630 likes | 707 Views
CE80N Introduction to Networks & The Internet. Dr. Chane L. Fullmer UCSC Winter 2002. Next Week…. Jan 29 (T) Inside the Internet Chapter 17, Clients + Servers = Distributed Computing Chapter 18, Names for Computers Due Jan 29: Written Assignment #1 Web search engine evaluation
E N D
CE80NIntroduction to Networks&The Internet Dr. Chane L. Fullmer UCSC Winter 2002
Next Week… Jan 29 (T) Inside the Internet • Chapter 17, Clients + Servers = Distributed Computing • Chapter 18, Names for Computers • Due Jan 29: Written Assignment #1 Web search engine evaluation (10% of course grade) Jan 31 (Th) Midterm Exam (20% of course grade) CE80N -- Lecture #7
Reading • Chapter 16 – • TCP: Software For Reliable Communications • Chapter 19 – • Why The Internet Works Well CE80N -- Lecture #7
A Packet Switching System Can Be Overrun • Packet switching allows multiple computers to communicate without initial delay. • Requires that the computers divide data into small packets • Requires additional communication software to ensure that data is delivered reliably. CE80N -- Lecture #7
Figure 16.1 An example internet with four networks connected by routers. Figure 16.2 Cars from two roads merging onto another road are analogous to packets from two networks merging onto a third network.
Our goals: understand principles behind transport layer services: multiplexing demultiplexing reliable data transfer flow control congestion control instantiation and implementation in the Internet Overview: transport layer services multiplexing/demultiplexing connectionless transport: UDP principles of reliable data transfer connection-oriented transport: TCP reliable transfer flow control connection management principles of congestion control TCP congestion control Transport Layer CE80N -- Lecture #7
provide logical communication between processes running on different hosts transport protocols run in end systems transport vs network layer services: network layer: data transfer between end systems transport layer: data transfer between processes 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 Transport services and protocols CE80N -- Lecture #7
Internet transport services: reliable, in-order unicast delivery (TCP) congestion flow control connection setup unreliable (“best-effort”), unordered unicast or multicast delivery: UDP services not available: real-time bandwidth guarantees 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 Transport-layer protocols CE80N -- Lecture #7
Segment- unit of data exchanged between transport layer entities aka TPDU: transport protocol data unit M M M M application transport network application transport network application transport network H n Multiplexing/demultiplexing Demultiplexing: delivering received segments to correct app layer processes receiver P3 P4 application-layer data segment header P1 P2 segment H t M segment CE80N -- Lecture #7
multiplexing/demultiplexing: based on sender, receiver port numbers, IP addresses source, dest port #s in each segment Note: well-known port numbers for specific applications Multiplexing: Multiplexing/demultiplexing 32 bits gathering data from multiple app processes, encapsulating data with header (later used for demultiplexing) source port # dest port # other header fields application data (message) TCP/UDP segment format CE80N -- Lecture #7
Source IP: C Dest IP: B source port: x dest. port: 80 Source IP: C Dest IP: B source port: y dest. port: 80 Source IP: A Dest IP: B source port: x dest. port: 80 source port:23 dest. port: x source port: x dest. port: 23 Multiplexing/demultiplexing: examples Web client host C server B host A port use: simple telnet app Web server B Web client host A port use: Web server CE80N -- Lecture #7
“no frills,” “bare bones” Internet transport protocol “best effort” service, UDP segments may be: lost delivered out of order to app connectionless: no handshaking between UDP sender, receiver each UDP segment handled independently of others 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] CE80N -- Lecture #7
often used for streaming multimedia apps loss tolerant rate sensitive other UDP uses: DNS SNMP reliable transfer over UDP: add reliability at application layer application-specific error recover! UDP: more 32 bits source port # dest port # Length, in bytes of UDP segment, including header checksum length Application data (message) UDP segment format CE80N -- Lecture #7
