Speech on HS-DSCH

7/30/2019 Speech on HS-DSCH

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Examiner at the Royal

Institute of Technology:

Mats Bengtsson, Ph.D.

Supervisor at Ericsson: Supervisor at the Royal

Institute of Technology:

Stefan Parkvall, Ph.D. Lei Bao

Access Technologies and Signal Processing Signals, Sensors & Systems

Ericsson AB Royal Institute of Technology

Kista, Sweden Stockholm, Sweden

Speech on HS-DSCH

Niklas Lithammer

Master of Science Thesis in Signal Processing and Digital Communication

Access Technologies and Signal Processing at

Ericsson Research, Ericsson AB

Department of Signals, Sensors & Systems atRoyal Institute of Technology

IR-SB-EX-0322

December 2003


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OpenREPORT (M. SC. THESIS) 1 (56)

Prepared (also subject responsible if other) No.

Niklas Lithammer EAB/TU-03:000381 UenApproved Checked Date Rev Reference

EAB/TUR Mikael Hk S.P. 2003-12-17 A

Speech on HS-DSCH

Abstract

The third generation mobile communication system, based on WCDMA, is beingdeployed around the world. In the latest specification, Release 5, the support for

packet data is significantly improved. This is achieved by introducing the High-Speed Downlink Shared Channel, HS-DSCH. This channel offers increasedcapacity, reduced delays and high peak rates, made possible thanks to fastadjustments of the transmission parameters. The main principles are channeldependent scheduling, fast link adaptation and fast hybrid ARQ with softcombining.

The HS-DSCH delivers particularly good performance for best-effort dataservices, however some of the high-speed benefits can also be used to providea speech service, called HS-speech. Instead of using a high bit rate, thechannel-dependent scheduling and fast link adaptation are used to provide aquality of service, on a best-effort channel.

This work will examine the possibilities with the HS-speech approach andmeasure how this would affect data traffic throughput, which the HS-DSCHnormally is used for. The point is not to host only speech but to investigate thepossibility to transmit speech in conjunction with data traffic on a shareddownlink channel.

Transmitting speech on the HS-DSCH is more power efficient than speechservices on dedicated channels. On average the power consumption can nearlybe halved, but still delivering the same or even better speech quality. Thesebenefits are under the assumption that the associated DPCHs are excluded forthe HS-speech. When the communication system is used both for speech andweb traffic, the speech users will consume resources leading to a capacitydegradation for the web users. This degradation will be comparable regardless

of the channel type used to carry the voice service.


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Acknowledgements

This work was carried out at the department of Access Technologies at EricssonResearch in Kista, Stockholm.I would in particular like to thank my supervisor Stefan Parkvall for introducingme to the topic of HSDPA. He has been an enormous resource for me.I would also like to thank all of my co-workers at Access Technologies foranswering questions.Niklas Jaldn has been my opponent on this work. A special thanks for proofreading this report and giving constructive criticism and feedback.Lei Bao for reading the report.And to Azadeh Shakorian and all my family.

Niklas LithammerStockholm, November 2003


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Contents

1 Background.......................................................................................................... 9 1.1 Purpose of project.......................................................................................101.2 Outline ........................................................................................................10

2 WCDMA...............................................................................................................11 2.1 Cells and systems.......................................................................................11 2.2 WCDMA......................................................................................................12 2.3 Evolving WCDMA .......................................................................................13

2.3.1 Link adaptation.........................................................................142.3.2 Channel reports........................................................................142.3.3 Scheduling ...............................................................................142.3.4 Fast hybrid ARQ.......................................................................15

2.4 Channels in WCDMA..................................................................................15 3

Traffic Models.....................................................................................................19

3.1 Web-browsing.............................................................................................19 3.2 Speech........................................................................................................20 3.3 Summary ....................................................................................................20

4 Why Voice on HS-DSCH?..................................................................................21 4.1 Voice services on DCH...............................................................................22 4.2 Voice services on HS-DSCH ...................................................................... 22

4.2.1 Link adaptation.........................................................................244.2.2 Channel reports........................................................................254.2.3 Scheduling of voice users ........................................................254.2.4 Fast hybrid ARQ.......................................................................26

4.3 Summary ....................................................................................................26 4.4 Potential advantages..................................................................................27

5 Scheduling Algorithms......................................................................................29 5.1 Assumptions ...............................................................................................305.2 Scheduling variables...................................................................................305.3 Scheduling algorithms for voice..................................................................30

5.3.1 Maximum C/I scheduler, MAX..................................................31 5.3.2 Round Robin scheduler, RR ....................................................31

5.4 Web browsing scheduler ............................................................................315.5 Retransmission power allocation for voice .................................................32

5.5.1 Retransmission power allocation algorithm..............................325.6 Expectations ...............................................................................................33

6 Simulations.........................................................................................................35 6.1 System simulator ........................................................................................356.2 Definitions...................................................................................................36 6.3 Simulations .................................................................................................38

6.3.1 Delay results ............................................................................38 6.3.2 Bit rate results ..........................................................................40 6.3.3 Power consumption..................................................................416.3.4 Packet loss rate........................................................................436.3.5 Retransmission power allocation .............................................456.3.6 Comparison of voice scheduling algorithms.............................47

6.4 Conclusions ................................................................................................47


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7 Conclusions and Future Work..........................................................................49 7.1 Conclusions ................................................................................................497.2 Future work.................................................................................................49

Appendix A: Additional Simulation Plots .................................................................51 Appendix B: Abbreviations........................................................................................55 References...................................................................................................................56


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List of figures

Figure 1: A fading channel .................................................................................. 12Figure 2: Code tree ............................................................................................. 16Figure 3: Data transmission for web-browsing users .......................................... 19Figure 4: Data transmission for speech users ..................................................... 20Figure 5: Speech frame....................................................................................... 23Figure 6: The TTI error rate for the MCS used.................................................... 25Figure 7: Illustration of the retransmission power allocation algorithm................ 33Figure 8: 90th percentile of normalized user delay............................................... 39Figure 9: 90th and 50th percentiles of user bit rate ............................................... 41Figure 10: Average transmit power per speech user........................................... 42Figure 11: Speech frame error rate at load load.................................................. 44Figure 12: Speech frame error rate at higher load .............................................. 44Figure 13: 95th percentile of packet error rate ..................................................... 45Figure 14: Decreased power consumption for HS-speech users. ....................... 46Figure 15: Decreased amount of availible power for the HS-DSCH.................... 51Figure 16: Increased usage of the nonreserved code space .............................. 52Figure 17: Decreased bit rate for web-browsing users........................................ 53Figure 18: 90th percentile of normalized user delay............................................. 53Figure 19: Power distribution............................................................................... 54Figure 20: Decreasing retransmission frequency................................................ 54


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List of tables

Table 1: Speech properties for the two different speech types . ......................... 26

Table 2: Code reservation for the three differnt traffic combinations................... 37


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1 Background

Around the world, telecommunication companies are launching third generationmobile communication systems, based on Wideband Code Division MultipleAccess, WCDMA. In the latest release of the WCDMA standard one of theimprovements are called High Speed Downlink Packet Access, HSDPA. Thisnew concept of high-speed downlink services will meet increased demands forpacket services in the third generation mobile communication systems. TheHSDPA service is introduced to improve the support for best-effort packet datatraffic. The most important part of the HSDPA is the transport channel, the HS-DSCH, which is responsible for transmitting the bits. HS-DSCH stands for HighSpeed Downlink Shared Channel and is a channel that can be shared betweenseveral users.

The HSDPA supports a highly efficient usage of the available resources.

