It is important to understand that TCP's strategy is to control congestion once it happens, as opposed to trying to avoid congestion in the first place. In fact, TCP repeatedly increases the load it imposes on the network in an effort to find the point at which congestion occurs, and then it backs off from this point. Said another way, TCP needs to create losses to find the available bandwidth of the connection. An appealing alternative is to predict when congestion is about to happen and then to reduce the rate at which hosts send data just before packets start being discarded. We call such a strategy congestion avoidance to distinguish it from congestion control, but it's probably most accurate to think of "avoidance" as a subset of "control."
This section describes three different congestion-avoidance mechanisms. The first two take a similar approach: They put a small amount of additional functionality into the router to assist the end node in the anticipation of congestion. The third mechanism is very different from the first two: It attempts to avoid congestion purely from the end nodes.
The first mechanism was developed for use on the Digital Network Architecture (DNA), a connectionless network with a connection-oriented transport protocol. This mechanism could, therefore, also be applied to TCP and IP. As noted above, the idea here is to more evenly split the responsibility for congestion control between the routers and the end nodes. Each router monitors the load it is experiencing and explicitly notifies the end nodes when congestion is about to occur. This notification is implemented by setting a binary congestion bit in the packets that flow through the router, hence the name DECbit. The destination host then copies this congestion bit into the ACK it sends back to the source. Finally, the source adjusts its sending rate so as to avoid congestion. The following discussion describes the algorithm in more detail, starting with what happens in the router.
A single congestion bit is added to the packet header. A router sets this bit in a packet if its average queue length is greater than or equal to 1 at the time the packet arrives. This average queue length is measured over a time interval that spans the last busy+idle cycle, plus the current busy cycle. (The router is busy when it is transmitting and idle when it is not.) Figure 1 shows the queue length at a router as a function of time. Essentially, the router calculates the area under the curve and divides this value by the time interval to compute the average queue length. Using a queue length of 1 as the trigger for setting the congestion bit is a trade-off between significant queuing (and hence higher throughput) and increased idle time (and hence lower delay). In other words, a queue length of 1 seems to optimize the power function.
Now turning our attention to the host half of the mechanism, the source records how many of its packets resulted in some router setting the congestion bit. In particular, the source maintains a congestion window, just as in TCP, and watches to see what fraction of the last window's worth of packets resulted in the bit being set. If less than 50% of the packets had the bit set, then the source increases its congestion window by one packet. If 50% or more of the last window's worth of packets had the congestion bit set, then the source decreases its congestion window to 0.875 times the previous value. The value 50% was chosen as the threshold based on analysis that showed it to correspond to the peak of the power curve. The "increase by 1, decrease by 0.875" rule was selected because additive increase/multiplicative decrease makes the mechanism stable.
Random Early Detection (RED)
A second mechanism, called random early detection (RED), is similar to the DECbit scheme in that each router is programmed to monitor its own queue length and, when it detects that congestion is imminent, to notify the source to adjust its congestion window. RED, invented by Sally Floyd and Van Jacobson in the early 1990s, differs from the DECbit scheme in two major ways.
The first is that rather than explicitly sending a congestion notification message to the source, RED is most commonly implemented such that it implicitly notifies the source of congestion by dropping one of its packets. The source is, therefore, effectively notified by the subsequent timeout or duplicate ACK. In case you haven't already guessed, RED is designed to be used in conjunction with TCP, which currently detects congestion by means of timeouts (or some other means of detecting packet loss such as duplicate ACKs). As the "early" part of the RED acronym suggests, the gateway drops the packet earlier than it would have to, so as to notify the source that it should decrease its congestion window sooner than it would normally have. In other words, the router drops a few packets before it has exhausted its buffer space completely, so as to cause the source to slow down, with the hope that this will mean it does not have to drop lots of packets later on.
Note that RED is also sometimes used with an explicit feedback
scheme simply by marking a packet instead of dropping it.
