Speech On HS-DSCH

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 at Royal Institute of Technology
IR-SB-EX-0322
December 2003
Examiner at the Royal Institute of Technology: Mats Bengtsson, Ph.D. Supervisor at Ericsson: Stefan Parkvall, Ph.D. Access Technologies and Signal Processing Ericsson AB Kista, Sweden Supervisor at the Royal Institute of Technology: Lei Bao Signals, Sensors & Systems Royal Institute of Technology Stockholm, Sweden
Open REPORT (M. SC. THESIS)

Prepared (also subject responsible if other) No.
1 (56)
Niklas Lithammer
Approved Checked
EAB/TU-03:000381 Uen
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Speech on HS-DSCH
Abstract The third generation mobile communication system, based on WCDMA, is being deployed around the world. In the latest specification, Release 5, the support for packet data is significantly improved. This is achieved by introducing the HighSpeed Downlink Shared Channel, HS-DSCH. This channel offers increased capacity, reduced delays and high peak rates, made possible thanks to fast adjustments of the transmission parameters. The main principles are channel dependent scheduling, fast link adaptation and fast hybrid ARQ with soft combining. The HS-DSCH delivers particularly good performance for best-effort data services, however some of the high-speed benefits can also be used to provide a speech service, called HS-speech. Instead of using a high bit rate, the channel-dependent scheduling and fast link adaptation are used to provide a quality of service, on a best-effort channel. This work will examine the possibilities with the HS-speech approach and measure how this would affect data traffic throughput, which the HS-DSCH normally is used for. The point is not to host only speech but to investigate the possibility to transmit speech in conjunction with data traffic on a shared downlink channel. Transmitting speech on the HS-DSCH is more power efficient than speech services on dedicated channels. On average the power consumption can nearly be halved, but still delivering the same or even better speech quality. These benefits are under the assumption that the associated DPCHs are excluded for the HS-speech. When the communication system is used both for speech and web traffic, the speech users will consume resources leading to a capacity degradation for the web users. This degradation will be comparable regardless of the channel type used to carry the voice service.

2 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17

3 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Acknowledgements This work was carried out at the department of Access Technologies at Ericsson Research in Kista, Stockholm. I would in particular like to thank my supervisor Stefan Parkvall for introducing me 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 for answering questions. Niklas Jaldn has been my opponent on this work. A special thanks for proof reading this report and giving constructive criticism and feedback. Lei Bao for reading the report. And to Azadeh Shakorian and all my family. Niklas Lithammer Stockholm, November 2003

4 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17

5 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Contents 1 Background ..........................................................................................................9 1.1 Purpose of project.......................................................................................10 1.2 Outline ........................................................................................................10 WCDMA...............................................................................................................11 2.1 Cells and systems.......................................................................................11 2.2 WCDMA......................................................................................................12 2.3 Evolving WCDMA .......................................................................................13 2.3.1 Link adaptation.........................................................................14 2.3.2 Channel reports........................................................................14 2.3.3 Scheduling ...............................................................................14 2.3.4 Fast hybrid ARQ.......................................................................15 2.4 Channels in WCDMA..................................................................................15 Traffic Models.....................................................................................................19 3.1 Web-browsing.............................................................................................19 3.2 Speech........................................................................................................20 3.3 Summary ....................................................................................................20 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.........................................................................24 4.2.2 Channel reports........................................................................25 4.2.3 Scheduling of voice users ........................................................25 4.2.4 Fast hybrid ARQ.......................................................................26 4.3 Summary ....................................................................................................26 4.4 Potential advantages ..................................................................................27 Scheduling Algorithms......................................................................................29 5.1 Assumptions ...............................................................................................30 5.2 Scheduling variables...................................................................................30 5.3 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 ............................................................................31 5.5 Retransmission power allocation for voice .................................................32 5.5.1 Retransmission power allocation algorithm..............................32 5.6 Expectations ...............................................................................................33 Simulations.........................................................................................................35 6.1 System simulator ........................................................................................35 6.2 Definitions ...................................................................................................36 6.3 Simulations .................................................................................................38 6.3.1 Delay results ............................................................................38 6.3.2 Bit rate results ..........................................................................40 6.3.3 Power consumption..................................................................41 6.3.4 Packet loss rate........................................................................43 6.3.5 Retransmission power allocation .............................................45 6.3.6 Comparison of voice scheduling algorithms.............................47 6.4 Conclusions ................................................................................................47

6 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Conclusions and Future Work ..........................................................................49 7.1 Conclusions ................................................................................................49 7.2 Future work.................................................................................................49
Appendix A: Additional Simulation Plots .................................................................51 Appendix B: Abbreviations ........................................................................................55 References ...................................................................................................................56

7 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
List of figures Figure 1: A fading channel .................................................................................. 12 Figure 2: Code tree ............................................................................................. 16 Figure 3: Data transmission for web-browsing users .......................................... 19 Figure 4: Data transmission for speech users ..................................................... 20 Figure 5: Speech frame ....................................................................................... 23 Figure 6: The TTI error rate for the MCS used .................................................... 25 Figure 7: Illustration of the retransmission power allocation algorithm................ 33 Figure 8: 90th percentile of normalized user delay............................................... 39 Figure 9: 90th and 50th percentiles of user bit rate ............................................... 41 Figure 10: Average transmit power per speech user........................................... 42 Figure 11: Speech frame error rate at load load.................................................. 44 Figure 12: Speech frame error rate at higher load .............................................. 44 Figure 13: 95th percentile of packet error rate ..................................................... 45 Figure 14: Decreased power consumption for HS-speech users. ....................... 46 Figure 15: Decreased amount of availible power for the HS-DSCH.................... 51 Figure 16: Increased usage of the nonreserved code space .............................. 52 Figure 17: Decreased bit rate for web-browsing users........................................ 53 Figure 18: 90th percentile of normalized user delay............................................. 53 Figure 19: Power distribution............................................................................... 54 Figure 20: Decreasing retransmission frequency ................................................ 54

8 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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

9 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Background
Around the world, telecommunication companies are launching third generation mobile communication systems, based on Wideband Code Division Multiple Access, WCDMA. In the latest release of the WCDMA standard one of the improvements are called High Speed Downlink Packet Access, HSDPA. This new concept of high-speed downlink services will meet increased demands for packet services in the third generation mobile communication systems. The HSDPA service is introduced to improve the support for best-effort packet data traffic. The most important part of the HSDPA is the transport channel, the HSDSCH, which is responsible for transmitting the bits. HS-DSCH stands for High Speed Downlink Shared Channel and is a channel that can be shared between several users. The HSDPA supports a highly efficient usage of the available resources. Therefore this service may be useful for transmitting speech in a more effective way than with dedicated channels. HSDPA is however designed to offer high bit rate in a best-effort sense, while a speech service requires some quality of service, QoS. These two types of traffic, packet and speech respectively, have different demands on a communication system. Users browsing the Internet are familiar with the best-effort concept, which is based on a variable bit rate and with low requirements on delay. Speech users on the other hand, require a fixed bit rate and only tolerate small variations in delay. Despite the differences of the two traffic types, several of the high-speed benefits can be used to fulfil the demands of a speech service. For example, instead of maximizing the bit throughput the resources can be distributed in such manner that the power consumption can be lowered, and the delays can be kept small, fulfilling the demands of a speech service. The algorithm responsible for this resource allocation is called a scheduler. The scheduler can take advantage of the high-speed concepts in a number of ways, for instance can the higher bit rate possible be used to transmit the speech faster than with a dedicated speech channel, i.e. with an instantaneous higher bit rate. However this is not the only improvement, second is the possibility to retransmit if the speech packet cannot be correctly decoded. Besides this, the fast transmission also creates a freedom to schedule the users when the channel conditions are favourable, resulting in an efficient link usage. In this work the HS-DSCH will be shared between two types of users: speechand web browsing-users. The speech users will always be prioritized ahead of web 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 and explored in this work. The speech scheduler has to be efficient, i.e. not to exhaust the power resources nor to increase the web users delays too much. Except these restrictions, the performance of the communication system and the speech users interference on the web traffic has to be supervised. In this work the HS-speech will be compared to speech transmitted on dedicated channels, so called DCH-speech. These are the problems addressed by this thesis.

10 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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 which performance and capacity a speech on HS-DSCH system can deliver. When using a communication system for data traffic, the introduction of speech will decrease the experienced capacity for data users. How big this degradation will be, will be studied and these results will be compared between the two cases of speech users, i.e. the ones on dedicated channels and the ones on the HSDSCH respectively.
1.2 Outline
The basic concepts of WCDMA are introduced in Chapter 2. Chapter 3 describes the different traffic models used in the simulator. In chapter 4 the DCH-speech service together with the HS-speech service are presented. Chapter 5 presents the scheduling algorithms used in this work. Thereafter the simulator and the simulations are presented and discussed in chapter 6. Finally conclusions are drawn in chapter 7.

