11. Transport Layer 3-11 Connectionless demux (cont) DatagramSocket serverSocket = new DatagramSocket(6428); Client IP:B P2 client IP: A P1 P3 server IP: C SP: 6428 DP: 9157 SP: 9157 DP: 6428 SP: 6428 DP: 5775 SP: 5775 DP: 6428 SP provides “return address”

12. Transport Layer 3-12 Connection-oriented demux r TCP socket identified by 4-tuple: m source IP address m source port number m dest IP address m dest port number r recv host uses all four values to direct segment to appropriate socket r Server host may support many simultaneous TCP sockets: m each socket identified by its own 4-tuple r Web servers have different sockets for each connecting client m non-persistent HTTP will have different socket for each request

13. Transport Layer 3-13 Connection-oriented demux (cont) Client IP:B P1 client IP: A P1P2P4 server IP: C SP: 9157 DP: 80 SP: 9157 DP: 80 P5P6P3 D-IP:C S-IP: A D-IP:C S-IP: B SP: 5775 DP: 80 D-IP:C S-IP: B

14. Transport Layer 3-14 Connection-oriented demux: Threaded Web Server Client IP:B P1 client IP: A P1P2 server IP: C SP: 9157 DP: 80 SP: 9157 DP: 80 P4 P3 D-IP:C S-IP: A D-IP:C S-IP: B SP: 5775 DP: 80 D-IP:C S-IP: B

15. Transport Layer 3-15 Chapter 3 outline r 3.1 Transport-layer services r 3.2 Multiplexing and demultiplexing r 3.3 Connectionless transport: UDP r 3.4 Principles of reliable data transfer r 3.5 Connection-oriented transport: TCP m segment structure m reliable data transfer m flow control m connection management r 3.6 Principles of congestion control r 3.7 TCP congestion control

16. Transport Layer 3-16 UDP: User Datagram Protocol [RFC 768] r “no frills,” “bare bones” Internet transport protocol r “best effort” service, UDP segments may be: m lost m delivered out of order to app r connectionless: m no handshaking between UDP sender, receiver m each UDP segment handled independently of others Why is there a UDP? r no connection establishment (which can add delay) r simple: no connection state at sender, receiver r small segment header r no congestion control: UDP can blast away as fast as desired

17. Transport Layer 3-17 UDP: more r often used for streaming multimedia apps m loss tolerant m rate sensitive r other UDP uses m DNS m SNMP r reliable transfer over UDP: add reliability at application layer m application-specific error recovery! source port #dest port # 32 bits Application data (message) UDP segment format length checksum Length, in bytes of UDP segment, including header

18. Transport Layer 3-18 UDP checksum Sender: r treat segment contents as sequence of 16-bit integers r checksum: addition (1’s complement sum) of segment contents r sender puts checksum value into UDP checksum field Receiver: r compute checksum of received segment r check if computed checksum equals checksum field value: m NO - error detected m YES - no error detected. But maybe errors nonetheless? More later …. Goal: detect “errors” (e.g., flipped bits) in transmitted segment

19. Transport Layer 3-19 Internet Checksum Example r Note m When adding numbers, a carryout from the most significant bit needs to be added to the result r Example: add two 16-bit integers 1 1 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 1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0 1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1 wraparound sum checksum

20. Transport Layer 3-20 Chapter 3 outline r 3.1 Transport-layer services r 3.2 Multiplexing and demultiplexing r 3.3 Connectionless transport: UDP r 3.4 Principles of reliable data transfer r 3.5 Connection-oriented transport: TCP m segment structure m reliable data transfer m flow control m connection management r 3.6 Principles of congestion control r 3.7 TCP congestion control

21. Transport Layer 3-21 Principles of Reliable data transfer r important in app., transport, link layers r top-10 list of important networking topics! r characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

22. Transport Layer 3-22 Principles of Reliable data transfer r important in app., transport, link layers r top-10 list of important networking topics! r characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

23. Transport Layer 3-23 Principles of Reliable data transfer r important in app., transport, link layers r top-10 list of important networking topics! r characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

