3-1 TCP Protocol r point-to-point: m one sender, one receiver r reliable, in-order byte steam: m no “message boundaries” r pipelined: m TCP congestion and flow control set window size r send & receive buffers r Connection-oriented: m handshaking (exchange of control msgs) init’s sender, receiver state before data exchange r flow controlled: m sender will not overwhelm receiver source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number Receive window Urg data pnter checksum F SR PAU head len not used Options (variable length) URG: urgent data (generally not used) ACK: ACK # valid PSH: push data now (generally not used) RST, SYN, FIN: connection estab (setup, teardown commands) # bytes rcvr willing to accept counting by bytes of data (not segments!) Internet checksum (as in UDP)

3-2 TCP reliable data transfer r TCP creates rdt service on top of IP’s unreliable service r Pipelined segments r Cumulative acks r TCP uses single retransmission timer r Retransmissions are triggered by: m timeout events m duplicate acks r Initially consider simplified TCP sender: m ignore duplicate acks m ignore flow control, congestion control data rcvd from app: r Create segment with seq # r seq # is byte-stream number of first data byte in segment r start timer if not already running (think of timer as for oldest unacked segment)  expiration interval: TimeOutInterval timeout: r retransmit segment that caused timeout r restart timer Ack rcvd: r If acknowledges previously unacked segments m update what is known to be acked m start timer if there are outstanding segments

3-3 TCP: retransmission scenarios Host A Seq=92, 8 bytes data ACK=100 loss timeout lost ACK scenario Host B X Seq=92, 8 bytes data ACK= 100 time SendBase = 100 Host A Seq=100, 20 bytes data ACK=100 time premature timeout Host B Seq=92, 8 bytes data ACK =12 0 Seq=92, 8 bytes data Seq=92 timeout ACK =12 0 Seq=92 timeout SendBase = 120 SendBase = 120 Sendbase = 100 Host A Seq=92, 8 bytes data ACK=100 loss timeout Cumulative ACK scenario Host B X Seq=100, 20 bytes data AC K=1 20 time SendBase = 120 r Time-out period often relatively long: m long delay before resending lost packet r Detect lost segments via duplicate ACKs. m Sender often sends many segments back-to-back m If segment is lost, there will likely be many duplicate ACKs. r If sender receives 3 ACKs for the same data, it supposes that segment after ACKed data was lost: r fast retransmit: resend segment before timer expires Fast Retransmit

3-4 TCP Flow Control r receive side of TCP connection has a receive buffer: r app process may be slow at reading from buffer r speed-matching service: matching the send rate to the receiving app’s drain rate sender won’t overflow receiver’s buffer by transmitting too much, too fast flow control (Suppose TCP receiver discards out-of-order segments) r spare room in buffer = RcvWindow = RcvBuffer-[LastByteRcvd - LastByteRead] r Rcvr advertises spare room by including value of RcvWindow in segments r Sender limits unACKed data to RcvWindow m guarantees receive buffer doesn’t overflow m Ensure lastByteSent – LastByteAcked <= RcvWindow

3-5 TCP Connection Management Recall: TCP sender, receiver establish “connection” before exchanging data segments r initialize TCP variables: m seq. #s  buffers, flow control info (e.g. RcvWindow ) r client: connection initiator Socket clientSocket = new Socket("hostname","port number"); r server: contacted by client Socket connectionSocket = welcomeSocket.accept(); Three way handshake: Step 1: client host sends TCP SYN segment to server m specifies initial seq # m no data Step 2: server host receives SYN, replies with SYNACK segment m server allocates buffers m specifies server initial seq. # Step 3: client receives SYNACK, replies with ACK segment, which may contain data

3-6 TCP Connection Management (cont.) 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. Step 3: client receives FIN, replies with ACK. m Enters “timed wait” - will respond with ACK to received FINs Step 4: server, receives ACK. Connection closed. client FIN server ACK FIN close closed timed wait

3-7 Principles of Congestion Control Congestion: r informally: “too many sources sending too much data too fast for network to handle” r different from flow control! r manifestations: r lost packets (buffer overflow at routers) r long delays (queueing in router buffers) unlimited shared output link buffers Host A in : original data Host B ou t r two senders, two receivers r one router, infinite buffers r no retransmission r large delays when congested Causes/costs of congestion: scenario 1 r one router, finite buffers r sender retransmission of lost packet r always: (goodput) r “perfect” retransmission only when loss: r retransmission of delayed (not lost) packet makes larger (than perfect case) for same Causes/costs of congestion: scenario 2 finite shared output link buffers Host A in : original data Host B out ' in : original data, plus retransmitted data Scenario 3 : r four senders r multihop paths r timeout/retransmit

3-8 Approaches towards congestion control End-end congestion control: r no explicit feedback from network r congestion inferred from end-system observed loss, delay r approach taken by TCP Network-assisted congestion control: r routers provide feedback to end systems m single bit indicating congestion (SNA, DECbit, TCP/IP ECN, ATM) m explicit rate sender should send at Two broad approaches towards congestion control: Another “cost” of congestion: r when packet dropped, any “upstream transmission capacity used for that packet was wasted!

3-9 TCP congestion control: additive increase, multiplicative decrease r Approach: increase transmission rate (window size), probing for usable bandwidth, until loss occurs m additive increase: increase CongWin by 1 MSS every RTT until loss detected m multiplicative decrease: cut CongWin in half after loss r sender limits transmission: r LastByteSent-LastByteAcked r  CongWin r Roughly, r CongWin is dynamic, function of perceived network congestion rate = CongWin RTT Bytes/sec How does sender perceive congestion? r loss event = timeout or 3 duplicate acks  TCP sender reduces rate ( CongWin ) after loss event three mechanisms: r AIMD r slow start r conservative after timeout events

3-10 TCP Slow Start  When connection begins, CongWin = 1 MSS m Example: MSS = 500 bytes & RTT = 200 msec m initial rate = 20 kbps r available bandwidth may be >> MSS/RTT m desirable to quickly ramp up to respectable rate r When connection begins, increase rate exponentially until first loss event:  double CongWin every RTT  done by incrementing CongWin for every ACK received r Summary: initial rate is slow but ramps up exponentially fast Host A one segment RTT Host B time two segments four segments

3-11 Summary: TCP Congestion Control  When CongWin is below Threshold, sender in slow-start phase, window grows exponentially.  When CongWin is above Threshold, sender is in congestion-avoidance phase, window grows linearly.  When a triple duplicate ACK occurs, Threshold set to CongWin/2 and CongWin set to Threshold.  When timeout occurs, Threshold set to CongWin/2 and CongWin is set to 1 MSS.