Computer Networks CSE 434 Fall 2009

Computer NetworksCSE 434 Fall 2009 Sandeep K. S. Gupta Arizona State University http://impact.asu.edu/cse434fa09.html http://impact.asu.edu Research Experience for Undergraduates (REU)

Announcements and Agenda • Transport Layer (Cont.) • TCP • Reliable data transfer • Flow Control • Connection Mgmt. • Congestion Control • Quiz • Project Presentations – Dec 2 and Dec 7 – Schedule has been posted.

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

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!

Go-back-N: overview sender: up to N unACKed pkts in pipeline receiver: only sends cumulative ACKs doesn’t ACK pkt if there’s a gap sender: has timer for oldest unACKed pkt if timer expires: retransmit all unACKed packets Selective Repeat: overview sender: up to N unACKed packets in pipeline receiver: ACKs individual pkts sender: maintains timer for each unACKed pkt if timer expires: retransmit only unACKed packet Pipelining Protocols

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

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

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 pointer 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)

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 implementer time TCP seq. #’s and ACKs 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 simple telnet scenario

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

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

Example RTT estimation:

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

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

data rcvd from app: create segment with seq # seq # is byte-stream number of first data byte in segment start timer if not already running (think of timer as for oldest unACKed segment) expiration interval: TimeOutInterval timeout: retransmit segment that caused timeout restart timer ACK rcvd: if acknowledges previously unACKed segments update what is known to be ACKed start timer if there are outstanding segments TCP sender events:

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum loop (forever) { switch(event) event: data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data) event: timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer event: ACK received, with ACK field value of y if (y > SendBase) { SendBase = y if (there are currently not-yet-acknowledged segments) start timer } } /* end of loop forever */ TCP sender(simplified) • Comment: • SendBase-1: last • cumulatively ACKed byte • Example: • SendBase-1 = 71;y= 73, so the rcvrwants 73+ ;y > SendBase, sothat new data is ACKed

Host A Host B Seq=92, 8 bytes data ACK=100 Seq=92 timeout 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 ACK=100 ACK=120 Seq=92, 8 bytes data Sendbase = 100 SendBase = 120 ACK=120 Seq=92 timeout SendBase = 100 SendBase = 120 premature timeout

Host A Host B Seq=92, 8 bytes data ACK=100 Seq=100, 20 bytes data timeout X loss ACK=120 time Cumulative ACK scenario TCP retransmission scenarios (more) SendBase = 120

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, ACKing both in-order segments Immediately send duplicate ACK, indicating seq. # of next expected byte Immediate send ACK, provided that segment starts at lower end of gap Event at Receiver Arrival of in-order segment with expected seq #. All data up to expected seq # already ACKed Arrival of in-order segment with expected seq #. One other segment has ACK pending Arrival of out-of-order segment higher-than-expect seq. # . Gap detected Arrival of segment that partially or completely fills gap

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 for that segment If sender receives 3 ACKs for same data, it assumes that segment after ACKed data was lost: fast retransmit:resend segment before timer expires Fast Retransmit

Host A Host B seq # x1 seq # x2 seq # x3 ACK x1 X seq # x4 seq # x5 ACK x1 ACK x1 ACK x1 triple duplicate ACKs resend seq X2 timeout time

Fast retransmit algorithm: event: ACK received, with ACK field value of y if (y > SendBase) { SendBase = y if (there are currently not-yet-acknowledged segments) start timer } else { increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) { resend segment with sequence number y } a duplicate ACK for already ACKed segment fast retransmit

receive side of TCP connection has a receive buffer: speed-matching service: matching send rate to receiving application’s drain rate flow control sender won’t overflow receiver’s buffer by transmitting too much, too fast (currently) unused buffer space application process IP datagrams TCP data (in buffer) TCP Flow Control • app process may be slow at reading from buffer

(suppose TCP receiver discards out-of-order segments) unused buffer space: = rwnd = RcvBuffer-[LastByteRcvd - LastByteRead] receiver: advertises unused buffer space by including rwnd value in segment header sender: limits # of unACKed bytes to rwnd guarantees receiver’s buffer doesn’t overflow (currently) unused buffer space application process IP datagrams TCP data (in buffer) rwnd RcvBuffer TCP Flow control: how it works

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 host sends TCP SYN segment to server specifies initial seq # no data Step 2:server host receives SYN, replies with SYNACK segment server allocates buffers specifies server initial seq. # Step 3: client receives SYNACK, replies with ACK segment, which may contain data TCP Connection Management

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.)