This feedback mechanism is implemented by treating two bits
in the IP
TOS field as Explicit Congestion Notification (ECN) bits.
One bit is set by the source to indicate that it is ECN-capable,
that is, able to react to a congestion notification. The other bit
is set by routers along the end-to-end path when congestion is
encountered, as indicated by the RED algorithm. The latter bit is
also echoed back to the source by the destination host. TCP running
on the source responds to the ECN bit set in exactly the same way it
responds to a dropped packet.
The second difference between RED and DECbit is in the details of how RED decides when to drop a packet and what packet it decides to drop. To understand the basic idea, consider a simple FIFO queue. Rather than wait for the queue to become completely full and then be forced to drop each arriving packet (the tail drop policy of the previous section), we could decide to drop each arriving packet with some drop probability whenever the queue length exceeds some drop level. This idea is called early random drop. The RED algorithm defines the details of how to monitor the queue length and when to drop a packet.
In the following paragraphs, we describe the RED algorithm as originally proposed by Floyd and Jacobson. We note that several modifications have since been proposed both by the inventors and by other researchers; some of these are discussed in Further Reading. However, the key ideas are the same as those presented below, and most current implementations are close to the algorithm that follows.
First, RED computes an average queue length using a weighted running
average similar to the one used in the original TCP timeout computation.
AvgLen is computed as
AvgLen = (1 - Weight) x AvgLen + Weight x SampleLen
where 0 <
Weight < 1 and
SampleLen is the length of the
queue when a sample measurement is made. In most software
implementations, the queue length is measured every time a new packet
arrives at the gateway. In hardware, it might be calculated at some
fixed sampling interval.
The reason for using an average queue length rather than an
instantaneous one is that it more accurately captures the notion of
congestion. Because of the bursty nature of Internet traffic, queues can
become full very quickly and then become empty again. If a queue is
spending most of its time empty, then it's probably not appropriate to
conclude that the router is congested and to tell the hosts to slow
down. Thus, the weighted running average calculation tries to detect
long-lived congestion, as indicated in the right-hand portion of
Figure 2, by filtering out short-term changes in the queue
length. You can think of the running average as a low-pass filter, where
Weight determines the time constant of the filter. The question of
how we pick this time constant is discussed below.
Second, RED has two queue length thresholds that trigger certain
MaxThreshold. When a packet arrives
at the gateway, RED compares the current
AvgLen with these two
thresholds, according to the following rules:
if AvgLen <= MinThreshold queue the packet if MinThreshold < AvgLen < MaxThreshold calculate probability P drop the arriving packet with probability P if MaxThreshold <= AvgLen drop the arriving packet
If the average queue length is smaller than the lower threshold, no
action is taken, and if the average queue length is larger than the
upper threshold, then the packet is always dropped. If the average queue
length is between the two thresholds, then the newly arriving packet is
dropped with some probability
P. This situation is depicted in
Figure 3. The approximate relationship between
AvgLen is shown in Figure 4. Note that the probability
of drop increases slowly when
AvgLen is between the two thresholds,
MaxP at the upper threshold, at which point it jumps to
unity. The rationale behind this is that, if
AvgLen reaches the
upper threshold, then the gentle approach (dropping a few packets) is
not working and drastic measures are called for: dropping all arriving
packets. Some research has suggested that a smoother transition from
random dropping to complete dropping, rather than the discontinuous
approach shown here, may be appropriate.
Although Figure 4 shows the probability of drop as a
function only of
AvgLen, the situation is actually a little more
complicated. In fact,
P is a function of both
AvgLen and how
long it has been since the last packet was dropped. Specifically, it is
computed as follows:
TempP = MaxP x (AvgLen - MinThreshold) / (MaxThreshold - MinThreshold) P = TempP/(1 - count x TempP)
TempP is the variable that is plotted on the y-axis in
count keeps track of how many newly arriving
packets have been queued (not dropped), and
AvgLen has been between
the two thresholds.
