Chapter 3 Transport Layer Computer Networking: A Top Down Approach 6th edition Jim Kurose, Keith Ross Addison-Wesley March 2012 A note on the use of these ppt slides: We’re making these slides freely available to all (faculty, students, readers). They’re in PowerPoint form so you see the animations; and can add, modify, and delete slides (including this one) and slide content to suit your needs. They obviously represent a lot of work on our part. In return for use, we only ask the following: If you use these slides (e.g., in a class) that you mention their source (after all, we’d like people to use our book!) If you post any slides on a www site, that you note that they are adapted from (or perhaps identical to) our slides, and note our copyright of this material. Thanks and enjoy! JFK/KWR All material copyright 1996-2013 J.F Kurose and K.W. Ross, All Rights Reserved Transport Layer

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 3-packet pipelining increases utilization by a factor of 3! Transport Layer

Pipelined protocols: overview Go-back-N: sender can have up to N unacked packets in pipeline receiver only sends cumulative ack doesn’t ack packet if there’s a gap sender has timer for oldest unacked packet when timer expires, retransmit all unacked packets Selective Repeat: sender can have up to N unack’ed packets in pipeline rcvr sends individual ack for each packet sender maintains timer for each unacked packet when timer expires, retransmit only that unacked packet Transport Layer

Chapter 3 outline 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 Transport Layer

TCP: Overview RFCs: 793,1122,1323, 2018, 2581 point-to-point: one sender, one receiver reliable, in-order byte steam: no “message boundaries” pipelined: TCP congestion and flow control set window size full duplex data: bi-directional data flow in same connection MSS: maximum segment size connection-oriented: handshaking (exchange of control msgs) inits sender, receiver state before data exchange flow controlled: sender will not overwhelm receiver Transport Layer

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

TCP seq. numbers, ACKs sequence numbers: source port # dest port # sequence number acknowledgement number checksum rwnd urg pointer outgoing segment from sender sequence numbers: byte stream “number” of first byte in segment’s data acknowledgements: 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 window size N sender sequence number space source port # dest port # sequence number acknowledgement number checksum rwnd urg pointer incoming segment to sender sent ACKed sent, not-yet ACKed (“in-flight”) usable but not yet sent not usable A Transport Layer

simple telnet scenario TCP seq. numbers, ACKs Host A Host B 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 Transport Layer

TCP round trip time, timeout 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 Transport Layer

TCP round trip time, timeout EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT exponential weighted moving average influence of past sample decreases exponentially fast typical value:  = 0.125 RTT: gaia.cs.umass.edu to fantasia.eurecom.fr RTT (milliseconds) sampleRTT EstimatedRTT time (seconds) Transport Layer

Chapter 3 outline 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 Transport Layer

TCP reliable data transfer TCP creates rdt service on top of IP’s unreliable service pipelined segments cumulative acks single retransmission timer retransmissions triggered by: timeout events duplicate acks Transport Layer

TCP sender events: 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 ack acknowledges previously unacked segments update what is known to be ACKed start timer if there are still unacked segments Transport Layer

TCP sender (simplified) create segment, seq. #: NextSeqNum pass segment to IP (i.e., “send”) NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer data received from application above L wait for event NextSeqNum = InitialSeqNum SendBase = InitialSeqNum retransmit not-yet-acked segment with smallest seq. # start timer timeout if (y > SendBase) { SendBase = y /* SendBase–1: last cumulatively ACKed byte */ if (there are currently not-yet-acked segments) start timer else stop timer } ACK received, with ACK field value y Transport Layer

TCP fast retransmit 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. TCP fast retransmit if sender receives 3 ACKs for same data (“triple duplicate ACKs”), resend unacked segment with smallest seq # likely that unacked segment lost, so don’t wait for timeout (“triple duplicate ACKs”), Transport Layer

TCP fast retransmit X fast retransmit after sender Host A Host B timeout Seq=92, 8 bytes of data Seq=100, 20 bytes of data X ACK=100 ACK=100 ACK=100 ACK=100 Seq=100, 20 bytes of data fast retransmit after sender receipt of triple duplicate ACK Transport Layer

Chapter 3 outline 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 Transport Layer

receiver protocol stack TCP flow control application process application may remove data from TCP socket buffers …. application OS TCP socket receiver buffers … slower than TCP receiver is delivering (sender is sending) TCP code IP code receiver controls sender, so sender won’t overflow receiver’s buffer by transmitting too much, too fast flow control from sender receiver protocol stack Transport Layer

TCP flow control receiver “advertises” free buffer space by including rwnd value in TCP header of receiver-to-sender segments RcvBuffer size set via socket options (typical default is 4096 bytes) many operating systems autoadjust RcvBuffer sender limits amount of unacked (“in-flight”) data to receiver’s rwnd value guarantees receive buffer will not overflow to application process buffered data free buffer space RcvBuffer rwnd TCP segment payloads receiver-side buffering Transport Layer

Chapter 3 outline 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 Transport Layer

Connection Management before exchanging data, sender/receiver “handshake”: agree to establish connection (each knowing the other willing to establish connection) agree on connection parameters application application connection state: ESTAB connection variables: seq # client-to-server server-to-client rcvBuffer size at server,client connection state: ESTAB connection Variables: seq # client-to-server server-to-client rcvBuffer size at server,client network network Socket clientSocket = newSocket("hostname","port number"); Socket connectionSocket = welcomeSocket.accept(); Transport Layer

