SIMULATION ANALYSIS OF PACKET SCHEDULING ALGORITHM FOR

VOICE, WWW AND VIDEO STREAMING SERVICES IN UMTS

DOWNLINK FDD MODE

Marko Porjazoski, Borislav Popovski

Department of Telecommunications, Faculty of Electrical Engineering, University St.s Cyril and Methodius

Keywords: UMTS, WCDMA, Radio Resource Management (RRM), Packet Scheduling, DCH, DSCH

Abstract: UMTS provides a new and important featur

e allowing negotiation of the property of the radio bearer. We

have focused on data transmission in WCDMA systems using packet scheduling for DCH and DSCH, in

case when voice, video streaming and WWW services are engaged. Network exploiting the DCH and

DSCH as a data transport channels can provide higher throughput than a network without DSCH, if a good

combination of resource sharing between the DCH and DSCH is select.

1 INTRODUCTION

The third-generation system known as the Universal

Mobile Telecommunication System (UMTS)

introduces very variable data rates on the air

interface, as well as the independence of the radio

access infrastructure and the service platform.

UMTS allows user application to negotiate bearer

characteristics that are most appropriate for carrying

information. For users this makes available a wide

spectrum of services through the Wideband Code

Division Multiple Access (WCDMA). The variable

bit rate and variety of traffic on the air interface have

presented new possibilities for radio resource

utilization (Holma, Toskala, 2001)(Laiho et al.,

2001).

The expression Radio Resource Utilization

RU) covers all functionality for handling the air

interface resources of a radio access network. These

functions together are responsible for supplying

optimum coverage, offering the maximum planned

capacity, guaranteeing the required quality of service

(QoS) and ensuring efficient use of physical and

transport resources (3GPP TS 25.211). The Radio

Resource Management (RRM) consists of Power

Control (PC). Handover control (HC), congestion

control (typically subdivided into admission control

(AC), load control (LC) and packet data scheduling)

and the resource manager (RM).

In this paper we are focused on simulation

anal

ysis of performances of packet scheduling

algorithm proposed in (Sallent et al., 2001). In the

simulations 7 omnidirectional

BS’s (Base Stations) are assumed with variable

num

ber of users uniformly distributed around the

scenario, using voice, video and WWW services.

The paper is organized as fallows: at the

begi

nning (section 2), we describe downlink

transport channels (Downlink Dedicated Channel-

DCH and Downlink Shared Channel-DSCH) used

for voice and data transmission in our simulations;

in third section some aspects of interference and

code management are described with the formulas

for interference and load factor estimation; section

four is description of packet scheduling algorithm to

be simulated; section five presents simulation

scenario, in sections six simulation results are

discussed and section seven concludes the paper.

2 TRANSPORT CHANNELS

UMTS have a reach set of dedicated channels and

shared channels. These channels can support

multimedia applications ranging from voice to best-

effort data.

Two transport channels caring a downlink

ransmission are (3GPP TS 25.211):

a) Dedicated Channel (DCH) – is available

exclusi

vely to a specific user. Dedicated Channel is

devoted to services with stringent transfer delay

requirements (voice and video). The transmission

rate of a DCH can be changed every 10 ms.

Porjazoski M. and Popovski B. (2004).

SIMULATION ANALYSIS OF PACKET SCHEDULING ALGORITHM FOR VOICE, WWW AND VIDEO STREAMING SERVICES IN UMTS DOWNLINK

FDD MODE.

In Proceedings of the First International Conference on E-Business and Telecommunication Networks, pages 67-73

DOI: 10.5220/0001396700670073

 SciTePress

b) Downlink Shared Channel (DSCH) – is

defined to support flexible multiplexing of bursty

data traffic (WWW) in WCDMA. DSCH is usually

used to support best-effort data services, as it cannot

provide guarantees for the service quality. Its

Transmission

capacity is divided up among several

users. The number of users multiplexed on DSCH

varies with time. Since the base station may transmit

to many users at one time, co-channel interference

exists.

Depending on the type of service to be

provided, transport channels should be managed and

allocated appropriately. In this paper we are focused

on conversational (voice), streaming (video) and

interactive (www) services, which quality

requirements may be expressed trough achieved bit

rate, percentage of lost packets, delay and the delay

jitter.