P increases slowly as
thereby making a drop increasingly likely as the time since the last
drop increases. This makes closely spaced drops relatively less likely
than widely spaced drops. This extra step in calculating
introduced by the inventorsof RED when they observed that, without it,
the packet drops were not well distributed in time but instead tended to
occur in clusters. Because packet arrivals from a certain connection are
likely to arrive in bursts, this clustering of drops is likely to cause
multiple drops in a single connection. This is not desirable, since only
one drop per round-trip time is enough to cause a connection to reduce
its window size, whereas multiple drops might send it back into slow
As an example, suppose that we set
MaxP to 0.02 and
initialized to zero. If the average queue length were halfway between
the two thresholds, then
TempP, and the initial value of
would be half of
MaxP, or 0.01. An arriving packet, of course, has a
99 in 100 chance of getting into the queue at this point. With each
successive packet that is not dropped,
P slowly increases, and by
the time 50 packets have arrived without a drop,
P would have
doubled to 0.02. In the unlikely event that 99 packets arrived without
P reaches 1, guaranteeing that the next packet is dropped. The
important thing about this part of the algorithm is that it ensures a
roughly even distribution of drops over time.
The intent is that, if RED drops a small percentage of packets when
MinThreshold, this will cause a few TCP
connections to reduce their window sizes, which in turn will reduce the
rate at which packets arrive at the router. All going well,
will then decrease and congestion is avoided. The queue length can be
kept short, while throughput remains high since few packets are dropped.
Note that, because RED is operating on a queue length averaged over
time, it is possible for the instantaneous queue length to be much
AvgLen. In this case, if a packet arrives and there is
nowhere to put it, then it will have to be dropped. When this happens,
RED is operating in tail drop mode. One of the goals of RED is to
prevent tail drop behavior if possible.
The random nature of RED confers an interesting property on the algorithm. Because RED drops packets randomly, the probability that RED decides to drop a particular flow's packet(s) is roughly proportional to the share of the bandwidth that that flow is currently getting at that router. This is because a flow that is sending a relatively large number of packets is providing more candidates for random dropping. Thus, there is some sense of fair resource allocation built into RED, although it is by no means precise.
Note that a fair amount of analysis has gone into setting the various
RED parameters—for example, and
Weight—all in the name of
optimizing the power function (throughput-to-delay ratio). The
performance of these parameters has also been confirmed through
simulation, and the algorithm has been shown not to be overly sensitive
to them. It is important to keep in mind, however, that all of this
analysis and simulation hinges on a particular characterization of the
network workload. The real contribution of RED is a mechanism by which
the router can more accurately manage its queue length. Defining
precisely what constitutes an optimal queue length depends on the
traffic mix and is still a subject of research, with real information
now being gathered from operational deployment of RED in the Internet.
Consider the setting of the two thresholds,
MaxThreshold. If the traffic is fairly bursty, then
should be sufficiently large to allow the link utilization to be
maintained at an acceptably high level. Also, the difference between the
two thresholds should be larger than the typical increase in the
calculated average queue length in one RTT. Setting
MinThreshold seems to be a reasonable rule of thumb given the
traffic mix on today's Internet. In addition, since we expect the
average queue length to hover between the two thresholds during periods
of high load, there should be enough free buffer space above
MaxThreshold to absorb the natural bursts that occur in Internet
traffic without forcing the router to enter tail drop mode.
We noted above that
Weight determines the time constant for the
running average low-pass filter, and this gives us a clue as to how we
might pick a suitable value for it. Recall that RED is trying to send
signals to TCP flows by dropping packets during times of congestion.
