15-744: computer networking l-4 tcp. l -4; 10-7-04© srinivasan seshan, 20042 tcp basics tcp...
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15-744: Computer Networking
L-4 TCP
L -4; 10-7-04© Srinivasan Seshan, 2004 2
TCP Basics
• TCP reliability• Congestion control basics• TCP congestion control• Assigned reading
• [JK88] Congestion Avoidance and Control• [CJ89] Analysis of the Increase and Decrease
Algorithms for Congestion Avoidance in Computer Networks
• [FF96] Simulation-based Comparisons of Tahoe, Reno, and SACK TCP
• [FHPW00] Equation-Based Congestion Control for Unicast Applications
L -4; 10-7-04© Srinivasan Seshan, 2004 3
Key Things You Should Know Already
• Port numbers• TCP/UDP checksum• Sliding window flow control
• Sequence numbers
• TCP connection setup
L -4; 10-7-04© Srinivasan Seshan, 2004 4
Overview
• TCP reliability: timer-driven
• TCP reliability: data-driven
• Congestion sources and collapse
• Congestion control basics
• TCP congestion control
• TCP modeling
L -4; 10-7-04© Srinivasan Seshan, 2004 5
Introduction to TCP• Communication abstraction:
• Reliable• Ordered• Point-to-point• Byte-stream• Full duplex• Flow and congestion controlled
• Protocol implemented entirely at the ends• Fate sharing
• Sliding window with cumulative acks• Ack field contains last in-order packet received• Duplicate acks sent when out-of-order packet received
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Evolution of TCP
1975 1980 1985 1990
1982TCP & IP
RFC 793 & 791
1974TCP described by
Vint Cerf and Bob KahnIn IEEE Trans Comm
1983BSD Unix 4.2
supports TCP/IP
1984Nagel’s algorithmto reduce overhead
of small packets;predicts congestion
collapse
1987Karn’s algorithmto better estimate
round-trip time
1986Congestion
collapseobserved
1988Van Jacobson’s
algorithmscongestion avoidance and congestion control(most implemented in
4.3BSD Tahoe)
19904.3BSD Renofast retransmitdelayed ACK’s
1975Three-way handshake
Raymond TomlinsonIn SIGCOMM 75
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TCP Through the 1990s
1993 1994 1996
1994ECN
(Floyd)Explicit
CongestionNotification
1993TCP Vegas
(Brakmo et al)real congestion
avoidance
1994T/TCP
(Braden)Transaction
TCP
1996SACK TCP(Floyd et al)
Selective Acknowledgement
1996Hoe
Improving TCP startup
1996FACK TCP
(Mathis et al)extension to SACK
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What’s Different From Link Layers?
• Logical link vs. physical link• Must establish connection
• Variable RTT• May vary within a connection
• Reordering• How long can packets live max segment lifetime
• Can’t expect endpoints to exactly match link• Buffer space availability
• Transmission rate• Don’t directly know transmission rate
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Timeout-based Recovery
• Wait at least one RTT before retransmitting• Importance of accurate RTT estimators:
• Low RTT unneeded retransmissions• High RTT poor throughput
• RTT estimator must adapt to change in RTT• But not too fast, or too slow!