In order to differentiate quality levels for

streaming services, we assume two layered video

application that is characterized by two different

flows: a basic layer, with the minimum requirements

for a proper visualization, and an enhancement layer,

that contains additional information to improve the

quality of the received images. Data traffic from

basic layer will be transmitted through DCH

channel, along with conversational (voice) traffic,

while the enhancement layer will be transmitted only

if there is capacity in the DSCH channels. It is

assumed that the possible retransmissions of the

basic layer can be carried out in the DSCH channel

together with the enhancement layer, and having a

higher precedence than the latter.

In case of interactive (WWW) service users, data

is transmitted trough DSCH

3 INTERFERENCE AND CODE

MANAGEMENT

Radio Resource Management (RRM) covers all

functionalities for handling the air interface

resources of a radio access network. These functions

are responsible for supplying optimum coverage,

offering the maximum planned capacity,

guaranteeing the required QoS and ensuring efficient

use of physical and transport resources (Laiho et al.,

2001)(3GPP TS 23.107). Good resource allocation

schemes will aim to assign many as possible links

with adequate SIR to mobile users. Resource

assignment is restricted by the interference caused

by the BS and terminals when they start using

assigned resources. RRM well not assign resources

to a terminal if this assignment would cause

excessive interference to other users. Decision about

who should transmit and its transmission parameters

(transport format and power level) are the

responsibility of the packet scheduler.

For the n users transmitting simultaneously at a

given cell, the following inequality for i-th user must

be satisfied (Sallent et al., 2001):

TiT

othiN

⋅

⎟

⎠

⎞

⎜

⎝

⎛

≥

⎥

⎦

⎤

⎢

⎣

⎡

−

×++

−

)(

(1)

(2)

∑

TiT

PT – base station transmitted power; PTi –

power devoted to i-th user; Ii-oth – intercell

interference observed by i-th user; Lp(di) – i-th user

path loss, r – channel coding rate; PN – background

noise; SFi – spreading factor, relating i-th user data

bit rate and chip rate; ρ - orthogonality between

codes used in downlink direction.

If we rearrange equation (1), we will obtain

condition that PTi have to satisfy:

⋅

⎟

⎠

⎞

⎜

⎝

⎛

×++

≥

−

dLP

othiN

ipTi

)(

(3)

Adding all n (users) inequalities, the total power

transmitted by the base station can be expressed as:

∑

⋅

⎟

⎠

⎞

⎜

⎝

⎛

−

=≥

max

)(

)1(

(4)

where ηDL is load factor defined as (Holma,

Toskala, 2001):

∑

−

⎟

⎠

⎞

⎜

⎝

⎛

⎥

⎦

⎤

⎢

⎣

⎡

ipothi

dLI

)(

. (5)

Another common restriction is the number of

available codewords that BSs can use. Beside

interference control, packet scheduler should

manage dynamical allocation of OVSF codes. The

system has SFmax orthogonal codes with maximum

ICETE 2004 - WIRELESS COMMUNICATION SYSTEMS AND NETWORKS

spreading factor SFmax = 512. According to the

properties of these codes, there availability is

guaranteed when the Kraft’s inequality is satisfied

(Minn, Siu, 2000):

∑

≤

max

(6)

where Rb,i is number of bits in transport block (TB)

for i-th user and Rb is minimal number of bits in TB

(corresponding to spreading factor SFmax = 512).

4 ALGORITHM DESCRIPTION

The scheduling strategy used in our simulations is

presented in (Sallent et al., 2001). Algorithm is

divided in two parts:

Kraft’s inequality

n + 1 users

(n +1, t) < P

Tmax

(n +1, t) <

granted

transmission

postponed

transmission

TF > 0

TF = TF - 1

4.1 Prioritization

All users intended to transmit information must be

classified according to a certain prioritization

criteria. Firs scheduler will order requests depending

on service class they belong to, from highest to

lowest priority level. Conversational service class

(voice) has highest priority. Second, in this

prioritization scale, is streaming video service.

WWW (interactive service) has lowest priority.

