Admission Control for VoIP over HSDPA in a
Mixed Traffic Scenario
André R. Braga , Stefan Wänstedt and Mårten Ericson
Abstract— Transporting voice over IP, instead of traditional CS
voice, is a necessary prerequisite for an all IP network. Among the
hurdles that need to be overcome are the QoS demands of VoIP,
which comprise timely and errorfree delivery of comparatively
low rate packet streams.
One way to provide the necessary QoS for VoIP is some
admission control strategy, where an incoming connection that
cannot have its quality requirements fulfilled is blocked.
This paper comprises a proposal and evaluation of a call
admission control framework aiming at improving system ca-
pacity and/or service priorization by providing QoS guarantees
for speech users in HSDPA. The main focus is the control of
packet delay, since it is a crucial requirement for real-time
services. The results show that admission control is able to
provide performance improvements and mitigation of the effects
from overloaded conditions, in a mixed traffic scenario comprised
of VoIP and Web users.
Index Terms— VoIP, HSDPA, QoS, all-IP.
I. I NTRODUCTION
The interest for VoIP is increasing because of potential
operational cost reductions for the operators, in the sense that
the Circuit Switched (CS) part of the core network can be
removed and the operators will need only one single network
for all types of communication, both speech and data [1]. This
is one of the steps in the path towards the all-IP concept [2].
Since HSDPA was mainly conceived to cope with best effort
services, the main goal of this work is to provide means
for Quality of Service (QoS) guarantees to VoIP with the
adoption of Admission Control (AC) algorithms. The VoIP
service discussed in this work may be a CS replacement.
Hence, a VoIP user will expect the same quality, capacity and
coverage provided by the CS service. Quality is in this case
primarily determined by delay (jitter) and frame losses. Due
to tight delay constraints, a packet that is delayed too much
will be useless to the application, and thereby be lost.
It is expected that a reliable AC scheme is able to provide
a transport channel suitable for VoIP users even under high
load conditions where the QoS limit is affected. Users, to
which the system will not be able to provide these transport
characteristics, are blocked.
The AC framework proposed in this work was firstly pro-
posed in [3] focusing a single service scenario, where only
the VoIP is provided in the system. The results obtained are
André R. Braga is with the Wireless Telecommunications Research Group
(GTEL), Federal University of Ceará (UFC), CP 6005 - Campus do Pici
- 60455-760, Fortaleza-CE , Brazil. E-mail: andrerb@gtel.ufc.br. Stefan
Wänstedt and Mårten Ericson are with Ericsson Research, PO Box 920, S-971
28, Luleå , Sweden. E-mails: {stefan , marten}@ericsson.com
85-89748-04-9/06/$25.00 © 2006 IEEE
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interesting and, as the framework suggests, this works extends
the evaluation to a mixed traffic scenario.
The remainder of this paper consists of the proposal and
description of AC algorithms in Section II The simulation sce-
nario and results are presented in Section III Some conclusions
and final considerations are accomplished in Section IV
II. A DMISSION C ONTROL A LGORITHM
In UTRAN, the RNC is in charge of performing AC for new
users and also those in process of handover. The resources in
each cell, controlled by the RNC, are monitored by means of
measurement reports from the Node B to the RNC through
the Node B Application Part (NBAP) signaling [4].
AC is responsible for, based on received measurement
reports concerning performance indicators from the Node B,
deciding which users that can be admitted without compro-
mising or degrading the performance of users with services of
the same or higher priority classes.
The AC employed in this work considers the delay as the
resource to be shared among users in the system. Loss rate
may play a role, but can be assumed to be low due to fast
retransmissions. Figure 1 represents the AC scheme proposed
considering both VoIP and Web traffics. It is based on a
resource usage indicator and a specified threshold. The new
sessions are accepted or blocked depending on the current
delay resource usage plus the estimated resource demand for
each user. If the sum reaches values over the pre-defined
threshold, the user is blocked. One should keep in mind that
the resource usage only considers VoIP users. Therefore, in
a mixed services scenario, the other services should respect
the delay resource limitation from VoIP, considering it the
service with the highest priority. It is also possible to consider
a priority margin for VoIP users, i.e. the access to the system
can be even more restricted to other service classes.
VoIP Threshold
Priority Margin
Web Threshold
Accepted VoIP Calls
Accepted Web Calls
Blocked VoIP Calls
Blocked Web Calls
Resource Usage
Admission Events
Fig. 1. Session admission control scheme applied.
71 ITS2006
 Two different schemes used to estimate the delay resource
usage were studied in this work.
A. Measured Delay Based (MDB)
A natural way to monitor a specific system behavior is to
perform measurements. Therefore, the delay must be measured
in the sense that only one metric should be able to represent
reliably the system performance, concerning delay experienced
