1 / 46

Reliable Byte-Stream (TCP)

This outline covers the connection establishment, sliding window, flow control mechanisms, and adaptive timeout strategies in TCP/IP protocols. It explores end-to-end services, demultiplexing in UDP, and the importance of acknowledgments and timeouts in reliable data transmission. The overview delves into flow control versus congestion control, data link reliability, and the sliding window concept for efficient data transfer. It highlights critical strategies for maintaining optimal network performance, reliability, and data integrity.

ckaminski
Download Presentation

Reliable Byte-Stream (TCP)

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. Reliable Byte-Stream (TCP) Outline Connection Establishment/Termination Sliding Window Revisited Flow Control Adaptive Timeout CS 332

  2. End-to-End Protocols • Underlying best-effort network • drops messages • re-orders messages • delivers duplicate copies of a given message • limits messages to some finite size • delivers messages after an arbitrarily long delay CS 332

  3. End-to-End Protocols • Common end-to-end services • guarantee message delivery • deliver messages in the same order they are sent • deliver at most one copy of each message • support arbitrarily large messages • support synchronization (between sender and receiver) • allow the receiver to flow control the sender • support multiple application processes on each host CS 332

  4. Simple Demultiplexer (UDP) • Extends host-to-host service into process-to-process • Unreliable and unordered datagram service • Adds multiplexing • No flow control • Endpoints identified by ports (why not PID?) • servers have well-known ports (clients don’t need this) • Often just starting point • see /etc/services on Unix/Linux • Implemented as message queue CS 332

  5. Simple Demultiplexer (UDP) • Header format • Note 16 bit port number (so only 64K ports) • Process really identified via <port,host> pair • Checksum (optional in IPv4, mandatory in IPv6) • pseudo header + UDP header + data • Pseudo header: Protocol number Source IP Dest IP UDP length field Why? 0 16 31 SrcPort DstPort Checksum Length Data CS 332

  6. Application process Application process W rite Read bytes bytes … … TCP TCP Send buffer Receive buffer … Segment Segment Segment T ransmit segments TCP Overview • Full duplex • Flow control: keep sender from overrunning receiver • Congestion control: keep sender from overrunning network • Connection-oriented • Byte-stream • app writes bytes • TCP sends segments • app reads bytes CS 332

  7. Flow Control vs Congestion Control • Flow Control • Prevent sender from overloading receiver • End-to-end issue • Congestion Control • Prevent too much data from being injected into network • Concerned with how hosts and network interact CS 332

  8. Data Link Reliability (text 2.5) Wherein we look at reliability issues on a point-to-point link! Error correcting codes can’t handle all possible errors (without introducing lots of overhead--including this is not designing for normal situation), so badly garbled frames are dropped. We need a way to recover from these lost frames. CS 332

  9. Acks and Timeouts • Acknowledgement (ACK) • Small frame sent to peer indicating receipt of frame • No data • Piggybacking • Timeout • If ACK not received within reasonable time, original frame is retransmitted • Automatic Repeat Request (ARQ) • General strategy of using ACKS and timeouts to implement reliable delivery CS 332

  10. Acknowledgements & Timeouts CS 332

  11. Acknowledgements & Timeouts CS 332

  12. A Subtlety… • Consider scenarios (c) and (d) in previous slide. • Receiver receives two good frames (duplicate) • It may deliver both to higher layer protocol (not good!) • Solution: 1-bit sequence number in frame header CS 332

  13. Stop-and-Wait Sender Receiver • Problem: keeping the pipe full • Example • 1.5Mbps link x 45ms RTT = 67.5Kb (8KB) • 1KB frames implies 1/8th link utilization (Next slide) CS 332

  14. Bandwidth x Delay Product • Sending a 1KB packet in 45ms implies sending at rate of (1024 x 8)/0.045 = 182 Kbps, or 1/8 of bandwidth. • Bandwidth-delay: The number of bits that fits in the pipe in a single round trip. (I.e. the amount of data that could be “in transit” at any given time.) • Goal: Want to be able to send this much data before getting first ACK. (keeping the pipe full) CS 332

  15. Sender Receiver … ime T … Sliding Window • Allow multiple outstanding (un-ACKed) frames • Upper bound on un-ACKed frames, called window CS 332