24. Transport Layer 3-24 Reliable data transfer: getting started send side receive side rdt_send(): called from above, (e.g., by app.). Passed data to deliver to receiver upper layer udt_send(): called by rdt, to transfer packet over unreliable channel to receiver rdt_rcv(): called when packet arrives on rcv-side of channel deliver_data(): called by rdt to deliver data to upper

25. Transport Layer 3-25 Reliable data transfer: getting started We’ll: r incrementally develop sender, receiver sides of reliable data transfer protocol (rdt) r consider only unidirectional data transfer m but control info will flow on both directions! r use finite state machines (FSM) to specify sender, receiver state 1 state 2 event causing state transition actions taken on state transition state: when in this “state” next state uniquely determined by next event event actions

26. Transport Layer 3-26 Rdt1.0: reliable transfer over a reliable channel r underlying channel perfectly reliable m no bit errors m no loss of packets r separate FSMs for sender, receiver: m sender sends data into underlying channel m receiver read data from underlying channel Wait for call from above packet = make_pkt(data) udt_send(packet) rdt_send(data) extract (packet,data) deliver_data(data) Wait for call from below rdt_rcv(packet) sender receiver

27. Transport Layer 3-27 Rdt2.0: channel with bit errors r underlying channel may flip bits in packet m checksum to detect bit errors r the question: how to recover from errors: m acknowledgements (ACKs): receiver explicitly tells sender that pkt received OK m negative acknowledgements (NAKs): receiver explicitly tells sender that pkt had errors m sender retransmits pkt on receipt of NAK  new mechanisms in rdt2.0 (beyond rdt1.0 ): m error detection m receiver feedback: control msgs (ACK,NAK) rcvr->sender

28. Transport Layer 3-28 rdt2.0: FSM specification Wait for call from above snkpkt = make_pkt(data, checksum) udt_send(sndpkt) extract(rcvpkt,data) deliver_data(data) udt_send(ACK) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) rdt_rcv(rcvpkt) && isACK(rcvpkt) udt_send(sndpkt) rdt_rcv(rcvpkt) && isNAK(rcvpkt) udt_send(NAK) rdt_rcv(rcvpkt) && corrupt(rcvpkt) Wait for ACK or NAK Wait for call from below sender receiver rdt_send(data) 

29. Transport Layer 3-29 rdt2.0: operation with no errors Wait for call from above snkpkt = make_pkt(data, checksum) udt_send(sndpkt) extract(rcvpkt,data) deliver_data(data) udt_send(ACK) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) rdt_rcv(rcvpkt) && isACK(rcvpkt) udt_send(sndpkt) rdt_rcv(rcvpkt) && isNAK(rcvpkt) udt_send(NAK) rdt_rcv(rcvpkt) && corrupt(rcvpkt) Wait for ACK or NAK Wait for call from below rdt_send(data) 

30. Transport Layer 3-30 rdt2.0: error scenario Wait for call from above snkpkt = make_pkt(data, checksum) udt_send(sndpkt) extract(rcvpkt,data) deliver_data(data) udt_send(ACK) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) rdt_rcv(rcvpkt) && isACK(rcvpkt) udt_send(sndpkt) rdt_rcv(rcvpkt) && isNAK(rcvpkt) udt_send(NAK) rdt_rcv(rcvpkt) && corrupt(rcvpkt) Wait for ACK or NAK Wait for call from below rdt_send(data) 

31. Transport Layer 3-31 rdt2.0 has a fatal flaw! What happens if ACK/NAK corrupted? r sender doesn’t know what happened at receiver! r can’t just retransmit: possible duplicate Handling duplicates: r sender retransmits current pkt if ACK/NAK garbled r sender adds sequence number to each pkt r receiver discards (doesn’t deliver up) duplicate pkt Sender sends one packet, then waits for receiver response stop and wait

32. Transport Layer 3-32 rdt2.1: sender, handles garbled ACK/NAKs Wait for call 0 from above sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_send(data) Wait for ACK or NAK 0 udt_send(sndpkt) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) ) sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) rdt_send(data) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt) udt_send(sndpkt) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) ) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt) Wait for call 1 from above Wait for ACK or NAK 1  