• Spurious timeouts• “Conservation of packets” principle – more than
a window worth of packets in flight
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Initial Round-trip Estimator
• Round trip times exponentially averaged:• New RTT = (old RTT) + (1 - ) (new sample)• Recommended value for : 0.8 - 0.9
• 0.875 for most TCP’s
• Retransmit timer set to RTT, where = 2• Every time timer expires, RTO exponentially backed-off• Like Ethernet
• Not good at preventing spurious timeouts
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Jacobson’s Retransmission Timeout
• Key observation:• At high loads round trip variance is high
• Solution:• Base RTO on RTT and standard deviation or
RRTT• rttvar = * dev + (1- )rttvar
• dev = linear deviation • Inappropriately named – actually smoothed linear
deviation
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Retransmission Ambiguity
A B
ACK
SampleRTT
Original transmission
retransmission
RTO
A B
Original transmission
retransmissionSampleRTT
ACKRTOX
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Karn’s RTT Estimator
• Accounts for retransmission ambiguity• If a segment has been retransmitted:
• Don’t count RTT sample on ACKs for this segment
• Keep backed off time-out for next packet• Reuse RTT estimate only after one successful
transmission
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Timestamp Extension
• Used to improve timeout mechanism by more accurate measurement of RTT
• When sending a packet, insert current timestamp into option• 4 bytes for seconds, 4 bytes for microseconds
• Receiver echoes timestamp in ACK• Actually will echo whatever is in timestamp
• Removes retransmission ambiguity• Can get RTT sample on any packet
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Timer Granularity
• Many TCP implementations set RTO in multiples of 200,500,1000ms
• Why?• Avoid spurious timeouts – RTTs can vary
quickly due to cross traffic• Make timers interrupts efficient
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Delayed ACKS
• Problem:• In request/response programs, you send
separate ACK and Data packets for each transaction
• Solution:• Don’t ACK data immediately• Wait 200ms (must be less than 500ms – why?)• Must ACK every other packet• Must not delay duplicate ACKs
L -4; 10-7-04© Srinivasan Seshan, 2004 17
Overview
• TCP reliability: timer-driven
• TCP reliability: data-driven
• Congestion sources and collapse
• Congestion control basics
• TCP congestion control
• TCP modeling
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TCP Flavors
• Tahoe, Reno, Vegas differ in data-driven reliability
• TCP Tahoe (distributed with 4.3BSD Unix)• Original implementation of Van Jacobson’s
mechanisms (VJ paper)• Includes:
• Slow start • Congestion avoidance• Fast retransmit
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Fast Retransmit
• What are duplicate acks (dupacks)?• Repeated acks for the same sequence
• When can duplicate acks occur?• Loss• Packet re-ordering• Window update – advertisement of new flow control
window• Assume re-ordering is infrequent and not of large
magnitude• Use receipt of 3 or more duplicate acks as indication of
loss• Don’t wait for timeout to retransmit packet
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Fast Retransmit
Time
Sequence NoDuplicate Acks
RetransmissionX
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Multiple Losses
Time
Sequence No Duplicate Acks
RetransmissionX
X
XX
Now what?
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Time
Sequence NoX
X
XX
Tahoe
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TCP Reno (1990)
• All mechanisms in Tahoe• Addition of fast-recovery
• Opening up congestion window after fast retransmit
• Delayed acks• Header prediction
• Implementation designed to improve performance• Has common case code inlined
• With multiple losses, Reno typically timeouts because it does not receive enough duplicate acknowledgements
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Reno
Time
Sequence NoX
X
XX
Now what? timeout
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NewReno
• The ack that arrives after retransmission (partial ack) should indicate that a second loss occurred
• When does NewReno timeout?• When there are fewer than three dupacks for
first loss• When partial ack is lost
• How fast does it recover losses?• One per RTT
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NewReno
Time
Sequence NoX
X
XX
Now what? partial ackrecovery
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SACK
• Basic problem is that cumulative acks provide little information• Ack for just the packet received
• What if acks are lost? carry cumulative also• Not used
• Bitmask of packets received • Selective acknowledgement (SACK)
• How to deal with reordering
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SACK
Time
Sequence NoX
X
XX
Now what? – sendretransmissions as soonas detected
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Performance Issues
• Timeout >> fast rexmit• Need 3 dupacks/sacks• Not great for small transfers
• Don’t have 3 packets outstanding
• What are real loss patterns like?
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Overview
• TCP reliability: timer-driven
• TCP reliability: data-driven
• Congestion sources and collapse
• Congestion control basics
• TCP congestion control
• TCP modeling
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Congestion
• Different sources compete for resources inside network
• Why is it a problem?• Sources are unaware of current state of resource• Sources are unaware of each other• In many situations will result in < 1.5 Mbps of
throughput (congestion collapse)
10 Mbps
100 Mbps
1.5 Mbps
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Causes & Costs of Congestion
• Four senders – multihop paths• Timeout/retransmit
Q: What happens as rate increases?