Therefore this service may be useful for transmitting speech in a more effectiveway than with dedicated channels. HSDPA is however designed to offer high bitrate in a best-effort sense, while a speech service requires some quality ofservice, QoS. These two types of traffic, packet and speech respectively, havedifferent demands on a communication system. Users browsing the Internet arefamiliar with the best-effort concept, which is based on a variable bit rate andwith low requirements on delay. Speech users on the other hand, require a fixedbit rate and only tolerate small variations in delay.

Despite the differences of the two traffic types, several of the high-speed benefitscan be used to fulfil the demands of a speech service. For example, instead ofmaximizing the bit throughput the resources can be distributed in such mannerthat the power consumption can be lowered, and the delays can be kept small,

fulfilling the demands of a speech service. The algorithm responsible for thisresource allocation is called a scheduler. The scheduler can take advantage ofthe high-speed concepts in a number of ways, for instance can the higher bit ratepossible be used to transmit the speech faster than with a dedicated speechchannel, i.e. with an instantaneous higher bit rate. However this is not the onlyimprovement, second is the possibility to retransmit if the speech packet cannotbe correctly decoded. Besides this, the fast transmission also creates a freedomto schedule the users when the channel conditions are favourable, resulting inan efficient link usage.

In this work the HS-DSCH will be shared between two types of users: speech-and web browsing-users. The speech users will always be prioritized ahead ofweb traffic, due to that the web traffic is of best effort type. This speech service

on the HS-DSCH is called HS-speech and is a new concept introduced andexplored in this work. The speech scheduler has to be efficient, i.e. not toexhaust the power resources nor to increase the web users delays too much.Except these restrictions, the performance of the communication system and thespeech users interference on the web traffic has to be supervised. In this workthe HS-speech will be compared to speech transmitted on dedicated channels,so called DCH-speech. These are the problems addressed by this thesis.


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1.1 Purpose of project

The purpose of the project is to propose a speech scheduler for the HS-speech

and to evaluate it by means of simulations. The results will reveal whichperformance and capacity a speech on HS-DSCH system can deliver. Whenusing a communication system for data traffic, the introduction of speech willdecrease the experienced capacity for data users. How big this degradation willbe, will be studied and these results will be compared between the two cases ofspeech users, i.e. the ones on dedicated channels and the ones on the HS-DSCH respectively.

1.2 Outline

The basic concepts of WCDMA are introduced in Chapter 2. Chapter 3 describesthe different traffic models used in the simulator. In chapter 4 the DCH-speechservice together with the HS-speech service are presented. Chapter 5 presents

the scheduling algorithms used in this work. Thereafter the simulator and thesimulations are presented and discussed in chapter 6. Finally conclusions aredrawn in chapter 7.


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2 WCDMA

WCDMA is one radio interface chosen for the third generation mobilecommunication system. This chapter gives a brief description of WCDMA,focusing on the issues relevant to this report. The reader with interests inWCDMA is referred to [1] for more details.

This chapter will describe the main principles of mobile communication systems,and mainly how WCDMA is developed for supporting high-speed data traffic.The description includes an introduction to the basic and most importantprinciples, which will be used in this work. This chapter also covers the differentchannels that will be used in this work and gives an introduction to the codeusage of WCDMA.

2.1 Cells and systems

Mobile communication systems have a structure of base stations covering anarea called a cell, and mobile terminals scattered in the cell. All the terminals arecommunicating with the base stations using the same shared radio frequencyspectrum. In order to separate different users transmissions there are a numberof alternatives. Two multiple access techniques are to separate different users intime or frequency. This is called Time Divisions Multiple Access, TDMA, andFrequency Divisions Multiple Access, FDMA, respectively. A third option is usedin Code Division Multiple Access, CDMA; in this case all transmissions use thesame frequency, and they are all simultaneous in time, instead different usersare separated by using different codes. These codes are called OrthogonalVariable Spreading Factor codes, OVSF, and they have particularly goodorthogonality properties useful in CDMA. These codes are used in downlink only.

In a FDMA system, different cells are assigned different frequencies. In CDMAon the other hand all cells use the same frequency. This is usually known asfrequency reuse one. The available frequency spectrum is limited and willtherefore be reused, which causes interference between different cells.

The medium between the base station and the terminal is the radio channel. Thechannel affects the propagating signal in different ways, e.g., fading. When asignal is subject to fading, the receiver experiences a time-varying signalstrength. Fading is due to the fact that scattered transmitted radio wavesinterfere, and at some locations is subject to constructive interference and atsome places destructive interference. When the receiving antenna moves thesignal strength will vary.

In Figure 1 an example of a fading channel is shown. As can be seen theamplitude is varying a lot, sometimes the channel is favourable and sometimesnot.


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Figure 1: An example of fading channel recorded after the receiver (the RAKE receiver),observe how the attenuation can decrease several dB in just a short moment. Thisexample is from a Typical Urban channel.

2.2 WCDMA

WCDMA, offers multiple access by allotting each user a unique code. Beforeevery transmission, the signal is multiplied with the user specific code and byusing the same code at the receiver it is possible to reproduce the desired userssignal and to suppress undesired signals. The code multiplication causes abandwidth expansion, which will be proportional to the spreading factor, SF, i.e.the number of chips per information bit. In WCDMA the chip rate is kept constant

at 3.84Mcps (Mega chips per second). Therefore different bit rates are achievedby using different spreading factors. The OVSF codes preserves orthogonality incases with different spreading factors.

In WCDMA the transmit power is varied in order not to use unnecessary power,so called power control. The variable transmitting power aims at that when auser has a disadvantageous channel quality more transmission power isallocated to that user. Consequently, more of the shared power resources arespent on users with unfavourable channel qualities. From a user point-of-viewthe system will be experienced as fair. But this fair resource distribution is not theoptimal when it comes to system throughput. By distributing the resources inanother way, the system throughput can be increased.

As with the downlink the uplink is also power controlled, i.e. the userequipments, UEs, transmit power is regulated. This is mainly done in order toreceive all signals with the same strength, and to avoid the signals from far awayusers to drown due to other users standing close to the base station.


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2.3 Evolving WCDMA

In the new Release 5 of WCDMA, there is a concept called HSPDA, which

introduces a number of improvements. Some examples of improvements are anincreased overall throughput and an improved support for best-effort serviceswith high peak bit rate. The higher throughput is achieved by spending moreresources on users with favourable channel qualities, instead of users withunfavourable channel qualities. This will of course be unfair in short termcompared to ordinary WCDMA, but it is very useful for best-effort services. Theaverage user throughput is increased, but the drawback is an increased varianceamong the user throughput. A best-effort service has no, or few, serviceguaranties. As the term states, the delivered throughput, or experienced delay, isa matter of available resources for the moment.

Instead of varying the transmit power as in the previous release of WCDMA, thebit rate is varied in the new release 5. Varying the modulation and coding

scheme adjust the bit rate. The key point is to achieve certain energy perreceived bit at the user terminal. This desired ratio could be fulfilled in two ways.First, as done in the previous release of WCDMA, this was achieved by varyingthe transmit power, and keeping the bit rate constant. In Release 5, the bit rate isvaried while the transmit power is kept constant. This results in a higher bit ratefor users with favourable channels and lower for user with unfavourablechannels. This adaptation due to the fading channel is called fast link adaptationand is done for every transmission time interval, TTI, which equals 2ms.

The link adaptation needs some information about the channel quality. Thereforeeach UE estimates the experienced channel and reports it back to the basestation. These estimates are called channel reports.

Further system performance can be gained if users are scheduled. A scheduleris an algorithm, which arranges in which order all users are allocated to thechannel. In order for the scheduler to choose the user with the most favourablechannel quality, the channel reports are used here as well.

