an introduction to computer networks

Univ. of Tehran Introduction to Computer Network

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An IntroductionAn Introduction to

Computer NetworksComputer Networks

University of TehranDept. of EE and Computer Engineering

By:Dr. Nasser Yazdani

Lecture 11: Congestion Congestion ControlControl


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Outline

Allocating resource among competing users. Bandwidth on the line Buffers on the routers

Congestion control and resource allocation, two side of the same coin.

Different congestion control Policies. Reacting to Congestion Avoiding Congestion


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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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Issues Two sides of the same coin

pre-allocate resources so to avoid congestion control congestion if (and when) is occurs

Two points of implementation hosts at the edges of the network (transport protocol) routers inside the network (queuing discipline)

Underlying service model best-effort (assume for now) multiple qualities of service (later)

Destination1.5-Mbps T1 link

Router

Source2

Source1

100-Mbps FDDI

10-Mbps Ethernet


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Why is Congestion Bad? Wasted bandwidth: retransmission of

dropped packets. Poor user service : unpredictable delay,

reduced throughput. Increased load can even result in lower network throughput. » Switched nets: heavy traffic -> long queues ->

lost packets ->retransmits » Ethernet: high demand -> many collisions » compare with highways: too much traffic slows

down throughput Solutions? Redesign the network.

» Add capacity to congested links » Reroute traffic. » What are the options?


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EvaluationWe need to know how good is a

resource allocation/congestion aviodness mechanism. Fairness Power of network (ratio of throughput to

delay), Maximize this ratio. Distributedness Efficiency Cost

Optimal load Load

Thr

ough

put/d

elay


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Evaluation (Fairness) What is fair resource allocation?

Equal share of bandwidth How about the length of paths?

Given a set of flow throughput (x1,x2,…, xn)

f(x1,x2,…, xn)= (Σi=1n xi)2/n Σi=1

nxi2

Fairness is always between 0, 1 and 1 completely fair and 0 completely unfair.


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Possible Solutions Redesign the network.

» Add capacity to congested links» Very slow solution: takes days to months!

Reroute traffic.» Alternate paths are not always available» Also too slow: takes 10s of seconds» In practice, most routing algorithms are

static Adjust the load in the network. Load

balancing» What are the options?


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Principles of Congestion Control

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 (queuing in router buffers)

a top-10 problem!


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Causes/costs of congestion: scenario 1

two senders, two receivers

one router, infinite buffers

no retransmission large delays

when congested maximum

achievable throughput


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Another scenario four senders multihop paths timeout/retransmit

in

Q: what happens as and increase ?

in


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Causes/costs of congestion: scenario 3

Another “cost” of congestion: when packet dropped, any “upstream transmission capacity

used for that packet was wasted!


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Framework Connectionless flows

sequence of packets sent between source/destination pair

maintain soft state at the routers

Router

Source2

Source1

Source3

Router

Router

Destination2

Destination1


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Framework (cont) Taxonomy: different approaches for congestion Ctrl.

router-centric versus host-centric: Router-Centric: each router takes the responsibility for dropping

packets, and informing the generating host the amount of traffic allowed to be sent.

Host-Centric: the end hosts observe the amount of traffic that is successfully getting through the network and adjust their transmission rates accordingly.

reservation-based versus feedback-based Reservation-Based: the end host asks the network for a certain

amount of bandwidth when requesting a connection (or flow). If the network does not have enough bandwidth it will reject the connection.

Feedback-Based: the end hosts send data without first reserving any capacity and then adjust their sending rate according to the feedback they received

Explicit feedback Implicit feedback.


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Framework (cont) window-based versus rate-based:

Window-Based: the receiver advertises a window to the sender. The window corresponds to how much buffer space the receiver has and it limits how much data the sender can transmit. Network also can do that, like X.25.

Rate-Based: How many bit the sender can send or the network can absorb.