11 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
WCDMA
WCDMA is one radio interface chosen for the third generation mobile communication system. This chapter gives a brief description of WCDMA, focusing on the issues relevant to this report. The reader with interests in WCDMA 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 important principles, which will be used in this work. This chapter also covers the different channels that will be used in this work and gives an introduction to the code usage of WCDMA.
2.1 Cells and systems

Mobile communication systems have a structure of base stations covering an area called a cell, and mobile terminals scattered in the cell. All the terminals are communicating with the base stations using the same shared radio frequency spectrum. In order to separate different users transmissions there are a number of alternatives. Two multiple access techniques are to separate different users in time or frequency. This is called Time Divisions Multiple Access, TDMA, and Frequency Divisions Multiple Access, FDMA, respectively. A third option is used in Code Division Multiple Access, CDMA; in this case all transmissions use the same frequency, and they are all simultaneous in time, instead different users are separated by using different codes. These codes are called Orthogonal Variable Spreading Factor codes, OVSF, and they have particularly good orthogonality properties useful in CDMA. These codes are used in downlink only. In a FDMA system, different cells are assigned different frequencies. In CDMA on the other hand all cells use the same frequency. This is usually known as frequency reuse one. The available frequency spectrum is limited and will therefore be reused, which causes interference between different cells. The medium between the base station and the terminal is the radio channel. The channel affects the propagating signal in different ways, e.g., fading. When a signal is subject to fading, the receiver experiences a time-varying signal strength. Fading is due to the fact that scattered transmitted radio waves interfere, and at some locations is subject to constructive interference and at some places destructive interference. When the receiving antenna moves the signal strength will vary. In Figure 1 an example of a fading channel is shown. As can be seen the amplitude is varying a lot, sometimes the channel is favourable and sometimes not.

12 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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. This example is from a Typical Urban channel.
2.2 WCDMA
WCDMA, offers multiple access by allotting each user a unique code. Before every transmission, the signal is multiplied with the user specific code and by using the same code at the receiver it is possible to reproduce the desired users signal and to suppress undesired signals. The code multiplication causes a bandwidth 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 achieved by using different spreading factors. The OVSF codes preserves orthogonality in cases 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 a user has a disadvantageous channel quality more transmission power is allocated to that user. Consequently, more of the shared power resources are spent on users with unfavourable channel qualities. From a user point-of-view the system will be experienced as fair. But this fair resource distribution is not the optimal when it comes to system throughput. By distributing the resources in another way, the system throughput can be increased. As with the downlink the uplink is also power controlled, i.e. the user equipments, UEs, transmit power is regulated. This is mainly done in order to receive all signals with the same strength, and to avoid the signals from far away users to drown due to other users standing close to the base station.

13 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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 an increased overall throughput and an improved support for best-effort services with high peak bit rate. The higher throughput is achieved by spending more resources on users with favourable channel qualities, instead of users with unfavourable channel qualities. This will of course be unfair in short term compared to ordinary WCDMA, but it is very useful for best-effort services. The average user throughput is increased, but the drawback is an increased variance among the user throughput. A best-effort service has no, or few, service guaranties. As the term states, the delivered throughput, or experienced delay, is a matter of available resources for the moment. Instead of varying the transmit power as in the previous release of WCDMA, the bit 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 per received 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 varying the transmit power, and keeping the bit rate constant. In Release 5, the bit rate is varied while the transmit power is kept constant. This results in a higher bit rate for users with favourable channels and lower for user with unfavourable channels. This adaptation due to the fading channel is called fast link adaptation and is done for every transmission time interval, TTI, which equals 2ms. The link adaptation needs some information about the channel quality. Therefore each UE estimates the experienced channel and reports it back to the base station. These estimates are called channel reports. Further system performance can be gained if users are scheduled. A scheduler is an algorithm, which arranges in which order all users are allocated to the channel. In order for the scheduler to choose the user with the most favourable channel 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 each TTI, the system is shared by allotting different time slots to different users, so called time multiplexing. In this case the available power resource is assigned to one user in each TTI, selected by the scheduler. This focusing on users with favourable channels will increase the throughput. Second, if several transmissions are made simultaneously, i.e. in the same TTI, the system is said to be code multiplexed as well. When the channel is code multiplexed different users are assigned different codes in order to share the channel. This extra multiplexing is particularly useful when transmitting small sized packets, i.e. the user 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 is requested. The algorithm responsible for these extra transmissions is the Hybrid ARQ process, which is crucial for minimizing the delay times.

14 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
The HSDPA works as follows. All active user terminals report their experienced channel quality to the base station. In the base station the scheduler ranks the users 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 algorithm specifies a suitable bit rate based on the channel quality and the amount of available 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 retransmission algorithm, 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, is chosen based on the instantaneous channel condition. A modulation and coding scheme is a specific set of parameters, which specifies a certain signal constellation and coding rate. Different MCSs have different data rate. A high order MCS has high data rate, but the link adaptation algorithm must not choose a higher MCS than the instantaneous channel quality permits. This is because the error probability will become high. If a too high MCS is selected, and if an error 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 it back to the base station. These channel reports are called Channel Quality Indicator, 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 and transmission time. Therefore the channel condition measurements are only estimates and may result in a suboptimum decision by the scheduler or by the link 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 use instantaneous radio conditions in the scheduling. The scheduler has to decide for every TTI, which users to get access to the HS-DSCH. In addition to radio conditions, based on the CQI reports, the scheduler can also take traffic priorities or other scheduling criteria into account. For example, a retransmission is prioritized ahead of a new transmission.

15 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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 are spent on the same data. With the HS-DSCH the retransmissions are carried out by the base station using fast Hybrid ARQ techniques. The basic principle of fast Hybrid ARQ is that received blocks that cannot be correctly decoded in the UE are not discarded, but instead stored and soft combined with subsequently received retransmissions. Since the erroneous data is stored and soft combined, the past time slots are not completely wasted. Each combination means that more energy per bit is received, which will decrease the error probability for each retransmission. By soft combining, the throughput deterioration is not as comprehensive as if not using fast Hybrid ARQ. Two methods can be used for retransmissions; they are called Chase combining and Incremental Redundancy. In case of Chase combining each retransmission is an identical copy of the original transmission. Incremental Redundancy, on the other hand, allows for different coding and even different MCSs in each transmission attempt. Chase combining is a special case of Incremental Redundancy. 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 are broadcast channels and some are user specific, i.e. transmitted with unique codes intended for unique users, called dedicated channels. The following list presents some channels relevant for this work. Common Pilot Information Channel, CPICH, broadcasts a predefined bit sequence used for channel estimation. Together with the known transmit power for the CPICH the UEs estimates the instantaneous channel conditions and reports them back in the CQIs. High Speed Downlink Shared Channel, HS-DSCH, is a channel which supports all the techniques described in section 2.3. When communicating via the HS-DSCH the receiving UE needs certain pieces of information prior to the transmission, in order to receive and decode it. The information needed by the receiver are for example how many and which codes that will be used and when the transmission will take place. This kind of information is sent on a channel called HS-SCCH. High Speed Shared Control Channel, HS-SCCH, is broadcasting control information. UEs are according to the standard supposed to be able to receive up to four HS-SCCHs simultaneously. This will allow for four UEs to receive data simultaneously in each cell. Remember how code multiplexing can be used to transmit to several users in the same TTI. Dedicated Physical Channel, DPCH, associated with the HS-DSCH. This is actually two channels, one for downlink and one for uplink. The downlink DPCH carries information used in the uplink power control. The base station

16 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
uses this power controls to regulate the UEs transmit power, in order to receive all uplink transmissions with the same strength. In downlink, there is a predefined set of codes available. This channelizationcode resource is called a code tree, and is illustrated in Figure 2. The code tree is shared between all users active in downlink. The first node in the tree includes the mandatory CPICH, which uses a high spreading factor of 256. Some code space will also be reserved for HS-SCCHs (one or several), which each uses a spreading factor of 128. The rest of the tree may be used dynamically or be reserved for specific transmissions. One example of such specific usage is the HS-DSCH, which uses codes with a spreading factor of 16. In Figure 2, the example 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 for associated DPCHs or other channels present. For example if the communication system is used for speech on dedicated channels as well, these channels will use 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=1 SF=2 SF=4 SF=8 SF=16
Figure 2: Illustration of the code tree for downlink. In this example eight codes with spreading 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 channels at the same time. The HS-DSCH is associated with two DPCHs, one for uplink and one for downlink signalling, besides these channels there is also one HSSCCH active. A numerical example with a user receiving downlink traffic on the HS-DSCH may look like this: If the transmission requires 2 HS-DSCH codes, one associated DPCH and one HS-SCCH, it will altogether include the part
1 SFHS DSCH
1 SFDPCH
1 SFHS SCCH
2 1 1 35 + + = 16 256 128 256
of the code tree, due to the fact that an active downlink DPCH will reserve one code with spreading factor 256, i.e. SFDPCH = 256. Hence if eight codes are reserved for the HS-DSCH, as in Figure 2, six of them will be unused in this example. Notice that the HS-DSCH codes may be assigned to different users in each TTI, depending on the schedulers decision.