The HS-DSCH can be shared in two ways. First if only one user is active in eachTTI, the system is shared by allotting different time slots to different users, socalled time multiplexing. In this case the available power resource is assigned toone user in each TTI, selected by the scheduler. This focusing on users withfavourable channels will increase the throughput. Second, if severaltransmissions are made simultaneously, i.e. in the same TTI, the system is saidto be code multiplexed as well. When the channel is code multiplexed differentusers are assigned different codes in order to share the channel. This extra

multiplexing is particularly useful when transmitting small sized packets, i.e. theuser is only assigned the amount of codes that is needed for the transmission.

If a transmission to a receiving UE fails, a retransmission from the base station isrequested. The algorithm responsible for these extra transmissions is the HybridARQ process, which is crucial for minimizing the delay times.


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The HSDPA works as follows. All active user terminals report their experiencedchannel quality to the base station. In the base station the scheduler ranks theusers based on the reported channel qualities, and selects one user to receive

data in the upcoming TTI. Once a user is selected, the link adaptation algorithmspecifies a suitable bit rate based on the channel quality and the amount ofavailable power. Thereafter the transmission is carried out on the HS-DSCH,using the specified bit rate and assigned power.

The link adaptation, channel reports, scheduler and the retransmissionalgorithm, Hybrid ARQ, are discussed briefly in the subsequent sections below.

2.3.1 Link adaptation

Link adaptation means that a suitable modulation and coding scheme, MCS, ischosen based on the instantaneous channel condition. A modulation and codingscheme is a specific set of parameters, which specifies a certain signal

constellation and coding rate. Different MCSs have different data rate. A highorder MCS has high data rate, but the link adaptation algorithm must not choosea higher MCS than the instantaneous channel quality permits. This is becausethe error probability will become high. If a too high MCS is selected, and if anerror occurs a retransmission is carried out which will lower the throughput.

2.3.2 Channel reports

Both the scheduler and the link adaptation need information about the channel.Therefore each UE estimates the experienced channel conditions and reports itback to the base station. These channel reports are called Channel QualityIndicator, CQI.

The measurements carried out in the UE will contain some measurement errors.In addition, there will be a delay introduced due to measurement periods andtransmission time. Therefore the channel condition measurements are onlyestimates and may result in a suboptimum decision by the scheduler or by thelink adaptation.

2.3.3 Scheduling

The key in achieving a high bit rate is to transmit when the channel is favourable.This is called fast channel dependent scheduling, and the benefit is to useinstantaneous radio conditions in the scheduling. The scheduler has to decidefor every TTI, which users to get access to the HS-DSCH.

In addition to radio conditions, based on the CQI reports, the scheduler can alsotake traffic priorities or other scheduling criteria into account. For example, aretransmission is prioritized ahead of a new transmission.


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2.3.4 Fast hybrid ARQ

In case a transport block is erroneously received it is retransmitted a few ms

later. The retransmissions decrease the throughput, since multiple time slots arespent on the same data. With the HS-DSCH the retransmissions are carried outby the base station using fast Hybrid ARQ techniques. The basic principle of fastHybrid ARQ is that received blocks that cannot be correctly decoded in the UEare not discarded, but instead stored and soft combined with subsequentlyreceived retransmissions. Since the erroneous data is stored and soft combined,the past time slots are not completely wasted. Each combination means thatmore energy per bit is received, which will decrease the error probability for eachretransmission. By soft combining, the throughput deterioration is not ascomprehensive as if not using fast Hybrid ARQ.

Two methods can be used for retransmissions; they are called Chase combiningand Incremental Redundancy. In case of Chase combining each retransmission

is an identical copy of the original transmission. Incremental Redundancy, on theother hand, allows for different coding and even different MCSs in eachtransmission attempt. Chase combining is a special case of IncrementalRedundancy. The hybrid ARQ can be seen as an implicit link adaptation,because the coding rate is adjusted based on the result of the decoding.

2.4 Channels in WCDMA

In a WCDMA system there are several different channels present. Some arebroadcast channels and some are user specific, i.e. transmitted with uniquecodes intended for unique users, called dedicated channels. The following listpresents some channels relevant for this work.

Common Pilot Information Channel, CPICH, broadcasts a predefined bitsequence used for channel estimation. Together with the known transmitpower for the CPICH the UEs estimates the instantaneous channelconditions and reports them back in the CQIs.

High Speed Downlink Shared Channel, HS-DSCH, is a channel whichsupports all the techniques described in section 2.3. When communicatingvia the HS-DSCH the receiving UE needs certain pieces of information priorto the transmission, in order to receive and decode it. The informationneeded by the receiver are for example how many and which codes that willbe used and when the transmission will take place. This kind of informationis sent on a channel called HS-SCCH.

High Speed Shared Control Channel, HS-SCCH, is broadcasting controlinformation. UEs are according to the standard supposed to be able toreceive up to four HS-SCCHs simultaneously. This will allow for four UEs toreceive data simultaneously in each cell. Remember how code multiplexingcan be used to transmit to several users in the same TTI.

Dedicated Physical Channel, DPCH, associated with the HS-DSCH. This isactually two channels, one for downlink and one for uplink. The downlinkDPCH carries information used in the uplink power control. The base station


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uses this power controls to regulate the UEs transmit power, in order toreceive all uplink transmissions with the same strength.

In downlink, there is a predefined set of codes available. This channelization-code resource is called a code tree, and is illustrated in Figure 2. The code treeis shared between all users active in downlink. The first node in the tree includesthe mandatory CPICH, which uses a high spreading factor of 256. Some codespace will also be reserved for HS-SCCHs (one or several), which each uses aspreading factor of 128. The rest of the tree may be used dynamically or bereserved for specific transmissions. One example of such specific usage is theHS-DSCH, which uses codes with a spreading factor of 16. In Figure 2, theexample reservation for HS-DSCH consists of 8 codes. According to Figure 2,the left part of the code tree, except for mandatory channels, can be used forassociated DPCHs or other channels present. For example if the communicationsystem is used for speech on dedicated channels as well, these channels willuse codes from the free (non reserved) space. A speech user receiving data on

a dedicated speech channel will reserve 1 code with spreading factor 128.

SF=16

SF=8

SF=4

SF=2

SF=1

Figure 2: Illustration of the code tree for downlink. In this example eight codes withspreading factor 16 are highlighted, and they are reserved for HS-DSCH transmissions.

A transmission to a single UE, using the HS-DSCH, will involve several channelsat the same time. The HS-DSCH is associated with two DPCHs, one for uplinkand one for downlink signalling, besides these channels there is also one HS-SCCH active. A numerical example with a user receiving downlink traffic on theHS-DSCH may look like this:

If the transmission requires 2 HS-DSCH codes, one associated DPCH and oneHS-SCCH, it will altogether include the part

256

35

128

1

256

1

16

21112 =++=++

SCCHHSDPCHDSCHHSSFSFSF

of the code tree, due to the fact that an active downlink DPCH will reserve onecode with spreading factor 256, i.e. SFDPCH = 256. Hence if eight codes arereserved for the HS-DSCH, as in Figure 2, six of them will be unused in thisexample. Notice that the HS-DSCH codes may be assigned to different users ineach TTI, depending on the schedulers decision.


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Remember that the chip rate is kept constant in WCDMA; therefore a highspreading factor will result in a low bit rate. That is why the channels used forsignalling have a high SF while the codes for HS-DSCH have a low SF.