  16. £ SWS … … LAR LFS Sliding Window: Sender • Assign sequence number to each frame (SeqNum) • Maintain three state variables: • send window size (SWS) • last acknowledgment received (LAR) • last frame sent (LFS) • Maintain invariant: LFS - LAR≤SWS • Advance LAR when ACK arrives • Buffer up to SWS frames (must be prepared to retransmit frames until they are ACKed) CS 332

  17. LFR Sliding Window: Receiver • Maintain three state variables • receive window size (RWS) (upper bound on # out-of-order frames) • largest frame acceptable (LFA) (sequence # of) • last frame received (LFR) • Maintain invariant: LFA - LFR≤RWS • Frame SeqNum arrives: • if LFR < SeqNum≤LFA accept • if SeqNum≤LFR or SeqNum > LFA discard • Send cumulative ACKs £ RWS … … LFA CS 332

  18. Note: • When packet loss occurs, pipe is no longer kept full! • Longer it takes to notice lost packet, worse the condition becomes • Possible solutions: • Send NACKs • Selective acknowledgements (just ACK exactly those frames received, not highest frame received) • Not used: too much added complexity CS 332

  19. Sequence Number Space • SeqNum field is finite; sequence numbers wrap around • Sequence number space must be larger than number of outstanding frames (I.e. stop-and-wait had sequence # space size == 2) • I.e. if sequence number space is of size 8 (say 0..7), and number of outstanding frames is allowed to be 10, then sender can send sequence numbers 0,1,2,3,4,5,6,7,0,1 all at once. Now if receiver sends back an ACK with sequence number 1, which packet 1 is it ACKing? CS 332

  20. Sequence Number Space • Even SWS < SequenceSpaceSize is not sufficient • suppose 3-bit SeqNum field (0..7) (so SequenceSpaceSize = 8) • Let SWS=RWS=7 • sender transmit frames 0..6 • Frames arrive successfully, but ACKs are lost • sender retransmits 0..6 • receiver expecting 7, 0..5, but believes it is receiving the second incarnation of 0..5 (because the receiver has at this point updated its various pointers) • SWS≤(SequenceSpaceSize+1)/2 is rule (if SWS=RWS) • Intuitively, SeqNum “slides” between two halves of sequence number space CS 332

  21. Easy to overlook… • Relationship between window size and sequence number space depends on assumption that frames are not reordered in transit (easy to assume on point-to-point link). CS 332

  22. Back to Chapter 5… CS 332

  23. Data Link Versus Transport • Transport potentially connects many different hosts • need explicit connection establishment and termination • Transport has potentially different RTT (over different routes and at different times, even on scale of minutes) • need adaptive timeout mechanism • Transport has potentially long delay in network • need to be prepared for arrival of very old packets • Transport has potentially different capacity at destination • need to accommodate different node capacity • Transport has potentially different network capacity • need to be prepared for network congestion CS 332

  24. The “End-to-End” Argument • Consider TCP vs X.25 • TCP: Consider underlying IP network unreliable and use sliding window to provide end-to-end in-order reliable delivery • X.25: Use sliding window within network on hop-by-hop basis (which should guarantee end-to-end). Several problems with this: • No guarantee that added hop preserves service • In link from A to B to C, no guarantee that B behaves perfectly (nodes known to introduce errors and mix packet order) CS 332

  25. End-to-End • “A function should not be provided in the lower levels of the system unless it can be completely and correctly implemented at that level” • Does allow for functions to be incompletely provided at lower levels for optimization • E.g. detecting and retransmitting single corrupt packet across one hop preferable to retransmitting entire file end-to-end. • See reading assignment on class homework page CS 332

  26. Segment Format CS 332

  27. Data (SequenceNum) Sender Receiver Acknowledgment + AdvertisedWindow Segment Format (cont) • Each connection identified with 4-tuple: • (SrcPort, SrcIPAddr, DestPort, DestIPAddr) • Sliding window and flow control • acknowledgment, SequenceNum, AdvertisedWindow • Flags • SYN, FIN, RESET, PUSH, URG, ACK • Checksum • pseudo header + TCP header + data CS 332

  28. Connection Establishment and Termination Active participant Passive participant (client) (server) Note: SequenceNum contains the sequence number of the first data byte contained in the segment. ACK field always gives the sequence number of the next data byte expected. (Except for the SYN segments) SYN, SequenceNum = x , y 1 + SYN + ACK, SequenceNum = x Acknowledgment = ACK, Acknowledgment = y + 1 CS 332