17 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Remember that the chip rate is kept constant in WCDMA; therefore a high spreading factor will result in a low bit rate. That is why the channels used for signalling have a high SF while the codes for HS-DSCH have a low SF.

18 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17

19 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Traffic Models
A traffic model is a collection of parameters defining a model supposed to mimic the real world. It is basically a model for how traffic is generated. In this work two traffic models are used. The first is a web-browsing traffic model supposed to illustrate a typical behaviour for a user surfing the Internet. The second one is the speech model, trying to mimic a speech user. These two models can be translated to a system of parameters used in the simulator. Examples of such parameters 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 the similarities and differences.
3.1 Web-browsing
The web-browsing model is based on user sessions of random length. The creation of new sessions follows a Poisson process. For more details on Poisson processes the reader is referred to [6]. One session contains requests of objects (web pages) of lognormal distributed size. Between the requests there are an exponentially distributed reading time. It is supposed to illustrate a user surfing the Internet requesting a new web page when done reading the previous one. When using a best-effort service for web browsing, the most interesting concern is to get the requested data as fast as possible. The experienced delay will depend on current traffic load, e.g. on the amount of available resources. In this traffic model the requested data is of random size. As a consequence of the limited resources the requested data may have to be divided into smaller fragments. This fragmentation can be different at different requests, even though the requested amount of data is the same. In Figure 3, an example of several requests of different sizes is illustrated.
Random time Random quantity
Time
Figure 3: Illustration of data transmission for web-browsing users. Requests for data from a 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 of the requested data, are random every occasion. The simulations will only concern downlink traffic.

20 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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. The voice can be coded to a bit rate of 12.2kbps, with acceptable quality. In a typical encoder the voice will be sampled and a 20ms segment of sound will be gather in one block. This block can be transmitted and decoded separately from subsequent blocks. These simple facts are the fundamentals of the speech traffic model. 12.2kbps is divided in 20ms block, each containing 244 bits. These blocks are to be delivered to the receiver every 20ms. The call length is exponentially distributed, and the average length for a typical call is 90s. Thereafter the user is finished and leaves the system. The creation of new users follows a Poisson process. As long as the session (call) is active the user is supposed to be talking. In the real world there are pauses and moments of 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 by this discontinuity. In Figure 4, the behaviour of the speech user traffic model is illustrated. There are no requests for data, but instead there is a constant flow of downlink traffic as long as the user is active. Compare the different traffic situations between speech user and web-browsing user.
Fixed time
Fixed size
Time
Figure 4: The speech user is passively receiving data. The time interval between different transmissions 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 bits entering the system at the base station and with purpose to be delivered to the receiving UEs. Web-browsing users are interested in short delay once they have requested data, whereas speech users most important concern is a continuous reception of data. The two traffic models only concerns downlink traffic. The speech encoder in the UE needs to receive a new packet of data every 20ms. If not, the output from the decoder will be noise or silence, experienced as annoying by the listener (i.e. the user). On the other hand, a user receiving data traffic will not notice a lost packet, because it will be retransmitted until it is received successfully. The only consequence is a bigger delay before the download is completed.

21 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Why Voice on HS-DSCH?
The main topic of this chapter is to highlight the difference between transmitting speech on dedicated channels, DCH-speech, compared to speech on HSDSCH, so called HS-speech. Dedicated channels, DCHs, for voice have been designed to fulfil the demands of voice services, while the HS-DSCH is developed to deliver good service for best-effort data traffic. The main usage of the HS-DSCH is for data traffic, but one interesting part to investigate is how the data 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 HSDSCH. More details on the dedicated channels can be found in [1]. A speech user has some characteristics according to the traffic model, as discussed in the previous chapter. Examples are the generated bit rate, typical session 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 to experience a continuous speech session. These facts result in 50 blocks of speech per second, each containing 244 bits. The 20ms time slots are called speech frames, SpFs. In each of these SpFs 20ms of speech is transmitted to the receiving UE. The most important distinction between the two types of speech users is how fast, i.e. bit rate and when in the SpF the speech transmission will take place. If the communication system should deliver adequate speech quality, there is a limited amount of SpFs that can be lost. Used in this work is the demand that 95% 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 exceed 1%. 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 of SpFER uses packet error rate, PER, but which will comprehend the same measurement. This chapter begins with highlighting the differences between speech on dedicated channels and speech transmitted on the HS-DSCH. The basic concepts from chapter 2 will once more be discussed and useful features will be highlighted. Next are a summary of the differences and similarities with the two speech user types presented. Thereafter is an inventory about the possible advantages with transmitting speech over the data channel presented.
The statistical term percentile will be defined in section 6.2

22 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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 been developed. In a WCDMA system, speech is transmitted on dedicated channels. These dedicated channels will in downlink reserve 1 code with spreading factor 128. The reservation is valid as long as the user is active, i.e. as long as the call is ongoing. To admit a new speech user into the communication system, there have 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 2 for an illustration of the code tree.) When transmitting speech on dedicated channels the limit concerning how many simultaneous users that can be served are a function of the available amount of power and codes. Typically the power will 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 not to waste power in the base station, because there is no advantage in achieving an unnecessary high success rate of received SpFs. The only concern is if the data in the SpF gets through or not. An unnecessary high power usage in the base station will make the interference higher at surrounding cells, and it will make the reception harder for users with poorer channel conditions. Another reason is that a user standing close to the base station will need less power than a 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 when transmitting 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 limiting resources, i.e. codes and power, differently compared to the dedicated channels. On dedicated channels a speech packet with 20ms of speech is transmitted in 20ms. In the HS-DSCH the smallest TTI is 2ms, therefore the speech packet can be transmitted in a tenth of the time, compared to a dedicated speech channel. If the packet is received erroneously, there is enough time to retransmit. When using the HS-DSCH for web-browsing traffic, the channel is assigned to one user per 2ms. This time multiplexing is the fundamental way of sharing the HS-DSCH and it is particularly good if the packets to transmit are large, i.e. all of the codes have to be assigned to a single receiver. On the other hand, if the packets are small as they typically are with speech, there is need for code multiplexing 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 demand more than one HS-SCCH, which is enough when only transmitting to a single user at the time. Therefore the simulations with HS-speech users have a bigger part of the code tree reserved for overhead channels, i.e. four HS-SCCHs rather than one. When the channel is shared between speech and web-browsing the maximum number of simultaneous users in each TTI is four. The HS-speech users are always prioritised ahead of web-browsing users, and only if there are not four such users the rest of the resources will be given to a web user.

23 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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 for voice 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 be numbered from 1 to 10. Each user active in the system will have a user specific time structure, defining the starting point for the SpF. An illustration of the time structure and an example of how different users are relative to each other is shown in Figure 5. Observe that each user has its own speech frame time structure. 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 is ten TTIs in each SpF and four simultaneous users in parallel, this results in a total of 40 TTIs per SpF. Therefore a theoretical upper limit to the maximum capacity will be 40 users receiving speech over HS-DSCH, but this will demand no 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 limit of 40 / 1.5 = 26 simultaneous users. The 40 TTIs available in each SpF are also shown in Figure 5. Within the time structure each user gets a start TTI defining when the users 20ms time frame starts. The first TTI is numbered as number one and will be called the start TTI. In every start TTI the buffer in the base station with data to transmit 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 goes by the TTI number is calculated modulo 10, and when the result is equal to a users start TTI, the user gets a new speech packet and is ready to be scheduled for next transmission.
Start of speech frame Transmission
End of speech frame Retransmission
U2 U1 1
U3 2
U5 U4 4
U6 5
U1 7
10
TTI nr
Figure 5: Illustrating two users U1 and U2 with their start TTI at TTI number one, they both 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 TTI number two, four and five respectively, i.e. U3s SpF will start at TTI two. The time gap available for the first transmission and how this first choice will decide the retransmission time is also shown. The simple connection is TTI number one with seven, TTI number two with eight and so forth.

24 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
When transmitting packets on the HS-DSCH, a lost packet will cause a retransmission. If this retransmission should be useful the delayed packet has to be 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 a lost speech packet, the delay before an ACK/NAK is returned is crucial. (ACK means an acknowledgement and NAK a negative acknowledgement.) In this work, the assumption of a 12ms delay was used, i.e. if a user is scheduled in his start TTI, then a retransmission can take place in TTI number seven. The delay time will also limit how long the scheduler can postpone a user before being forced to schedule him, in order to provide the possibility to retransmit. The last moment for retransmission is TTI number ten and therefore the first transmission has to be carried out in one of the first four TTIs. This transmissionsretransmissions relation is also illustrated in Figure 5. The 12ms delay includes decoding and signalling of ACK/NAK, and it is the total time elapsed before the base station knows there is time for a retransmission. The introduction in chapter 2.3 to the principles of HS-DSCH will in this chapter be more specific. The basic methods that enable high packet bit rate will once more be discussed and further analysed, in order to describe how they have been used in this work. Remember that the focus will not be on high bit rate, but rather on the demands introduced with a speech service, e.g. few packet errors after one retransmission. Some of the trade-offs needed to support the QoS will be discussed in the upcoming subsections below. 4.2.1 Link adaptation In this work, the packets with speech will always have the same size. Therefore a 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 packets the 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 given channel condition and a fixed amount of power. When the packet to transmit has a fixed size, this kind of maximization is not needed. Instead the bit rate is kept fixed 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 to interference energy. In Figure 6, the TTI error rate for the MCS used when transmitting the speech packets 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 small degradation in C/I can result in a lost packet. A term often used in this context is switch point, which is the C/I value that results in 10% error rate. From Figure 6 it can be concluded that the switch point for this MCS is about 13.5dB. The Block Error Rate curve is based on simulations in an AWGN channel, and this curve is used under the assumption that the channel conditions do not vary too much under a single transmission, i.e. under one TTI of 2ms. This assumption is valid in this work thanks to that the simulated users are moving with a low speed (3km/h).