33. Transport Layer 3-33 rdt2.1: receiver, handles garbled ACK/NAKs Wait for 0 from below sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq0(rcvpkt) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) Wait for 1 from below rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq0(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && (corrupt(rcvpkt) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq1(rcvpkt) rdt_rcv(rcvpkt) && (corrupt(rcvpkt) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt)

34. Transport Layer 3-34 rdt2.1: discussion Sender: r seq # added to pkt r two seq. #’s (0,1) will suffice. Why? r must check if received ACK/NAK corrupted r twice as many states m state must “remember” whether “current” pkt has 0 or 1 seq. # Receiver: r must check if received packet is duplicate m state indicates whether 0 or 1 is expected pkt seq # r note: receiver can not know if its last ACK/NAK received OK at sender

35. Transport Layer 3-35 rdt2.2: a NAK-free protocol r same functionality as rdt2.1, using ACKs only r instead of NAK, receiver sends ACK for last pkt received OK m receiver must explicitly include seq # of pkt being ACKed r duplicate ACK at sender results in same action as NAK: retransmit current pkt

36. Transport Layer 3-36 rdt2.2: sender, receiver fragments Wait for call 0 from above sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_send(data) udt_send(sndpkt) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) ) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) Wait for ACK 0 sender FSM fragment Wait for 0 from below rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK1, chksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && (corrupt(rcvpkt) || has_seq1(rcvpkt)) udt_send(sndpkt) receiver FSM fragment 

37. Transport Layer 3-37 rdt3.0: channels with errors and loss New assumption: underlying channel can also lose packets (data or ACKs) m checksum, seq. #, ACKs, retransmissions will be of help, but not enough Approach: sender waits “reasonable” amount of time for ACK r retransmits if no ACK received in this time r if pkt (or ACK) just delayed (not lost): m retransmission will be duplicate, but use of seq. #’s already handles this m receiver must specify seq # of pkt being ACKed r requires countdown timer

38. Transport Layer 3-38 rdt3.0 sender sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) start_timer rdt_send(data) Wait for ACK0 rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) ) Wait for call 1 from above sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) start_timer rdt_send(data) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,0) ) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,1) stop_timer udt_send(sndpkt) start_timer timeout udt_send(sndpkt) start_timer timeout rdt_rcv(rcvpkt) Wait for call 0from above Wait for ACK1  rdt_rcv(rcvpkt)   

39. Transport Layer 3-39 rdt3.0 in action

40. Transport Layer 3-40 rdt3.0 in action

41. Transport Layer 3-41 Performance of rdt3.0 r rdt3.0 works, but performance stinks r ex: 1 Gbps link, 15 ms prop. delay, 8000 bit packet: m U sender : utilization – fraction of time sender busy sending m 1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link m network protocol limits use of physical resources!

42. Transport Layer 3-42 rdt3.0: stop-and-wait operation first packet bit transmitted, t = 0 senderreceiver RTT last packet bit transmitted, t = L / R first packet bit arrives last packet bit arrives, send ACK ACK arrives, send next packet, t = RTT + L / R

43. Transport Layer 3-43 Pipelined protocols Pipelining: sender allows multiple, “in-flight”, yet-to- be-acknowledged pkts m range of sequence numbers must be increased m buffering at sender and/or receiver r Two generic forms of pipelined protocols: go-Back-N, selective repeat

44. Transport Layer 3-44 Pipelining: increased utilization first packet bit transmitted, t = 0 senderreceiver RTT last bit transmitted, t = L / R first packet bit arrives last packet bit arrives, send ACK ACK arrives, send next packet, t = RTT + L / R last bit of 2 nd packet arrives, send ACK last bit of 3 rd packet arrives, send ACK Increase utilization by a factor of 3!

