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Abstract
This paper analyzes the trade-off of providing deterministic QoS guarantees versus achieving
a good bandwidth utilization. This issue has been adressed with different philosophies under the
IntServ and DiffServ approaches. This paper compares the achievable bandwidth utilization for
IntServ and DiffServ approaches using, on one hand, the formulas given by the network calculus
and, on the other, simulations of a realistic workload on a DiffServ architecture. Results allow,
not only to compare the IntServ and DiffServ approaches, but also how pessimistic are the
bounds provided by network calculus.
1 Introduction
Transporting multimedia traffic while guaranteeing a requiered Quality of Service (QoS) is a
chalenging problem that has recived a lot of attention in recent years. Two main approaches
has been developed: integrated services (IntServ) and differentiated services (DiffServ). IntServ
architecture is based on reserving network resources between individual flows, while DiffServ archi-
tecture is based on provisioning network resources between traffic aggregates. Designing efficient
algorithms for traffic control and resource allocation (or provisioning) in these approaches is very
difficult since VBR (Variable Bit Rate) video is both delay-sensitive and has a high degree of
burstiness.
Although both the IntServ and DiffServ approaches can offer different service classes, the un-
derlying discussion or the main trade-off between these two approaches is, basically, the one of
deterministic guarantees versus bandwidth utilization. IntServ relies on admission control and can
offer deterministic bandwidths and end-to-end delays to individual flows at the cost of placing strict
resource reservations that guarantee the worst case scenario. Since this case occurrs very seldom, a
lot of bandwidth is wasted. On the other hand, DiffServ prioritizes flows according to their service
class and provides a much better bandwidth utilization, because no admission control is performed.
The cost is a higher degree of uncertainty: it cannot offer guarantees to individual flows, the max-
imum delay and jitter are difficult to calculate, packet losses may occurr etc. It is obvious that
higher priority classes will get a better service in DiffServ than in a flat service class, but the key
question is: although deterministic guarantees cannot be offered, is it possible to quantify or to put
put bounds on the QoS parameters provided by DiffServ?. The answer to this question is, perhaps,
the main goal of a discipline known as Network Calculus. Some recent advances in network calculus
[3] have established a delay bound for the Expedited Forwarding (EF) behaviour of DiffServ. On
the other hand the realtion between reserved bandwidth and delay bound were also established by
[12][13].
∗
This work was supported by the Spanish Government Research Office (CICYT) under grant TIC99-1043-C03-02
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The primary goal of this paper is to compare the achievable bandwidth utilization for IntServ
and DiffServ approaches using, on one hand, the formulas given by the network calculus using a
realistic workload with QoS demands (maximum delay) and, on the other, simulations of a DiffServ
architecture. Results using network calculus show that IntServ cannot guarantee beyond a 40%
of bandwidth utilization but, surprisingly, DiffServ EF achieves even less than that. This rises
another interesting issue: the goodness of theoretical bounds. From the above results, it is not
clear whether the utilization bound for EF provided by the network calculus is not very accurate or
whether DiffServ provides a good average case delay but a very bad worst case delay. That makes
interesting to know the delay distribution. According to this, a second goal of this paper is to check
how good are the formulas provided by network calculus. This is done through simulation. Results
show that utilization bounds provided by network calculus are in general quite pessimistic.
This paper is organized as follows: section 2 presents the main results of network calculus, sec-
tion 3 describes the simultaion tools and workload description, section 4 describes the experiments
and main results and, finally, section 5 concludes the paper.
