1. 3: Transport Layer3a-1 Chapter 3: Transport Layer Chapter goals: r understand principles behind transport layer services: m multiplexing/demultiplex ing m reliable data transfer m flow control m congestion control r instantiation and implementation in the Internet m UDP m TCP Chapter Overview: r transport layer services r multiplexing/demultiplexing r connectionless transport: UDP r principles of reliable data transfer r connection-oriented transport: TCP m reliable transfer m flow control m connection management r principles of congestion control r TCP congestion control

2. 3: Transport Layer3a-2 Transport services and protocols r provide logical communication between app’ processes running on different hosts r transport protocols run in end systems r transport vs network layer services: r network layer: data transfer between end systems r transport layer: data transfer between processes m relies on, enhances, network layer services application transport network data link physical 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 logical end-end transport

3. 3: Transport Layer3a-3 Transport-layer protocols Internet transport services: r reliable, in-order unicast delivery (TCP) m congestion m flow control m connection setup r unreliable (“best-effort”), unordered unicast or multicast delivery: UDP r services not available: m real-time m bandwidth guarantees m reliable multicast application transport network data link physical 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 logical end-end transport

4. 3: Transport Layer3a-4 application transport network M P2 application transport network Multiplexing/demultiplexing Recall: segment - unit of data exchanged between transport layer entities m aka TPDU: transport protocol data unit receiver H t H n Demultiplexing: delivering received segments to correct app layer processes segment M application transport network P1 MMM P3 P4 segment header application-layer data

5. 3: Transport Layer3a-5 Multiplexing/demultiplexing multiplexing/demultiplexing: r based on sender, receiver port numbers, IP addresses m source, dest port #s in each segment m recall: well-known port numbers for specific applications gathering data from multiple app processes, enveloping data with header (later used for demultiplexing) source port #dest port # 32 bits application data (message) other header fields TCP/UDP segment format Multiplexing:

6. 3: Transport Layer3a-6 Multiplexing/demultiplexing: examples host A server B source port: x dest. port: 23 source port:23 dest. port: x port use: simple telnet app Web client host A Web server B Web client host C Source IP: C Dest IP: B source port: x dest. port: 80 Source IP: C Dest IP: B source port: y dest. port: 80 port use: Web server Source IP: A Dest IP: B source port: x dest. port: 80

7. 3: Transport Layer3a-7 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

8. 3: Transport Layer3a-8 UDP: more r often used for streaming multimedia apps m loss tolerant m rate sensitive r other UDP uses (why?): m DNS m SNMP r reliable transfer over UDP: add reliability at application layer m application-specific error recover! source port #dest port # 32 bits Application data (message) UDP segment format length checksum Length, in bytes of UDP segment, including header

9. 3: Transport Layer3a-9 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 nonethless? More later …. Goal: detect “errors” (e.g., flipped bits) in transmitted segment

10. 3: Transport Layer3a-10 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

11. 3: Transport Layer3a-11 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number rcvr window size ptr urgent data 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)

12. 3: Transport Layer3a-12 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

13. 3: Transport Layer3a-13 TCP: reliable data transfer simplified sender, assuming wait for event wait for event event: data received from application above event: timer timeout for segment with seq # y event: ACK received, with ACK # y create, send segment retransmit segment ACK processing one way data transfer no flow, congestion control

14. 3: Transport Layer3a-14 TCP: reliable data transfer 00 sendbase = initial_sequence number 01 nextseqnum = initial_sequence number 02 03 loop (forever) { 04 switch(event) 05 event: data received from application above 06 create TCP segment with sequence number nextseqnum 07 start timer for segment nextseqnum 08 pass segment to IP 09 nextseqnum = nextseqnum + length(data) 10 event: timer timeout for segment with sequence number y 11 retransmit segment with sequence number y 12 compue new timeout interval for segment y 13 restart timer for sequence number y 14 event: ACK received, with ACK field value of y 15 if (y > sendbase) { /* cumulative ACK of all data up to y */ 16 cancel all timers for segments with sequence numbers < y 17 sendbase = y 18 } 19 else { /* a duplicate ACK for already ACKed segment */ 20 increment number of duplicate ACKs received for y 21 if (number of duplicate ACKS received for y == 3) { 22 /* TCP fast retransmit */ 23 resend segment with sequence number y 24 restart timer for segment y 25 } 26 } /* end of loop forever */ Simplified TCP sender

15. 3: Transport Layer3a-15 TCP ACK generation [RFC 1122, RFC 2581] Event in-order segment arrival, no gaps, everything else already ACKed in-order segment arrival, no gaps, one delayed ACK pending out-of-order segment arrival 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 send duplicate ACK, indicating seq. # of next expected byte immediate ACK if segment starts at lower end of gap

16. 3: Transport Layer3a-16 TCP: retransmission scenarios Host A Seq=92, 8 bytes data ACK=100 loss timeout time lost ACK scenario Host B X Seq=92, 8 bytes data ACK=100 Host A Seq=100, 20 bytes data ACK=100 Seq=92 timeout time premature timeout, cumulative ACKs Host B Seq=92, 8 bytes data ACK=120 Seq=92, 8 bytes data Seq=100 timeout ACK=120