45. Transport Layer 3-45 Pipelining Protocols Go-back-N: big picture: r Sender can have up to N unacked packets in pipeline r Rcvr only sends cumulative acks m Doesn’t ack packet if there’s a gap r Sender has timer for oldest unacked packet m If timer expires, retransmit all unacked packets Selective Repeat: big pic r Sender can have up to N unacked packets in pipeline r Rcvr acks individual packets r Sender maintains timer for each unacked packet m When timer expires, retransmit only unack packet

46. Transport Layer 3-46 Selective repeat: big picture r Sender can have up to N unacked packets in pipeline r Rcvr acks individual packets r Sender maintains timer for each unacked packet m When timer expires, retransmit only unack packet

47. Transport Layer 3-47 Go-Back-N Sender: r k-bit seq # in pkt header r “window” of up to N, consecutive unack’ed pkts allowed r ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK” m may receive duplicate ACKs (see receiver) r timer for each in-flight pkt r timeout(n): retransmit pkt n and all higher seq # pkts in window

48. Transport Layer 3-48 GBN: sender extended FSM Wait start_timer udt_send(sndpkt[base]) udt_send(sndpkt[base+1]) … udt_send(sndpkt[nextseqnum-1]) timeout rdt_send(data) 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) base = getacknum(rcvpkt)+1 If (base == nextseqnum) stop_timer else start_timer rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) base=1 nextseqnum=1 rdt_rcv(rcvpkt) && corrupt(rcvpkt) 

49. Transport Layer 3-49 GBN: receiver extended FSM ACK-only: always send ACK for correctly-received pkt with highest in-order seq # m may generate duplicate ACKs  need only remember expectedseqnum r out-of-order pkt: m discard (don’t buffer) -> no receiver buffering! m Re-ACK pkt with highest in-order seq # Wait udt_send(sndpkt) default rdt_rcv(rcvpkt) && notcurrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpkt) expectedseqnum++ expectedseqnum=1 sndpkt = make_pkt(expectedseqnum,ACK,chksum) 

50. Transport Layer 3-50 GBN in action

51. Transport Layer 3-51 Selective Repeat r receiver individually acknowledges all correctly received pkts m buffers pkts, as needed, for eventual in-order delivery to upper layer r sender only resends pkts for which ACK not received m sender timer for each unACKed pkt r sender window m N consecutive seq #’s m again limits seq #s of sent, unACKed pkts

52. Transport Layer 3-52 Selective repeat: sender, receiver windows

53. Transport Layer 3-53 Selective repeat data from above : r if next available seq # in window, send pkt timeout(n): r resend pkt n, restart timer ACK(n) in [sendbase,sendbase+N]: r mark pkt n as received r if n smallest unACKed pkt, advance window base to next unACKed seq # sender pkt n in [rcvbase, rcvbase+N-1] r send ACK(n) r out-of-order: buffer r in-order: deliver (also deliver buffered, in-order pkts), advance window to next not-yet-received pkt pkt n in [rcvbase-N,rcvbase-1] r ACK(n) otherwise: r ignore receiver

54. Transport Layer 3-54 Selective repeat in action

55. Transport Layer 3-55 Chapter 3 outline r 3.1 Transport-layer services r 3.2 Multiplexing and demultiplexing r 3.3 Connectionless transport: UDP r 3.4 Principles of reliable data transfer r 3.5 Connection-oriented transport: TCP m segment structure m reliable data transfer m flow control m connection management r 3.6 Principles of congestion control r 3.7 TCP congestion control

56. Transport Layer 3-56 TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581 r full duplex data: m bi-directional data flow in same connection m MSS: maximum segment size r connection-oriented: m handshaking (exchange of control msgs) init’s sender, receiver state before data exchange r flow controlled: m sender will not overwhelm receiver r point-to-point: m one sender, one receiver r reliable, in-order byte steam: m no “message boundaries” r pipelined: m TCP congestion and flow control set window size r send & receive buffers

57. Transport Layer 3-57 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number Receive window Urg data pnter checksum F SR PAU head len not used Options (variable length) URG: urgent data (generally not used) ACK: ACK # valid PSH: push data now (generally not used) RST, SYN, FIN: connection estab (setup, teardown commands) # bytes rcvr willing to accept counting by bytes of data (not segments!) Internet checksum (as in UDP)