by all packets from all users in the cell.
The measurement reports from the Node B are already
standardized and must use an exponential filtering approach
based on ARMA filters for measurements from the physical
layer [4]. The problem with this approach is that the filtering
process tracks the average and the average delay for VoIP
traffic does not increase significantly with increasing load,
while the user satisfaction is highly affected by the load.
In [5], a variation of the exponential filtering is considered
to be implemented in the RNC, to filter the received reports.
It is called Attack-Decay (AD) filter. The update of the
filtered measurement is done in accordance with the following
expressions:
c.
Dn = Mn − Fn−1 ,
if Dn > 0 then Fn = Fn−1 + Dn .fup ,
else if Dn < 0 then Fn = Fn−1 + Dn .fdown ,
where Fn−1 is the old filtered measurement result, Fn is the
updated filtered measurement and Mn is the new sample that
is feeding the filter. This filter has the advantage of being
capable to track any percentile just by setting the fup and
fdown , attack and decay factors, respectively.
That is the motivation of using the AD filter to evaluate
the delay measurements of VoIP packets. Each transmitted or
discarded packet provides a delay sample that feeds the AD
filter. The updated filtered value in each cell is reported to the
RNC every report interval, in order to be used for admission
purposes. The values for fup and fdown assumed in this work
were 0.5 and 0.05, respectively.
B. Estimated Delay Based (EDB)
Sending delay measurement reports from the Node B to the
RNC may not be feasible due to limitations in the UTRAN
architecture. Consequently, a method to indirectly estimate
the delay in a certain queue based on some other metrics is
required. In [6], an AC scheme is proposed combining power
and traffic measurements, like the number of transport blocks
in the queue and the service rate of the base station. By using
those metrics, an estimate of the delay can be calculated and
compared to a limiting threshold in the admission process.
In [7], the provided bit rate calculation for each service
class is standardized. It is calculated using the total number
of bits which have been considered successfully transmitted
by the MAC-hs in Node B during the last measurement
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period divided by the measurement period. This interval is
also standardized to 100ms. Using this approach, the Node
B reports to the RNC the provided bit rate for each service
priority class, by using the NBAP signaling over the Iub
interface at every report interval.
This information can be used, together with the buffer status,
to provide an approximation of how much time it takes to
transmit the remaining bits in the buffer. That gives an esti-
mation of the delay experienced by the last transmitted packet
in the queue. Using Equation 4 the delay for each priority
class p can be estimated considering Bp and Rp as being the
number of bits in the buffer for all users and the provided bit
rate (both regarding the service class), respectively.
Bp
dˆp = . (4)
Rp
III. S IMULATION R ESULTS
A. Simulations setup
The AC schemes were evaluated using a WCDMA radio
network simulator featuring the High Speed Downlink Shared
Channel (HS-DSCH), based on a modular simulation toolbox
used for radio network simulations in MATLAB
(1) The studied scenario is comprised of both VoIP and Web
services and all the traffic is handled by the HS-DSCH. The
(2) scenario anticipates the evolution towards a point where all
kinds of data have converged into the all-IP environment.
Since the speech traffic is the most important and popular
in wireless networks, the average offered traffic distribution
(3) considered in the AC evaluations is composed of 75% of real-
time speech users (VoIP traffic) and 25% of non-real-time
interactive users (Web browsing).
The AC framework was considered using two different
policies. One of them does not consider a priority margin
during the admission procedure, i.e. both services have equal
thresholds (55ms), with the AC policy trying to protect the
already connected VoIP users. Another analysis comprised the
adoption of a lower admission threshold for web users, using
a priority margin of 10ms. Hence, the admission threshold
for web users considered was 45ms. Both the Estimated and
Measured Delay based AC schemes were considered to be
compared as well.
The admission threshold of 55ms for the VoIP service was
considered because it is about half the value of the discard
timer, in the sense that it tries to improve the performance of
the system by decreasing the discard rate and, consequently
the Frame Erasure Rate (FER). The estimation for the delay
resource needed by each new session is fixed to 2ms, since it
is expected that each new user will increase the delay for all
other users by one HSDPA TTI.
In both the MDB and EDB AC schemes the report of the
delay resource to the RNC is made every 100ms. This means
that the metrics used by the AC are updated with this interval.
For a not blocked user, the satisfaction for the VoIP service
is reached when the user has a FER of less than 1% since, for
the AMR codec, 2% should be provided (1% each downlink
and uplink) for a good perceived voice quality [8]. Blocked
users are considered as not satisfied. For Web data users, the
72
 TABLE I 1
M AIN SIMULATION PARAMETERS .