  29. CLOSED Active open /SYN Passive open Close Close Opening connection LISTEN SYN/SYN + ACK Send/ SYN SYN/SYN + ACK SYN_RCVD SYN_SENT ACK SYN + ACK/ACK Close /FIN ESTABLISHED Close /FIN FIN/ACK FIN_WAIT_1 CLOSE_WAIT Closing connection FIN/ACK ACK Close /FIN ACK + FIN/ACK FIN_WAIT_2 CLOSING LAST_ACK Timeout after two ACK ACK segment lifetimes FIN/ACK TIME_WAIT CLOSED State Transition Diagram event/action CS 332

  30. Sending application Receiving application TCP TCP LastByteWritten LastByteRead LastByteAcked LastByteSent NextByteExpected LastByteRcvd Sliding Window Revisited • Sending side • LastByteAcked ≤ LastByteSent • LastByteSent ≤ LastByteWritten • buffer bytes between LastByteAcked and LastByteWritten • Receiving side • LastByteRead < NextByteExpected • NextByteExpected ≤ LastByteRcvd +1 • buffer bytes between LastByteRead and LastByteRcvd CS 332

  31. Flow Control • Send buffer size: MaxSendBuffer • Receive buffer size: MaxRcvBuffer • Receiving side • LastByteRcvd - LastByteRead≤MaxRcvBuffer • AdvertisedWindow = MaxRcvBuffer - (LastByteRcvd - LastByteRead) • Sending side • LastByteSent - LastByteAcked≤AdvertisedWindow • EffectiveWindow = AdvertisedWindow - (LastByteSent - LastByteAcked) • LastByteWritten - LastByteAcked≤MaxSendBuffer • block sender if (LastByteWritten - LastByteAcked) + y > MaxSenderBuffer CS 332

  32. Flow Control • Always send ACK in response to arriving data segment • This response contains latest Acknowledge and AdvertisedWindow fields even if they haven’t changed • Slow receiving process may cause AdvertisedWindow = 0. • Problem: How does the sending side know when the advertised window is no longer 0? • It can’t get this info, since receiver only sends window advertisements in response to received packets, and sender can’t send anything because it believes the window size is zero. • Solution: Persist when AdvertisedWindow= 0 • Periodically send a probe segment with one byte of data. Although most won’t be accepted, they trigger responses, and eventually one will come back with a nonzero advertised window. CS 332

  33. Protection Against Wrap Around • 32-bit SequenceNum Bandwidth Time Until Wrap Around T1 (1.5 Mbps) 6.4 hours Ethernet (10 Mbps) 57 minutes T3 (45 Mbps) 13 minutes FDDI, FastEther (100 Mbps) 6 minutes OC-3 (155 Mbps) 4 minutes OC-12 (622 Mbps) 55 seconds OC-48 (2.5 Gbps) 14 seconds CS 332

  34. Keeping the Pipe Full Results below assume RTT of 100 ms, typical for cross-country link • 16-bit AdvertisedWindow Bandwidth Delay x Bandwidth Product T1 (1.5 Mbps) 18KB Ethernet (10 Mbps) 122KB T3 (45 Mbps) 549KB FDDI, FastEther (100 Mbps) 1.2MB OC-3 (155 Mbps) 1.8MB OC-12 (622 Mbps) 7.4MB OC-48 (2.5 Gbps) 29.6MB CS 332

  35. TCP Extensions • Implemented as header options • Store timestamp in outgoing segments • Extend sequence space with 32-bit timestamp: PAWS (Protection Against Wrapped Sequence Numbers) • Shift (scale) advertised window CS 332

  36. Nagle Algorithm • 1 byte data segment generates 41 byte packets (20 for IP header + 20 for TCP header). • Small packets are called tinygrams • On LANs, usually not an issue, but on WANs, this can be a problem (it adds congestion) • Solution: Nagle Algorithm (RFC 896, Nagle, 1984): When a TCP connection has outstanding data that has not yet been Acked, small segments cannot be sent until the outstanding data is acknowledged. CS 332

  37. Nagle Algorithm (continued) • Nagle is self-clocking: the faster the ACKs come back, the faster the data is sent. But on slow WAN, where tinygrams can be a problem, fewer segments are sent. • Ex. On LAN, time for single byte to be sent, ACKed and echoed is around 16ms. To generate data at this rate, you need to be typing around 60 characters per second (so on LAN you don’t kick in Nagle) • On WAN, you’ll often kick in Nagle CS 332