58. Transport Layer 3-58 TCP seq. #’s and ACKs Seq. #’s: m byte stream “number” of first byte in segment’s data ACKs: m seq # of next byte expected from other side m cumulative ACK Q: how receiver handles out-of-order segments m A: TCP spec doesn’t say, - up to implementor Host A Host B Seq=42, ACK=79, data = ‘C’ Seq=79, ACK=43, data = ‘C’ Seq=43, ACK=80 User types ‘C’ host ACKs receipt of echoed ‘C’ host ACKs receipt of ‘C’, echoes back ‘C’ time simple telnet scenario

59. Transport Layer 3-59 TCP Round Trip Time and Timeout Q: how to set TCP timeout value? r longer than RTT m but RTT varies r too short: premature timeout m unnecessary retransmissions r too long: slow reaction to segment loss Q: how to estimate RTT?  SampleRTT : measured time from segment transmission until ACK receipt m ignore retransmissions  SampleRTT will vary, want estimated RTT “smoother”  average several recent measurements, not just current SampleRTT

60. Transport Layer 3-60 TCP Round Trip Time and Timeout EstimatedRTT = (1-  )*EstimatedRTT +  *SampleRTT r Exponential weighted moving average r influence of past sample decreases exponentially fast  typical value:  = 0.125

61. Transport Layer 3-61 Example RTT estimation:

62. Transport Layer 3-62 TCP Round Trip Time and Timeout Setting the timeout  EstimtedRTT plus “safety margin”  large variation in EstimatedRTT -> larger safety margin r first estimate of how much SampleRTT deviates from EstimatedRTT: TimeoutInterval = EstimatedRTT + 4*DevRTT DevRTT = (1-  )*DevRTT +  *|SampleRTT-EstimatedRTT| (typically,  = 0.25) Then set timeout interval:

63. Transport Layer 3-63 Chapter 3 outline r 3.1 Transport-layer services r 3.2 Multiplexing and demultiplexing r 3.3 Connectionless transport: UDP r 3.4 Principles of reliable data transfer r 3.5 Connection-oriented transport: TCP m segment structure m reliable data transfer m flow control m connection management r 3.6 Principles of congestion control r 3.7 TCP congestion control

64. Transport Layer 3-64 TCP reliable data transfer r TCP creates rdt service on top of IP’s unreliable service r Pipelined segments r Cumulative acks r TCP uses single retransmission timer r Retransmissions are triggered by: m timeout events m duplicate acks r Initially consider simplified TCP sender: m ignore duplicate acks m ignore flow control, congestion control

65. Transport Layer 3-65 TCP sender events: data rcvd from app: r Create segment with seq # r seq # is byte-stream number of first data byte in segment r start timer if not already running (think of timer as for oldest unacked segment)  expiration interval: TimeOutInterval timeout: r retransmit segment that caused timeout r restart timer Ack rcvd: r If acknowledges previously unacked segments m update what is known to be acked m start timer if there are outstanding segments

66. Transport Layer 3-66 TCP sender (simplified) 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 */ Comment: SendBase-1: last cumulatively ack’ed byte Example: SendBase-1 = 71; y= 73, so the rcvr wants 73+ ; y > SendBase, so that new data is acked

67. Transport Layer 3-67 TCP: retransmission scenarios Host A Seq=100, 20 bytes data ACK=100 time premature timeout Host B Seq=92, 8 bytes data ACK=120 Seq=92, 8 bytes data Seq=92 timeout ACK=120 Host A Seq=92, 8 bytes data ACK=100 loss timeout lost ACK scenario Host B X Seq=92, 8 bytes data ACK=100 time Seq=92 timeout SendBase = 100 SendBase = 120 SendBase = 120 Sendbase = 100

68. Transport Layer 3-68 TCP retransmission scenarios (more) Host A Seq=92, 8 bytes data ACK=100 loss timeout Cumulative ACK scenario Host B X Seq=100, 20 bytes data ACK=120 time SendBase = 120