Rate-Based can support video. Rate-Based still is an open problem


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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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What’s Really Happening? Knee – point after

which throughput increases

very slow delay increases fast

Cliff – point after which throughput starts to

decrease very fast to zero (congestion collapse)

delay approaches infinity

Load

Load

Th

rou

ghp

ut

De

lay

knee cliff

congestioncollapse

packetloss


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Goals

Operate near the knee point Remain in equilibrium How to maintain equilibrium?

Don’t put a packet into network until another packet leaves. How do you do it?

Use ACK: send a new packet only after you receive and ACK (self-clocking)

This maintains the number of packets in network “constant”


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Self-clocking

PrPb

Ar

Ab

ReceiverSender

As


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How Do You Do It? Detect when network

approaches/reaches knee point Stay there

Questions How do you get there? What if you overshoot (i.e., go over knee

point) ? Possible solution:

Increase window size until you notice congestion

Decrease window size if network congested


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TCP Congestion Control Idea

assumes best-effort network (FIFO or FQ routers)each source determines network capacity for itself

uses implicit feedback ACKs pace transmission (self-clocking)

Challenge determining the available capacity in the

first place adjusting to changes in the available

capacity


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Additive Increase/Multiplicative Decrease

Objective: adjust to changes in the available capacity

New state variable per connection: CongestionWindow (cwnd)

Idea: increase Cwnd when congestion goes down decrease Cwnd when congestion goes up

EffectiveWindow = MaxWindow – (LastByteSent – LastByteAcked)

MaxWindow = min(cwnd, AdvertisedWindow)

LastByteAcked LastByteSent

sequence number increases


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AIMD (cont)

Question: how does the source determine whether or not the network is congested?

Answer: a timeout occurs timeout signals that a packet was lost packets are seldom lost due to transmission error lost packet implies congestion


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AIMD (cont)

In practice: increment a little for each ACKIncrement = MSS * (MSS/cwnd)cwnd += Increment

MSS= Maximum segment size

Source Destination

…

Algorithm increment Cwnd by one packet

per RTT (linear or additive increase)

divide Cwnd by two whenever a timeout occurs (multiplicative decrease)


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AIMD (cont) Trace: saw tooth behavior

60

20

1.0 2.0 3.0 4.0 5.0 6.0 7.0 8.0 9.0

KB

Time (seconds)

70

304050

10

10.0

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TCP: Slow Start Question? How much is the initial cwnd size?

Small cwnd implies less network utilization Big cwnd means congestion again What is optimal of cwnd.

Solution: Start with cwnd =1 and Quickly increase cwnd until network congested get a rough estimate of the optimal of cwnd

Each time a segment is acknowledged increment cwnd by one (cwnd++)

Increase of cwnd is exponential


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Slow Start Example

TCP slows down the increase of cwnd when cwnd >=ssthresh

Ssthresh - slow start threshold value.

After ssthresh, TCP change from slow start to congestion avoidance!

ACK for segment 1

segment 1cwnd = 1

cwnd = 2 segment 2segment 3

ACK for segments 2 + 3

cwnd = 4 segment 4segment 5segment 6segment 7

ACK for segments 4+5+6+7

cwnd = 8

source destination


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Slow Start/Congestion Avoidance Example

Assume that ssthresh = 8

Roundtrip times

Cw

nd (

in s

egm

ents

)

ssthresh

cwnd = 1

cwnd = 2

cwnd = 4

cwnd = 8

cwnd = 9

cwnd = 10

Client Server

0

2

4

6

8

10

12

14

t=0

t=2

t=4

t=6

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Putting Everything Together:TCP PseudocodeInitially:cwnd = 1;ssthresh = infinite;

New ack received:if (cwnd < ssthresh) /* Slow Start*/ cwnd = cwnd + 1;else /* Congestion Avoidance */ cwnd = cwnd + 1/cwnd;

Timeout:/* Multiplicative decrease */ssthresh = win/2;cwnd = 1; /* Again slow start */