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: compute checksum of received segment check if computed checksum equals checksum field value: NO - error detected YES - no error detected. But maybe errors nonetheless? UDP checksum Goal: detect “errors” (e.g., flipped bits) in transmitted segment CE80N -- Lecture #7
TCP Helps IP Guarantee Delivery • When hardware in a router or network system fails, other routers start sending datagrams. • As a result, some datagrams: • Arrive in a different order than they were sent • Checked by TCP • Put in order • Checked for duplicates by TCP CE80N -- Lecture #7
TCP Provides A Connection Between Computer Programs • TCP software makes it possible for two computer programs to communicate across the Internet. • Establishes a connection • Exchanges data • Terminates communication CE80N -- Lecture #7
The Magic Of Recovering Lost Datagrams • TCP includes an identification of each datagram. • Ignores duplicate copies • Recovers lost datagrams • Uses timers • Sends an acknowledgement back to the source CE80N -- Lecture #7
TCP Retransmission Is Automatic • TCP adapts to work everywhere on the Internet • Retransmits after a short time if destination computer is close • Retransmits after a longer period if destination computer is far • Measures delays • Adjusts the timeout factor automatically CE80N -- Lecture #7
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 receive duplicate ACKs (see receiver) • timer for each in-flight pkt • timeout(n): retransmit pkt n and all higher seq # pkts in window CE80N -- Lecture #7
receiver simple: ACK-only: always send ACK for correctly-received pkt with highest in-order seq # may generate duplicate ACKs need only remember expectedseqnum out-of-order pkt: discard (don’t buffer) -> no receiver buffering! ACK pkt with highest in-order seq # GBN: receiver CE80N -- Lecture #7
GBN inaction CE80N -- Lecture #7
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 CE80N -- Lecture #7
Selective repeat: sender, receiver windows CE80N -- Lecture #7
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 CE80N -- Lecture #7
Selective repeat in action CE80N -- Lecture #7
Example: seq #’s: 0, 1, 2, 3 window size=3 receiver sees no difference in two scenarios! incorrectly passes duplicate data as new in (a) Q: what relationship between seq # size and window size? Selective repeat: dilemma CE80N -- Lecture #7
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: Overview RFCs: 793, 1122, 1323, 2018, 2581
32 bits source port # dest port # sequence number acknowledgement number head len not used rcvr window size U A P R S F checksum ptr urgent data 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) CE80N -- Lecture #7
Seq. #’s: byte stream “number” of first byte in segment’s data ACKs: seq # of next byte expected from other side cumulative ACK Q: how receiver handles out-of-order segments A: TCP spec doesn’t say, - up to implementor Host B Host A User types ‘C’ Seq=42, ACK=79, data = ‘C’ host ACKs receipt of ‘C’, echoes back ‘C’ Seq=79, ACK=43, data = ‘C’ host ACKs receipt of echoed ‘C’ Seq=43, ACK=80 time simple telnet scenario TCP seq. #’s and ACKs CE80N -- Lecture #7
TCP: reliable data transfer event: data received from application above simplified sender, assuming • one way data transfer • no flow, congestion control create, send segment wait for event event: timer timeout for segment with seq # y wait for event retransmit segment event: ACK received, with ACK # y ACK processing CE80N -- Lecture #7
00sendbase = 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 */ TCP: reliable data transfer Simplified TCP sender CE80N -- Lecture #7
TCP ACK generation[RFC 1122, RFC 2581] 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 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 CE80N -- Lecture #7
Host A Host B Seq=92, 8 bytes data ACK=100 timeout X loss Seq=92, 8 bytes data ACK=100 time time lost ACK scenario TCP: retransmission scenarios Host A Host B Seq=92, 8 bytes data Seq=100, 20 bytes data Seq=92 timeout ACK=100 ACK=120 Seq=100 timeout Seq=92, 8 bytes data ACK=120 premature timeout, cumulative ACKs CE80N -- Lecture #7
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 flow control sender won’t overrun receiver’s buffers by transmitting too much, too fast TCP Flow Control RcvBuffer= size or TCP Receive Buffer RcvWindow = amount of spare room in Buffer receiver buffering CE80N -- Lecture #7
