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TCP. 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
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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 startsat 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. 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
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
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 “smoother” use several recent measurements, not just current SampleRTT TCP Round Trip Time and Timeout Computer Science, FSU
Setting the timeout EstimtedRTT plus “safety margin” Timeout is doubled every time a segment is timeout 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| Computer Science, FSU
TCP flow/congestion control • Sometimes sender shouldn’t send a pkt whenever its ready • Receiver not ready (e.g., buffers full) • React to congestion • Many unACK’ed pkts, may mean long end-end delays, congested networks • Network itself may provide sender with congestion indication • Avoid congestion • Sender transmits smoothly to avoid temporary network overloads
TCP • To react to these, TCP has only one knob – the size of the send window • Reduce or increase the size of the send window • In our project, the size is fixed • The size of the send window is determined by two things: • The size of the receiver window the receiver told him in the TCP segment • His own perception about the level of congestion in the network
flow control TCP Flow Control • 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 sender won’t overrun receiver’s buffers by transmitting too much, too fast RcvBuffer= size of TCP Receive Buffer RcvWindow = amount of spare room in Buffer receiver buffering
What is Congestion? • Informally: “too many sources sending too much data too fast for network to handle” • Different from flow control, caused by the network not by the receiver • How does the sender know whether there is congestion? Manifestations: • Lost packets (buffer overflow at routers) • Long delays (queuing in router buffers)
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
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:
TCP Congestion Control • Idea • Each source determines network capacity for itself • Uses implicit feedback, adaptive congestion window • ACKs pace transmission (self-clocking) • Challenge • Determining the available capacity in the first place • Adjusting to changes in the available capacity
Additive Increase/Multiplicative Decrease • Objective: Adjust to changes in available capacity • A state variable per connection: CongWin • Limit how much data source has is in transit • MaxWin = MIN(RcvWindow, CongWin) • Algorithm • Increase CongWin when congestion goes down (no losses) • Increment CongWin by 1 pkt per RTT (linear increase) • Decrease CongWin when congestion goes up (timeout) • Divide CongWin by 2 (multiplicative decrease)
Window-based, implicit, end-end control 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: Computer Science, FSU
two “phases” slow start congestion avoidance important variables: Congwin threshold: defines threshold between slow start phase and congestion avoidance 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
Why Slow Start? • Objective • Determine the available capacity in the first place • Idea • Begin with congestion window = 1 pkt • Double congestion window each RTT • Increment by 1 packet for each ack • Exponential growth but slower than one blast • Used when • First starting connection • Connection goes dead waiting for a timeout
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 one segment RTT initialize: Congwin = 1 for (each segment ACKed) Congwin++ until (loss event OR CongWin > threshold) two segments four segments
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
Fairness goal: if N TCP sessions share same bottleneck link, each should get 1/N of link capacity TCP connection 1 bottleneck router capacity R TCP connection 2 TCP Fairness
Two competing sessions: Additive increase gives slope of 1, as throughput increases multiplicative decrease decreases throughput proportionally Why is TCP fair? equal bandwidth share R However, TCP is not perfectly fair. It biases towards flows with small RTT. Connection 2 throughput loss: decrease window by factor of 2 congestion avoidance: additive increase Connection 1 throughput R