0.9
Parameter Value Unit

Satisfied VoIP users ratio

Parallel HARQ Processes 6 - 0.8
Maximum HARQ RTx 5 -
HARQ type Chase Combining -
Max. # of Code Mult. 4 - 0.7
# Codes for HSDPA 8 -
Max. BS Tx Power 20 W
Common Channels Power 3 W 0.6
SCH Power 0.2 W
No AC
CPICH Power 1.2619 W
Est. Web ACth=55ms
HS-SCCH Power 0.8 W 0.5 Est. Web ACth=45ms
AMR Codec Rate 7.95 kbps Mes. Web ACth=55ms
Mean Holding Time 30 s Mes. Web ACth=45ms
Compressed Header Size 24 bits 0.4
VoIP Discard Threshold 130 ms 0.7 0.8 0.9 1 1.1 1.2 1.3 1.4 1.5 1.6
Normalized Offered Load

Fig. 2. VoIP satisfaction ratio using PF.

satisfaction is reached when the average packet throughput
(packet size divided by the waiting time) is at least 64kbps. 1

The QoS limit assumed to define the system capacity is 90%

of satisfied users for each service. 0.95
Satisfied Web users ratio

0.9
B. AC applied to Proportional Fair (PF)
The performance is presented in Figures 2 and 3 for VoIP
0.85
and Web services, respectively, and using the PF scheduler.
It is possible to notice, in Figure 2, that the AC does not
increase the offered load capacity for VoIP, but improves the 0.8

satisfaction ratio for very high loads. The improvements are No AC

Est. Web ACth=55ms
highlighted when a priority margin is considered. Although, 0.75 Est. Web ACth=45ms
it is reasonable to consider that the AC mitigates the effects Mes. Web ACth=55ms
Mes. Web ACth=45ms
of overloaded states (up to 10% of increase in the satisfaction 0.7
ratio). 0.7 0.8 0.9 1 1.1 1.2 1.3 1.4 1.5 1.6
Normalized Offered Load
When the performance of Web traffic is analyzed in Figure
3, the blocking has a high influence on the increase in the user Fig. 3. Web satisfaction ratio using PF.
dissatisfaction (up to 25% of reduction in satisfaction ratio).
When a margin is adopted for VoIP users in the admission,
the satisfaction ratio for the web service decreases even more, possible to notice a decrease in the served load for the same
due to higher imposed blocking rates. In fact it decreases the satisfaction ratio.
Web offered load capacity, but considering that the VoIP is Figure 5 also shows that, in the simulated loads using the
the limiting service, there is no decrease in the overall system PF, the satisfaction ratio of admitted web users remains above
capacity. In fact, each AC configuration provides a different 90%, meaning that it is also possible to provide some load
trade-off between quality for VoIP and Web. balancing for connected users by means of smart scheduling
When only the connected users are analyzed, some improve- algorithms, which can control the access to transmission slots
ments can also be perceived by using the AC. In Figure 4, it is for the already admitted users. Although it is out of the scope
possible to notice that no capacity improvements are obtained of this work.
by using the AC (considering the target 90% of satisfaction In Figures 6 and 7, the blocking rates regarding both
ratio). Nevertheless, the improvement in the user satisfaction services are presented. Since the Measured Delay Based AC
for high loads reaches up to around 8% when the Measured provides a higher resource usage evaluation, it protects more
Based AC without priority margin is considered. the VoIP service through higher blocking. When using no
However, the performance of connected Web users, in fact, margin for VoIP, the blocking rates for both services are
is not much affected by the AC, which is presented in Figure comparable. But when adopting some protection margin, the
5. Despite the blocking imposed to new web calls, there is blocking rates for the web service are remarkably higher. It
no salient reduction in the served load, considering that a big is this limitation that improves the performance for the VoIP
decrease in the number of web users would lead to a huge service. The adoption of a priority margin leads also to a
reduction in the system served load, since this service has reduction in the blocking rates for VoIP users.
much more payload to be served by the network. Actually, The effect of the AC on the system throughput is presented
only at high loads and using more restricted AC schemes, it is in Figure 8. It shows that when more priority is given to