25 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Figure 6: The TTI error rate for the MCS used. The switch point (at 10% TTI error rate) is about 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 work was known to be transmitting with a power of 2W. The experienced C/I for this transmission is reported back to the base station and used to calculate the needed amount of transmit power. If the UE is reporting a good channel the amounts of transmit power can be lowered, but still keeping the success rate unchanged. The point is to use as much power to achieve a sufficiently high C/I value 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 once used in the base station, because of the transmission delay and the measurement error made in the UE. When a transmission takes place, the power its assigned is calculated based on the delayed CQI report. The channel conditions actually experienced at the moment of transmission will be measured in the UE and reported back later on. Once this report is received the difference between the estimated channel and the experienced one can be approximated, and this information can be used to lower 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 channel qualities between different users. 4.2.3 Scheduling of voice users The scheduler has to decide for every TTI which users that should get access to the HS-DSCH. This decision can be based on different rules, for example the channel conditions reported by the CQIs. In addition to radio conditions, the scheduler also can take traffic priorities into account. This will be very useful in this work, because of the real time demands introduced when transmitting speech. For example, retransmissions are given a higher priority over scheduling of new data.

26 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
The scheduling is tightly connected to the algorithm that allocates power to the transmissions. Therefore it would be optimal to always transmit when the channel 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 of the 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 the speech packets, which will keep the speech users satisfied. This goal will be achieved if there are enough packets received successfully after two transmissions, i.e. with one retransmission. A first transmission that cant be decoded in the UE, will nevertheless contribute to the packet transmission. No transmissions are a waste of energy thanks to the soft 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 that the sum of the two transmissions most of the time will result in a decodable packet. The hybrid ARQ process will offer a possibility to by purpose use less power in the first transmission than a normal transmission would. A high initial success rate (corresponding to a low retransmission rate) will use unnecessary much energy for a big amount of the users, because the received signal strength will be unnecessary high. On the other hand, if the success rate is kept lower the users with good channels will still be satisfied, while the users with unfavourable channels only have to request a retransmission, which will result in a decodable packet at a later moment. As long as this delayed delivery is kept within the stated time structure, the speech users will not notice any difference. The hybrid ARQ offers a possibility to lower the used energy when transmitting speech, 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 important specifications. This summary is presented to highlight the differences and similarities of the two types of speech users. Property Bit rate Transmission time 95 percentile of SpFER Retransmissions Outer loop power control
th
Speech
DCH-speech 12.2kbps 20ms (one SpF) 1% No Yes
HS-speech 12.2kbps 2ms (one TTI) 1% (SpFER equal to PER) Yes No
Table 1: Speech properties for the two different speech types.

27 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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 speech will use different amounts of resources in different manners. This will affect the resources available for best-effort data traffic. The comparison between the two types of speech users will all be variations on the theme of link efficiency. The question is how to use the available time, code and power resources most optimum, i.e. how can the resources available be used most efficient to transmit speech to receiving UEs. A basic ide when building a communication system is the benefit in saving resources possibly needed by someone else and also to host as many users as possible, seen from a system operators point-of-view. More users will generate more 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. at transmission. For example, if a HS-speech user is using a lot of power in the TTIs he is scheduled it can nevertheless result in lower average power consumption, compared to a user consuming less power but doing it every TTI. The available amount of power is limited; therefore the two types of speech users will be compared according to used energy. The point is to host HSspeech users in an efficient way, hopefully more power efficient than the DCHspeech 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. As mentioned before, the quality measurement used in this work is that a system with 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 same packet loss rate, as the speech frame error rate for DCH-speech users. One question now arises, if we have the possibility to retransmit, should the algorithm deliberately use too little power, because we know that after a retransmission almost every packet is received successfully, thanks to the hybrid ARQ, or should more resources be spent in the first transmission, to achieve a higher success rate in the first attempt. There is obviously a trade-off needed.

28 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Code aspect A speech user on a dedicated channel will consume one code with spreading factor 128 every TTI. This is to be compared with a minimum HS-DSCH transmission using one code with spreading factor 16, for a single TTI. Therefore sending speech on HS-DSCH will consume a smaller part of the code tree, if the retransmission frequency is kept lower than 25%. Because 1/16 every tenth TTI is the same as 1/160, and 1/128 is the same as 1.25/160. In this code usage calculation the associated DPCH and the HS-SCCH have been excluded. A more useful comparison than the one discussed above, is two study how the two speech user types affect the web-browsing users, which also are present in the communication system. System advantages A base station complete with all hardware for WCMDA may be unnecessary if the sector it is covering is small. Therefore the use of smaller and simpler equipments might come handy if it nevertheless is capable of delivering al kinds of services to mobile communicating users. Imagine a simpler equipment only capable of communicating via the HS-DSCH, this would be a great simplification if it anyhow could provide voice services. The main task should still be data traffic 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 all the way from the source to the receiver as a packet based service, without needing the circuit switched service like todays voice communication systems do. These simplifications are used in downlink only, therefore the base station still have to support different kinds of services in uplink. A speech user, for example, have to transmit speech too and not only receive.

29 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Scheduling Algorithms
A scheduler is an algorithm that decides which user that will receive data in the following time slots. The scheduling algorithm picks a user or users based on some criterion. When several users are ready to use the HS-DSCH, they have to be scheduled in order to use the shared channel in an efficient way. When sharing the channel between the two different user-types, web-browsing and HS-speech, two different schedulers are needed. The first one will schedule the HS-speech users. This will prioritise the HS-speech users ahead of web-browsing users. If there are resources left over, the second schedulers only concern is to schedule the 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 speech users is increasing there will be some competition in getting the packets through. To highlight this situation two scheduling algorithms for speech have been tested. Together with the scheduling there is an algorithm responsible for the power allocation. Once a user is scheduled for transmission, a proper amount of power will be assigned to that user. A HS-speech user will be assigned enough power as to achieve a predefined C/I value at the receiving UE, compared to a webbrowsing user that will be assigned all power available. This difference is due to that a speech user has a fixed bit rate, while a web-browsing user has a variable bit rate. Remember that there are two different methods for achieving certain energy 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 of power 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 some variables will be defined, which are used in the scheduling algorithms. This is followed by the description of the two speech scheduling algorithms examined in this work. Thereafter the web browsing scheduler is presented, and a retransmission power allocation algorithm for HS-speech users is described. Last is a section about expectations on the speech schedulers.

30 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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 a speech user has not been scheduled in his third TTI, the scheduler will automatically chose him no matter of the original scheduling criterions, i.e. the user will be scheduled in his forth TTI. This behaviour will guarantee the possibility 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 transmissionretransmission relation. There is also a fundamental rule to prioritise a retransmission before scheduling new transmissions. If this has not been the case, a lost first transmission would have been a complete waste of resources if a second transmission never occurs. With this restriction the point is to give resources to already initiated transmissions before starting new ones.
5.2 Scheduling variables

To measure the performance of the different schedulers and to express selection criterions, some variables are defined. First is the delay time d, the number of TTIs a user has to wait before getting scheduled. A few TTI delays will not necessarily be bad, the point is to try to schedule at good channel conditions. On the other hand, delays may pile up a lot of users in the same TTIs, resulting in users being even more delayed. Or even worse, that the users never get the chance to be scheduled, because of the maximum number of simultaneous users or exhausted power resources. Once the scheduler has made its decision the delay time is updated for usage in the next TTI, according to
if scheduled # 0 d ( j) = " ( j) !d + 1 if not scheduled

where d(j) is the j:th users delay time.
(1)
Second is the maximum number of simultaneous users. This variable has been fixed to four, because the 3GPP standard requires a UE to be able to decode four HS-SCCHs simultaneously.
( j) Third is the estimated channel quality, 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 in every TTI. The selection follows different criterions for the different schedulers. Next the two algorithms for voice are presented. Both the speech algorithms will be repeated several times in each TTI, until there is no power left or there are no more users with data waiting.