σ + nLi Lmaxj
Di = + Σnj=1 R≤p (1)
R Cj
where Li is the maximum packet size for the session i, Lmaxj is the maximum packet size in
node j, Cj th bandwidth of the link j, and n the number of nodes. In order to simplify this delay
expression, we can use the Ctot and Dtot parameters for definining the network. Where Ctot is nLi
Lmax
and Dtot is Σnj=1 Cj j . Note that equation 1 can be used only when the bandwidth reservation R
is smaller than the peak reservation p. When R > p another expression that does not depend on
the flow parameters must be used. In summary, the delay equations are:
σ + Ctot
D= Dtot R≤p (2)
R
M + Ctot
D= Dtot R≥p≥ρ (3)
R
With these equations, the control admission test becomes very simple. For a new connection i
with a given end-to-end delay Di , it is necessary to calculate the bandwidth reservation that makes
the equation 2 or 3 less than D, and the sum of the bandwidth for all channels at the node less
than the total link bandwidth Cj
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2.2 Differentiated services and EF traffic
Differentiated services is based on traffic aggregates [11]. Each of the aggregates that crosses a
router belongs to a per-hop-behaviour (PHB). Per-hop-behaviours indicate how to schedule agregate
packets and the amount of resources that can be used. Two PHBs have been standarized Assured
Forwarding (AF) [6] and Expedited Forwarding (EF) [10]. The first one basically provides a better-
than-best-effort service while the second one can offer service with stringent QoS guarantees. To
classify packets into differents agregates a 6-bits IP-header field is used. This field is named DSCP
[2]. Although inside a DiffServ domain all the EF packets are seen like a traffic aggregate, at the
domain boundaries each data flow is conformed following an arrival curve. Similarly to IntServ,
there exists a network calculus due to Le Boudec [3] that provides an equation to obtain the
maximum delay when each microflow of the aggreate leaky-bucket regulated at the input. In this
formula, the end-to-end delay bound is a function of the traffic aggregate characteristics and the
maximum number of hops that the connections cross. This delay bound can be calculated by means
of this equation:
e+τ
D1 = (4)
1 − (h − 1)ν
where e is the node latency and h is a bound on the number of hops used by any flow. The
terms ν and τ are traffic aggregate parameters and correspond to the utilization factor and the
scaled burstiness. If each microflow is conformed according to the token bucket parameters (σ, ρ),
and there are m microflows that cross the node, ν and τ can be written as:
1 X
ν= ρi (5)
rm im
and,
1 X
τ= σi (6)
rm im
Figure 1: The bound D (in seconds) versus the utilization factor (ν)
Figure 1 shows how varies the the bound D (vs. the utilization factor) in equation 4 with the
number of hops (H ). The parameters of the simulation are:
• e = 2 1500B
rm ,
• Lmax = 1000b,
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• rm = 149, 760M b/s, and
• Cm = +∞
As it can be seen, for 10 hops the bound is valid only for small utilization factors due to the
1
equation 4 explodes at ν < h−1 . However, it does not mean that the worst case delay does grow
to infinity [5]. In some cases the network may be unbounded; in some other cases (such as the
unidirectional ring, there is always a finite bound for all ν < 1. In any case, in the general case, it
can be assumed that the network is unbounded.
Although DiffServ does not use admission control tests, it is possible to guarantee a maximum
given delay bound for an aggregate by restricting the admitted workload to meet equation 4. For
a new connection i with an end-to-end delay Dmaxi it is necessary to calculate D1 (adding the
terms (σi , ρi ) to τ and ν respectively) and in order to guarantee all the flows, verify that:
3 Simulation environment
This section describes the workload used to evaluate the bounds given by network calculus and the
simulations tools used to obtain real results about delay and utilization bounds.
• Timing measures.
• Packet loss.
1
This program and the C source code of some of the experiments and admision control tests of the services
disciplines are avaiable from http://www.disca.upv.es/ehheror/RTNOU.html
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Figure 2: Ingress node configuration
• Bandwidth utilization.
• Queues depth.
• etc.
For more details about the DUSTIN simulation tool see [15]
This traffic traces (table 1) are a part of the well-know MPEG1 traces studied by O.Rose from
the University of Wurzburg [14]. These sequences have been encoded with the Berkeley MPEG-
encoder (version 1.3) with a capture rate of 25fps.
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Flow arrivals are generated according to a Poisson process with a parameter α and their duration
are exponentially distributed with mean 1/β. The ratio α/β characterizes the load offered to the
link (the average number of flows that would exist at any time at a link with no capacity limitation).
Each flow has traffic characteristics, which are chosen ramdomly from the characteristics of the six
types of traffic. A million flows were generated in each simulation run.
2. DiffServ test: Although DiffServ does not use primarily admission control tests, it is possible
to guarantee a maximum given delay bound for an aggragate by restricting the admitted
workload to meet equation 4. This is performed by the RTNOU simulator using the equation
7.
The goal of this experiment is to compute the bandwidth utilization level in a single node
(considered to be the botteleneck) provided by the above admission control tests when the QoS
requirement is given by an end-to-end delay requirement of the flow of d (excluding propagation
delays) that was uniformly distributed in [50ms, 1s].
The bandwidth utilization level is obtained by dynamically calculating the mean bandwidth
level (the sum of the mean bandwidth of the chanels accepted) of the flows accepted by the above
formula. When a channel is accepted or leaves the node, the temporal mean bandwidth level
is updated, so this value gives a dynamic vision of the bandwidth utilization. Therefore, the
bandwidth utilization level is obtained as the average of the mean bandwidth level divided by the
link bandwidth.