73
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 0.35
1 No AC
Est. Web ACth=55ms
0.3 Est. Web ACth=45ms
Mes. Web ACth=55ms
Satisfied Connected VoIP users ratio

0.9 Mes. Web ACth=45ms

0.25

Web blocking rate

0.8
0.2

0.7 0.15

0.6 0.1

No AC
Est. Web ACth=55ms 0.05
0.5 Est. Web ACth=45ms
Mes. Web ACth=55ms
Mes. Web ACth=45ms 0
0.4 0.7 0.8 0.9 1 1.1 1.2 1.3 1.4 1.5 1.6
0.75 0.8 0.85 0.9 0.95 1 1.05 1.1 1.15 1.2 Normalized Offered Load
Normalized Served Load
Fig. 7. Web blocking rate using PF.
Fig. 4. VoIP satisfaction ratio of connected users using PF.
1.2
No AC
Est. Web ACth=55ms
1.15
1 Est. Web ACth=45ms
Mes. Web ACth=55ms
1.1 Mes. Web ACth=45ms
0.995
Normalized Served Load
Satisfied Connected Web users ratio

1.05
0.99
1
0.985
0.95
0.98
0.9
0.975
0.85
0.97

No AC 0.8
0.965 Est. Web ACth=55ms
Est. Web ACth=45ms 0.75
0.96 Mes. Web ACth=55ms 0.7 0.8 0.9 1 1.1 1.2 1.3 1.4 1.5 1.6
Mes. Web ACth=45ms Normalized Offered Load
0.955
0.75 0.8 0.85 0.9 0.95 1 1.05 1.1 1.15 1.2
Normalized Served Load Fig. 8. Effect of AC in the served load using PF.

Fig. 5. Web satisfaction ratio of connected users using PF.

the VoIP service, the higher is the reduction in the system
throughput. It establishes a trade-off between VoIP quality and
0.18 spectral efficiency without considering the QoS limit.
No AC
Est. Web ACth=55ms
0.16
Est. Web ACth=45ms
Mes. Web ACth=55ms
C. AC applied to Round Robin (RR)
0.14 Mes. Web ACth=45ms
Now, considering the , Figure 9 shows improved perfor-
mance for overloaded situations (up to around 15% of increase
VoIP blocking rate

0.12
in the satisfaction ratio), even though no explicit capacity
0.1
improvement was perceived. The problem also depends on
0.08 the fact that the web performance using the RR is not as
good, and the blocking imposed leads to a high degradation
0.06
of this service, which can be seen in Figure 10. The more
0.04 restricted AC configuration (Measured Delay with priority
margin) causes the QoS of Web users to be the capacity
0.02
limitation in the system.
0 Figures 11 and 12 show the performance of the admitted
0.7 0.8 0.9 1 1.1 1.2 1.3 1.4 1.5 1.6
Normalized Offered Load users. Now, the AC is capable of increasing the satisfaction
ratio better than by using the PF. This is because the RR
Fig. 6. VoIP blocking rate using PF. distributes equally the resources among all the users, with the
drawback of decreasing the efficiency of the network, since it

74
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 0.98
No AC
1
Est. Web ACth=55ms
0.97 Est. Web ACth=45ms

Satisfied Connected Web users ratio

0.95 Mes. Web ACth=55ms
0.96 Mes. Web ACth=45ms

0.9
Satisfied VoIP users ratio

0.95
0.85
0.94
0.8
0.93
0.75
0.92
0.7 No AC
Est. Web ACth=55ms 0.91
Est. Web ACth=45ms
0.65
Mes. Web ACth=55ms
Mes. Web ACth=45ms 0.9
0.92 0.93 0.94 0.95 0.96 0.97 0.98 0.99
0.8 0.9 1 1.1 1.2 1.3 1.4 1.5 1.6 Normalized Served Load
Normalized Offered Load
Fig. 12. Web satisfaction ratio of connected users using RR.
Fig. 9. VoIP satisfaction ratio using RR.

does not consider the channel conditions.