  38. Disabling the Nagle Algorithm • Why would you want to? • X Window system: small messages (mouse movements) need to be delivered without delay • Typing one of the terminals special function keys during interactive login • Function keys normally generate multiple bytes of data, beginning with ASCII escape character. If TCP gets data a byte at a time, it can potentially send first byte and then hold the rest of the characters. The server wouldn’t generate the ACK until it received the rest of the command, so Nagle would kick in, meaning rest of bytes not sent for 200ms, which can be a noticeable delay. • With sockets API, the TCP_NODELAY option disables Nagle • Host Requirements RFCs (1122, 1123) specify that there must be a way for an app to disable Nagle on an individual TCP connection. CS 332

  39. Adaptive Retransmission(Original Algorithm) • Measure SampleRTT for each segment/ACK pair • Compute weighted average of RTT • a between 0.8 and 0.9 (recommended value 0.9) • Note a in this range has a strong smoothing effect • Set timeout based on EstRTT • TimeOut=2xEstRTT (rather conservative) CS 332

  40. Karn/Partridge Algorithm Sender Receiver Sender Receiver • Problem: ACK doesn’t acknowledge a transmission (it acks a receive) • Do not sample RTT when retransmitting • Double timeout after each retransmission (exponential backoff) Original transmission Original transmission TT TT ACK Retransmission SampleR SampleR Retransmission ACK Why? CS 332

  41. A Problem • Problem with both these approaches: they can’t keep up with wide RTT fluctuations, thus causing unnecessary retransmissions • When the network is already loaded, unnecessary retransmissions add to the network load (as Stevens notes, “It is the network equivalent of pouring gasoline on a fire”) • What’s needed: keep track of the variance in RTT measurements AND use smooth RTT estimator. CS 332

  42. Jacobson/ Karels Algorithm • New Calculations for average RTT • Diff = sampleRTT - EstRTT • EstRTT = EstRTT + (gx Diff) • Recommended value for g is 0.125 • EstRTT is just the smoothed RTT as before • Dev = Dev + h ( |Diff| - Dev) • Recommended value for h is 0.25 • Dev is the smoothed mean deviation (easier to compute mean that standard deviation, which requires a square root) • TimeOut = EstRTT + 4xDev • Larger gain for the deviation makes the TimeOut value increase faster when the RTT changes. • Notes • algorithm only as good as granularity of clock (500ms on Unix) • accurate timeout mechanism important to congestion control (later) Note these values? CS 332

  43. TCP Interactive Data Flow • Material here is from TCP/IP Illustrated, Vol. 1 • Study by Caceres, et. al. (1991) : • On a packet count basis, about half of all TCP segments contain bulk data (ftp, email, Usenet news) • Half contain interactive data (telnet, rlogin) • On byte count basis, ratio is around 90% bulk transfer, 10% interactive. • Bulk data tends to be full size (normally 512 bytes of data), interactive is much smaller (90% of telnet and rlogin packets carry less than 10 bytes of data). CS 332

  44. More recent data • 1997 MCI study • Web: 75% of bytes, 70% of packets • FTP: 5% of bytes, 3% of packets • SMTP (email): 5% of bytes, 5% of packets • DNS, NNTP, Telnet/rlogin: < 2% each • 2000 CAIDA study • Napster, streaming audio, online games show up • 2004 top application? • P2P: 62% of Internet traffic. BitTorrent: approx. 35% (CacheLogic) • Update – 2007, Cringley says 50% BitTorrent (5% of users) CS 332

  45. Rlogin and Telnet • Surprisingly, each interactive keystroke typically generates a packet (as opposed to a line generating a packet). • Moreover, a single rlogin keystroke can generate 4 segments (though usually 3) • Interactive keystroke from client • ACK of keystroke from server (typically piggybacked in echo of data byte) see next slide • Echo of data byte from server • ACK of echoed byte from client CS 332

  46. Delayed ACKs • Normally, TCP does not send an ACK the instant it receives data. Instead, it delays the ACK, hoping to have data going in other direction on which it can piggyback the ACK. • Most implementations use a 200ms delay (delays ACK up to 200ms before sending the ACK by itself) • This is why in previous slide, ACK would normally piggyback with the echoed character CS 332

More Related