Figure 3 shows the number of accepted flows for IntServ and DiffServ-EF when the workload
is varied from 0 to 500 flows. Figure 4.1 shows the mean bandwidth utilization level. Since the
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equation for DiffServ-EF varies with the number of hops, the results are shown for a 2, 5 and 10
hops.
The results of this simulation reveal in first place that the mean bandwidth utilization level to
guarantee a maximum dterministic delay is very low. Up to a 40% for IntServ, and surprsingly
even less for DiffServ-EF.
The main reason of this result is the named aggregate effect. This effect is caused when we try
to guarantee maximum delay to individual flows inside the aggregate. Note in equation 4 that the
delay bound D is a bound for all the flows inside it, and so, if we have to guarantee individual
flows, in the admision control test (equation 7) we have to impose the aggregate’s maximum delay
is less than the minimum of the Dmax for all the flows, so as to guatentee all of them.
It is not clear whether the utilization bound for EF provided by the network calculus is not
very accurate or whether DiffServ provides a good average case delay but a very bad worst case
delay. That makes interesting to know the delay distribution.
The differeces between 2, 5 and 10 hops is due to the dependency between the number of hops
and the utilization factor, as we had explain en section 2.2. The equation 4 tries to capture that
the more nodes a flow can cross, the more unforseeable the traffic is, and so the more burstiness it
causes.
Another conclusion from this experiment is that, for both approaches, IntServ and DiffServ,
there is a critical load that causes the tests to start rejecting channels (in this sample, approximately
100). When load is higher than 200, the utilization increases very slowly and the number of rejected
workloads is very high.
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Figure 5: Load file description
The workload information is taken from a text file by the RTNOU simulation tool. As it can
be seen in figure 4.2, this file contains information about the video, its QoS parameters (maximum
delay), its traffic characteristics (the token bucket parameters (σ, ρ, peak)), and timing information
(start time and the duration).
Figure 4.2 shows the simulation tool configuration. The topology used in this experiment is very
simple. Ten video servers connected to an ingress node, that is considered to be the botteleneck.
Each server provides a different number of simultaneous connections (or flows) depending on the
simulaton parameters. The number of simultaneous connections varies in the interval [100..500] in
steps of 50 connections. There is a load distributor which is reponsible of distributing the workload
among the different video servers.
Each video server has so many video players as different individual flows it has to send, so each
video player represents an individual flow. This the way in which aggregate flows are generated. The
number of video players is set as a simulation parameter. All the metrics, bandwidth utilization,
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packet loss and maximum delays, are measured at the output of the ingress node.
Figure 7.a shows the bandwidth utilization results of this experiment together with the results
of the teorical bounds (see section 4.1). As it can be shown, real bandwidth utilization results
are much more better than theorical results. It grows linearly with the number of the flows, and
reaches up to a the 99% of bandwith utilization with 500 simultaneous flows.
This result reveals that a EF service is able to guarantee the maximum delay bound with
independence of the amount of traworst case analysisffic and, thus providing a full utilization of
the channels. The penalty that has to be paid for that is packet losses: packets are dropped when
the arrival rate is higher than the link capacity. However the interesting result is that a higher
utilization can be achieved with a very low loss rate. Figure 7.b shows the packet loss rate. It can
be seen that up to 300 (which represents the 80% of the link utilization) flows the loss rate is null.
When the number of simultaneous flows increases so does the packet loss rate, but in any case the
maximum rate is very low (0.082%).
Finally, and due to the drop action, and as it can be seen in figure 8.a, there are no missing
deadlines since the queue depth does not grow enough to produce high delays at the ingress node.
Although there are no missing deadlines, the maximum delay observed is the 78ms, which is inside
the maximum delay range ([50ms, 1s]). So, in order to know how far is this worst maximum delay
to the average delay, it is necessary to show the maximum delay distribution. Figure 8.b shows
the maximum delay distribution with 500 simultaneous flows. As it can be seen the average case is
around 30ms, which is less than a half the worst maximum delay. This result shows that it will be
intersting to use statistical methods for the maximum delay analysis instead of worst case analysis,
so as to provide better utilization results.
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traditional theory of the worst case analysis is being renewed introducing some statistical methods
for the analysis [4]. On the other hand, it can be also concluded that multimedia applications
should be designed in flexible way to tolerate some samll amount of packet losses, since this seems
the best way to get a better resource utilization.
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