Second prioritization rule is based on number of

basic layer TB in user’s buffer waiting for

retransmission (This rule can be applied only for

DSCH allocation, because there is no TB to be

retransmitted in DCH). When two or more users

have the same number of TB to be (re)transmitted, a

third prioritization level based on the Service Credit

(SCr) is considered.

The SCr is associated with an active link or user

and it computes the difference between the bit rate

requested by the user and the bit rate that system

provides to him. So it calculates the amount of

service that system owes to the user. The SCr value

of each active connection must be updated every

TTI (Transport Time Interval), following the

expression:

)1()1()( −−+−= kNumTx

kSCrkSCr

(7)

where SCr(k) is the Service Credit for TTI=k,

SCr(k-1) is the Service Credit in the previous TTI,

RG is the guaranteed bit rate measured in bits/TTI,

TB is the number of bits in Transport Block for the

considered RAB (radio Access Bearer) and

NumTx(k-1) is the number of successfully

transmitted Transport Blocks in the previous TTI.

4.2 Resource Allocation and

Availability Check

Once requests are ordered, the next step will be to

decide whether or not they are accepted for

transmission and which is the accepted TF. The

limitations dealing with interference and code

availability are taken into account in this phase. For

this purpose we have to estimate the expected load

factor and transmitted power level once all the

requests are accepted. So, the expected load factor in

system with n active transmissions in frame t can be

estimated using equation (8).

Figure 1 : Resource allocation process

∑

−

⎟

⎠

⎞

⎜

⎝

⎛

⎥

⎦

⎤

⎢

⎣

⎡

×−

ipothi

dLtI

)()1(

),(

. (8)

The expected power is given by:

SIMULATION ANALYSIS OF PACKET SCHEDULING ALGORITHM FOR VOICE, WWW AND VIDEO

STREAMING SERVICES

∑

⋅

⎟

⎠

⎞

⎜

⎝

⎛

−

tnP

)(

)),(

),(

(9)

Here we have to mention that some differences

between real and estimated load factor value may

occur, as a consequence of inaccuracies in the

measurement of the other-to-own-cell interference

and path loss.

Algorithm execution follows the flow shown on

Figure 1. Assuming n already granted transmissions

and initially selected TF for n+1 request, the Kraft’s

inequality is evaluated, the expected load factor is

compared with a threshold φ and expected

transmission power should be below the maximum

transmitted power. If all tree conditions are satisfied,

transmission is granted for this request during one

TTI, otherwise, the transport format is reduced by

one (i.e. transmission bit rate is reduced). If this is

not possible, the request should wait for the next

frame.

5 SIMULATION SCENARIO

The system model includes 7 omnidirectional base

stations. Distance between two neighboring base

stations is 1 km. Maximum transmitted power by the

base station is 43 dBm. Mobile users are uniformly

distributed in the scenario moving with speed of 50

km/h. At each position update (every TTI) we

assume that mobile user will change his direction to

left or to right in 45° with probability 0.1 for each

side, and probability 0.8 that he will stay on previous

course. Path loss model used in this simulation is

adopted from (3GPP TS 25 942) and path loss

calculations are made according to (10).

L = 128.1 + 37.6 Log10(R) (10)

R is a distance from BS to ME. Path loss

calculated buy (10) shall in no circumstances be less

than free space path loss – FSPL = 20log(4πR/λ) . If

during calculations L become smaller than FSPL,

then FSPL should be considered instead L as path

loss.

After L (FSPL) is calculated, log-normally

distributed shadowing (LogF) with standard

deviation of 10 dB should be added, so that the

resulting path loss is the following:

Pathloss = L + LogF (11)

Number of voice, video and www users is taken

to be on of the parameters which will be changing

during simulations.

Traffic generation model proposed in (Perez-

Romero, 2002)(3FPP TR 101 112-UMTS 30.03) is

used, including following parameters for:

- Conversational Service (Voice traffic): On-Off

model with 0.3 activity factor. In the active period

voice users generates 160 bits in 10 ms (16 kbps).

- Interactive Service (WWW traffic): Session

arrival process – Poisson process; Number of packet

call requests per session – geometrically distributed

with mean 5 calls per session; Reading time between

packet calls – geometrically distributed with a mean

33 [s]; Number of packets within a packet call –

geometrically distributed with a mean value 25; Inter

arrival time between packets – geometrically

distributed with a mean 0.0625 [s] (for 64 kbps);

Packet size – Pareto Distribution (with cut-off), max

packet size 66666 bytes with parameters α=1.1,

k=81.5.

