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Chapter 6 Congestion Control and Resource

Allocation

Outline6.1 Issues in Resource Allocation6.2 Queuing Discipline6.3 TCP Congestion Control6.4 Congestion Avoidance Mechanisms6.5 Quality of Service

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6.1 Issues in Resource Allocation• Two sides of the same coin

– pre-allocate resources so that 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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Framework• Connectionless flows

– sequence of packets sent between source/destination pair– maintain soft state (state information) at the routers

• Taxonomy for resource allocation mechanisms– router-centric versus host-centric– reservation-based versus feedback-based– window-based versus rate-based

Router

Source2

Source1

Source3

Router

Router

Destination2

Destination1

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Evaluation Criteria

• Effectiveness and Fairness• Power (ratio of throughput to delay)

Optimalload

Load

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6.2 Queuing Discipline• First-In-First-Out (FIFO)

– does not discriminate between traffic sources• Fair Queuing (FQ)

– explicitly segregates traffic based on flows– ensures no flow captures more than its share of capacity– variation: weighted fair queuing (WFQ)

• Problem?Flow 1

Flow 2

Flow 3

Flow 4

Round-robinservice

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FQ Algorithm

• For single flow• Suppose clock ticks each time a bit is transmitted• Let Pi denote the length of packet i• Let Si denote the time when start to transmit packet i• Let Fi denote the time when finish transmitting packet i• Fi = Si + Pi

• When does router start transmitting packet i?– if before router finished packet i - 1 from this flow, then

immediately after last bit of i - 1 (Fi-1)– if no current packets for this flow, then start transmitting when

arrives (call this Ai)• Thus: Fi = MAX (Fi - 1, Ai) + Pi

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FQ Algorithm (cont)• For multiple flows

– calculate Fi for each packet that arrives on each flow– treat all Fi’s as timestamps– next packet to transmit is one with lowest timestamp

• Not perfect: can’t preempt current packet• Example

• The link is never left idle as long as there is at least one packet in the queue (work-conserving)

• If the link is fully loaded and there are n flows sending data, then I cannot use more than 1/n of the link bandwidth.

Flow 1 Flow 2

(a) (b)

Output Output

F = 8 F = 10

F = 5

F = 10

F = 2

Flow 1(arriving)

Flow 2(transmitting)

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6.3 TCP Congestion Control

• Idea– TCP 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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6.3.1 Additive Increase/Multiplicative Decrease (AIMD)

• Objective: adjust to changes in the available capacity• New state variable per connection: CongestionWindow

– limits how much data source has in transit

MaxWin = MIN(CongestionWindow, AdvertisedWindow)

EffWin = MaxWin - (LastByteSent - LastByteAcked)

• Idea:– increase CongestionWindow when congestion goes down– decrease CongestionWindow when congestion goes up

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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)/CongestionWindow

CongestionWindow += Increment

• Algorithm– increment CongestionWindow by

one packet per RTT (linear increase)

– divide CongestionWindow by two whenever a timeout occurs (multiplicative decrease)

Source Destination

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AIMD (cont)

• Trace: sawtooth behavior

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6.3.2 Slow Start• Objective: determine the available

capacity in the first• Effectively increases the congestion

window exponentially, rather than linearly• Idea:

– begin with CongestionWindow = 1 packet– double CongestionWindow each RTT

(increment by 1 packet for each ACK)

• Why called “slow start” ? It is compared to the original behavior of TCP, not to the linear mechanism

Source Destination

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Slow Start (cont)• Exponential growth, but slower than all at once• Used in two different situations …

– when first starting connection– when connection goes dead waiting for timeout

• Trace

• Colored line = value of CongestionWindow, Solid bullets = timeouts, Hash marks = time when each packet is transmitted, vertical bars = time when a packet that was eventually retransmitted was first transmitted

• Problem: lose up to half a CongestionWindow’s worth of data if the source is aggressive at the beginning, as TCP is during exponential growth

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6.3.3 Fast Retransmit and Fast Recovery

• Problem: coarse-grained implementation of TCP timeouts lead to long periods during which the connection went dead and while waiting for a timer to expire

• Fast retransmit: use duplicate ACKs to trigger retransmission

• TCP waits for 3 duplicate ACKs

Packet 1

Packet 2

Packet 3

Packet 4

Packet 5

Packet 6

Retransmitpacket 3

ACK 1

ACK 2

ACK 2

ACK 2

ACK 6

ACK 2

Sender Receiver

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Results (Fast Retransmission)

• The long periods during which the congestion window stays flat and no packets are sent have been eliminated• Eliminated about half of the coarse-grained timeouts• Fast recovery

– skip the slow start phase that happens between when fast retransmit detects a lost packet and additive increase begins

– go directly to half the last successful CongestionWindow– Slow start is only used at the beginning of a connection and whenever a coarse-grained timeout occurs. At all other times, the

congestion window is following a pure AIMD pattern.

