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Midterm Review

Midterm Review. In class, 1:00-2:30 pm, Tu. 2/6 Closed Book One 8.5” by 11” sheet of paper permitted (single side). Lecture 1. Internet Architecture Network Protocols Network Edge A taxonomy of communication networks. A Taxonomy of Communication Networks.

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Midterm Review

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  1. Midterm Review In class, 1:00-2:30 pm, Tu. 2/6 Closed Book One 8.5” by 11” sheet of paper permitted (single side)

  2. Lecture 1 • Internet Architecture • Network Protocols • Network Edge • A taxonomy of communication networks

  3. A Taxonomy of Communication Networks • The fundamental question: how is data transferred through net (including edge & core)? • Communication networks can be classified based on how the nodes exchange information: Communication Networks SwitchedCommunication Network BroadcastCommunication Network Packet-SwitchedCommunication Network Circuit-SwitchedCommunication Network TDM FDM Datagram Network Virtual Circuit Network

  4. Sequence of A & B packets does not have fixed pattern  statistical multiplexing. In TDM each host gets same slot in revolving TDM frame. D E Packet Switching: Statistical Multiplexing 10 Mbs Ethernet C A statistical multiplexing 1.5 Mbs B queue of packets waiting for output link

  5. Packet Switching Network resources (e.g., bandwidth) divided into “pieces” for allocation Resource piece idle if not used by owning call (no sharing) Dividing link bandwidth into “pieces” Great for bursty data NOT efficient ! Circuit Switching: Excessive congestion: packet delay and loss protocols needed for reliable data transfer, congestion control Packet Switching versus Circuit Switching

  6. Datagram Packet Switching • Each packet is independently switched • Each packet header contains destination address which determines next hop • Routes may change during session • No resources are pre-allocated (reserved) in advance • Example: IP networks

  7. Virtual-Circuit Packet Switching • Hybrid of circuit switching and packet switching • All packets from one packet stream are sent along a pre-established path (= virtual circuit) • Each packet carries tag (virtual circuit ID), tag determines next hop • Guarantees in-sequence delivery of packets • However, packets from different virtual circuits may be interleaved • Example: ATM (Asynchronous Transfer Mode) networks

  8. Lecture 2 • Network access and physical media • Internet structure and ISPs • Delay & loss in packet-switched networks • Protocol layers, service models

  9. “Tier-3” ISPs and local ISPs last hop (“access”) network (closest to end systems) Tier-3: Turkish Telecom, Minnesota Regional Network Tier 3 ISP local ISP local ISP local ISP local ISP local ISP local ISP local ISP local ISP NAP Local and tier- 3 ISPs are customers of higher tier ISPs connecting them to rest of Internet Tier-2 ISP Tier-2 ISP Tier-2 ISP Tier-2 ISP Tier-2 ISP Internet structure: network of networks Tier 1 ISP Tier 1 ISP Tier 1 ISP

  10. 1. processing: check bit errors determine output link transmission A propagation B processing queueing Four sources of packet delay • 2. queueing • time waiting at output link for transmission • depends on congestion level of router

  11. 3. Transmission delay: R=link bandwidth (bps) L=packet length (bits) time to send bits into link = L/R 4. Propagation delay: d = length of physical link s = propagation speed in medium (~2x108 m/sec) propagation delay = d/s transmission A propagation B processing queueing Delay in packet-switched networks Note: s and R are very different quantities!

  12. application: supporting network applications FTP, SMTP, STTP transport: host-host data transfer TCP, UDP network: routing of datagrams from source to destination IP, routing protocols link: data transfer between neighboring network elements PPP, Ethernet physical: bits “on the wire” application transport network link physical Internet protocol stack

  13. Principles of app layer protocols Web and HTTP FTP Electronic Mail: SMTP, POP3, IMAP DNS Socket Programming Web Caching Application Layer

  14. Nonpersistent HTTP At most one object is sent over a TCP connection. HTTP/1.0 uses nonpersistent HTTP Persistent HTTP Multiple objects can be sent over single TCP connection between client and server. HTTP/1.1 uses persistent connections in default mode HTTP connections • HTTP Message, Format, Response, Methods • HTTP cookies

  15. Nonpersistent HTTP issues: requires 2 RTTs per object OS must work and allocate host resources for each TCP connection but browsers often open parallel TCP connections to fetch referenced objects Persistent HTTP server leaves connection open after sending response subsequent HTTP messages between same client/server are sent over connection Persistent without pipelining: client issues new request only when previous response has been received one RTT for each referenced object Persistent with pipelining: default in HTTP/1.1 client sends requests as soon as it encounters a referenced object as little as one RTT for all the referenced objects Response Time of HTTP

  16. FTP client contacts FTP server at port 21, specifying TCP as transport protocol Client obtains authorization over control connection Client browses remote directory by sending commands over control connection. When server receives a command for a file transfer, the server opens a TCP data connection to client After transferring one file, server closes connection. TCP control connection port 21 TCP data connection port 20 FTP client FTP server FTP: separate control, data connections • Server opens a second TCP data connection to transfer another file. • Control connection: “out of band” • FTP server maintains “state”: current directory, earlier authentication