- Video Streaming Basic and Enhancement

Service: CBR model; bit rate 32 kbps (each 40 ms

packet with 1280 bits is generated);

In our simulations we have adopted: 40 ms for

TTI (Transport Time interval), 320 bits Transport

Block Size for WWW and Video streaming service

and 160 bits Transport Block Size for conversational

service. DCH used for transmission of voice and

video streaming basic layer will have two transport

formats (TFs): TF0 – no transmission and TF1

transmission of 4 Transport Blocks (TBs) in TTI.

Transport formats for DSCH, used for transport of

WWW and video streaming enhancement layer, are

listed in Table 1.

Table 1: Transport Formats for Downlink Shared Channel

(DSCH)

TB sizes, bits 320 bits (payload) + 16 bits

(MAC/RLC header)

TF0, bits

0×320

TF1, bits

1×320 (8 Kb/s)

TF2, bits

2×320 (16 Kb/s)

TF3, bits

4×320 (32 Kb/s)

TF4, bits

8×320 (64 Kb/s)

TFS

TF5, bits

16×320 (128 Kb/s)

TTI, ms 40

Duration of this simulation corresponds to 3 min

in real time. Life time of voice and video services

packets was set to 1s and for www service to 10s. If

the packet remains in the buffer longer then his life

time it will be discarded.

ICETE 2004 - WIRELESS COMMUNICATION SYSTEMS AND NETWORKS

6 RESULTS

Results from the simulation of packet scheduling

algorithm described in this paper are shown in the

figures below. We have considered average user bit

rate, packet loss, delay and delay jitter as measures

that will help as to estimate system behavior and

performances. Figures provide a clear view of

system behavior in regard of number of users in the

scenario and the load factor threshold. Load factor

and power estimations, as it's shown in equation (8)

and (9), are based on interference measured in

previous TTI.

As a result of prioritization of different service

classes, it is obvious that best performances system

will provide to voice users (as service class with

highest priority), then to video users and at the

finally to www users

As you can see from the results, average bit rate,

packet loss and delay for the voice users are nearly

constant irrespectively to number of video and www

users and load factor (unless a small degradation of

performances for load factor values around 1).

On the other hand, system performances

experienced by video streaming service users

(second level in prioritization scale), appears to be

more depend on number of voice users and load

factor. It can be a little confusing, the packet delay

decrease for load factor in the order of 1. The

explanation for this behavior is that number of lost

packets for video users tremendously increases, and

in calculation of average packet delay we don take

this packets into account. Real system behavior for

the video streaming service can be noticed from

average bit rate, packet loss and packet delay jitter.

And finally, www service is treated as best effort

service class. When there is available resources left

by voice and video users, it will be allocated to www

users. As a result of smaller granularity of transport

formats, www user traffic is more adaptive to

severe system conditions.

7 CONCLUSIONS

In this paper, the performance of packet scheduling

algorithm, used the WCDMA network area with

seven cells, exploiting the DCH and DSCH was

studied by simulations. It was shown that as the

traffic load is increasing, the network resource

utilization increases until it reach the maximum. If

load is further increased, the resource utilization

saturates and starts decreasing again.

For smaller load factor threshold values (0.8-

0.9), packet scheduler does not provide best

performances. Load factor is the limiting factor that

do not allow new transmissions to be granted. For

increased load threshold values (values from 0.9 to

0.95) system provides best performances: maximum

average user bit rate, minimal packet loss, delay and

jitter.

For higher numbers of users and load factor

values around 1, difference between estimated and

real value of load factor is growing, system is

becoming unstable, base station transmitted power

reach its maximum value faster and becomes main

obstacle in provisioning better performances.

The results suggest that there is an optimal set of

parameters (load factor values form 0.9 to 0.95 and

number of users around 150, 75 voice, 50 video and

25 www users), by which the network resources

utilization reaches the maximum.