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3 Traffic Models

A traffic model is a collection of parameters defining a model supposed to mimicthe real world. It is basically a model for how traffic is generated. In this work twotraffic models are used. The first is a web-browsing traffic model supposed toillustrate a typical behaviour for a user surfing the Internet. The second one isthe speech model, trying to mimic a speech user. These two models can betranslated to a system of parameters used in the simulator. Examples of suchparameters are bit rate and session time for a typical call and so on.

The outline of this chapter is to present the two traffic models used in this work:the web-browsing model, and the speech model and finally to summarize thesimilarities and differences.

3.1 Web-browsing

The web-browsing model is based on user sessions of random length. Thecreation of new sessions follows a Poisson process. For more details on Poissonprocesses the reader is referred to [6]. One session contains requests of objects(web pages) of lognormal distributed size. Between the requests there are anexponentially distributed reading time. It is supposed to illustrate a user surfingthe Internet requesting a new web page when done reading the previous one.

When using a best-effort service for web browsing, the most interesting concernis to get the requested data as fast as possible. The experienced delay willdepend on current traffic load, e.g. on the amount of available resources. In thistraffic model the requested data is of random size. As a consequence of thelimited resources the requested data may have to be divided into smaller

fragments. This fragmentation can be different at different requests, even thoughthe requested amount of data is the same. In Figure 3, an example of severalrequests of different sizes is illustrated.

Time

Random timeRandom quantity

Figure 3: Illustration of data transmission for web-browsing users. Requests for data froma user is drawn as an upward arrow, while downlink traffic to the requesting user is

drawn as a downward arrow. The reading time between different requests and the size ofthe requested data, are random every occasion. The simulations will only concerndownlink traffic.


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3.2 Speech

The human voice is complex to model, but in this case the important part, and

the one that will be summarized in the speech traffic model is really simple. Thevoice can be coded to a bit rate of 12.2kbps, with acceptable quality. In a typicalencoder the voice will be sampled and a 20ms segment of sound will be gatherin one block. This block can be transmitted and decoded separately fromsubsequent blocks. These simple facts are the fundamentals of the speechtraffic model. 12.2kbps is divided in 20ms block, each containing 244 bits. Theseblocks are to be delivered to the receiver every 20ms.

The call length is exponentially distributed, and the average length for a typicalcall is 90s. Thereafter the user is finished and leaves the system. The creation ofnew users follows a Poisson process. As long as the session (call) is active theuser is supposed to be talking. In the real world there are pauses and momentsof silence in the human speech. This can be modelled as a discontinuous voice

activity, but in this simulator the bit rate is kept fixed and will not be affected bythis discontinuity.

In Figure 4, the behaviour of the speech user traffic model is illustrated. Thereare no requests for data, but instead there is a constant flow of downlink trafficas long as the user is active. Compare the different traffic situations betweenspeech user and web-browsing user.

Time

Fixed time Fixed size

Figure 4: The speech user is passively receiving data. The time interval between differenttransmissions is fixed, and every transmission contains the same number of bits.

3.3 Summary

The two traffic models are, as seen by the simulator, only a stream of bitsentering the system at the base station and with purpose to be delivered to thereceiving UEs. Web-browsing users are interested in short delay once they haverequested data, whereas speech users most important concern is a continuousreception of data. The two traffic models only concerns downlink traffic.

The speech encoder in the UE needs to receive a new packet of data every20ms. If not, the output from the decoder will be noise or silence, experienced asannoying by the listener (i.e. the user). On the other hand, a user receiving datatraffic will not notice a lost packet, because it will be retransmitted until it isreceived successfully. The only consequence is a bigger delay before thedownload is completed.


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4 Why Voice on HS-DSCH?

The main topic of this chapter is to highlight the difference between transmittingspeech on dedicated channels, DCH-speech, compared to speech on HS-DSCH, so called HS-speech. Dedicated channels, DCHs, for voice have beendesigned to fulfil the demands of voice services, while the HS-DSCH isdeveloped to deliver good service for best-effort data traffic. The main usage ofthe HS-DSCH is for data traffic, but one interesting part to investigate is how thedata traffic will be affected if speech traffic is introduced in the same channel.Users surfing the Internet generate the data traffic normally present on the HS-DSCH. More details on the dedicated channels can be found in [1].

A speech user has some characteristics according to the traffic model, asdiscussed in the previous chapter. Examples are the generated bit rate, typicalsession length and so on. In this work the bit rate was assumed to be 12.2kbps.

A typical speech decoder needs to receive a packet every 20ms for the user toexperience a continuous speech session. These facts result in 50 blocks ofspeech per second, each containing 244 bits. The 20ms time slots are calledspeech frames, SpFs. In each of these SpFs 20ms of speech is transmitted tothe receiving UE. The most important distinction between the two types ofspeech users is how fast, i.e. bit rate and when in the SpF the speechtransmission will take place.

If the communication system should deliver adequate speech quality, there is alimited amount of SpFs that can be lost. Used in this work is the demand that95% of the users should not loose more than 1% of their SpFs. In other words,the 95th percentile1 of the speech frame error rate, SpFER, should not exceed1%. This quality measure will be used on both of the speech user types. There is

only a minor difference concerning the HS-speech users, which instead ofSpFER uses packet error rate, PER, but which will comprehend the samemeasurement.

This chapter begins with highlighting the differences between speech ondedicated channels and speech transmitted on the HS-DSCH. The basicconcepts from chapter 2 will once more be discussed and useful features will behighlighted. Next are a summary of the differences and similarities with the twospeech user types presented. Thereafter is an inventory about the possibleadvantages with transmitting speech over the data channel presented.

1The statistical term percentile will be defined in section 6.2


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4.1 Voice services on DCH

Over history, the most important service delivered by a mobile communication

system has been voice. Not until the past years, other demands have beendeveloped. In a WCDMA system, speech is transmitted on dedicated channels.These dedicated channels will in downlink reserve 1 code with spreading factor128. The reservation is valid as long as the user is active, i.e. as long as the callis ongoing. To admit a new speech user into the communication system, therehave to be code space available in the code tree. If it is enough code space free,a dedicated channel for the speech transmission can be set up. (Recall Figure 2for an illustration of the code tree.) When transmitting speech on dedicatedchannels the limit concerning how many simultaneous users that can be servedare a function of the available amount of power and codes. Typically the powerwill put an upper limit before the code resource is exhausted.

The speech users are power controlled in both uplink and downlink, in order to

keep the SpFER at a reasonable level. In downlink the power control is used notto waste power in the base station, because there is no advantage in achievingan unnecessary high success rate of received SpFs. The only concern is if thedata in the SpF gets through or not. An unnecessary high power usage in thebase station will make the interference higher at surrounding cells, and it willmake the reception harder for users with poorer channel conditions. Anotherreason is that a user standing close to the base station will need less power thana user standing far from the base station, near the border of the cell.

4.2 Voice services on HS-DSCH

A potential advantage of using a channel designed for data traffic whentransmitting speech is to provide a speech service in a more resource effective

way than with dedicated speech channels. The key point is to use the limitingresources, i.e. codes and power, differently compared to the dedicated channels.On dedicated channels a speech packet with 20ms of speech is transmitted in20ms. In the HS-DSCH the smallest TTI is 2ms, therefore the speech packet canbe transmitted in a tenth of the time, compared to a dedicated speech channel. Ifthe packet is received erroneously, there is enough time to retransmit.