69. Transport Layer 3-69 Chapter 3 outline r 3.1 Transport-layer services r 3.2 Multiplexing and demultiplexing r 3.3 Connectionless transport: UDP r 3.4 Principles of reliable data transfer r 3.5 Connection-oriented transport: TCP m segment structure m reliable data transfer m flow control m connection management r 3.6 Principles of congestion control r 3.7 TCP congestion control

70. Transport Layer 3-70 TCP Flow Control r receive side of TCP connection has a receive buffer: r speed-matching service: matching the send rate to the receiving app’s drain rate r app process may be slow at reading from buffer sender won’t overflow receiver’s buffer by transmitting too much, too fast flow control

71. Transport Layer 3-71 TCP Flow control: how it works (Suppose TCP receiver discards out-of-order segments)  spare room in buffer = RcvWindow = RcvBuffer-[LastByteRcvd - LastByteRead]  Rcvr advertises spare room by including value of RcvWindow in segments  Sender limits unACKed data to RcvWindow m guarantees receive buffer doesn’t overflow

72. Transport Layer 3-72 Chapter 3 outline r 3.1 Transport-layer services r 3.2 Multiplexing and demultiplexing r 3.3 Connectionless transport: UDP r 3.4 Principles of reliable data transfer r 3.5 Connection-oriented transport: TCP m segment structure m reliable data transfer m flow control m connection management r 3.6 Principles of congestion control r 3.7 TCP congestion control

73. Transport Layer 3-73 TCP Connection Management Recall: TCP sender, receiver establish “connection” before exchanging data segments r initialize TCP variables: m seq. #s  buffers, flow control info (e.g. RcvWindow ) r client: connection initiator Socket clientSocket = new Socket("hostname","port number"); r server: contacted by client Socket connectionSocket = welcomeSocket.accept(); Three way handshake: Step 1: client host sends TCP SYN segment to server m specifies initial seq # m no data Step 2: server host receives SYN, replies with SYNACK segment m server allocates buffers m specifies server initial seq. # Step 3: client receives SYNACK, replies with ACK segment, which may contain data

74. Transport Layer 3-74 TCP Connection Management (cont.) Closing a connection: client closes socket: clientSocket.close(); Step 1: client end system sends TCP FIN control segment to server Step 2: server receives FIN, replies with ACK. Closes connection, sends FIN. client FIN server ACK FIN close closed timed wait

75. Transport Layer 3-75 TCP Connection Management (cont.) Step 3: client receives FIN, replies with ACK. m Enters “timed wait” - will respond with ACK to received FINs Step 4: server, receives ACK. Connection closed. Note: with small modification, can handle simultaneous FINs. client FIN server ACK FIN closing closed timed wait closed

76. Transport Layer 3-76 TCP Connection Management (cont) TCP client lifecycle TCP server lifecycle

77. Transport Layer 3-77 Chapter 3 outline r 3.1 Transport-layer services r 3.2 Multiplexing and demultiplexing r 3.3 Connectionless transport: UDP r 3.4 Principles of reliable data transfer r 3.5 Connection-oriented transport: TCP m segment structure m reliable data transfer m flow control m connection management r 3.6 Principles of congestion control r 3.7 TCP congestion control

78. Transport Layer 3-78 Principles of Congestion Control Congestion: r informally: “too many sources sending too much data too fast for network to handle” r different from flow control! r manifestations: m lost packets (buffer overflow at routers) m long delays (queueing in router buffers) r a top-10 problem!

