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Simulation of Quality of Service Mechanisms in the UMTS Terrestrial Radio

Access Network

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 Simulation of Quality of Service Mechanisms in the
UMTS Terrestrial Radio Access Network
A. B. García, M. Álvarez-Campana, E. Vázquez and J. Berrocal
Departamento de Ingeniería de Sistemas Telemáticos, Ciudad Universitaria s/n, 28040 Madrid, Spain
{abgarcia, mac, enrique, berrocal}@dit.upm.es
Abstract - Supporting multiple traffic classes with different QoS
(Quality of Service) constraints in third generation mobile
systems is not a straightforward problem. This issue becomes
critical at the access network interfaces, where transmission
resources are usually expensive. One of the most important
interfaces is the one that connects each base station with its
controller, because there can be many instances of it. We have
approached the dimensioning of this interface from the
perspective of simulation, concretely for the UMTS (Universal
Mobile Telecommunications System) access network. At this
respect we have developed a simulation model capable of
representing multi-service scenarios inside this particular UTMS
interface. This tool can be fed with traffic of different types,
representing the different traffic classes defined for UMTS, and
is able to simulate traffic differentiation both at ATM
(Asynchronous Transfer Mode) and AAL2 (ATM Adaptation
Layer 2) level. In the paper we present simulation results
showing how the simulator can aid us in the task of access
network dimensioning with QoS constraints.
Keywords: UMTS; Radio Access Network; ATM; AAL2; Quality of
Service; Simulation.
1. INTRODUCTION
Third generation (3G) mobile communications systems will
offer high-speed mobile access to a great variety of services
in a world-wide scope. Some of these services require certain
QoS (Quality of Service) constraints to be met by the network
in order to function properly (for instance, multimedia
applications). In the case of UMTS (Universal Mobile
Telecommunications System), the 3GPP (3rd Generation
Partnership Project) has decided not to standardize a closed
group of services; instead, an open QoS architecture is
specified, including the definition of four QoS traffic classes
and a group of QoS parameters (tolerance to delay and losses
among them).
The UMTS QoS classes are: conversational, streaming,
interactive and background; they are defined in 3GPP
specification TS 23.107 [1], together with their QoS
parameter values. To efficiently accommodate the different
traffic classes, a careful network dimensioning process should
be carried out. This process should take into account several
factors, including the services demand forecast, traffic
parameters and QoS constraints of the applications, and
network topology.
In particular, the access network is one of the most critical
parts of the system (see [2]), since both the air interface and
the terrestrial transmission resources inside it can be
considered scarce, and the QoS constraints are tight.
For the first release of UMTS, 3GPP has decided to specify
ATM (Asynchronous Transfer Mode) as the transport
technology inside the access network (UTRAN – UMTS
Terrestrial Radio Access Network). ATM can be seen as a
more flexible technology than traditional circuit switching
methods. However, in order to take advantage of this
flexibility, new dimensioning methods have to be envisaged
for the access network, since models used in second
generation (2G) systems are no longer adequate.
Due to the many factors to be considered and to the
technological change with respect to 2G systems, analytical
solutions for the UTRAN dimensioning problem seem
difficult to obtain. This is why we have developed a
simulation model, suitable for the performance analysis and
dimensioning of the UTRAN interface connecting each base
station (or Node-B) and its controller (or RNC – Radio
Network Controller). After giving a brief description of the
simulation model, we will present some results obtained with
this tool under specific traffic and network scenarios.
2. SIMULATION MODEL
2.1. User Plane Protocol Stack