When using the HS-DSCH for web-browsing traffic, the channel is assigned toone user per 2ms. This time multiplexing is the fundamental way of sharing theHS-DSCH and it is particularly good if the packets to transmit are large, i.e. all ofthe codes have to be assigned to a single receiver. On the other hand, if thepackets are small as they typically are with speech, there is need for codemultiplexing as well, i.e. several users will share the 2ms TTI. In this case the

codes are distributed between several users. The code multiplexing will demandmore than one HS-SCCH, which is enough when only transmitting to a singleuser at the time. Therefore the simulations with HS-speech users have a biggerpart of the code tree reserved for overhead channels, i.e. four HS-SCCHs ratherthan one. When the channel is shared between speech and web-browsing themaximum number of simultaneous users in each TTI is four. The HS-speechusers are always prioritised ahead of web-browsing users, and only if there arenot four such users the rest of the resources will be given to a web user.


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Two types of time structures will be used when discussing voice over HS-DSCH.The first is based on 20ms speech frames (coinciding with the time structure forvoice on DCH). The second time structure is based on 2ms subframes. These

are called TTIs and each SpF will consist of ten TTIs. Those ten TTIs will benumbered from 1 to 10. Each user active in the system will have a user specifictime structure, defining the starting point for the SpF. An illustration of the timestructure and an example of how different users are relative to each other isshown in Figure 5. Observe that each user has its own speech frame timestructure.

In this work there is a maximum of four simultaneous users on the HS-DSCH,because every UE is supposed to be able to decode four HS-SCCHs. If there isten TTIs in each SpF and four simultaneous users in parallel, this results in atotal of 40 TTIs per SpF. Therefore a theoretical upper limit to the maximumcapacity will be 40 users receiving speech over HS-DSCH, but this will demandno retransmissions if every user should be satisfied. In practice it is most likely to

have a retransmission frequency in the order of 50%, resulting in an upper limitof 40 / 1.5 = 26 simultaneous users. The 40 TTIs available in each SpF are alsoshown in Figure 5.

Within the time structure each user gets a start TTI defining when the users20ms time frame starts. The first TTI is numbered as number one and will becalled the start TTI. In every start TTI the buffer in the base station with data totransmit is filled with 244 new bits, which should be delivered in the present SpF.These new data bits illustrate the output from the speech encoder. As time goesby the TTI number is calculated modulo 10, and when the result is equal to ausers start TTI, the user gets a new speech packet and is ready to be scheduledfor next transmission.

Transmission Retransmission

Start of speech frame End of speech frame

U2 U5

U1 U3 U4 U6 U1

1 2 3 4 5 6 7 8 9 10 TTI nr

Figure 5: Illustrating two users U1 and U2 with their start TTI at TTI number one, theyboth get scheduled but only U2s transmission is successful. Therefore U1 requests a

retransmission in TTI number seven. The speech frame illustrated is for U1 and U2.Shown is also four other users (U3, U4, U5 and U6) having their start TTIs in U1s TTInumber two, four and five respectively, i.e. U3s SpF will start at TTI two. The time gapavailable for the first transmission and how this first choice will decide the retransmissiontime is also shown. The simple connection is TTI number one with seven, TTI numbertwo with eight and so forth.


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When transmitting packets on the HS-DSCH, a lost packet will cause aretransmission. If this retransmission should be useful the delayed packet has tobe delivered in the present speech frame. A delayed speech packet received

after the intended speech frame will be useless for the speech decoder.Assuming that every HS-speech user should have the possibility to retransmit alost speech packet, the delay before an ACK/NAK is returned is crucial. (ACKmeans an acknowledgement and NAK a negative acknowledgement.) In thiswork, the assumption of a 12ms delay was used, i.e. if a user is scheduled in hisstart TTI, then a retransmission can take place in TTI number seven. The delaytime will also limit how long the scheduler can postpone a user before beingforced to schedule him, in order to provide the possibility to retransmit. The lastmoment for retransmission is TTI number ten and therefore the first transmissionhas to be carried out in one of the first four TTIs. This transmissions-retransmissions relation is also illustrated in Figure 5. The 12ms delay includesdecoding and signalling of ACK/NAK, and it is the total time elapsed before thebase station knows there is time for a retransmission.

The introduction in chapter 2.3 to the principles of HS-DSCH will in this chapterbe more specific. The basic methods that enable high packet bit rate will oncemore be discussed and further analysed, in order to describe how they havebeen used in this work. Remember that the focus will not be on high bit rate, butrather on the demands introduced with a speech service, e.g. few packet errorsafter one retransmission. Some of the trade-offs needed to support the QoS willbe discussed in the upcoming subsections below.

4.2.1 Link adaptation

In this work, the packets with speech will always have the same size. Thereforea specially designed MCS can be selected once and for all. This MCS is chosen

in order to minimize the used resources. When transmitting fixed sized packetsthe flexibility will be the power allocation, rather than the variable bit rate.Remember that a variable bit rate tries to maximize the throughput for a givenchannel condition and a fixed amount of power. When the packet to transmit hasa fixed size, this kind of maximization is not needed. Instead the bit rate is keptfixed and the used power is tuned.

A convenient measure for channel quality is the carrier to interference ratio, C/I,where C corresponds to the users specific energy and I correspond tointerference energy.

In Figure 6, the TTI error rate for the MCS used when transmitting the speechpackets can be seen. This curve matches experienced C/I values to a probability

that the TTI is received correctly. Notice how steep the curve is, a smalldegradation in C/I can result in a lost packet. A term often used in this context isswitch point, which is the C/I value that results in 10% error rate. From Figure 6 itcan be concluded that the switch point for this MCS is about 13.5dB. The BlockError Rate curve is based on simulations in an AWGN channel, and this curve isused under the assumption that the channel conditions do not vary too muchunder a single transmission, i.e. under one TTI of 2ms. This assumption is validin this work thanks to that the simulated users are moving with a low speed(3km/h).


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Figure 6: The TTI error rate for the MCS used. The switch point (at 10% TTI error rate) isabout 13.5dB. The curve shown is based on simulations with an AWGN channel.

4.2.2 Channel reports

The CQI reports are based on measurements on the CPICH, which in this workwas known to be transmitting with a power of 2W. The experienced C/I for thistransmission is reported back to the base station and used to calculate theneeded amount of transmit power. If the UE is reporting a good channel theamounts of transmit power can be lowered, but still keeping the success rateunchanged. The point is to use as much power to achieve a sufficiently high C/Ivalue at the receiving UE. In this work the switch point were used as a target

value for the C/I. The CQI reports will only be estimates of the channel onceused in the base station, because of the transmission delay and themeasurement error made in the UE.

When a transmission takes place, the power its assigned is calculated based onthe delayed CQI report. The channel conditions actually experienced at themoment of transmission will be measured in the UE and reported back later on.Once this report is received the difference between the estimated channel andthe experienced one can be approximated, and this information can be used tolower the power consumption, as seen later on in chapter 5.

The CQI reports can also be used by the scheduler as a ranking of the channelqualities between different users.

4.2.3 Scheduling of voice users

The scheduler has to decide for every TTI which users that should get access tothe HS-DSCH. This decision can be based on different rules, for example thechannel conditions reported by the CQIs. In addition to radio conditions, thescheduler also can take traffic priorities into account. This will be very useful inthis work, because of the real time demands introduced when transmittingspeech. For example, retransmissions are given a higher priority over schedulingof new data.


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The scheduling is tightly connected to the algorithm that allocates power to thetransmissions. Therefore it would be optimal to always transmit when thechannel is favourable, but the time structure for speech limits this freedom, the

speech packet has to be delivered in the present speech frame. This is one ofthe trade-offs needed to fit a speech service on a best-effort channel.

4.2.4 Fast hybrid ARQ

The key goal with voice over HS-DSCH is to achieve a high success rate for thespeech packets, which will keep the speech users satisfied. This goal will beachieved if there are enough packets received successfully after twotransmissions, i.e. with one retransmission.

