CSC 250

1. CSC 250 Introduction to Networking Fundamentals Class Meeting 7

2. Approach: incrementally develop sender, receiver sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer but control info will flow on both directions! use finite state machines (FSM) to specify sender, receiver So far: rdt 1.0 / perfectly reliable channel rdt 2.0 / corruption of data bits in packets imperfect / why? Rdt 2.1 / corrupt ACKs & NACs and retransmission Rdt 2.2 / a further refinement; no NAK packets Review of Reliable Data Transfer

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

4. New assumption: underlying channel can also lose packets (data or ACKs) checksum, seq. #, ACKs, retransmissions will be of help, but not enough Q: how to deal with loss? sender waits until certain data or ACK lost, then retransmits yuck: drawbacks? Approach: sender waits “reasonable” amount of time for ACK retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost): retransmission will be duplicate, but use of seq. #’s already handles this receiver must specify seq # of pkt being ACKed requires countdown timer rdt3.0: channels with errors and loss

5. rdt3.0 in action

6. rdt3.0 in action

7. rdt3.0 works, but performance stinks example: 1 Gbps link, 15 ms e-e prop. delay, 1KB packet: Performance of rdt3.0 L (packet length in bits) 8kb/pkt T = = = 8 microsec transmit R (transmission rate, bps) 10**9 b/sec • U sender: utilization – fraction of time sender busy sending • 1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link • network protocol limits use of physical resources!

8. 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 RTT last packet bit arrives, send ACK ACK arrives, send next packet, t = RTT + L / R

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

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

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

12. GBN inaction

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

14. Selective repeat: sender, receiver windows

15. 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 sender Selective repeat 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

16. checksum timer sequence number ACK NAK pipelining window Summary of Reliable Data Transfer Mechanism

17. 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 Chapter 3 outline

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

19. 32 bits source port # dest port # sequence number acknowledgement number head len not used Receive window U A P R S F checksum Urg data pnter Options (variable length) application data (variable length) TCP segment structure URG: urgent data (generally not used) counting by bytes of data (not segments!) ACK: ACK # valid PSH: push data now (generally not used) # bytes rcvr willing to accept RST, SYN, FIN: connection estab (setup, teardown commands) Internet checksum (as in UDP)

20. Q: how to set TCP timeout value? longer than RTT but RTT varies too short: premature timeout unnecessary retransmissions too long: slow reaction to segment loss Q: how to estimate RTT? 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 TCP Round Trip Time and Timeout

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

22. Example RTT estimation:

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

24. 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 Chapter 3 outline

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

26. Timeouts only Duplicate ACKs TCP rdt Implementation

27. 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 Chapter 3 outline

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

29. (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 guarantees receive buffer doesn’t overflow TCP Flow control: how it works

30. 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 3.6Principles of congestion control 3.7 TCP congestion control Chapter 3 outline

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

32. two senders, two receivers one router, infinite buffers no retransmission large delays when congested maximum achievable throughput lout lin : original data unlimited shared output link buffers Host A Host B Causes/costs of congestion: scenario 1

33. Costs of a Congested Network • large queuing delays

34. one router, finite buffers sender retransmission of lost packet Causes/costs of congestion: scenario 2 Host A lout lin : original data l'in : original data, plus retransmitted data Host B finite shared output link buffers

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

36. Costs of a Congested Network • large queuing delays • two retransmission costs • more work – the retransmissions necessary to achieve good throughput • unneeded retransmissions: link carries multiple copies of packets that are okay

37. four senders multihop paths timeout/retransmit l l in in Host A Host B Causes/costs of congestion: scenario 3 Q:what happens as and increase ? lout lin : original data l'in : original data, plus retransmitted data finite shared output link buffers

38. Host A Host B Causes/costs of congestion: scenario 3 lout Another “cost” of congestion: • when packet dropped, any “upstream transmission capacity used for that packet was wasted!

39. Costs of a Congested Network • large queuing delays • two retransmission costs • more work – the retransmissions necessary to achieve good throughput • unneeded retransmissions: link carries multiple copies of packets that are okay • upstream packet retransmissions wasted

40. 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 sender should send at Approaches towards congestion control Two broad approaches towards congestion control:

41. 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 Chapter 3 outline

42. TCP congestion control issues • how to limit sending rate • how to recognize congestion • an appropriate algorithm

43. end-end control (no network assistance) sender limits transmission: LastByteSent-LastByteAcked  CongWin Roughly, CongWin is dynamic, function of perceived network congestion How does sender perceive congestion? loss event = timeout or 3 duplicate acks TCP sender reduces rate (CongWin) after loss event three mechanisms: AIMD slow start conservative after timeout events CongWin rate = Bytes/sec RTT TCP Congestion Control

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

45. multiplicative decrease: cut CongWin in half after loss event TCP AIMD additive increase: increase CongWin by 1 MSS every RTT in the absence of loss events: probing Long-lived TCP connection

46. When connection begins, increase rate exponentially until first loss event: double CongWin every RTT done by incrementing CongWin for every ACK received Summary: initial rate is slow but ramps up exponentially fast Host A Host B one segment RTT two segments four segments time Host-to-host Activity

47. After 3 dup ACKs: CongWin is cut in half window then grows linearly But after timeout event: CongWin instead set to 1 MSS; window then grows exponentially to a threshold, then grows linearly Refinement Philosophy: • 3 dup ACKs indicates network capable of delivering some segments • timeout before 3 dup ACKs is “more alarming”

48. 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.

49. 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

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