79. Transport Layer 3-79 Chapter 3 outline r 3.1 Transport-layer services r 3.2 Multiplexing and demultiplexing r 3.3 Connectionless transport: UDP r 3.4 Principles of reliable data transfer r 3.5 Connection-oriented transport: TCP m segment structure m reliable data transfer m flow control m connection management r 3.6 Principles of congestion control r 3.7 TCP congestion control

80. Transport Layer 3-80 TCP congestion control: additive increase, multiplicative decrease r Approach: increase transmission rate (window size), probing for usable bandwidth, until loss occurs m additive increase: increase CongWin by 1 MSS every RTT until loss detected m multiplicative decrease: cut CongWin in half after loss time congestion window size Saw tooth behavior: probing for bandwidth

81. Transport Layer 3-81 TCP Congestion Control: details r sender limits transmission: LastByteSent-LastByteAcked  CongWin r Roughly,  CongWin is dynamic, function of perceived network congestion How does sender perceive congestion? r loss event = timeout or 3 duplicate acks  TCP sender reduces rate ( CongWin ) after loss event three mechanisms: m AIMD m slow start m conservative after timeout events rate = CongWin RTT Bytes/sec

82. Transport Layer 3-82 TCP Slow Start  When connection begins, CongWin = 1 MSS m Example: MSS = 500 bytes & RTT = 200 msec m initial rate = 20 kbps r available bandwidth may be >> MSS/RTT m desirable to quickly ramp up to respectable rate r When connection begins, increase rate exponentially fast until first loss event

83. Transport Layer 3-83 TCP Slow Start (more) r When connection begins, increase rate exponentially until first loss event:  double CongWin every RTT  done by incrementing CongWin for every ACK received r Summary: initial rate is slow but ramps up exponentially fast Host A one segment RTT Host B time two segments four segments

84. Transport Layer 3-84 Refinement: inferring loss r After 3 dup ACKs:  CongWin is cut in half m window then grows linearly r But after timeout event:  CongWin instead set to 1 MSS; m window then grows exponentially m to a threshold, then grows linearly  3 dup ACKs indicates network capable of delivering some segments  timeout indicates a “more alarming” congestion scenario Philosophy:

85. Transport Layer 3-85 Refinement Q: When should the exponential increase switch to linear? A: When CongWin gets to 1/2 of its value before timeout. Implementation: r Variable Threshold r At loss event, Threshold is set to 1/2 of CongWin just before loss event

86. 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.

87. Transport Layer 3-87 TCP throughput r What’s the average throughout of TCP as a function of window size and RTT? m Ignore slow start r Let W be the window size when loss occurs. r When window is W, throughput is W/RTT r Just after loss, window drops to W/2, throughput to W/2RTT. r Average throughout:.75 W/RTT

88. Transport Layer 3-88 TCP Futures: TCP over “long, fat pipes” r Example: 1500 byte segments, 100ms RTT, want 10 Gbps throughput r Requires window size W = 83,333 in-flight segments r Throughput in terms of loss rate:  ➜ L = 2·10 -10 Wow r New versions of TCP for high-speed

89. Transport Layer 3-89 Fairness goal: if K TCP sessions share same bottleneck link of bandwidth R, each should have average rate of R/K TCP connection 1 bottleneck router capacity R TCP connection 2 TCP Fairness

90. Transport Layer 3-90 Why is TCP fair? Two competing sessions: r Additive increase gives slope of 1, as throughout increases r multiplicative decrease decreases throughput proportionally R R equal bandwidth share Connection 1 throughput Connection 2 throughput congestion avoidance: additive increase loss: decrease window by factor of 2 congestion avoidance: additive increase loss: decrease window by factor of 2

91. Transport Layer 3-91 Fairness (more) Fairness and UDP r Multimedia apps often do not use TCP m do not want rate throttled by congestion control r Instead use UDP: m pump audio/video at constant rate, tolerate packet loss r Research area: TCP friendly Fairness and parallel TCP connections r nothing prevents app from opening parallel connections between 2 hosts. r Web browsers do this r Example: link of rate R supporting 9 connections; m new app asks for 1 TCP, gets rate R/10 m new app asks for 11 TCPs, gets R/2 !

92. Transport Layer 3-92 Chapter 3: Summary r principles behind transport layer services: m multiplexing, demultiplexing m reliable data transfer m flow control m congestion control r instantiation and implementation in the Internet m UDP m TCP Next: r leaving the network “edge” (application, transport layers) r into the network “core”