A first transmission that cant be decoded in the UE, will nevertheless contributeto the packet transmission. No transmissions are a waste of energy thanks to thesoft combining in the hybrid ARQ. Using less power in the first transmission will

create a high retransmission rate, but this wont be any problem thanks to thatthe sum of the two transmissions most of the time will result in a decodablepacket. The hybrid ARQ process will offer a possibility to by purpose use lesspower in the first transmission than a normal transmission would.

A high initial success rate (corresponding to a low retransmission rate) will useunnecessary much energy for a big amount of the users, because the receivedsignal strength will be unnecessary high. On the other hand, if the success rateis kept lower the users with good channels will still be satisfied, while the userswith unfavourable channels only have to request a retransmission, which willresult in a decodable packet at a later moment. As long as this delayed deliveryis kept within the stated time structure, the speech users will not notice anydifference.

The hybrid ARQ offers a possibility to lower the used energy when transmittingspeech, but this advantage comes with a drawback for the web-browsing users,e.g. a lot of retransmissions will use more codes.

4.3 Summary

In Table 1 the two types of speech users are compared according to importantspecifications. This summary is presented to highlight the differences andsimilarities of the two types of speech users.

Property Speech DCH-speech HS-speech

Bit rate 12.2kbps 12.2kbps

Transmission time 20ms (one SpF) 2ms (one TTI)

95th percentile of SpFER 1% 1% (SpFER equal to PER)

Retransmissions No Yes

Outer loop power control Yes No

Table 1: Speech properties for the two different speech types.


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4.4 Potential advantages

Several comparisons can be made when choosing between transmitting speech

on dedicated channels or on the HS-DSCH. The two ways of transmitting speechwill use different amounts of resources in different manners. This will affect theresources available for best-effort data traffic. The comparison between the twotypes of speech users will all be variations on the theme of link efficiency. Thequestion is how to use the available time, code and power resources mostoptimum, i.e. how can the resources available be used most efficient to transmitspeech to receiving UEs.

A basic ide when building a communication system is the benefit in savingresources possibly needed by someone else and also to host as many users aspossible, seen from a system operators point-of-view. More users will generatemore traffic, which in turn will generate more profit.

Different criterions and aspects will be used when comparing the speech users;some of them are discussed in the following key-points.

Power aspect

A major difference is that the DCH-speech users are using power every TTI,while the HS-speech users only consume power when they are scheduled, i.e. attransmission. For example, if a HS-speech user is using a lot of power in theTTIs he is scheduled it can nevertheless result in lower average powerconsumption, compared to a user consuming less power but doing it every TTI.

The available amount of power is limited; therefore the two types of speechusers will be compared according to used energy. The point is to host HS-

speech users in an efficient way, hopefully more power efficient than the DCH-speech users.

To compare how the two types of speech users influence the system capacity,we have to make sure that the speech users are of equal satisfaction. Asmentioned before, the quality measurement used in this work is that a systemwith satisfied speech users has a 95th percentile of the speech frame error rate(corresponding to a packet error rate for HS-speech users) of less than 1%.Therefore the HS-speech scheduling algorithms are tuned to achieve the samepacket loss rate, as the speech frame error rate for DCH-speech users. Onequestion now arises, if we have the possibility to retransmit, should the algorithmdeliberately use too little power, because we know that after a retransmissionalmost every packet is received successfully, thanks to the hybrid ARQ, or

should more resources be spent in the first transmission, to achieve a highersuccess rate in the first attempt. There is obviously a trade-off needed.


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Code aspect

A speech user on a dedicated channel will consume one code with spreadingfactor 128 every TTI. This is to be compared with a minimum HS-DSCHtransmission using one code with spreading factor 16, for a single TTI. Thereforesending speech on HS-DSCH will consume a smaller part of the code tree, if theretransmission frequency is kept lower than 25%. Because 1/16 every tenth TTIis the same as 1/160, and 1/128 is the same as 1.25/160. In this code usagecalculation the associated DPCH and the HS-SCCH have been excluded.

A more useful comparison than the one discussed above, is two study how thetwo speech user types affect the web-browsing users, which also are present inthe communication system.

System advantages

A base station complete with all hardware for WCMDA may be unnecessary ifthe sector it is covering is small. Therefore the use of smaller and simplerequipments might come handy if it nevertheless is capable of delivering al kindsof services to mobile communicating users. Imagine a simpler equipment onlycapable of communicating via the HS-DSCH, this would be a great simplificationif it anyhow could provide voice services. The main task should still be datatraffic services, but if a user requests voice service then it should be possible.

In the future, it would be an excellent simplification to transmit speech packets allthe way from the source to the receiver as a packet based service, withoutneeding the circuit switched service like todays voice communication systemsdo.

These simplifications are used in downlink only, therefore the base station stillhave to support different kinds of services in uplink. A speech user, for example,have to transmit speech too and not only receive.


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5 Scheduling Algorithms

A scheduler is an algorithm that decides which user that will receive data in thefollowing time slots. The scheduling algorithm picks a user or users based onsome criterion.

When several users are ready to use the HS-DSCH, they have to be scheduledin order to use the shared channel in an efficient way. When sharing the channelbetween the two different user-types, web-browsing and HS-speech, twodifferent schedulers are needed. The first one will schedule the HS-speechusers. This will prioritise the HS-speech users ahead of web-browsing users. Ifthere are resources left over, the second schedulers only concern is to schedulethe web-browsing users.

Altogether there will be three scheduling algorithms presented in this chapter,

two for speech and one for web-browsing users. When the amount of speechusers is increasing there will be some competition in getting the packets through.To highlight this situation two scheduling algorithms for speech have beentested.

Together with the scheduling there is an algorithm responsible for the powerallocation. Once a user is scheduled for transmission, a proper amount of powerwill be assigned to that user. A HS-speech user will be assigned enough poweras to achieve a predefined C/I value at the receiving UE, compared to a web-browsing user that will be assigned all power available. This difference is due tothat a speech user has a fixed bit rate, while a web-browsing user has a variablebit rate. Remember that there are two different methods for achieving certainenergy per received bit (i.e. C/I), at the receiving UE. If the amount of power is

fixed the bit rate can be varied (as for web-browsing users) or the amount ofpower is varied while the bit rate is kept fixed (as for HS-speech).

The outline of this chapter begins with a few assumptions made and then somevariables will be defined, which are used in the scheduling algorithms. This isfollowed by the description of the two speech scheduling algorithms examined inthis work. Thereafter the web browsing scheduler is presented, and aretransmission power allocation algorithm for HS-speech users is described.Last is a section about expectations on the speech schedulers.


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5.1 Assumptions

The two speech scheduling algorithms have several aspects in common. One of

these similarities is that every user is guaranteed the possibility to retransmit. If aspeech user has not been scheduled in his third TTI, the scheduler willautomatically chose him no matter of the original scheduling criterions, i.e. theuser will be scheduled in his forth TTI. This behaviour will guarantee thepossibility to retransmit in the last TTI, before the 20ms speech frame is over,thanks to that the ACK/NAK delay is 12ms. Recall Figure 5 for the transmission-retransmission relation. There is also a fundamental rule to prioritise aretransmission before scheduling new transmissions. If this has not been thecase, a lost first transmission would have been a complete waste of resources ifa second transmission never occurs. With this restriction the point is to giveresources to already initiated transmissions before starting new ones.

5.2 Scheduling variables

To measure the performance of the different schedulers and to express selectioncriterions, some variables are defined.