UTRAN terrestrial protocols are structured in two layers
[3]: Radio Network Layer (RNL), consisting of all the
UMTS-specific protocols, and Transport Network Layer
(TNL), which includes generic protocols, and is in charge of
conveying RNL data across the terrestrial interfaces with the
necessary QoS.
Our simulator models the behavior of TNL at the User
Plane of the Iub interface (the one between each Node-B and
its controlling RNC). As we can see in Fig. 1, AAL2 (ATM
Adaptation Layer 2) and ATM protocols are used at the TNL
of Iub, above the physical layer (e.g. an E1 line).
We will suppose that each user session (whether packet
switched or circuit switched) makes use of a dedicated
transport channel (DCH), and the resulting DCH Frame
Protocol (DCH FP) frames are carried by an AAL2
connection across Iub.
 Peer protocols The simulation results shown later include two applications,
at the Radio
User Equipment Protocols voice and web browsing, representing the conversational and
interactive UMTS QoS classes respectively. An AMR
FP FP
RNL (Adaptive Multi Rate) codec in 12.2 mode with SID (Silence
Insertion Description) frames is used for voice. Web
TNL AAL2
AAL2 browsing has a download rate (speed at which a web
document is downloaded) of 64 kbit/s. The statistical
ATM ATM
distribution of burst state durations is exponential, while a
PHY
constant distribution has been used both for time between
PHY
packets and packet sizes. Table 1 shows the mean values for
Node-B RNC
each parameter distribution. These values have been derived
from the recommendations given in Annex A of UMTS
Fig. 1. Iub interface: User Plane protocol stack. technical report TR 25.933 [7], AMR codec specification [8],
an ETSI technical report to be used in UMTS evaluation
2.2. Traffic Characterization
phases [6], and relevant protocol overheads and
For us every user is in an active state (e.g. during a voice characteristics.
call or a web browsing session). We assume that the number
TABLE 1
of simultaneously active users of each kind is a previously TRAFFIC CHARACTERIZATION PARAMETERS: MEAN VALUES
known parameter, which can be obtained as a result of the
radio planning phase, for instance. Also, when the application Voice Web
exhibits an asymmetric behavior, we implicitly model Burst level parameters
downstream traffic, since typically the majority of the traffic
High state duration 3s 1.5 s
is sent to the user equipment.
Low state duration 3s 412 s
Since our goal is to study the TNL, the source model should
include not only the end applications’ statistical properties, Packet level parameters
but also the peculiarities of the radio protocols used to convey Time between packets (High and Low) 20 ms 40 ms
the data. We decided to use the same kind of source model Packet size (High) 40 B 325 B
for the four UMTS QoS classes. However, its Packet size (Low) 13 B 0B
parameterization is specific to each traffic class. We
distinguish two layers, called burst and packet, shown in Fig. 2.3. Simulator Architecture
2 together with the relevant model parameters. In contrast
with several traffic parameterization studies ([4], [5], [6]), our Fig. 3 shows the overall simulator architecture at the
model does not include a session level, since, as stated above, sending side. There is also a receiving module in charge of
every user is supposed to be inside an active session. computing the relevant source, AAL2 and ATM results.
Each source follows a “High-Low” (or “generic” ON-OFF)
•Standard AAL2 segmentation
pattern, modeling the variable-rate nature of many and multiplexing
•Supports Timer_CU
applications. The packet generation process parameters have •Does not introduce losses
to be set separately for each state. A packet represents a DCH
FP frame at Iub; in the rest of the paper the terms packet and Src. AAL2
Group 1 Mux.
frame will be used without distinction. ATM
•Sends cells at the beginning
of cell slots
VCC 1 •Scheduling: FIFO or PRIOR
High state Low state
duration duration
L1 cells
Src. AAL2 To physical
Burst Group n Mux. Line
ATM (B bit/s)
P cells
VCC m
Time between packets, Time between packets,
High state Low state From other
AAL2 mux’s
Packet size, Packet size,
High state Low state Lm cells