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Causes & Costs of Congestion
• When packet dropped, any “upstream transmission capacity used for that packet was wasted!
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Congestion Collapse
• Definition: Increase in network load results in decrease of useful work done
• Many possible causes• Spurious retransmissions of packets still in flight
• Classical congestion collapse• How can this happen with packet conservation• Solution: better timers and TCP congestion control
• Undelivered packets• Packets consume resources and are dropped elsewhere in
network• Solution: congestion control for ALL traffic
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Other Congestion Collapse Causes
• Fragments• Mismatch of transmission and retransmission units• Solutions
• Make network drop all fragments of a packet (early packet discard in ATM)
• Do path MTU discovery
• Control traffic• Large percentage of traffic is for control
• Headers, routing messages, DNS, etc.
• Stale or unwanted packets• Packets that are delayed on long queues• “Push” data that is never used
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Where to Prevent Collapse?
• Can end hosts prevent problem?• Yes, but must trust end hosts to do right thing• E.g., sending host must adjust amount of data it
puts in the network based on detected congestion
• Can routers prevent collapse?• No, not all forms of collapse• Doesn’t mean they can’t help • Sending accurate congestion signals• Isolating well-behaved from ill-behaved sources
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Congestion Control and Avoidance
• A mechanism which:• Uses network resources efficiently• Preserves fair network resource allocation• Prevents or avoids collapse
• Congestion collapse is not just a theory• Has been frequently observed in many
networks
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Overview
• TCP reliability: timer-driven
• TCP reliability: data-driven
• Congestion sources and collapse
• Congestion control basics
• TCP congestion control
• TCP modeling
L -4; 10-7-04© Srinivasan Seshan, 2004 39
Objectives
• Simple router behavior • Distributedness
• Efficiency: Xknee = xi(t)
• Fairness: (xi)2/n(xi2)
• Power: (throughput/delay)• Convergence: control system must be
stable
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Basic Control Model
• Let’s assume window-based control• Reduce window when congestion is
perceived• How is congestion signaled?
• Either mark or drop packets• When is a router congested?
• Drop tail queues – when queue is full• Average queue length – at some threshold
• Increase window otherwise• Probe for available bandwidth – how?
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Linear Control
• Many different possibilities for reaction to congestion and probing• Examine simple linear controls• Window(t + 1) = a + b Window(t)• Different ai/bi for increase and ad/bd for
decrease• Supports various reaction to signals
• Increase/decrease additively• Increased/decrease multiplicatively• Which of the four combinations is optimal?
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Phase plots
• Simple way to visualize behavior of competing connections over time
Efficiency Line
Fairness Line
User 1’s Allocation x1
User 2’s Allocation
x2
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Phase plots
• What are desirable properties?• What if flows are not equal?
Efficiency Line
Fairness Line
User 1’s Allocation x1
User 2’s Allocation
x2Optimal point
Overload
Underutilization
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Additive Increase/Decrease
T0
T1
Efficiency Line
Fairness Line
User 1’s Allocation x1
User 2’s Allocation
x2
• Both X1 and X2 increase/decrease by the same amount over time• Additive increase improves fairness and additive
decrease reduces fairness
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Multiplicative Increase/Decrease
• Both X1 and X2 increase by the same factor over time• Extension from origin – constant fairness
T0
T1
Efficiency Line
Fairness Line
User 1’s Allocation x1
User 2’s Allocation
x2
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Convergence to Efficiency
xH
Efficiency Line
Fairness Line
User 1’s Allocation x1
User 2’s Allocation
x2
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Distributed Convergence to Efficiency
xH
Efficiency Line
Fairness Line
User 1’s Allocation x1
User 2’s Allocation
x2
a=0b=1
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Convergence to Fairness
xH
Efficiency Line
Fairness Line
User 1’s Allocation x1
User 2’s Allocation
x2
xH’
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Convergence to Efficiency & Fairness
xH
Efficiency Line
Fairness Line
User 1’s Allocation x1
User 2’s Allocation
x2
xH’
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Increase
Efficiency Line
Fairness Line
User 1’s Allocation x1
User 2’s Allocation
x2
xL
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Constraints
• Distributed efficiency• I.e., Window(t+1) > Window(t) during
increase• ai > 0 & bi ≥ 1• Similarly, ad < 0 & bd ≤ 1
• Must never decrease fairness• a & b’s must be ≥ 0• ai/bi > 0 and ad/bd ≥ 0
• Full constraints• ad = 0, 0 ≤ bd < 1, ai > 0 and bi ≥ 1
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What is the Right Choice?