First is the delay time d, the number of TTIs a user has to wait before gettingscheduled. A few TTI delays will not necessarily be bad, the point is to try toschedule at good channel conditions. On the other hand, delays may pile up a lotof users in the same TTIs, resulting in users being even more delayed. Or evenworse, that the users never get the chance to be scheduled, because of themaximum number of simultaneous users or exhausted power resources. Oncethe scheduler has made its decision the delay time is updated for usage in thenext TTI, according to

!"#

+

=

schedulednotifdscheduledifd

j

j

10

)(

)((1)

where d(j) is the j:th users delay time.

Second is the maximum number of simultaneous users. This variable has beenfixed to four, because the 3GPP standard requires a UE to be able to decodefour HS-SCCHs simultaneously.

Third is the estimated channel quality,)( j

I

C , based on each users CQI report.

These channel estimates will be used to compare different users channel

conditions.

5.3 Scheduling algorithms for voice

The scheduling algorithm simply decides which users that will receive data inevery TTI. The selection follows different criterions for the different schedulers.Next the two algorithms for voice are presented. Both the speech algorithms willbe repeated several times in each TTI, until there is no power left or there are nomore users with data waiting.


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5.3.1 Maximum C/I scheduler, MAX

The maximum C/I scheduler, MAX, considers each users estimated channel

quality and selects the users with the highest values. This scheduler will try totake advantage of the channel reports available, and to make a wise decisionaccording to the channel quality ranking. This scheduling criterion will be efficientin distributing the resources. A user with a good channel will consume lesspower, making it possible to schedule several users simultaneously withoutexhausting the available power.

Expressed in mathematical form, the user is selected according to

$%

&'(

)=

)(

maxarg

j

j I

Ci . (2)

Where)( j

IC is the j:th users channel quality and i is the number of the selected

user.

5.3.2 Round Robin scheduler, RR

The Round Robin scheduler only considers each users delay time. The roundrobin method applied here, is to select the users with the longest waiting time.The main principle is to schedule every user in its first TTI. If that is not possiblea delayed user will get a higher priority in the next TTI, than a user experiencingits first TTI. This delay may occur if there are a lot of users or if they are not wellscattered in time, i.e. their first TTIs coincides a lot. This RR scheduler will notexplore the benefits of scheduling users with favourable channel qualities.

In mathematical form the algorithm becomes

[ ])(maxarg jj

di = (3)

where d(j) is the j:th users delay time.

The reader with interest in the fundamentals of Round Robin scheduling isreferred to [5].

5.4 Web browsing scheduler

The web-browsing users are scheduled with a max C/I criterion, equivalent toformula (2). The principle is to choose the user with the best channel estimate.This will maximize the throughput in each TTI. When a web-browsing user isscheduled he will be assigned all power available in the base station.Furthermore, based on the estimated channel quality the highest possible MCSis selected. This will maximize the bit throughput under a limited powerconstraint. Notice how the bit rate is varied while the amount of power is fixed.This is the opposite behaviour compared to the MAX voice scheduler, whichkeeps a constant bit rate while minimizing the used amount of power.


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5.5 Retransmission power allocation for voice

When allocating power to a transmission, the target value for the C/I at the

receiver is the switch point. The power allocation is based on the channelestimates, i.e. the CQI reports. If the first transmission fails it is probably causedby an optimistic CQI report, resulting in too little power. If the same powerallocation algorithm is used in a requested retransmission, the received energyin the first transmission is underestimated. Remember that the hybrid ARQ isstoring the energy from previously received transmissions.

In order to adjust the transmit power in the retransmission, an algorithm fortracking the changes in the CQI reports have been implemented. The algorithmwill be described in the following subsection. This power allocation algorithm willonly be used on HS-speech users. The web-browsing users are always assignedthe total available amount of power once they are scheduled, retransmission ornot.

5.5.1 Retransmission power allocation algorithm

As discussed earlier, the CQI reports are delayed and the channel qualitymeasurements are noisy. (Hence the delay is due to transmission and notintroduced by purpose.) This results in some drawbacks when the neededtransmit power is to be calculated.

First of all, the MAX scheduler prefers to schedule users that report too goodchannels. This will in average result in an underestimated amount of neededpower. (In some cases an improved channel condition cancels this powershortage, which results in a successful transmission anyhow.) In order toestimate the lack of power for the lost packets, all the CQI reports used for

calculating the transmit power are compared to the CQI reports returned fromthe moment of transmission. These reports will be received 4ms after thetransmission, but in time for a retransmission. The extra information is used tolower the amount of power assigned for the retransmission. Compared to thecase without retransmission power regulation the amount of power assigned fora retransmission is calculated based on the latest available CQI report only. Theordinary rule used is to assign as much power as to achieve the switch point C/I,C/Iswp, at the receiving UE. With the retransmission power allocation algorithmthe allocation is modified to the following rule: The C/Iswp is subtracted with theestimated C/I, C/Iest, from the first transmission. The C/I target value, C/I target, forthe retransmission can be expressed in mathematical terms as

estswpett I

C

sI

C

I

C=

arg (4)

where s is a safety margin, i.e. s(0,1).

This results in a lowered power allocation, but with a kept success rate. Thedifference is to estimate what the UE received in the first transmission, instead ofassuming that the accumulated energy in the hybrid ARQ is zero.


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The basics of the algorithm are illustrated in Figure 7. One drawback with thealgorithm is that the CQI report for the actual transmission time also containssome measurement errors. If there is no safety margin in the algorithm, the extra

information is of no use and will result in a lost packet anyhow. Just imaginewhat would happen if the second CQI report also signals a favourable channel,then the retransmission will get too low power and the packet will be lostanyhow.

The main reason to implement this algorithm is to lower the used power in theretransmissions. Compared to the ordinary power allocation one drawback withthis algorithm is that an extra CQI report is used, i.e. an extra measurement errorhas to be dealt with. Nevertheless the opportunities and benefits take over hand.The point is to use the extra information with care, and if used the powerconsumption will be lowered.

Tx1 Tx2

CQI1 CQI2 CQI3

Time

Figure 7: Illustration of the retransmission power allocation algorithm. CQI1 is used tocalculate the assigned amount of power for transmission one (Tx1). The CQI receivedfrom the moment of transmission (CQI2) is used to estimate the received C/I at thereceiver. This estimation is used in conjunction with CQI3 to assign a proper amount ofpower for the retransmission (Tx2) to be successful.

5.6 Expectations

The two speech scheduling algorithms are supposed to deliver the sameperformance at low loads, for instance comparable power consumption andsimilar affection on web users. This can be expected because if there are one ortwo users in a TTI, then the difference between choosing the one with the bestchannel or the one with the shortest delay time is almost the same. If there aretwo users and both of them gets scheduled then the order is not that important,both of the users will be able to retransmit and the difference in schedulingalgorithm will be hard to notice.

The overall difference will be noticeable at higher loads approaching the capacitylimit of speech users. In a case with only speech users the main limiting factorwill be the packet loss rate. The capacity limit will be reached when too manyusers are unsatisfied.

The retransmission power allocation algorithm is expected to lower the overallconsumed power for HS-speech users.


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6 Simulations

There are several ways of measuring the capacity of the HS-DSCH. Twomethods will be used in this work. The first is to plot the normalized user delayversus system throughput and the second will be user bit rate versus systemthroughput. These two plots will show how and how much the speech users willinterfere with the web traffic.

The capacity measurements have to be associated with some kind of qualitymeasurement on the speech. Therefore the amount of satisfied speech users willbe monitored according to the speech quality measurements discussed inchapter 4.