31 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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 to take advantage of the channel reports available, and to make a wise decision according to the channel quality ranking. This scheduling criterion will be efficient in distributing the resources. A user with a good channel will consume less power, making it possible to schedule several users simultaneously without exhausting the available power. Expressed in mathematical form, the user is selected according to
)C ( j) & i = arg max ' $. j (I %
(2)
( j) Where C 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 round robin 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 possible a delayed user will get a higher priority in the next TTI, than a user experiencing its first TTI. This delay may occur if there are a lot of users or if they are not well scattered in time, i.e. their first TTIs coincides a lot. This RR scheduler will not explore the benefits of scheduling users with favourable channel qualities. In mathematical form the algorithm becomes
i =arg max d ( j )
j
[ ]
(3)
where d(j) is the j:th users delay time. The reader with interest in the fundamentals of Round Robin scheduling is referred to [5].
5.4 Web browsing scheduler

The web-browsing users are scheduled with a max C/I criterion, equivalent to formula (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 is scheduled he will be assigned all power available in the base station. Furthermore, based on the estimated channel quality the highest possible MCS is selected. This will maximize the bit throughput under a limited power constraint. 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, which keeps a constant bit rate while minimizing the used amount of power.

32 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
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 channel estimates, i.e. the CQI reports. If the first transmission fails it is probably caused by an optimistic CQI report, resulting in too little power. If the same power allocation algorithm is used in a requested retransmission, the received energy in the first transmission is underestimated. Remember that the hybrid ARQ is storing the energy from previously received transmissions. In order to adjust the transmit power in the retransmission, an algorithm for tracking the changes in the CQI reports have been implemented. The algorithm will be described in the following subsection. This power allocation algorithm will only be used on HS-speech users. The web-browsing users are always assigned the total available amount of power once they are scheduled, retransmission or not. 5.5.1 Retransmission power allocation algorithm As discussed earlier, the CQI reports are delayed and the channel quality measurements are noisy. (Hence the delay is due to transmission and not introduced by purpose.) This results in some drawbacks when the needed transmit power is to be calculated. First of all, the MAX scheduler prefers to schedule users that report too good channels. This will in average result in an underestimated amount of needed power. (In some cases an improved channel condition cancels this power shortage, which results in a successful transmission anyhow.) In order to estimate 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 from the moment of transmission. These reports will be received 4ms after the transmission, but in time for a retransmission. The extra information is used to lower the amount of power assigned for the retransmission. Compared to the case without retransmission power regulation the amount of power assigned for a retransmission is calculated based on the latest available CQI report only. The ordinary 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 algorithm the allocation is modified to the following rule: The C/Iswp is subtracted with the estimated C/I, C/Iest, from the first transmission. The C/I target value, C/Itarget, for the retransmission can be expressed in mathematical terms as
C C C = s I t arg et I swp I est

where s is a safety margin, i.e. s(0,1).
(4)
This results in a lowered power allocation, but with a kept success rate. The difference is to estimate what the UE received in the first transmission, instead of assuming that the accumulated energy in the hybrid ARQ is zero.

33 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
The basics of the algorithm are illustrated in Figure 7. One drawback with the algorithm is that the CQI report for the actual transmission time also contains some 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 imagine what 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 lost anyhow. The main reason to implement this algorithm is to lower the used power in the retransmissions. Compared to the ordinary power allocation one drawback with this algorithm is that an extra CQI report is used, i.e. an extra measurement error has 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 power consumption will be lowered.
Tx1 CQI1 CQI2 CQI3
Tx2
Time
Figure 7: Illustration of the retransmission power allocation algorithm. CQI1 is used to calculate the assigned amount of power for transmission one (Tx1). The CQI received from the moment of transmission (CQI2) is used to estimate the received C/I at the receiver. This estimation is used in conjunction with CQI3 to assign a proper amount of power for the retransmission (Tx2) to be successful.
5.6 Expectations
The two speech scheduling algorithms are supposed to deliver the same performance at low loads, for instance comparable power consumption and similar affection on web users. This can be expected because if there are one or two users in a TTI, then the difference between choosing the one with the best channel or the one with the shortest delay time is almost the same. If there are two 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 scheduling algorithm will be hard to notice. The overall difference will be noticeable at higher loads approaching the capacity limit of speech users. In a case with only speech users the main limiting factor will be the packet loss rate. The capacity limit will be reached when too many users are unsatisfied. The retransmission power allocation algorithm is expected to lower the overall consumed power for HS-speech users.

34 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17

35 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Simulations
There are several ways of measuring the capacity of the HS-DSCH. Two methods will be used in this work. The first is to plot the normalized user delay versus system throughput and the second will be user bit rate versus system throughput. These two plots will show how and how much the speech users will interfere with the web traffic. The capacity measurements have to be associated with some kind of quality measurement on the speech. Therefore the amount of satisfied speech users will be monitored according to the speech quality measurements discussed in chapter 4. The simulations are presented both from a web-browsing user point-of-view and as a comparison of the two speech user types. The interesting part with the webbrowsing users is how they will be influenced when the communication system is used for speech as well. One major difference between the two types of speech users is that the DCH-speech users have dedicated channels, only affecting the web-browsing users indirectly. Compared to HS-speech users, which will use the same channel as the web-browsing users, therefore limiting the available resources in a more direct way. Both speech user types will be prioritised ahead of web-browsing users, and throughout this chapter the MAX scheduler for HSspeech has been used. This chapter begins with a short introduction to the simulator, some assumptions are made and then the simulations carried out are discussed and their results are presented, for example delay figures and capacity results. Several comparisons between DCH-speech and HS-speech users will be made. The conclusions from the simulations are summarized and discussed in the last section.
6.1 System simulator

In this work a MATLAB based simulator was used, which simulates a WCDMA communication system with HS-DSCH functionality. The time resolution used is on slot basis 1/1500 seconds, each HS-DSCH TTI consists of 3 slots. Included in the simulator are models for propagation, the physical layer and the traffic models. The propagation model involves calculations of path gains between base stations and UEs. They are based on geographical environment model with antennas, user movements, distance attenuation and fading. Each site consists of three sectors and the area is wrapped at the borders to illustrate a closed area. The speed of the users is fixed to 3km/h and the channel model used in the simulations is called Typical Urban, defined in the 3GPP standard. The physical layer model estimates the received signal strength at the mobile terminals and at the base stations. These C/I values are mapped to block error rate values, defining the chance of successful reception. The traffic models discussed in chapter three are also included in the simulator.

36 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
For every 10ms the simulator updates some basic variables like the number of users, creates new users, terminates finished calls, generates new web page requests 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 hybrid ARQ process. Each TTI consists of 3 slots, in each of those the fast fading is calculated, power is assigned and collection of slot performance is made. The process of creating new users and terminating finished ones may result in a time varying number of users, but the target value for the number of speech users will always be ten. Several of the simulations will study different behaviours of the communication system as a function of an increasing traffic load, which is done by increasing the number of web users in each sector. The communication system simulations were run for several hundreds of seconds, in order to reach a steady state not influenced by short time differences. 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 of bits delivered from the system to the receiving UEs. In order to compare these numbers with other simulations and other kinds of communicating systems, the total number of bits is normalized with time, the number of sectors per site and the number of MHz the bandwidth is occupying. Therefore the term known as system throughput is defined as the total number of bits that the system has delivered per second, sector and MHz. A site is normally divided into three sectors, each covering a third of the area, usually 120 degrees. In WCDMA the bandwidth 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 a measurement connected to the communication system. Associated with individually users is a term called normalized user delay. It is defined as the total waiting time divided by the sum of all the bits the user has received. 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 the download is complete. The normalized user delay can then be calculated as the total waiting time divided by the sum of all delivered bits. This term will be measured in s/Mbit. Tightly connected to the normalized user delay is the user bit rate, measured in kbps. 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 the normalized user delay. Some of the simulation results will be presented with CDFs, cumulative distribution functions. A CDF is defined to be, for a distribution X, the probability P(X x), for all values of x. Also the statistical measurement percentile is used, which returns the value x for which, given p, P(X x) = p is true.

37 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
The two types of speech users will influence the web-browsing users in different manners. In order to compare these two situations from the web-browsing users point-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-browsing users. A transmission to a HS-speech user will reserve one code with spreading factor 16 (recall Figure 2). If there are twelve codes reserved for HS-DSCH, then a web-browsing users possible amount of codes is decreased with one for every scheduled HS-speech user. In the following simulations there is usually ten HSspeech users active in every sector. This will lead to an average of one code reserved for HS-speech every TTI, and if retransmissions are included there is an even bigger part reserved. These code conditions for web-browsing users, should be the same if the communication system instead is used for DCHspeech and web browsing. Therefore the amount of codes reserved for the HSDSCH, when there are DCH-speech users present, is lowered from twelve to ten. This will make the two cases of speech users comparable from a webbrowsing 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 Web Web & DCH-speech Web & HS-speech HS-DSCH 12 codes with SF 16 10 codes with SF 16 12 codes with SF 16 HS-SCCH 1 code with SF 128 1 code with SF 128 4 codes with SF 128
Table 2: Code reservation for HS-DSCH and HS-SCCH at the three differnt traffic combinations.
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 an associated downlink DPCH, but no information needed for the HS-DSCH transmission is carried in this channel; instead necessary signalling is carried by the HS-SCCH. The downlink DPCH is only carrying information from the base station to the UE that concerns power control in the uplink. This information is of course essential in a communication system and is assumed to be present anyhow. 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 has only been excluded for HS-speech users. The web surfing users have theirs associated channels left.