Suppose that a router drops a packet from some TCP connection and then
immediately forwards some more packets from the same connection. When
those packets arrive at the receiver, it starts sending duplicate ACKs
to the sender. When the sender sees enough duplicate ACKs, it will
reduce its window size. So, from the time the router drops a packet
until the time when the same router starts to see some relief from the
affected connection in terms of a reduced window size, at least one
round-trip time must elapse for that connection. There is probably not
much point in having the router respond to congestion on time scales
much less than the round-trip time of the connections passing through
it. As noted previously, 100 ms is not a bad estimate of average
round-trip times in the Internet. Thus,
Weight should be chosen such
that changes in queue length over time scales much less than 100 ms are
Since RED works by sending signals to TCP flows to tell them to slow down, you might wonder what would happen if those signals are ignored. This is often called the unresponsive flow problem. Unresponsive flows use more than their fair share of network resources and could cause congestive collapse if there were enough of them, just as in the days before TCP congestion control. Some of the techniques described in the next section can help with this problem by isolating certain classes of traffic from others. There is also the possibility that a variant of RED could drop more heavily from flows that are unresponsive to the initial hints that it sends.
Source-Based Congestion Avoidance
Unlike the two previous congestion-avoidance schemes, which depended on cooperation from routers, we now describe a strategy for detecting the incipient stages of congestion—before losses occur—from the end hosts. We first give a brief overview of a collection of related mechanisms that use different information to detect the early stages of congestion, and then we describe a specific mechanism in some detail.
The general idea of these techniques is to watch for a sign from the network that some router's queue is building up and that congestion will happen soon if nothing is done about it. For example, the source might notice that as packet queues build up in the network's routers, there is a measurable increase in the RTT for each successive packet it sends. One particular algorithm exploits this observation as follows: 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.
A second algorithm does something similar. The decision as to whether or not to change the current window size is based on changes to both the RTT and the window size. The window is adjusted once every two round-trip delays based on the product
(CurrentWindow - OldWindow) x (CurrentRTT - OldRTT)
If the result is positive, the source decreases the window size by one-eighth; if the result is negative or 0, the source increases the window by one maximum packet size. Note that the window changes during every adjustment; that is, it oscillates around its optimal point.
Another change seen as the network approaches congestion is the flattening of the sending rate. A third scheme takes advantage of this fact. 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 one-half the throughput achieved when only one packet was in transit—as was the case at the beginning of the connection—the algorithm decreases the window by one packet. This scheme calculates the throughput by dividing the number of bytes outstanding in the network by the RTT.
A fourth mechanism, the one we are going to describe in more detail, is similar to this last algorithm in that it looks at changes in the throughput rate or, more specifically, changes in the sending rate. However, it differs from the third algorithm in the way it calculates throughput, and instead of looking for a change in the slope of the throughput it compares the measured throughput rate with an expected throughput rate. The algorithm, TCP Vegas, is not widely deployed in the Internet, but the strategy it uses has been adopted by other implementations that are now being deployed.
The intuition behind the Vegas algorithm can be seen in the trace of standard TCP given in Figure 5. The top graph shown in Figure 5 traces the connection's congestion window; it shows the same information as the traces given earlier in this section. The middle and bottom graphs depict new information: The middle graph shows the average sending rate as measured at the source, and the bottom graph shows the average queue length as measured at the bottleneck router. All three graphs are synchronized in time. In the period between 4.5 and 6.0 seconds (shaded region), the congestion window increases (top graph). We expect the observed throughput to also increase, but instead it stays flat (middle graph). This is because the throughput cannot increase beyond the available bandwidth. Beyond this point, any increase in the window size only results in packets taking up buffer space at the bottleneck router (bottom graph).
A useful metaphor that describes the phenomenon illustrated in Figure 5 is driving on ice. The speedometer (congestion window) may say that you are going 30 miles an hour, but by looking out the car window and seeing people pass you on foot (measured sending rate) you know that you are going no more than 5 miles an hour. The extra energy is being absorbed by the car's tires (router buffers).
TCP Vegas uses this idea to measure and control the amount of extra data this connection has in transit, where by "extra data" we mean data that the source would not have transmitted had it been trying to match exactly the available bandwidth of the network. The goal of TCP Vegas is to maintain the "right" amount of extra data in the network. Obviously, if a source is sending too much extra data, it will cause long delays and possibly lead to congestion. Less obviously, if a connection is sending too little extra data, it cannot respond rapidly enough to transient increases in the available network bandwidth. TCP Vegas's congestion-avoidance actions are based on changes in the estimated amount of extra data in the network, not only on dropped packets. We now describe the algorithm in detail.