Packet
•Sends cells at no more than PCR
•Scheduling: FIFO or PRIOR

Fig. 2. Source model in two layers.

Fig. 3. Simulator logical architecture.
 Although we have represented only one physical line, configured with specific values of VCC bit rate and number
several lines can be actually configured, each with its own bit of users. In all cases, the VCC buffer size was set to a number
rate (B bit/s). This bit rate determines the duration of one of ATM cells equal to the number of users. The figures of this
ATM cell slot (the time it takes to transmit one cell at B bit/s). section are referred to FLR (Frame Loss Ratio) as QoS
The physical-line server will start sending each ATM cell parameter. Analogous results could be obtained taking
(when available) at the beginning of a slot, and has a limited different objective parameters, such as frame delay (95th
buffer space of P cells. percentile).
We can define one or several (m) ATM VCCs (Virtual Fig. 4 shows the frame loss ratio for voice traffic as a
Channel Connections) to be multiplexed into the same function of the VCC capacity (normalized to the peak bit rate
physical line, each one with an independent PCR (Peak Cell of one voice source). The simulation results are compared
Rate). Other parameters that can be configured for a VCC are with the estimated values provided by an analytical fluid-flow
the buffer size (Li cells for the ith VCC) and an absolute approximation [15]; these analytical estimations are plotted
priority which is meaningful among all the VCCs sharing the with dashed lines. From this kind of results we can derive the
same physical line. If different priorities are assigned to minimum VCC capacity necessary for a given number of
several VCCs, the physical-line server will perform PRIOR voice users in order to meet specific FLR requirements, as
(absolute priorities) scheduling accordingly. can be seen in Fig. 5. Similar results are provided in Fig. 6 for
The tool also allows us to define as many source groups as web traffic. This kind of curves can be useful for
we want. A source group consists of a set of independent and dimensioning purposes with QoS constraints.
identical traffic generators with the relevant parameters set as
described in a previous subsection. Each group is assigned an
Normalized VCC capacity
AAL2 multiplexer that performs standard AAL2 0 10 20 30 40 50 60 70 80 90 100 110 120
segmentation and multiplexing tasks, including Timer_CU 1E+00
Analytical
handling, as specified by ITU-T [9]. These two processes do Simulation

not introduce any loss. However, the value of the AAL2 1E-01

Timer_CU can affect the delay experienced by the frames.

Frame Loss Ratio

We can also multiplex the output of several AAL2 1E-02

multiplexers (i.e. the data of several source groups) into an

ATM VCC. Again, in a similar manner as stated above for 1E-03
VCCs, each source group being multiplexed in a VCC can be
25 users 50 75 100 125 150
assigned an absolute priority. This can be seen as a basic 1E-04
AAL2-level traffic differentiation, in line (although with a
different method) with what has also been proposed in some 1E-05
related papers, for instance, [10], [11], [12], [13] and [14].
Fig. 4. Frame loss ratio as a function of VCC capacity (voice).
3. SIMULATION RESULTS

In this section we present results obtained with the simulator

described above. All the experiments were performed with a
2500
physical-line bit rate B = 1 984 000 bit/s, corresponding to an
E1 line, a physical line buffer length P = 10 cells, and FLR < 10
-5
Minimum required capacity (kbit/s)

2000
Timer_CU = 1 ms. On one hand, these results show how the
-3
values of several parameters (e.g. the PCR of ATM VCCs) FLR < 10

affect the main QoS parameters of the application frames in 1500

-1
FLR < 10
the studied UTRAN interface (basically loss ratio and delay).
On the other hand, a post-processing of the data allows us to 1000

obtain dimensioning rules such as the minimum (with a

certain tolerance) PCR that, under specific traffic scenarios, 500
guarantees that the QoS constraints of the applications are
met. 0
25 50 75 100 125 150
3.1. Single Traffic Class Scenarios
Number of voice users
A set of simulations has been performed including users of
Fig. 5. Minimum VCC capacity to meet FLR requirements (voice).
a single traffic class in each one. Each experiment was
 TABLE 3
500
QOS PARAMETER VALUES OBTAINED IN SINGLE CLASS AND SHARED VCC SCENARIOS

Shared VCC
Minimum required capacity (kbit/s)

Single
400 Class 100% 95% 90%
FLR < 10-5
FLR 4.2×10-5 2.8×10-6 5.5×10-6 1.5×10-5
FLR < 10-4
300 Frame Delay