Agreeing to establish a connection 2-way handshake: Q: will 2-way handshake always work in network? variable delays retransmitted messages (e.g. req_conn(x)) due to message loss message reordering can’t “see” other side Let’s talk ESTAB OK ESTAB choose x req_conn(x) ESTAB acc_conn(x) ESTAB Transport Layer

TCP 3-way handshake client state server state LISTEN SYNSENT SYNbit=1, Seq=x choose init seq num, x send TCP SYN msg SYN RCVD ESTAB SYNbit=1, Seq=y ACKbit=1; ACKnum=x+1 choose init seq num, y send TCP SYNACK msg, acking SYN ACKbit=1, ACKnum=y+1 received SYNACK(x) indicates server is live; send ACK for SYNACK; this segment may contain client-to-server data received ACK(y) indicates client is live ESTAB Transport Layer

TCP: closing a connection client, server each close their side of connection send TCP segment with FIN bit = 1 respond to received FIN with ACK on receiving FIN, ACK can be combined with own FIN simultaneous FIN exchanges can be handled Transport Layer

TCP: closing a connection client state server state ESTAB ESTAB FIN_WAIT_1 FINbit=1, seq=x can no longer send but can receive data clientSocket.close() CLOSE_WAIT FIN_WAIT_2 ACKbit=1; ACKnum=x+1 wait for server close can still send data can no longer send data LAST_ACK TIMED_WAIT FINbit=1, seq=y CLOSED timed wait for 2*max segment lifetime CLOSED ACKbit=1; ACKnum=y+1 Transport Layer

Chapter 3 outline 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 Transport Layer

Principles of congestion control 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! Transport Layer

Causes/costs of congestion: scenario 1 original data: lin throughput: lout two senders, two receivers one router, infinite buffers output link capacity: R no retransmission Host A unlimited shared output link buffers Host B R/2 delay lin R/2 lout lin maximum per-connection throughput: R/2 large delays as arrival rate, lin, approaches capacity Transport Layer

Approaches towards congestion control two broad approaches towards congestion control: 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 for sender to send at Transport Layer

Chapter 3 outline 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 Transport Layer

TCP congestion control: additive increase multiplicative decrease approach: sender increases transmission rate (window size), probing for usable bandwidth, until loss occurs additive increase: increase cwnd by 1 MSS every RTT until loss detected multiplicative decrease: cut cwnd in half after loss additively increase window size … …. until loss occurs (then cut window in half) AIMD saw tooth behavior: probing for bandwidth congestion window size cwnd: TCP sender time Transport Layer

TCP Slow Start Host A Host B when connection begins, increase rate exponentially until first loss event: initially cwnd = 1 MSS double cwnd every RTT done by incrementing cwnd for every ACK received summary: initial rate is slow but ramps up exponentially fast one segment RTT two segments four segments time Transport Layer

TCP: detecting, reacting to loss loss indicated by timeout: cwnd set to 1 MSS; window then grows exponentially (as in slow start) to threshold, then grows linearly loss indicated by 3 duplicate ACKs: TCP RENO dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks) Transport Layer

TCP: switching from slow start to CA Q: when should the exponential increase switch to linear? A: when cwnd gets to 1/2 of its value before timeout. Implementation: variable ssthresh on loss event, ssthresh is set to 1/2 of cwnd just before loss event Transport Layer

TCP Congestion Control Slow start: Cwnd starts at 1MSS, cwnd doubles every RTT if no loss or triple duplicate ACK until reaches the ssthresh Congestion avoidance: Additive increase multiple decrease (AIMD): after slow start to the ssthresh, cwnd and ssthresh increases linearly TCP Tahoe: Time out: cwnd reset to 1MSS, ssthresh cut down to half of the existing ssthresh 3 duplicate ACKs: the same as time out TCP Reno: 3 duplicate ACKs: cwnd and ssthresh cut down to half and increase linearly Transport Layer

a. Identify the intervals of time when TCP slow start is operating. Consider Figure below. Assuming TCP Reno is the protocol experiencing the behavior shown above, answer the following questions. In all cases, you should provide a short discussion justifying your answer. a. Identify the intervals of time when TCP slow start is operating. b. Identify the intervals of time when TCP congestion avoidance is operating. c. After the 16th transmission round, is segment loss detected by a triple duplicate ACK or by a timeout? d. After the 22nd transmission round, is segment loss detected by a triple duplicate ACK or by a timeout? e. What is the initial value of ssthresh at the first transmission round? f. What is the value of ssthresh at the 18th transmission round? g. What is the value of ssthresh at the 24th transmission round? h. During what transmission round is the 70th segment sent? i. Assuming a packet loss is detected after the 26th round by the receipt of a triple duplicate ACK, what will be the values of the congestion window size and of ssthresh? Transport Layer

Chapter 3: summary principles behind transport layer services: multiplexing, demultiplexing reliable data transfer flow control congestion control instantiation, implementation in the Internet UDP TCP next: leaving the network “edge” (application, transport layers) into the network “core” Transport Layer