38 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
In all simulations made, the HS-SCCH has been assumed to transmit with the fixed power of 0.4W. This power is necessary for the HS-DSCH transmission and 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 made dynamic, 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 user standing far away. A decreased power consumption for the control channels would be a benefit for the whole communication system, but these possibilities will 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. A convenient way of showing the statistics is to plot different percentiles. The 90 th percentile 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 degradation experienced when introducing speech users can easily be shown. The first simulations studied, in subsection 6.3.1 and 6.3.2, will discuss how the two types of speech users affect the web-browsing users. Therefore these two subsections will contain three different curves in each plot, each as a result of one of the three simulated situations. The three situations are: only web users, web plus DCH-speech and web plus HS-speech. These three curves will be labelled according two the simulated combination of traffic, e.g. web & speech means a simulation with web and DCH-speech users. The plots showing the results will always show how the web-users experience the different situations. The rest of the simulations, discussed in the after coming subsections will compare the two speech user types to each other. This section is concluded with a specific study of the HS-speech users retransmission power allocation. First is the increased normalized user delay for web-browsing users studied. 6.3.1 Delay results When sharing the communication system among web-browsing users and speech users, the speech users will always be prioritized. Therefore the webbrowsing users will experience a greater delay compared to the basic situation when they are alone using the communication system. This prioritizing is made regardless of the speech users type. Figure 8 shows the normalized user delay for web-browsing users. Note how it is increasing, and how the unlimited growth begins earlier for the web users when the system is hosting speech as well. The two types of speech users affect the delay results in a comparable manner.

39 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
The quality perceived by individual web-browsing users can be measured by the normalized user delay. In these simulations a user is classified as satisfied if the normalized user delay is kept below 60s/Mbit. For a communication system to deliver adequate quality, the amount of satisfied users should be at least 90%. From Figure 8 it can be concluded that the communication systems capacity is lowered from 287kbps/sector/MHz to 234kbps/sector/MHz, when used in conjunction with DCH-speech, and just a little bit lower when used with HSspeech (233kbps/sector/MHz). The result indicates that the two types of speech users will have a comparable impact on the web-browsing users normalized delay. The increased delays for the web-browsing users are because of a smaller amount of resources available. The speech users will consume both codes and power. More discussion about the limiting factors can be found in the next subsection, when the bit rate degradation is studied. When the communication system is used with web and DCH-speech, the number of codes for HS-DSCH is ten, compared to twelve when used with HSspeech and web-browsing. This difference is introduced to make the prerequisites for web-browsing users comparable no matter of the type of speech users present. Recall Table 2 for a summary of the code reservation. In this first simulation, the HS-speech users are not optimised in order to minimize the influence on the web-browsing users. Later on such optimisations will be made, and the results will be presented. The most important consequences of these optimizations will not affect the normalized user delay, but other results.
Figure 8: 90 percentile of normalized user delay versus system throughput, for webbrowsing users. The three curves shown are from three different simulations. For reference the system is simulated with only web users, the rightmost curve. The next two almost identical curves are from simulations with web users and the two types of speech users respectively.
th

40 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
6.3.2 Bit rate results Similar to the delay results, the bit rate experienced by the web-browsing users will be affected once speech users are introduced in the communication system. In Figure 9 the bit rate degradation can be seen for the three different traffic situations. When studying the case with only web-browsing users, the reason to the decreased bit rate at higher loads will be a combination of several factors. For example, the available amount of power decreases with increasing web load. This is because the associated DPCHs consume power. If there are more users waiting to be scheduled, there are more associated channels active. A second reason is that, at high loads there can be a shortage of free codes for the associated DPCHs. This will delay the users with data waiting before even being able to become scheduled. When the delays increase the bit rate will decrease. As with the normalized delay, the web-browsing users will suffer from decreased performance when the communication system is used with both speech and web traffic. The two types of speech users will have similar affect on the webbrowsing users bit rate, at least when comparing the degradation up two the system capacity limit. As stated in the previous subsection the system capacity limit is about 230kbps/sector/MHz when used with speech and web-browsing. The rest of the curves exceeding the capacity limit, indicate what happens when the system is heavily loaded. A conclusion that can be drawn from the last parts of the curves is that the DCH-speech user will have a smaller impact on the webbrowsing traffic performance, compared to the case with HS-speech users. An earlier exhaustion of the code space available for associated DPCHs causes the drawback for the HS-speech users. Plots showing the decreased amount of power and the increased code usage can be found in the appendix. If the retransmission frequency is varied for the HS-speech users, it will influence the capacity achieved by web-browsing users. With a higher amount of retransmissions the HS-speech users will consume a larger part of the HSDSCH codes. Therefore the user bit rates will be affected for the web users. A plot showing how the bit rate is decreased as a consequence of a higher retransmissions frequency can be found in the appendix. So far the two types of speech users are comparable, at least from a webbrowsing users perspective. The throughput and capacity of the communication system will experience a similar degradation regardless of the speech user type present. In the upcoming subsections the two speech user types will be compared more explicitly to each other. One of the interesting characteristics is the speech users power consumption, which will be studied in the next subsection. There is also an advantage with HS-speech when the packet loss rate is compared to DCH-speech users. These results will be presented later on.

41 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Figure 9: 90th and 50th percentiles of user bit rate versus system throughput, for webbrowsing users. The highest curve indicates the capacity limit achieved when the communication system only hosts web users. The two lower curves (with solid lines) indicate the degradation when speech users are introduced.
6.3.3 Power consumption In this subsection together with the two upcoming ones, the focus will be on speech users. The comparisons made will be functions of the web traffic load. The two types of speech users are different in power efficiency. When the system is loaded with web users plus DCH-speech users and web users plus HS-speech users respectively, the power consumption per speech user will differ. The power consumption is calculated as the average over all speech users, where each users consumption is calculated as the sum of all transmissions power divided by the number of speech frames. In the appendix some plots showing the power distribution can be found, for comparison between the two speech user types. In Figure 10, the average power consumption can be seen for the two types of speech users, as a function of the web users load. For HS-speech users there are two curves shown, the upper one includes the power for the HS-SCCH, i.e. the power needed for control information, and in the lower curve this power is excluded. Remember that each transmission on the HS-DSCH requires a HSSCCH. In case of code multiplexing each receiving user has its own HS-SCCH, which consumes 0.4W each.

42 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Figure 10: Average transmit power per speech user. The power consumption is increasing with increasing web traffic load, due to increased interference. The two curves shown for HS-speech are with and without the HS-SCCH power respectively. The two types of speech users show a similar behaviour at increasing web traffic load, but the advantage of HS-speech is obvious. The reason for the increased power consumption for both cases is the increased amount of interference present at higher loads. The increased interference leads to more unfavourable channel conditions, and is caused by the greater number of web users each having an associated DPCH active. The overall conclusion is that the HS-speech users consume less power than the DCHspeech users, even as low as half the power if the HS-SCCH power is excluded.
One parameter affecting the used power for HS-speech users is the number of retransmissions. The trade-off between using a lot of power in the first transmission compared to using too little will decide the amount of retransmissions. The overall power consumption is not necessarily minimized with a high or low retransmission frequency. The hybrid ARQ is designed to help in the situation of failure in the first transmission; therefore a rather high retransmission frequency can be used. However the best value, i.e. lowest power consumption for HS-speech users, may not be the best situation for webbrowsing users. A high retransmission frequency may use a smaller amount of power, but will have a drawback in the usage of more HS-DSCH codes. Therefore the power trade-off is not just a power trade-off, but rather a trade-off concerning the whole communication system. DCH-speech users in soft handover will consume power from both cells, i.e. the users that are moving between two cells. In the simulations above the amount of power in the retransmission is not optimised. Instead the retransmission power is estimated as to achieve enough C/I at the receiver, as if there would not have been any first transmission. The affect of optimising the retransmit power will be investigated in subsection 6.3.5, according to the retransmission power allocation algorithm described in chapter 5.5.1.