3400

3500

3600

3700

3800

3900

4000

4100

4200

0.8 0.85 0.9 0.95 1

Load Factor

Voice users bit rate (bps)

75 voice users

100 voice users

125 voice users

150 voice users

Figure 2: Average bit rate for voice service users in

scenario with constant number of video streaming users

(50) and constant number of www users (25)

5000

10000

15000

20000

25000

0.8 0.85 0.9 0.95 1

Load Factor

video users bit rate (bps)

75 voice users

100 voice users

125 voice users

150 voice users

Figure 3: Average bit rate for 50 video streaming service

users (basic and enhancement layer) in scenario with

variable number of voice users and fixed number of www

users (25)

SIMULATION ANALYSIS OF PACKET SCHEDULING ALGORITHM FOR VOICE, WWW AND VIDEO

STREAMING SERVICES

200

400

600

800

1000

1200

0.8 0.85 0.9 0.95 1

Load Factor

www users bit rate (bps)

75 voice users

100 voice users

125 voice users

150 voice users

Figure 4: Average bit rate for 25 www service users in

scenario with variable number of voice users and fixed

number of video streaming service users (50)

3500

3600

3700

3800

3900

4000

4100

0.8 0.85 0.9 0.95 1

Load Factor

Voice users bit rate (bps)

25 video users

50 video users

75 video users

Figure 5: Average bit rate for 75 voice service users in

scenario with variable number of video streaming users

and fixed number of www service users (25)

0.8 0.85 0.9 0.95 1

Load Factor

Voice users Packet loss (%)

75 voice users

100 voice users

125 voice users

150 voice users

Figure 6: Packet loss for voice users in scenario with fixed

number of video streaming users (50) and number of

www service users (25)

100

0.8 0.85 0.9 0.95 1

Load factor

Video users packet loss

75 voice users

100 voice users

125 voice users

150 voice users

Figure 7: Packet loss for voice users in scenario with fixed

number of video streaming users (50) and number of

www service users (25)

0.1

0.2

0.3

0.4

0.5

0.6

0.7

0.8

0.8 0.85 0.9 0.95 1

Load Factor

Average delay (s)

75 voice users

50 video users

25 www users

Figure 8: Average packet delay in scenario with 75 voice

service users, 50 video streaming users and 25 www

service users

0.2

0.21

0.22

0.23

0.24

0.25

0.8 0.85 0.9 0.95 1

Load Factor

video users delaj jitter s)

25 video users

50 video users

75 video users

Figure 9: Packet delay jitter for video streaming users in

scenario with variable number of voice service users, 50

video streaming users and 25 www service users

ICETE 2004 - WIRELESS COMMUNICATION SYSTEMS AND NETWORKS

REFERENCES

Holma H., Toskala A., 2001. WCDMA for UMTS, John

Wiley & Sons.

Laiho J., Wacher A., Novosad T., 2001. Radio Network

Planning and Optimisation for UMTS, John Wiley &

Sons.

3GPP TS 23.107. Quality of Service (QoS) concept and

architecture

3GPP TS 25.211. Physical channels and mapping of

transport channels onto physical channels (FDD)

Sallent O., Pérez-Romero J., Casadevall F., Agustí R.,

2001. An Emulator Framework for a New Radio

Resource Management for QoS guaranteed Services in

W-CDMA Systems, In: IEEE Journal on Selected

Areas in Communications, Vol.19, No. 10, October

2001, pp. 1893-1904.

Minn T., Siu K.Y., 2000. Dynamic Assignment of

Orthogonal Variable-Spreading-Factor Codes in W-

CDMA, In: IEEE Journal on Selected Area in

Communications, August 2000, pp. 1429-1440.

Perez-Romero J., Sallent O., Agusti R., 2002. Downlink

Packet Scahouling for Two-Layered Streaming Video

Service in UMTS, In: IST Mobile & Wireless

Telecommunication Summit 2002, Thessaloniki –

Greece, June 2002, pp. 212-216.

3FPP TR 101 112-UMTS 30.03. Selection procedures for

the choice of radio transmission technologies of the

UMTS

3GPP TS 25 942 . RF system scenarios

SIMULATION ANALYSIS OF PACKET SCHEDULING ALGORITHM FOR VOICE, WWW AND VIDEO

STREAMING SERVICES