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. Note:with small modification, can handle simultaneous FINs. TCP Connection Management (cont.) client server closing FIN ACK closing FIN ACK timed wait closed closed

TCP Connection Management (cont) TCP server lifecycle TCP client lifecycle

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

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

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

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

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 R/2 in in in R/2 R/2 out out out R/3 lout lout lout R/4 R/2 R/2 R/2 lin lin lin a. b. c. Causes/costs of congestion: scenario 2 “costs” of congestion: • more work (retrans) for given “goodput” • unneeded retransmissions: link carries multiple copies of pkt

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

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!

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:

ABR: available bit rate: “elastic service” if sender’s path “underloaded”: sender should use available bandwidth if sender’s path congested: sender throttled to minimum guaranteed rate RM (resource management) cells: sent by sender, interspersed with data cells bits in RM cell set by switches (“network-assisted”) NI bit: no increase in rate (mild congestion) CI bit: congestion indication RM cells returned to sender by receiver, with bits intact Case study: ATM ABR congestion control

two-byte ER (explicit rate) field in RM cell congested switch may lower ER value in cell sender’ send rate thus maximum supportable rate on path EFCI bit in data cells: set to 1 in congested switch if data cell preceding RM cell has EFCI set, sender sets CI bit in returned RM cell Case study: ATM ABR congestion control

TCP congestion control: • goal: TCP sender should transmit as fast as possible, but without congesting network • Q: how to find rate just below congestion level • decentralized: each TCP sender sets its own rate, based on implicit feedback: • ACK: segment received (a good thing!), network not congested, so increase sending rate • lost segment: assume loss due to congested network, so decrease sending rate

loss, so decrease rate X TCP congestion control: bandwidth probing • “probing for bandwidth”: increase transmission rate on receipt of ACK, until eventually loss occurs, then decrease transmission rate • continue to increase on ACK, decrease on loss (since available bandwidth is changing, depending on other connections in network) ACKs being received, so increase rate X X X TCP’s “sawtooth” behavior X sending rate time • Q: how fast to increase/decrease? • details to follow

sender limits rate by limiting number of unACKed bytes “in pipeline”: cwnd: differs from rwnd(how, why?) sender limited bymin(cwnd,rwnd) roughly, cwndis dynamic, function of perceived network congestion ACK(s) cwnd rate = bytes/sec RTT TCP Congestion Control: details LastByteSent-LastByteAcked  cwnd cwnd bytes RTT

segment loss event: reducing cwnd timeout: no response from receiver cut cwnd to 1 3 duplicate ACKs: at least some segments getting through (recall fast retransmit) cut cwnd in half, less aggressively than on timeout TCP Congestion Control: more details ACK received: increase cwnd • slowstart phase: • increase exponentially fast (despite name) at connection start, or following timeout • congestion avoidance: • increase linearly

when connection begins, cwnd = 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 increase rate exponentially until first loss event or when threshold reached double cwnd every RTT done by incrementing cwnd by 1 for every ACK received time TCP Slow Start Host A Host B one segment RTT two segments four segments

ssthresh:cwnd threshold maintained by TCP on loss event: set ssthreshto cwnd/2 remember (half of) TCP rate when congestion last occurred when cwnd >= ssthresh: transition from slowstart to congestion avoidance phase new ACK cwnd = cwnd+MSS dupACKcount = 0 transmit new segment(s),as allowed L cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0 cwnd > ssthresh timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment slow start congestion avoidance timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment duplicate ACK dupACKcount++ Transitioning into/out of slowstart L

TCP: congestion avoidance AIMD • when cwnd > ssthresh grow cwnd linearly • increase cwnd by 1 MSS per RTT • approach possible congestion slower than in slowstart • implementation: cwnd = cwnd + MSS/cwnd for each ACK received • ACKs: increase cwnd by 1 MSS per RTT: additive increase • loss: cut cwnd in half (non-timeout-detected loss ): multiplicative decrease AIMD: Additive Increase Multiplicative Decrease

loss: timeout loss: timeout loss: timeout cwnd > ssthresh slow start congestion avoidance fast recovery TCP congestion control FSM: overview new ACK loss: 3dupACK loss: 3dupACK

Computer Networks CSE 434 Fall 2009