43 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
6.3.4 Packet loss rate When simulating a communication system with speech users, the quality of the delivered speech must be supervised. As discussed earlier the speech users will be classified as satisfied if the 95th percentile of the packet error rate is less than 1%. This will be equivalent to the speech frame error rate, SpFER, for DCHspeech users. The error rates are called differently but they will comprehend the same quality measurement. To loose a packet containing 20ms of speech for HS-speech users, is considered the same as to loose a speech frame with 20ms of speech for DCH-speech users. Two examples of packet loss plots can be seen in Figure 11; the loss rate is shown as a CDF. As can bee seen (in the leftmost plot) the HS-speech users have a little worse quality than the DCH-speech users, when it comes to the 95th percentile at least. The consequence of a poorer 95th percentile for HS-speech users is fewer satisfied users, but the difference is quite small, and there would not be any problem assigning some more power to all users, resulting in a lower packet error rate. The leftmost plot showed in Figure 11, is from a simulation with few web-browsing users, not using all of the resources available in the base station. Therefore there would not be any problem to lower the error rate and still keeping the same capacity for web-browsing users. In Figure 11, it can also be seen that 80% of the HS-speech users have a packet error rate better than the DCH-speech users. Furthermore there are also 10% of the HS-speech users that seldom lose any of their packets. At higher webbrowsing loads, as seen in Figure 12, the HS-speech users will be even more satisfied, and more than 80% of the users seldom lose any packets. The rather differently looking plots in Figure 11 and 12 are because the DCHspeech users are power controlled in a different manner than the HS-speech users. A DCH-speech user is power controlled using an outer loop control, in order to achieve a 95th percentile of 1% for the SpFER. An outer loop power control algorithm works as follows: A speech user receiving several correct speech frames will get less and less power until a speech frame is lost. At this moment the assigned amount of power is increased and hopefully the next speech frame is received successfully. As seen in Figure 11 and 12, 10% and 80% respectively of the HS-speech users never lose any packets; this implies that they get unnecessary much power. A possible enhancement would be to implement a similar power-regulating algorithm, as for the DCH-speech users. Such an algorithm could use the information about packet success rate for lowering the used power in an eventual retransmission. The benefit would be even lowered power usage, and a potential lowering of the number of users receiving every packet correct. The unnecessary high success rate also indicates that an outer power control mechanism, for the initial transmission, would save resources.

44 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Figure 11: Speech frame error rate for DCH-speech and packet error rate for HS-speech. th The 95 percentile of SpFER and PER respectivly is for DCH-speech about 1.1% while for HS-speech it is about 1.6%.
Figure 12: At a higher web-browsing load, the 95 percentile for DCH-speech has increased to 1.8%, while there is a decrease to 0.1% for HS-speech. The improved satisfaction for HS-speech, compared to Figure 11, indiucates that they get too much resources.
th
In Figure 11 and 12 the number of web-browsing users was fixed at two different values. If the load of web-browsing users is varied, the 95th percentile of the packet error rate will vary too. In Figure 13, the 95th percentile for packet error rate and speech frame error rate for HS-speech and DCH-speech respectively, is shown. The two types of speech users are showing an opposite behaviour, when the web-browsing load is increased. The DCH-speech users will experience an increased number of unsatisfied users, while the HS-speech users will be more and more satisfied. The increased amount of unsatisfied DCH-speech users is a consequence of the increased interference. The DCH-speech users are limited to a maximum amount of transmit power. Therefore the possibility to cope with the increased interference is limited at higher system loads.

45 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
The HS-speech users will, with increasing web-browsing load, be more and more satisfied. This indicates that a bigger part of the first transmission power will be useful, i.e. resulting in a higher received C/I at the UE. This improvement is thanks to steadier CQI reports, which result in better channel estimates. These conclusions are supported by the fact that the retransmission frequency is lowered with increasing traffic load. One plot showing the lowered retransmission frequency can be found in the appendix.
Figure 13: 95th percentile of packet error rate, for HS-speech users, and frame error rate for speech users. Plotted as a function of the number of web-browsing users. Notice the opposite behaviour, which is caused by the power limitation for DCH-speech users and the improved CQI quality for HS-speech.
6.3.5 Retransmission power allocation The simplest way to allocate power to a retransmission is to assign the amount of power as if there have not been any earlier transmissions. This underestimation of the already accumulated C/I will make the power consumption unnecessary high. Therefore a retransmission power allocation algorithm, see chapter 5.5.1, has been tested, and the results are pleasant. When the retransmission power is regulated, the consumption can easily be lowered by 5-10%. In Figure 14, the decreased average power usage can be seen, not including the power for HS-SCCH. The decreased power is a good enhancement for the HS-speech users, but the lowered resource usage cannot be noticed as an improved bit rate for web-browsing users.

46 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
In the base station, a scheduled user will be assigned power, based on the estimated channel quality. The amount of power is calculated based on a target value for the C/I, at the receiving mobile. If the channel quality estimation were perfect, this power allocation would result in a retransmission frequency of 10%, because the target value is the switch point. Due to measurement errors, delays in the CQI reports and fading channels, the allocated power cannot achieve this high success rate. If a transmission is successful or not, also depends on the interference, which can vary a lot when there are web-browsing users active. A scheduled web-browsing user is assigned all of the available power, left over in the base station. This behaviour will make the interference bursty at low system loads. At higher loads the interference level will be steadier, thanks to that the base stations always transmits at full power, therefore the channel estimates will be more accurate, resulting in a lowered retransmission frequency, as was seen in the previous subsection. When using the retransmission power allocation algorithm the 95th percentile of the PER is slightly increased. This drawback can easily be coped with by using different safety margins depending on traffic load. At higher loads the 95th percentile is well below the 1% limit, thanks to the steady interference level. Recall from Figure 12 that the HS-speech users were unnecessary satisfied. There for the drawback of an increased PER is kept within the stated satisfaction limit of speech users.
Figure 14: Decreased power consumption for HS-speech users, when the retransmission power allocation algorithm is used. The transmit power does not include the power for HS-SCCH.

47 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
6.3.6
Comparison of voice scheduling algorithms
The two speech scheduling algorithms have been compared both in simulations with only speech users, and with both speech and web-browsing users. The performance and capacity results for the two schedulers are as expected quite similar. The theoretical expected advantage, of exploring the users different channel qualities, is not visible in this work. The reason is that the scheduling algorithms will rank the users in a similar manner, due to the small number of speech users. One conclusion, that can be drawn, is that to improve HS-speech the focus should not be on scheduling, but rather on something else, e.g. filtering of the channel reports or power control.
6.4 Conclusions
At low loads, under the system capacity limit, the two types of speech users will have a similar impact on the web-browsing users. For example is the normalized user delay for the web-browsing users, decreased in a comparable manner when speech is introduced in the communication system, regardless of the speech user type. Once exceeding the capacity limit the DCH-speech users is of advantage. The benefit at high system loads is thanks to more codes available for associated DPCHs. This is despite a larger amount of power and in average more HSDSCH codes available for web users, when the system is hosting HS-speech users as well. When comparing the two types of speech users the most obvious difference is how power effective the HS-speech users are compared to DCH-speech users. In average the DCH-speech user is consuming twice the transmit power compared to a HS-speech user. This big advantage is decreased if the HSSCCHs power is included, but nevertheless the HS-speech users consume less power. The HS-speech users are also favourable when the packet error rate is compared to the speech frame error rate for DCH-speech users. Although it should be remembered, that the HS-speech users do not have an outer loop power control algorithm. If such an algorithm would be implemented the power consumption most probably would be lowered even more. However one drawback with such an algorithm is that the packet error rate would increase a little bit, at least for the users that never loose any packets. This conclusion is drawn based on the assumption that, if both of the speech user types were power controlled in the same manner, the packet error rate curves would look more similar. Resulting in an increased packet loss rate for the most fortunate users, i.e. the users that seldom looses any packets.

48 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
A conclusion that can be drawn, when comparing the two proposed voice schedulers, is that the scheduling algorithm is of low importance. The two algorithms used in this work are theoretically very different, but they result in similar results, at least when the number of speech users is as low as ten. From this it can be concluded that further improvements of the HS-speech service, should be focused on other aspects than the actual scheduler. For example may the channel estimates be filtered and predicted ahead in time. Resulting in more accurate input two the scheduler, which in turn may highlight the theoretical difference of the to scheduling algorithms. The schedulers is believed to be more important if the speech packets were larger and transmitted more seldom, this is however not explored in this work.

49 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Conclusions and Future Work
The overall conclusions are summarized in this Chapter. It is divided into conclusions and future work.
7.1 Conclusions
The overall conclusion is that the two different methods of sending speech are comparable, when viewed from a web-browsing user point-of-view. Both the normalized user delay and bit rate for the web-browsing users are affected in the same manner regardless of the speech type. When the communication system is used with both speech and web traffic the system capacity is lowered from 278kbps/MHz/sector to about 230kbps/MHz/sector. When comparing the two types of speech users the HS-speech version is of advantage. The power consumption is lower for HS-speech users, and they are also favourable when the packet error rate is compared to the speech frame error rate for DCH-speech users. The speech quality will be better for a bigger amount of the speech users if they are of HS-speech type. When using the retransmission power allocation algorithm the power efficiency is getting even bigger. A lowering of the consumed power with 5-10% is possible. When using the HS-DSCH for both web-browsing traffic and speech, the speech users, if reasonable many, will stay satisfied with their speech quality even if the load of web users is high. Remark that the HS-speech users do not have any associated DPCHs these channels have been excluded.
7.2 Future work

One enhancement of the speech scheduler would be to make some kind of inventory about the distribution of the start TTIs, which the HS-speech users have. This information could be used to avoid jammed TTIs, which results in a longer average queue time before being scheduled. Combining this information with the knowledge about the average consumed power, by different users, would result in a priority scheme. The point with such a scheme is that a user with a favourable channel, probably standing close to the base station, can be scheduled together with similar users, whereas users demanding a lot of power resources should if possible get scheduled in TTIs with low load, just to make the transmission success probability higher. A retransmission to a user with a favourable channel costs less, than a retransmission to a user with an unfavourable channel, at least in power.