First, define a given flow's
BaseRTT to be the RTT of a packet when
the flow is not congested. In practice, TCP Vegas sets
the minimum of all measured round-trip times; it is commonly the RTT of
the first packet sent by the connection, before the router queues
increase due to traffic generated by this flow. If we assume that we are
not overflowing the connection, then the expected throughput is given by
ExpectedRate = CongestionWindow / BaseRTT
CongestionWindow is the TCP congestion window, which we assume
(for the purpose of this discussion) to be equal to the number of bytes
Second, TCP Vegas calculates the current sending rate,
This is done by recording the sending time for a distinguished packet,
recording how many bytes are transmitted between the time that packet is
sent and when its acknowledgment is received, computing the sample RTT
for the distinguished packet when its acknowledgment arrives, and
dividing the number of bytes transmitted by the sample RTT. This
calculation is done once per round-trip time.
Third, TCP Vegas compares
ExpectedRate and adjusts
the window accordingly. We let
Diff = ExpectedRate -
ActualRate. Note that
Diff is positive or 0 by definition, since
ActualRate >ExpectedRate implies that we need to change
the latest sampled RTT. We also define two thresholds, , roughly corresponding to having too little and too much extra
data in the network, respectively. When
Diff < , TCP Vegas
increases the congestion window linearly during the next RTT, and when
Diff > , TCP Vegas decreases the congestion window linearly
during the next RTT. TCP Vegas leaves the congestion window unchanged
Diff < .
Intuitively, we can see that the farther away the actual throughput gets from the expected throughput, the more congestion there is in the network, which implies that the sending rate should be reduced. The threshold triggers this decrease. On the other hand, when the actual throughput rate gets too close to the expected throughput, the connection is in danger of not utilizing the available bandwidth. The threshold triggers this increase. The overall goal is to keep between and extra bytes in the network.
Figure 6 traces the TCP Vegas congestion-avoidance algorithm.
The top graph traces the congestion window, showing the same information
as the other traces given throughout this chapter. The bottom graph
traces the expected and actual throughput rates that govern how the
congestion window is set. It is this bottom graph that best illustrates
how the algorithm works. The colored line tracks the
while the black line tracks the
ActualRate. The wide shaded strip
gives the region between the and thresholds; the top of
the shaded strip is KBps away from
ExpectedRate, and the
bottom of the shaded strip is KBps away from
The goal is to keep the
ActualRate between these two thresholds,
within the shaded region. Whenever
ActualRate falls below the shaded
region (i.e., gets too far from
ExpectedRate), TCP Vegas decreases
the congestion window because it fears that too many packets are being
buffered in the network. Likewise, whenever
ActualRate goes above
the shaded region (i.e., gets too close to the
Vegas increases the congestion window because it fears that it is
underutilizing the network.
Because the algorithm, as just presented, compares the difference
between the actual and expected throughput rates to the and
thresholds, these two thresholds are defined in terms of KBps.
However, it is perhaps more accurate to think in terms of how many extra
buffers the connection is occupying in the network. For example, on a
connection with a
BaseRTT of 100 ms and a packet size of 1 KB, if
= 30 KBps and = 60 KBps, then we can think
of as specifying that the
connection needs to be occupying at least 3 extra buffers in the network
and as specifying that the connection should occupy no more than
6 extra buffers in the network. In practice, a setting of to 1
buffer and to 3 buffers works well.
Finally, you will notice that TCP Vegas decreases the congestion window linearly, seemingly in conflict with the rule that multiplicative decrease is needed to ensure stability. The explanation is that TCP Vegas does use multiplicative decrease when a timeout occurs; the linear decrease just described is an early decrease in the congestion window that should happen before congestion occurs and packets start being dropped.