  17. uses TCP to reliably transfer email message from client to server, port 25 direct transfer: sending server to receiving server three phases of transfer handshaking (greeting) transfer of messages closure command/response interaction commands: ASCII text response: status code and phrase messages must be in 7-bit ASCII Electronic Mail: SMTP [RFC 2821]

  18. no server has all name-to-IP address mappings local name servers: each ISP, company has local (default) name server host DNS query first goes to local name server authoritative name server: for a host: stores that host’s IP address, name can perform name/address translation for that host’s name Why not centralize DNS? single point of failure traffic volume distant centralized database maintenance doesn’t scale! DNS name servers

  19. Root name server: may not know authoritative name server may know intermediate name server: who to contact to find authoritative name server local name server dns.eurecom.fr intermediate name server dns.nwu.edu DNS example root name server 6 2 3 7 5 4 1 8 authoritative name server dns.cs.nwu.edu requesting host surf.eurecom.fr www.cs.nwu.edu

  20. recursive query: puts burden of name resolution on contacted name server heavy load? iterated query: contacted server replies with name of server to contact “I don’t know this name, but ask this server” local name server dns.eurecom.fr intermediate name server dns.umass.edu DNS: iterated queries root name server iterated query 2 3 4 7 5 6 1 8 authoritative name server dns.cs.umass.edu requesting host surf.eurecom.fr gaia.cs.umass.edu

  21. user sets browser: Web accesses via cache browser sends all HTTP requests to cache object in cache: cache returns object else cache requests object from origin server, then returns object to client Why web caching? Web caches (proxy server) Goal: satisfy client request without involving origin server origin server Proxy server HTTP request HTTP request client HTTP response HTTP response HTTP request HTTP response client origin server

  22. Install cache suppose hit rate is .4 Consequence 40% requests will be satisfied almost immediately 60% requests satisfied by origin server utilization of access link reduced to 60%, resulting in negligible delays (say 10 msec) total delay = Internet delay + access delay + LAN delay = .6*2 sec + .6*.01 secs + milliseconds < 1.3 secs Caching example (3) origin servers public Internet 1.5 Mbps access link institutional network 10 Mbps LAN institutional cache

  23. Transport-layer services Multiplexing and demultiplexing Connectionless transport: UDP Principles of reliable data transfer TCP Segment structures Flow control Congestion control Transport Layer

  24. UDP socket identified by two-tuple: (dest IP address, dest port number) When host receives UDP segment: checks destination port number in segment directs UDP segment to socket with that port number Demultiplexing • TCP socket identified by 4-tuple: • source IP address • source port number • dest IP address • dest port number • recv host uses all four values to direct segment to appropriate socket

  25. Why is there a UDP? no connection establishment (which can add delay) simple: no connection state at sender, receiver small segment header no congestion control: UDP can blast away as fast as desired UDP: User Datagram Protocol [RFC 768] 32 bits source port # dest port # checksum length Application data (message) UDP segment format

  26. Sender: treat segment contents as sequence of 16-bit integers checksum: addition (1’s complement sum) of segment contents sender puts checksum value into UDP checksum field Receiver: addition of all segment contents + checksum check if all bits are 1: NO - error detected YES - no error detected. But maybe errors nonetheless? More later …. UDP checksum Goal: detect “errors” (e.g., flipped bits) in transmitted segment 0110 0101 1011 0100 0110 0101 0100 1111 1’s complement sum: Addition: Addition: 1’s complement sum:

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

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

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

  30. What happens if ACK/NAK corrupted? sender doesn’t know what happened at receiver! can’t just retransmit: possible duplicate What to do? sender ACKs/NAKs receiver’s ACK/NAK? What if sender ACK/NAK lost? retransmit, but this might cause retransmission of correctly received pkt! Handling duplicates: sender adds sequence number to each pkt sender retransmits current pkt if ACK/NAK garbled receiver discards (doesn’t deliver up) duplicate pkt stop and wait rdt2.0 has a fatal flaw! Sender sends one packet, then waits for receiver response

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

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

  33. 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 it is certain that 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

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

  35. 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) • Single timer for all in-flight pkts • timeout(n): retransmit pkt n and all higher seq # pkts in window

  36. 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

  37. Selective repeat: sender, receiver windows

  38. 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)

  39. 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

  40. Time-out period often relatively long: long delay before resending lost packet Detect lost segments via duplicate ACKs. Sender often sends many segments back-to-back If segment is lost, there will likely be many duplicate ACKs. If sender receives 3 ACKs for the same data, it supposes that segment after ACKed data was lost: fast retransmit:resend segment before timer expires Fast Retransmit

  41. (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

  42. 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

  43. 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

  44. After 3 dup ACKs: CongWin is cut in half window then grows linearly But after timeout event: Enter “slow start” 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”

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

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