• Constraints limit us to AIMD• Can have multiplicative term in increase (MAIMD)• AIMD moves towards optimal point
x0
x1
x2
Efficiency Line
Fairness Line
User 1’s Allocation x1
User 2’s Allocation
x2
L -4; 10-7-04© Srinivasan Seshan, 2004 53
Overview
• TCP reliability: timer-driven
• TCP reliability: data-driven
• Congestion sources and collapse
• Congestion control basics
• TCP congestion control
• TCP modeling
L -4; 10-7-04© Srinivasan Seshan, 2004 54
TCP Congestion Control
• Motivated by ARPANET congestion collapse• Underlying design principle: packet conservation
• At equilibrium, inject packet into network only when one is removed
• Basis for stability of physical systems
• Why was this not working?• Connection doesn’t reach equilibrium• Spurious retransmissions• Resource limitations prevent equilibrium
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TCP Congestion Control - Solutions
• Reaching equilibrium• Slow start
• Eliminates spurious retransmissions• Accurate RTO estimation• Fast retransmit
• Adapting to resource availability• Congestion avoidance
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TCP Congestion Control
• Changes to TCP motivated by ARPANET congestion collapse
• Basic principles• AIMD• Packet conservation• Reaching steady state quickly• ACK clocking
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AIMD
• Distributed, fair and efficient• Packet loss is seen as sign of congestion and
results in a multiplicative rate decrease • Factor of 2
• TCP periodically probes for available bandwidth by increasing its rate
Time
Rate
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Implementation Issue
• Operating system timers are very coarse – how to pace packets out smoothly?
• Implemented using a congestion window that limits how much data can be in the network.• TCP also keeps track of how much data is in transit
• Data can only be sent when the amount of outstanding data is less than the congestion window.• The amount of outstanding data is increased on a “send” and
decreased on “ack”• (last sent – last acked) < congestion window
• Window limited by both congestion and buffering• Sender’s maximum window = Min (advertised window, cwnd)
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Congestion Avoidance
• If loss occurs when cwnd = W• Network can handle 0.5W ~ W segments• Set cwnd to 0.5W (multiplicative decrease)
• Upon receiving ACK• Increase cwnd by (1 packet)/cwnd
• What is 1 packet? 1 MSS worth of bytes• After cwnd packets have passed by
approximately increase of 1 MSS
• Implements AIMD
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Congestion Avoidance Sequence Plot
Time
Sequence No
Packets
Acks
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Congestion Avoidance Behavior
Time
CongestionWindow
Packet loss+ Timeout
Grabbingback
Bandwidth
CutCongestion
Windowand Rate
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Packet Conservation
• At equilibrium, inject packet into network only when one is removed• Sliding window and not rate controlled• But still need to avoid sending burst of packets
would overflow links• Need to carefully pace out packets• Helps provide stability
• Need to eliminate spurious retransmissions• Accurate RTO estimation• Better loss recovery techniques (e.g. fast retransmit)
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TCP Packet Pacing
• Congestion window helps to “pace” the transmission of data packets
• In steady state, a packet is sent when an ack is received• Data transmission remains smooth, once it is smooth• Self-clocking behavior
Pr
Pb
ArAb
ReceiverSender
As
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Reaching Steady State
• Doing AIMD is fine in steady state but slow…
• How does TCP know what is a good initial rate to start with?• Should work both for a CDPD (10s of Kbps or
less) and for supercomputer links (10 Gbps and growing)
• Quick initial phase to help get up to speed (slow start)
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Slow Start Packet Pacing
• How do we get this clocking behavior to start?• Initialize cwnd = 1• Upon receipt of every
ack, cwnd = cwnd + 1• Implications
• Window actually increases to W in RTT * log2(W)
• Can overshoot window and cause packet loss
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Slow Start Example
1
One RTT
One pkt time
0R
2
1R
3
4
2R
567
83R
91011
1213
1415
1
2 3
4 5 6 7
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Slow Start Sequence Plot
Time
Sequence No
.