Voice
5.0 ms 2.6 ms 3.0 ms 3.3 ms
(mean)
-3
FLR < 10
Frame Delay
8.5 ms 4.5 ms 5.0 ms 6.5 ms
200
(95th perc.)
FLR < 10-2

Web
FLR 8.2×10-5 1.8×10-4 4.2×10-4 1.2×10-3
100
25 50 75 100 125 150

Number of web users
Fig. 6. Minimum VCC capacity to meet FLR requirements (web).
3.2. Traffic Mix Scenarios
We have also performed simulation runs in multi-service
scenarios. Our goal in this case is to investigate the possible
bandwidth savings that can be achieved by multiplexing
different traffic classes in a single ATM VCC.
More specifically, in this section we show the results
corresponding to a mix of 50 voice users and 50 web users.
All of them are multiplexed into the same VCC, giving
absolute priority to voice users, since their delay requirements
are typically more stringent. The VCC buffer size is
L1 = 100 cells. Table 2 shows the QoS objectives at the Iub
interface established as criteria for our analysis. Note that the
FLR requirement is tighter for voice than for web. The reason
for this is that we have supposed that RLC (Radio Link
Control) error recovery procedures can be used for web
traffic, but not for voice because the end to end delay would
suffer inadmissible increments. Values at the last row of the
table refer to the minimum VCC capacity required to meet the
QoS requirements when using separated VCCs (i.e. in single-
class scenarios).
The experiments carried out for the shared VCC strategy
started by considering an aggregated capacity equal to the
sum of capacities required for separated VCCs with 50 users.
Then, we proceeded to decrease the aggregated capacity in
steps of 5% (of the initial value). The results are shown in
table 3. The single class column corresponds to the VCC
capacities given in the last row of Table 2.
TABLE 2
QOS OBJECTIVES AND MINIMUM VCC CAPACITY TO MEET THEM
FLR
Frame Delay (95th percentile)
Minimum VCC capacity (50 users)
The results prove that with a 5% of capacity reduction, all
the QoS requirements are still met (for both traffic classes).
However, for a 10% reduction, the web FLR exceeds the
maximum allowed value. Therefore, we can conclude that the
VCC sharing strategy does not lead to significant bandwidth
savings.
Nevertheless, a more in depth analysis of the results reveals
a beneficial effect of the shared VCC strategy, which is the
reduction of frame delay for voice traffic. This phenomenon
is a consequence of giving absolute priority to voice over web
traffic. Furthermore, we observe a reduction on the voice
frame delay variation (jitter), which is specially advantageous
when dealing with packed voice.
Fig. 7 represents the delay histograms for voice frames
obtained with the shared VCC strategy corresponding to
100%, 95% and 90% of the sum of capacities required for
separated VCCs. The histogram for the only-voice scenario is
shown in the upper left corner for comparison. As the
capacity of the shared VCC decreases, the jitter reduction
tends to vanish.
4. CONCLUSIONS
In this paper we have presented a simulation model suitable
for the dimensioning of the UMTS access network interface
between each Node-B and its controlling RNC. This tool
supports multi-service scenarios and implements basic forms
of both ATM and AAL2-level traffic differentiation. It
provides a wide range of results, including the most
significant performance measurements (frame loss ratio and
delay among them), allowing for QoS-aware dimensioning
figures to be obtained.
Several simulation results have been also shown, both for
single class and for multi-class scenarios. Variable rate nature
of some applications makes it possible to obtain statistical
multiplexing gain (i.e. bandwidth savings) if they use a
Voice Web
common transmission capacity. Bandwidth savings can be
-4
< 10 < 10-3 slightly increased if several traffic classes share resources.
< 25 ms n/a This strategy, however, poses a more important advantage,
756 kbit/s 257 kbit/s which is the possibility of reducing jitter for real time traffic.
 20% 20%
18% 18%
16% Only voice 16% Shared VCC (100%)
14% 14%