The simulations are presented both from a web-browsing user point-of-view andas a comparison of the two speech user types. The interesting part with the web-

browsing users is how they will be influenced when the communication system isused for speech as well. One major difference between the two types of speechusers is that the DCH-speech users have dedicated channels, only affecting theweb-browsing users indirectly. Compared to HS-speech users, which will use thesame channel as the web-browsing users, therefore limiting the availableresources in a more direct way. Both speech user types will be prioritised aheadof web-browsing users, and throughout this chapter the MAX scheduler for HS-speech has been used.

This chapter begins with a short introduction to the simulator, some assumptionsare made and then the simulations carried out are discussed and their resultsare presented, for example delay figures and capacity results. Severalcomparisons between DCH-speech and HS-speech users will be made. The

conclusions from the simulations are summarized and discussed in the lastsection.

6.1 System simulator

In this work a MATLAB based simulator was used, which simulates a WCDMAcommunication system with HS-DSCH functionality. The time resolution used ison slot basis 1/1500 seconds, each HS-DSCH TTI consists of 3 slots. Included inthe simulator are models for propagation, the physical layer and the trafficmodels.

The propagation model involves calculations of path gains between basestations and UEs. They are based on geographical environment model with

antennas, user movements, distance attenuation and fading. Each site consistsof three sectors and the area is wrapped at the borders to illustrate a closedarea. The speed of the users is fixed to 3km/h and the channel model used inthe simulations is called Typical Urban, defined in the 3GPP standard.

The physical layer model estimates the received signal strength at the mobileterminals and at the base stations. These C/I values are mapped to block errorrate values, defining the chance of successful reception.

The traffic models discussed in chapter three are also included in the simulator.


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For every 10ms the simulator updates some basic variables like the number ofusers, creates new users, terminates finished calls, generates new web pagerequests and so forth. Every 10ms contains 5 TTIs, for each TTI the scheduler

assigns users to the HS-DSCH, measures CQI reports and executes the hybridARQ process. Each TTI consists of 3 slots, in each of those the fast fading iscalculated, power is assigned and collection of slot performance is made.

The process of creating new users and terminating finished ones may result in atime varying number of users, but the target value for the number of speechusers will always be ten. Several of the simulations will study differentbehaviours of the communication system as a function of an increasing trafficload, which is done by increasing the number of web users in each sector.

The communication system simulations were run for several hundreds ofseconds, in order to reach a steady state not influenced by short timedifferences. The simulation time was also chosen in order to cover several

speech calls, because the average call length is 90s.

6.2 Definitions

One important result to observe when doing system simulations is the amount ofbits delivered from the system to the receiving UEs. In order to compare thesenumbers with other simulations and other kinds of communicating systems, thetotal number of bits is normalized with time, the number of sectors per site andthe number of MHz the bandwidth is occupying. Therefore the term known assystem throughput is defined as the total number of bits that the system hasdelivered per second, sector and MHz. A site is normally divided into threesectors, each covering a third of the area, usually 120 degrees. In WCDMA thebandwidth is 5MHz. Therefore the bit rate (in bits/second/site) is normalized with

the factor 15 = 3*5. System throughput (measured in kbps/sector/MHz) is ameasurement connected to the communication system.

Associated with individually users is a term called normalized user delay. It isdefined as the total waiting time divided by the sum of all the bits the user hasreceived. For example, if a user requests a web page while surfing the Internet,the normalized delay timer will start at the request moment and end once thedownload is complete. The normalized user delay can then be calculated as thetotal waiting time divided by the sum of all delivered bits. This term will bemeasured in s/Mbit.

Tightly connected to the normalized user delay is the user bit rate, measured inkbps. This ratio is calculated as the sum of all bits the user has received, divided

by the total waiting time that the user has experienced, i.e. the inverse of thenormalized user delay.

Some of the simulation results will be presented with CDFs, cumulativedistribution functions. A CDF is defined to be, for a distribution X, the probability

P(X x), for all values ofx. Also the statistical measurement percentile is used,

which returns the value x for which, given p, P(X x) = p is true.


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The two types of speech users will influence the web-browsing users in differentmanners. In order to compare these two situations from the web-browsing userspoint-of-view, the code reservation for the HS-DSCH will be set differently. When

the HS-speech users are present they are prioritised compared to web-browsingusers. A transmission to a HS-speech user will reserve one code with spreadingfactor 16 (recall Figure 2). If there are twelve codes reserved for HS-DSCH, thena web-browsing users possible amount of codes is decreased with one for everyscheduled HS-speech user. In the following simulations there is usually ten HS-speech users active in every sector. This will lead to an average of one codereserved for HS-speech every TTI, and if retransmissions are included there isan even bigger part reserved. These code conditions for web-browsing users,should be the same if the communication system instead is used for DCH-speech and web browsing. Therefore the amount of codes reserved for the HS-DSCH, when there are DCH-speech users present, is lowered from twelve toten. This will make the two cases of speech users comparable from a web-browsing users point-of-view. The code reservation for the HS-DSCH and the

HS-SCCH is summarized in Table 2.

Reserved code space for:

Traffic situation HS-DSCH HS-SCCH

Web 12 codes with SF 16 1 code with SF 128

Web & DCH-speech 10 codes with SF 16 1 code with SF 128

Web & HS-speech 12 codes with SF 16 4 codes with SF 128

Table 2: Code reservation for HS-DSCH and HS-SCCH at the three differnt trafficcombinations.

The web-browsing users have the lowest priority of them all, in other words,when all speech users have been taken care of, the rest of the resources, if any,are given to web browsing users.

According to the 3GPP standard, every HS-DSCH user should have anassociated downlink DPCH, but no information needed for the HS-DSCHtransmission is carried in this channel; instead necessary signalling is carried bythe HS-SCCH. The downlink DPCH is only carrying information from the basestation to the UE that concerns power control in the uplink. This information is ofcourse essential in a communication system and is assumed to be presentanyhow. Therefore the downlink DPCH has been excluded for HS-speech users,i.e. no code or power is reserved for such a channel. The associated DPCH hasonly been excluded for HS-speech users. The web surfing users have theirs

associated channels left.


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In all simulations made, the HS-SCCH has been assumed to transmit with thefixed power of 0.4W. This power is necessary for the HS-DSCH transmissionand will be included in the power consumption of the HS-speech users. In a

more advanced simulator the power allocation for the HS-SCCH may be madedynamic, with respect to which user it is intended for, i.e. to use power control.Compare the case with a user standing close to the base station with a userstanding far away. A decreased power consumption for the control channelswould be a benefit for the whole communication system, but these possibilitieswill not be explored in this work.

6.3 Simulations

The first case studied, is when using the HS-DSCH only for web browsing traffic.In order for the users to be satisfied the delays have to be kept small. Aconvenient way of showing the statistics is to plot different percentiles. The 90 thpercentile indicates the quality for most of the users and will be used frequently

in this work.

With the web-browsing capacity results as a starting point, the degradationexperienced when introducing speech users can easily be shown. The firstsimulations studied, in subsection 6.3.1 and 6.3.2, will discuss how the two typesof speech users affect the web-browsing users. Therefore these two subsectionswill contain three different curves in each plot, each as a result of one of thethree simulated situations. The three situations are: only web users, web plusDCH-speech and web plus HS-speech. These three curves will be labelledaccording two the simulated combination of traffic, e.g. web & speech means asimulation with web and DCH-speech users. The plots showing the results willalways show how the web-users experience the different situations.

The rest of the simulations, discussed in the after coming subsections willcompare the two speech user types to each other. This section is concluded witha specific study of the HS-speech users retransmission power allocation.

Documents

Speech on HS-DSCH