Q: how to set TCP timeout value? longer than RTT note: RTT will vary 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, cumulatively ACKed segments SampleRTT will vary, want estimated RTT “smoothed” use several recent measurements, not just current SampleRTT TCP Round Trip Time (RTT) & Timeout CE80N -- Lecture #7
Setting the timeout EstimtedRTT plus “safety margin” large variation in EstimatedRTT -> larger safety margin TCP Round Trip Time and Timeout EstimatedRTT = (1-x)*EstimatedRTT + x*SampleRTT • Exponential weighted moving average • influence of given sample decreases exponentially fast • typical value of x: 0.1 Timeout = EstimatedRTT + 4*Deviation Deviation = (1-x)*Deviation + x*|SampleRTT-EstimatedRTT| CE80N -- Lecture #7
Recall:TCP sender, receiver establish “connection” before exchanging data segments initialize TCP variables: seq. #s buffers, flow control info (e.g. RcvWindow) client: connection initiator Socket clientSocket = new Socket("hostname","port number"); server: contacted by client Socket connectionSocket = welcomeSocket.accept(); Three way handshake: Step 1:client end system sends TCP SYN control segment to server specifies initial seq # Step 2:server end system receives SYN, replies with SYNACK control segment ACKs received SYN allocates buffers specifies server-> receiver initial seq. Step 3: client receives SYNACK, replies with ACK ACKs SYN TCP Connection Management CE80N -- Lecture #7
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 server close FIN ACK close FIN ACK timed wait closed TCP Connection Management (cont.) CE80N -- Lecture #7
Step 3:client receives FIN, replies with ACK. Enters “timed wait” - will respond with ACK to received FINs Step 4:server, receives ACK. Connection closed. TCP Connection Management (cont.) client server closing FIN ACK closing FIN ACK timed wait closed closed CE80N -- Lecture #7
TCP Connection Management (cont) TCP server lifecycle TCP client lifecycle CE80N -- Lecture #7
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 CE80N -- Lecture #7
two senders, two receivers one router, infinite buffers no retransmission large delays when congested maximum achievable throughput Causes/costs of congestion: scenario 1 CE80N -- Lecture #7
four senders multihop paths timeout/retransmit l l in in Causes/costs of congestion: scenario 2 Q:what happens as and increase ? CE80N -- Lecture #7
Causes/costs of congestion: scenario 2 Another “cost” of congestion: • when packet dropped, any “upstream transmission capacity used for that packet was wasted! CE80N -- Lecture #7
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 explicit rate sender should send at Approaches towards congestion control Two broad approaches towards congestion control: CE80N -- Lecture #7
end-end control (no network assistance) transmission rate limited by congestion window size, Congwin, over segments: w * MSS throughput = Bytes/sec RTT TCP Congestion Control Congwin • w segments, each with MSS bytes sent in one RTT: CE80N -- Lecture #7
two “phases” slow start congestion avoidance important variables: Congwin threshold: defines threshold between two slow start phase, congestion control phase “probing” for usable bandwidth: ideally: transmit as fast as possible (Congwin as large as possible) without loss increaseCongwin until loss (congestion) loss: decreaseCongwin, then begin probing (increasing) again TCP congestion control: CE80N -- Lecture #7
exponential increase (per RTT) in window size (not so slow!) loss event: timeout (Tahoe TCP) and/or or three duplicate ACKs (Reno TCP) Slowstart algorithm time TCP Slowstart Host A Host B initialize: Congwin = 1 for (each segment ACKed) Congwin++ until (loss event OR CongWin > threshold) one segment RTT two segments four segments CE80N -- Lecture #7
TCP Congestion Avoidance Congestion avoidance /* slowstart is over */ /* Congwin > threshold */ Until (loss event) { every w segments ACKed: Congwin++ } threshold = Congwin/2 Congwin = 1 perform slowstart 1 1: TCP Reno skips slowstart (fast recovery) after three duplicate ACKs CE80N -- Lecture #7
AIMD: additive increase, multiplicative decrease increase window by 1 per RTT decrease window by factor of 2 on loss event TCP congestion avoidance CE80N -- Lecture #7