.
.
Packets
Acks
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Return to Slow Start
• If packet is lost we lose our self clocking as well• Need to implement slow-start and congestion
avoidance together
• When timeout occurs set ssthresh to 0.5w• If cwnd < ssthresh, use slow start• Else use congestion avoidance
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TCP Saw Tooth Behavior
Time
CongestionWindow
InitialSlowstart
Fast Retransmit
and Recovery
Slowstartto pacepackets
Timeoutsmay still
occur
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How to Change Window
• When a loss occurs have W packets outstanding
• New cwnd = 0.5 * cwnd• How to get to new state?
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Fast Recovery
• Each duplicate ack notifies sender that single packet has cleared network
• When < cwnd packets are outstanding• Allow new packets out with each new duplicate
acknowledgement• Behavior
• Sender is idle for some time – waiting for ½ cwnd worth of dupacks
• Transmits at original rate after wait• Ack clocking rate is same as before loss
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Fast Recovery
Time
Sequence NoSent for each dupack after
W/2 dupacks arriveX
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NewReno Changes
• Send a new packet out for each pair of dupacks• Adapt more gradually to new window
• Will not halve congestion window again until recovery is completed • Identifies congestion events vs. congestion
signals
• Initial estimation for ssthresh
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Rate Halving Recovery
Time
Sequence No
Sent after everyother dupack
X
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Delayed Ack Impact
• TCP congestion control triggered by acks• If receive half as many acks window
grows half as fast
• Slow start with window = 1• Will trigger delayed ack timer• First exchange will take at least 200ms• Start with > 1 initial window
• Bug in BSD, now a “feature”/standard
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Overview
• TCP reliability: timer-driven
• TCP reliability: data-driven
• Congestion sources and collapse
• Congestion control basics
• TCP congestion control
• TCP modeling
L -4; 10-7-04© Srinivasan Seshan, 2004 77
TCP Modeling
• Given the congestion behavior of TCP can we predict what type of performance we should get?
• What are the important factors• Loss rate
• Affects how often window is reduced
• RTT• Affects increase rate and relates BW to window
• RTO• Affects performance during loss recovery
• MSS • Affects increase rate
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Overall TCP Behavior
Time
Window
• Let’s concentrate on steady state behavior with no timeouts and perfect loss recovery
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Simple TCP Model
• Some additional assumptions• Fixed RTT• No delayed ACKs
• In steady state, TCP losses packet each time window reaches W packets• Window drops to W/2 packets• Each RTT window increases by 1 packetW/2
* RTT before next loss• BW = MSS * avg window/RTT = MSS * (W +
W/2)/(2 * RTT) = .75 * MSS * W / RTT
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Simple Loss Model
• What was the loss rate?• Packets transferred = (.75 W/RTT) * (W/2 *
RTT) = 3W2/8• 1 packet lost loss rate = p = 8/3W2
• W = sqrt( 8 / (3 * loss rate))
• BW = .75 * MSS * W / RTT• BW = MSS / (RTT * sqrt (2/3p))
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TCP Friendliness
• What does it mean to be TCP friendly?• TCP is not going away• Any new congestion control must compete with TCP
flows• Should not clobber TCP flows and grab bulk of link• Should also be able to hold its own, i.e. grab its fair share, or it
will never become popular
• How is this quantified/shown?• Has evolved into evaluating loss/throughput behavior• If it shows 1/sqrt(p) behavior it is ok• But is this really true?