17. 3: Transport Layer3a-17 TCP Flow Control receiver: explicitly informs sender of (dynamically changing) amount of free buffer space  RcvWindow field in TCP segment sender: keeps the amount of transmitted, unACKed data less than most recently received RcvWindow sender won’t overrun receiver’s buffers by transmitting too much, too fast flow control receiver buffering RcvBuffer = size or TCP Receive Buffer RcvWindow = amount of spare room in Buffer

18. 3: Transport Layer3a-18 TCP Round Trip Time and Timeout Q: how to set TCP timeout value? r longer than RTT m note: RTT will vary 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, cumulatively ACKed segments  SampleRTT will vary, want estimated RTT “smoother”  use several recent measurements, not just current SampleRTT

19. 3: Transport Layer3a-19 TCP Round Trip Time and Timeout EstimatedRTT = (1-x)*EstimatedRTT + x*SampleRTT r Exponential weighted moving average r influence of given sample decreases exponentially fast r typical value of x: 0.1 Setting the timeout  EstimtedRTT plus “safety margin”  large variation in EstimatedRTT -> larger safety margin Timeout = EstimatedRTT + 4*Deviation Deviation = (1-x)*Deviation + x*|SampleRTT-EstimatedRTT|

20. 3: Transport Layer3a-20 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 end system sends TCP SYN control segment to server m specifies initial seq # Step 2: server end system receives SYN, replies with SYNACK control segment m ACKs received SYN m allocates buffers m specifies server-> receiver initial seq. #

21. 3: Transport Layer3a-21 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

22. 3: Transport Layer3a-22 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 handly simultaneous FINs. client FIN server ACK FIN closing closed timed wait closed

23. 3: Transport Layer3a-23 TCP Connection Management (cont) TCP client lifecycle TCP server lifecycle

24. 3: Transport Layer3a-24 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!

25. 3: Transport Layer3a-25 Causes/costs of congestion: scenario 1 r two senders, two receivers r one router, infinite buffers r no retransmission r large delays when congested r maximum achievable throughput

26. 3: Transport Layer3a-26 Causes/costs of congestion: scenario 2 r one router, finite buffers r sender retransmission of lost packet

27. 3: Transport Layer3a-27 Causes/costs of congestion: scenario 2 r always: (goodput) r “perfect” retransmission only when loss: r retransmission of delayed (not lost) packet makes larger (than perfect case) for same in out = in out > in out “costs” of congestion: r more work (retrans) for given “goodput” r unneeded retransmissions: link carries multiple copies of pkt

28. 3: Transport Layer3a-28 Causes/costs of congestion: scenario 3 r four senders r multihop paths r timeout/retransmit in Q: what happens as and increase ? in

29. 3: Transport Layer3a-29 Causes/costs of congestion: scenario 3 Another “cost” of congestion: r when packet dropped, any “upstream transmission capacity used for that packet was wasted!

30. 3: Transport Layer3a-30 Approaches towards congestion control End-end congestion control: r no explicit feedback from network r congestion inferred from end-system observed loss, delay r approach taken by TCP Network-assisted congestion control: r routers provide feedback to end systems m single bit indicating congestion (SNA, DECbit, TCP/IP ECN, ATM) m explicit rate sender should send at Two broad approaches towards congestion control:

31. 3: Transport Layer3a-31 TCP Congestion Control r end-end control (no network assistance)  transmission rate limited by congestion window size, Congwin, over segments: r w segments, each with MSS bytes sent in one RTT: throughput = w * MSS RTT Bytes/sec Congwin

32. 3: Transport Layer3a-32 TCP congestion control: r two “phases” m slow start m congestion avoidance r important variables:  Congwin  threshold: defines threshold between two slow start phase, congestion control phase r “probing” for usable bandwidth:  ideally: transmit as fast as possible ( Congwin as large as possible) without loss  increase Congwin until loss (congestion)  loss: decrease Congwin, then begin probing (increasing) again

33. 3: Transport Layer3a-33 TCP Slowstart r exponential increase (per RTT) in window size (not so slow!) r loss event: timeout (Tahoe TCP) and/or or three duplicate ACKs (Reno TCP) initialize: Congwin = 1 for (each segment ACKed) Congwin++ until (loss event OR CongWin > threshold) Slowstart algorithm Host A one segment RTT Host B time two segments four segments

34. 3: Transport Layer3a-34 TCP Congestion Avoidance /* slowstart is over */ /* Congwin > threshold */ Until (loss event) { every w segments ACKed: Congwin++ } threshold = Congwin/2 Congwin = 1 perform slowstart Congestion avoidance 1 1: TCP Reno skips slowstart (fast recovery) after three duplicate ACKs

35. 3: Transport Layer3a-35 TCP Fairness Fairness goal: if N TCP sessions share same bottleneck link, each should get 1/N of link capacity TCP congestion avoidance: r AIMD: additive increase, multiplicative decrease m increase window by 1 per RTT m decrease window by factor of 2 on loss event AIMD TCP connection 1 bottleneck router capacity R TCP connection 2

36. 3: Transport Layer3a-36 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