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6.4 Congestion Avoidance Mechanisms

• TCP’s strategy– control congestion once it happens, not trying to avoid congestion in the first

place– repeatedly increase load in an effort to find the point at which congestion occurs,

and then back off– TCP needs to increase losses to find the available bandwidth of the connection

• 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 different mechanisms– router-centric: DECbit and RED Gateways – host-centric: TCP Vegas

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6.4.1 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 CongestionWindow 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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6.4.2 Random Early Detection (RED)

• Invented by Sally Floyd and Van Jacobson in the early 1990s, differs from the DECbit in two major ways

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

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

• The weighted running average calculation tries to detect long-lived congestion, by filtering out short-term changes in the queue length

MaxThreshold MinThreshold

AvgLen

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RED Details (cont)

• Weighted Running Average Queue Length

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RED Details (cont)

• Two queue length thresholds

if AvgLen <= MinThreshold then

enqueue the packet

if MinThreshold < AvgLen < MaxThreshold then

calculate probability P

drop arriving packet with probability P

if 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 CurveP(drop)

1.0

MaxP

MinThresh MaxThresh

AvgLen

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Tuning RED• Probability of dropping a particular flow’s packet(s) is roughly

proportional to the share of the bandwidth that flow is currently getting• MaxP is typically set to 0.02, meaning that when the average queue size

is halfway between the two thresholds, the gateway drops roughly one out of 50 packets.

• If traffic is bursty, then MinThreshold should be sufficiently large to allow link utilization to be maintained at an acceptably high level

• 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 for traffic on today’s Internet

• Penalty Box for Offenders

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6.4.3 Source-based Congestion Avoidance-- 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

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Algorithms By RTT Grows

• Algorithm 1: The congestion window normally increases as in TCP, but every two round-trip delays, the algorithm checks to see if the current RTT is greater than the average of the minimum and maximum RTTs seen so far. If it is, then the algorithm decreases the congestion window by one-eighth.

• Algorithm 2: The window is adjusted once every ywo round-trip delays based on the product

w = (CurrentWindow –OldWindow)x(CurrentRTT – OldRTT) If w > 0, the source decreases the window size by 1/8, If w <=0, the source increases the window by one maximum

packet size.

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Algorithms by sending rate flattens

• Algorithm 3: Every RTT, it increases the window size by one packet and compares the throughput achieved to the throughput when the window was one packet smaller.

• If the difference is less than ½ the throughput achieved when only one packet was in transit, the algorithm decreases the window by one packet.

• This scheme calculates the throughput by dividing the number of bytes outstanding in he network by the RTT.

• Algorithm 4: TCP Vegas. The goal is to maintain the “right” amount of extra data in the network. If a source is sending two much extra data, it will cause long delays and possibly lead to congestion. If a connection is sending too little extra data, it cannot responds rapidly enough to transient increases in the available network bandwidth.

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TCP Vegas

• Congestion window vs observed throughput rate (the three graphs are synchronized).

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TCP Vegas Algorithm

• Let BaseRTT be the minimum of all measured RTTs (commonly the RTT of the first packet)

• If not overflowing the connection, thenExpectRate = CongestionWindow/BaseRTT

• Source calculates sending rate (ActualRate) once per RTT• Source compares ActualRate with ExpectRate

Diff = ExpectedRate - ActualRateif Diff < (too little extra data)

increase CongestionWindow linearly in next RTTelse if Diff > (too much extra data)

decrease CongestionWindow linearly in next RTTelse

leave CongestionWindow unchanged

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TCP Vegas Algorithm (cont)• Parameters

= 1 packet = 3 packets

• The goal is to keep the ActualRate between these two thresholds, that is, within the region

• TCP Vegas does use multiplicative decrease when a timeout occurs; the linear decrease is an early decrease in the congestion window that, hopefully, happens before congestion occurs.

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