1
0.16
No AC
Est. Web ACth=55ms
0.95 0.14 Est. Web ACth=45ms
Mes. Web ACth=55ms
Satisfied Web users ratio

0.9 0.12 Mes. Web ACth=45ms

VoIP blocking rate

0.1
0.85

0.08
0.8

0.06
0.75
No AC
Est. Web ACth=55ms
0.04
0.7 Est. Web ACth=45ms
Mes. Web ACth=55ms 0.02
Mes. Web ACth=45ms
0.65
0.8 0.9 1 1.1 1.2 1.3 1.4 1.5 1.6 0
Normalized Offered Load 0.8 0.9 1 1.1 1.2 1.3 1.4 1.5 1.6
Normalized Offered Load

Fig. 10. Web satisfaction ratio using RR.

Fig. 13. VoIP blocking rate using RR.

In Figures 13 and 14, the blocking rates are assessed. It

1
can be seen that, the blocking when considering the priority
0.95 margin is increased for Web and reduced for VoIP, because
Satisfied Connected VoIP users ratio

the AC policy becomes more restrictive to Web users.

0.9 The results in Figure 15 regarding the effect of AC in the
served load have the same behavior as with the PF. Since more
0.85
access is given to VoIP, the spectral efficiency is decreased,
0.8
due to the lower bit rate requirement for this service. But the
reduction in the served load is even higher compared to the
0.75 PF case.
In Figure 16, the bar graphics show the absolute offered
0.7 No AC
capacity results (considering 90% of QoS limit for both
Est. Web ACth=55ms
Est. Web ACth=45ms services) comparing all the system configurations. It is clear
0.65
Mes. Web ACth=55ms
Mes. Web ACth=45ms
that, when the RR scheduler is considere, the employment of
the AC provides a capacity decrease. It is due to the fact that,
0.92 0.93 0.94 0.95 0.96 0.97 0.98 0.99
Normalized Served Load although the AC is able to improve the performance for the
VoIP service, the RR scheduler works not well for the Web
Fig. 11. VoIP satisfaction ratio of connected users using RR. service, leading the system to be limited by the Web service,
instead of by the VoIP.

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 0.25
Regarding the PF scheduler, since the improvements are
No AC achieved when the load is beyond the QoS limit for the VoIP
Est. Web ACth=55ms
Est. Web ACth=45ms
service, the capacity is not really affected.
0.2 Mes. Web ACth=55ms
Mes. Web ACth=45ms
IV. C ONCLUSIONS
Web blocking rate

0.15
This work proposes and evaluates the performance of an AC
algorithm with the aim to provide QoS control to users making
use of the VoIP service over the HSDPA. The algorithm is
0.1 based on the delay resource usage, which is calculated based
on either measurements or estimations.
It can be concluded that the improvements obtained by em-
0.05 ploying the AC are highly dependent on the scheduling policy
applied, when the VoIP service is provided. The scheduler
should be able to provide QoS guarantees to the admitted
0
0.8 0.9 1 1.1 1.2 1.3 1.4 1.5 1.6 users. If not, the capacity improvements considering the QoS
Normalized Offered Load
guarantees is limited.
The performance of both MDB and EDB AC was concluded
Fig. 14. Web blocking rate using RR.
to be comparable. The former imposes a higher protection
to the connected users and, consequently, provides higher
blocking. Regarding the EDB AC, it performed in some cases
0.99
No AC even better, because of a better trade-off between QoS and
Est. Web ACth=55ms blocking. With the advantage of not requiring any additional
0.98 Est. Web ACth=45ms
Mes. Web ACth=55ms signaling in the Iub interface.
Mes. Web ACth=45ms An interesting perspective that can be traced for the conti-
Normalized Served Load

0.97
nuity of this work is the evaluation of the admission control
0.96
framework together with advanced scheduling algorithms in
a mixed traffic scenario. Some other scenarios comprised of
0.95
different service proportions is an interesting issue as well.

0.94 ACKNOWLEDGEMENT
André Ribeiro Braga was supported by a grant from Erics-
0.93 son of Brazil - research branch under ERBB/UFC.17 technical
cooperation contract.
0.92
0.8 0.9 1 1.1 1.2 1.3 1.4 1.5 1.6
Normalized Offered Load R EFERENCES
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Fig. 16. Offered capacity.

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