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TCP Performance
• Can TCP saturate a link?• Congestion control
• Increase utilization until… link becomes congested
• React by decreasing window by 50%• Window is proportional to rate * RTT
• Doesn’t this mean that the network oscillates between 50 and 100% utilization?• Average utilization = 75%??• No…this is *not* right!
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TCP Congestion Control
Only W packets may be outstanding
Rule for adjusting W• If an ACK is received: W ← W+1/W• If a packet is lost: W ← W/2
Source Dest
maxW
2maxW
t
Window size
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Single TCP FlowRouter without buffers
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Summary Unbuffered Link
t
W Minimum window for full utilization
• The router can’t fully utilize the link• If the window is too small, link is not full• If the link is full, next window increase causes drop• With no buffer it still achieves 75% utilization
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TCP Performance
• In the real world, router queues play important role• Window is proportional to rate * RTT
• But, RTT changes as well the window
• Window to fill links = propagation RTT * bottleneck bandwidth
• If window is larger, packets sit in queue on bottleneck link
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TCP Performance
• If we have a large router queue can get 100% utilization• But, router queues can cause large delays
• How big does the queue need to be?• Windows vary from W W/2
• Must make sure that link is always full• W/2 > RTT * BW• W = RTT * BW + Qsize• Therefore, Qsize > RTT * BW
• Ensures 100% utilization• Delay?
• Varies between RTT and 2 * RTT
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Single TCP FlowRouter with large enough buffers for full link utilization
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Summary Buffered Link
t
W
Minimum window for full utilization
• With sufficient buffering we achieve full link utilization• The window is always above the critical threshold• Buffer absorbs changes in window size
• Buffer Size = Height of TCP Sawtooth• Minimum buffer size needed is 2T*C
• This is the origin of the rule-of-thumb
Buffer
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Example
• 10Gb/s linecard• Requires 300Mbytes of buffering.• Read and write 40 byte packet every 32ns.
• Memory technologies• DRAM: require 4 devices, but too slow. • SRAM: require 80 devices, 1kW, $2000.
• Problem gets harder at 40Gb/s• Hence RLDRAM, FCRAM, etc.
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Rule-of-thumb
• Rule-of-thumb makes sense for one flow• Typical backbone link has > 20,000 flows• Does the rule-of-thumb still hold?
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If flows are synchronized
• Aggregate window has same dynamics• Therefore buffer occupancy has same dynamics• Rule-of-thumb still holds.
2maxW
t
max
2
W
maxW
maxW
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If flows are not synchronized
ProbabilityDistribution
B
0
Buffer Size
W
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Central Limit Theorem
• CLT tells us that the more variables (Congestion Windows of Flows) we have, the narrower the Gaussian (Fluctuation of sum of windows)
• Width of Gaussian decreases with
• Buffer size should also decreases with
n
CT
n
BB n
21
n
1
n
1
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Required buffer size
2T C
n
Simulation
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Important Lessons
• How does TCP implement AIMD?• Sliding window, slow start & ack clocking• How to maintain ack clocking during loss recovery
fast recovery
• Modern TCP loss recovery• Why are timeouts bad?• How to avoid them? fast retransmit, SACK
• How does TCP fully utilize a link?• Role of router buffers
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Next Lecture
• TCP Vegas/alternative congestion control schemes• RED• Fair queuing• Core-stateless fair queuing/XCP• Assigned reading
• [BP95] TCP Vegas: End to End Congestion Avoidance on a Global Internet
• [FJ93] Random Early Detection Gateways for Congestion Avoidance
• [DKS90] Analysis and Simulation of a Fair Queueing Algorithm, Internetworking: Research and Experience
• [SSZ98] Core-Stateless Fair Queueing: Achieving Approximately Fair Allocations in High Speed Networks
• [KHR02] Congestion Control for High Bandwidth-Delay Product Networks