12% 12%

10% 10%

8% 8%

6% 6%

4% 4%

2%
2%

0% 0%
0,00 2,00 4,00 6,00 8,00 10,00 0,00 2,00 4,00 6,00 8,00 10,00

Frame delay (ms) Frame delay (ms)

20% 20%
18% 18%
16% Shared VCC (95%) 16% Shared VCC (90%)
14% 14%
12% 12%
10% 10%
8% 8%
6% 6%
4% 4%
2% 2%
0% 0%
0,00 2,00 4,00 6,00 8,00 10,00 0,00 2,00 4,00 6,00 8,00 10,00

Frame delay (ms) Frame delay (ms)

Fig. 7. Voice frame delay histogram: single class (voice) and shared scenarios (100%, 95% and 90% of VCC capacity sum).

[10] O. Isnard, J.-M. Calmel, A.-L. Beylot, and G. Pujolle, “Handling

REFERENCES
[1] 3GPP, “QoS Concept and Architecture,” 3GPP TS 23.107.
[2] S. Dixit, Y. Guo, and Z. Antoniou, “Resource Management and
Quality of Service in Third-Generation Wireless Networks,” IEEE
Communications Magazine, pp 125-133, February 2001.
[3] 3GPP, “UTRAN Overall Description,” 3GPP TS 25.401.
[4] A. Reyes-Lecuona, E. González-Parada, E. Casilari, J. C. Casasola, and
A. Díaz-Estrella, “A page-oriented WWW traffic model for wireless
system simulations,” Proceedings of the 16th International Teletraffic
Congress (ITC'16), Edinburgh, United Kingdom, pp. 1271-1280, June
1999.
[5] A. Klemm, C. Lindemann, and M. Lohmann, “Traffic Modeling and
Characterization for UMTS Networks,” Proceedings of the Globecom,
Internet Performance Symposium, San Antonio TX, November 2001.
[6] ETSI, “Universal Mobile Telecommunications System (UMTS);
Selection procedures for the choice of radio transmission technologies
of the UMTS,” TR 101 112 V3.2.0, April 1998.
[7] 3GPP, “IP Transport in UTRAN,” 3GPP TR 25.933.
[8] 3GPP, “AMR Speech Codec; General Description,” 3GPP TS 26.071.
[9] ITU-T Recommendation I.363.2, “B-ISDN ATM Adaptation layer
specification: Type 2 AAL,” November 2000.
Traffic Classes at AAL2 / ATM layer over the Logical Interfaces of the
UMTS Terrestrial Access Network,” 11th IEEE International
Symposium on Personal, Indoor and Mobile Radio Communication,
PIMRC, London, England, Vol. 2, pp. 1464-1468, September 2000.
[11] G. Eneroth, G. Fodor, G. Leijonhufvud, A. Rácz, and I. Szabó,
“Applying ATM/AAL2 as a Switching Technology in Third-Generation
Mobile Access Networks,” IEEE Communications, Vol. 37 No. 6, pp.
112-122, June 1999.
[12] S.-K. Yoo and H.-S. Park, “Quality-of-Service Provisioning for Mobile
Voice and Data Services over ATM Network using AAL2,” 3rd
ICACT, Muju, Korea, 2001.
[13] H. Lim, S. Lee, D. Lee, K. Kim, K. Song, and C. Oh, “A New AAL2
Scheduling Algorithm for Mobile Voice and Data Services over ATM,”
ITC-CSCC 2000, Pusan, Korea, vol. 1, pp. 229-232, July 2000.
[14] J.-L.C. Wu, C.-H. Huang, and R.-T. Sheu, “Performance study of
AAL2 protocol for low-bit-rate multimedia services,” 15th International
Conference on Information Networking, Proceedings, pp. 793–798,
2001.
[15] O. Hersent, D. Gurle, J.-P. Petit, “IP Telephony - Packet-based
multimedia communications systems,” Chapter 7, Addison-Wesley,
2000.

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