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The big picture

Time

cwnd

Timeout

Slow Start

CongestionAvoidance


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Slow Start (cont) Exponential growth, but slower than all at once Used…

when first starting connection when connection goes dead waiting for timeout

Trace

Problem: lose up to half a Cwnd’s worth of data

60

20

1.0 2.0 3.0 4.0 5.0 6.0 7.0 8.0 9.0

KB

70

304050

10


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Packet Loss Detection Wait for Retransmission Time Out (RTO) What’s the problem with this? Because RTO is performance killer In BSD TCP implementation, RTO is

usually more than 1 second the granularity of RTT estimate is 500 ms retransmission timeout is at least two times

of RTT Solution: Don’t wait for RTO to expire


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Fast Retransmit Resend a segment

after 3 duplicate ACKs

Remember, a duplicate ACK means that an out-of sequence segment was received

Notes: Duplicate ACKs due

packet reordering! If window is small

won’t get duplicate ACKs!

ACK 2

segment 1cwnd = 1

cwnd = 2 segment 2segment 3

ACK 4cwnd = 4 segment 4

segment 5segment 6segment 7

ACK 3

3 duplicateACKs

ACK 4ACK 4

• Set cwnd to 1 after each retransmit!!

source destination


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Results

Compare to slow start, it avoids long time out waiting in 4.0 – 5.0 period.

How avoid slow start at each fast retransmission?

60

20

1.0 2.0 3.0 4.0 5.0 6.0 7.0

KB

70

304050

10


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Fast Recovery After a fast-retransmit set cwnd to

ssthresh/2 i.e., don’t reset cwnd to 1, then, avoid

slow start again!. But when RTO expires still do cwnd

= 1 Fast Retransmit and Fast Recovery implemented by TCP Reno; most widely used version of TCP today

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Fast Retransmit and Fast Recovery

Prevent expensive timeouts No need to slow start again At steady state, cwnd oscillates

around the optimal window size.

Time

cwnd

Slow Start

CongestionAvoidance

Fast retransmit


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Congestion Control Summary

Architecture: end system detects congestion and slow down

Starting point: Slow start/congestion avoidance

packet drop detected by retransmission timeout RTO as congestion signal

Fast retransmission/fast recovery packet drop detected by (three) duplicate acks

Router support Binary feedback scheme: explicit signaling

Today Explicit Congestion Notification [RF99]


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Congestion Control vs. Congestion Avoidance

Congestion control goal Stay left of cliff

Congestion avoidance goal Stay left of knee

Load

Th

rou

ghp

ut knee cliff

congestioncollapse


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Congestion Avoidance TCP’s strategy

control congestion once it happens repeatedly increase load in an effort to find the point

at which congestion occurs, and then back off Alternative strategy

predict when congestion is about to happen reduce rate before packets start being discarded call this congestion avoidance, instead of congestion

control Two possibilities

router-centric: DECbit and RED Gateways host-centric: TCP Vegas


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DECbit Add binary congestion bit to each packet header Router

monitors average queue length over last busy+idle cycle

set congestion bit if average queue length > 1 attempts to balance throughout against delay

Queue length

Currenttime

TimeCurrent

cyclePrevious

cycleAveraginginterval


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End Hosts Destination echoes bit back to source Source records how many packets

resulted in set bit If less than 50% of last window’s worth

had bit set increase Cwnd by 1 packet

If 50% or more of last window’s worth had bit set decrease CongestionWindow by 0.875 times


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Random Early Detection (RED)

Notification is implicit just drop the packet (TCP will timeout) could make explicit by marking the packet

Early random drop rather than wait for queue to become full,

drop each arriving packet with some drop probability whenever the queue length exceeds some drop level


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RED Details Compute average queue length AvgLen

AvgLen = (1 - Weight) * AvgLen + Weight * SampleLen

0 < Weight < 1 (usually 0.002)SampleLen is queue length each time a packet

arrivesMaxThreshold MinThreshold

AvgLen

Queue


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RED Details (cont)

Two queue length thresholdsif AvgLen <= MinThreshold then

enqueue the packetif MinThreshold < AvgLen < MaxThreshold then

calculate probability P drop arriving packet with

probability Pif ManThreshold <= AvgLen then

drop arriving packet


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RED Details (cont) Computing probability P

TempP = MaxP * (AvgLen - MinThreshold)/ (MaxThreshold - MinThreshold)

P = TempP/(1 - count * TempP)

Drop Probability Curve P(drop)

1.0

MaxP

MinThresh MaxThresh

AvgLen

Count is the # of queued fromlast drop.


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Tuning RED P of a particular flow’s packet(s) is roughly

proportional to the share of the flow’s bandwidth. MaxP is typically 0.02, meaning when is halfway

between the two thresholds, router drops roughly one out of 50 packets.

If traffic is bursty, then MinTh. should be sufficiently large to allow link utilization to be acceptably high.

Difference between two thresholds should be larger than the typical increase in the calculated average queue length in one RTT; setting MaxThreshold to twice MinThreshold is reasonable.

Penalty Box for Offenders


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Sourced-based Cong. Avoidance

End hosts adapt traffic before any lost in the net. Watch for router queue’s lengths by checking RTT.

A collection of related algorithms In each 2 RTT check if RTT > (minRTT + maxRTT)/2 ->

decrease cwnd, cwnd -= cwnd/8. Increase otherwise. Check (currentWind –oldWind)x (CurrentRTT- oldRTT), if

the result is positive, decrease cwnd, cwnd -= cwnd/8. Increase 1 otherwise.

Check flattening of sending rate. Increase Cwnd by 1 in each RTT, compare throughput with previous one, if it is less than half of one packet, decrease cwnd by one.

TCP Vegas is like the last one with a difference to compare to expected rate.


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TCP Vegas Idea: source watches for some sign that

router’s queue is building up and congestion will happen too; e.g., RTT grows sending rate flattens

60

20

0.5 1.0 1.5 4.0 4.5 6.5 8.0

KB

Time (seconds)

Time (seconds)

70

304050

10

2.0 2.5 3.0 3.5 5.0 5.5 6.0 7.0 7.5 8.5

900

300100

0.5 1.0 1.5 4.0 4.5 6.5 8.0

Sen

ding

KB

ps

1100

500700

2.0 2.5 3.0 3.5 5.0 5.5 6.0 7.0 7.5 8.5

Time (seconds)0.5 1.0 1.5 4.0 4.5 6.5 8.0Q

ueue

siz

e in

rou

ter

5

10

2.0 2.5 3.0 3.5 5.0 5.5 6.0 7.0 7.5 8.5

•Top is cwn size•Middle is ave. rate in source•Bottom is ave. queue length in the bottleneck router•In Shade, Cwnd increase, but ave. rate stay flat.


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TCP Vegas Keep track of extra data in the network.

Extra data is the amount more than available bandwidth.

Maintain “Right” amount of extra data.


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Algorithm Let BaseRTT be the minimum of all measured

RTTs (commonly the RTT of the first packet) If not overflowing the connection, then

ExpectRate = Cwnd/BaseRTT Source calculates sending rate (ActualRate)

once per RTT Source compares ActualRate with ExpectRateDiff = ExpectedRate - ActualRateif Diff <

increase Cwnd linearlyelse if Diff >

decrease Cwnd linearlyelse

leave Cwnd unchanged


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Algorithm (cont) Parameters

= 1 packet = 3 packets

Top Cwn size

Even faster retransmit keep fine-grained timestamps for each packet check for timeout on first duplicate ACK

70605040302010

KB

Time (seconds)

0.5 1.0 1.5 2.0 2.5 3.0 3.5 4.0 4.5 5.0 5.5 6.0 6.5 7.0 7.5 8.0

0.5 1.0 1.5 2.0 2.5 3.0 3.5 4.0 4.5 5.0 5.5 6.0 6.5 7.0 7.5 8.0

CA

M K

Bps

240200160120

8040

Time (seconds)

• black line is Actual rate•Colored expected rate•Shade area is and

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