50 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
The DCH-speech users are power controlled, in order to keep the frame error rate on a reasonable level. One enhancement with the HS-speech user would be to implement a similar control function. The algorithm should record if a retransmission is successful or not, and use this information in the next retransmission, maybe to assign some more power than done previously. In this work, the HS-speech users were implemented with the 20ms speech frame, putting a hard limit to how many retransmissions that were possible. An eventual retransmission will return an ACK/NAK, so there is more information available, but not used. An ACK could be used to lower the power assigned in the next retransmission, in order to slowly decrease the amount of used power. On the other hand if a NAK is returned the assigned amount of retransmission power should be increased. The increase and decrease values should be in such proportions, that an increase in average should be followed by 99 decreases. That is to keep the packet error rate at a fixed level of 1%. This kind of power regulating would distribute the power more efficient. A similar discussion can be used to tune in the retransmission frequency, i.e. to control the amount of assigned power in the first transmissions. This kind of control would allow specifying different target values, depending on the traffic load situation. It could also be used to provide different algorithms, or dynamically changing ones, depending on the traffic load in the communication system. The reason for the power enhancement is obvious when the SpFER plots are studied. Remember that at high loads the HS-speech users are very satisfied. This is of course good, but the HS-speech users will consume a larger amount of power than necessary. For example, if 50% of the users never loose a packet then the conclusion is that they get to much power. The speech quality would of course decrease, but the difference between loosing 0 or 0.1% of the packets wouldnt be noticeable. The advance of introducing packet loss for some users is more resource to other ones. This would probably make the PER curve become more similar to the SpFER curve. In this work the 20ms speech frame time structure is one of the most fundamental. Therefore a different strategy would be to loose this restriction and maybe study 40ms speech blocks instead. As a consequence of this the speech packets would be bigger and the scheduling freedom would increase. There would also be more time for retransmissions available. A speech model based on 40ms blocks is believed to be more similar to the web browsing traffic model and therefore it might be more suited for the HS-DSCH. In the case with bigger packets and more time for transmission the scheduler most probably would become more important.

51 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Appendix A: Additional Simulation Plots

Decreased amount of available power When the load of web-browsing users is increased the number of active associated DPCHs will increase. These channels will consume power and therefore the available amount of power left over for the HS-DSCH will decrease, as seen in Figure 15. The curves plotted are the average values per base station. The uppermost curve in Figure 15 shows the decreased amount of power for HS-DSCH when the communication system only hosts web users. The two lower curves are from simulations with speech users as well. The reason to the lower amount of power available is the speech users consumption, which is prioritised ahead of web users.
Figure 15: Decreased amount of availible power for the HS-DSCH, which leads to a lowered bit rate.
Increased code usage In Figure 16, the increased code usage can be seen. The lowest curve shows the code usage at the basic simulation with only web-browsing users. As can be seen the available code space is exhausted at a load of 100 web-browsing users.

52 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
When introducing speech on dedicated channels, the code usage will contain both associated DPCHs for web users and the codes used for the speech channels. The code space available in this case will be larger than the basic situation, because of the fewer codes reserved for HS-DSCH transmissions. In this case the code space will also be exhausted at a load of 100 web users. The third curve shown is the case with HS-speech users. In this case the code space available for associated DPCHs are a bit smaller than in the basic case, due to the fact that four codes instead of one are reserved for HS-SCCHs. Therefore the used amount of codes will increase faster and the maximum is reached at 80, instead of 100 web users. The shortage of code space for associated DPCHs is the most important reason for the rapid bit rate degradation at high loads.
Figure 16: Increased usage of the nonreserved code space. The maximum is reached earlier in the case with web- and HS-speech-users compared to the other two simulation cases.
Decreased bit rate for web-browsing users When the HS-speech users retransmission frequency is varied, the amount of resources for the web-browsing users will be affected. Typically, the webbrowsing users bit rate is decreased if the retransmission frequency is increased. In Figure 17 the 90th and 50th percentile of the web-browsing users bit rate is shown. The upper curve is from a simulation with a lower retransmission frequency, approximately under 20%, and the lower curves are with a retransmission frequency of more than 65%. The decreased bit rate is a consequence of the smaller amount of HS-DSCH codes available for the web-browsing users. Although the extra amount of speech retransmissions does not only consume extra codes, there is also more control channels used. Remember that each user receiving data on the HSDSCH needs a HS-SCCH, as with the bit rate, the retransmission frequency also affects the normalized user delay, as seen in Figure 18.

53 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Figure 17: Decreased bit rate for web-browsing users, as a function of the retransmission frequency, rtf, for HS-speech users.
Figure 18: 90th percentile of normalized user delay. The system capacity, at 60s/Mbit is 213kbps/sector/MHz for the high retransmission frequency case and 236kbps/sector/MHz for the low retransmission frequency case.
Power distribution Power distribution for DCH-speech users compared to HS-speech users is shown in Figure 19. The average values for the two cases are 0.23W and 0.10W respectively.

54 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Figure 19: Example of power distribution for the two types of speech users, plotted as a CDF.
Lowered retransmission frequency In Figure 20 the retransmission frequency for HS-speech users can be seen. Notice how the frequency is lowered with increasing web-browsing load. The reason is a more steady interference level, resulting in better estimates of the channel quality, which in turn results in an increased received C/I.
Figure 20: Decreasing retransmission frequency with increasing web-browsing load.

55 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
Appendix B: Abbreviations
3GPP ACK ARQ CDF CDMA CQI C/I DCH DCH-speech FDMA HSDPA HS-DSCH HS-SCCH HS-speech kbps Mcps MCS NAK PER QoS RR SF SpF SpFER TDMA TTI UE WCDMA 3rd Generation partnership project (produces WCDMA standard) Acknowledgement Automatic Retransmission request Cumulative Distribution Function Code Division Multiple Access Channel Quality Indicator Carrier-to-Interference ratio Dedicated Channel Dedicated Channel speech Frequency Division Multiple Access High Speed Downlink Packed Access High Speed Downlink Shared Channel High-Speed Shared Control Channel High-Speed speech kilo bit per second Mega chips per second Modulation and Coding Scheme Negative Acknowledgement Packet Error Rate Quality of Service Round Robin Spreading Factor Speech Frame Speech Frame Error Rate Time Division Multiple Access Time Transmission Interval User Equipment Wideband CDMA

56 (56)
Niklas Lithammer
Approved Checked
Date Rev Reference
EAB/TUR Mikael Hk
S.P.
2003-12-17
References
[1] [2] [3] [4] H. Holma and A. Toskala, WCDMA for UMTS Radio Access For Third Generation Mobile Communications, John Wiley & Sons, 2000 J. G. Proakis and M. Salehi, Communication Systems Engineering, Prentice Hall, 1994 3GPP, UTRA High Speed Downlink Packet Access (HSDPA); Overall description; Stage 2, TS 25.308, 2003 S. Parkvall, E. Englund, P. Malm, T. Hedberg, M. Persson and J. Peisa, WCDMA evolved High-speed packet-data services, Ericsson Review, nr. 3 2003 U. Krner, Kteori och tillfrlitlighetsteori, Studentlitteratur, 1992 G. Blom, Sannolikhetsteori och statistikteori med tillmpningar, Studentlitteratur, 1989
[5] [6]

Speech On HS-DSCH

Diunggah oleh

Informasi Dokumen

Deskripsi Asli:

Judul Asli

Hak Cipta

Format Tersedia

Bagikan dokumen Ini

Bagikan atau Tanam Dokumen

Opsi Berbagi

Apakah menurut Anda dokumen ini bermanfaat?

Apakah konten ini tidak pantas?

Hak Cipta:

Format Tersedia

Speech On HS-DSCH

Diunggah oleh

Hak Cipta:

Format Tersedia

Speech on HS-DSCH

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

1.1 Purpose of project

Open REPORT (M. SC. THESIS)

2.1 Cells and systems

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

2.3 Evolving WCDMA

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

2.4 Channels in WCDMA

Open REPORT (M. SC. THESIS)

2 1 1 35 + + = 16 256 128 256

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Why Voice on HS-DSCH?

The statistical term percentile will be defined in section 6.2

Open REPORT (M. SC. THESIS)

4.1 Voice services on DCH

4.2 Voice services on HS-DSCH

Open REPORT (M. SC. THESIS)

Start of speech frame Transmission

End of speech frame Retransmission

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

DCH-speech 12.2kbps 20ms (one SpF) 1% No Yes

HS-speech 12.2kbps 2ms (one TTI) 1% (SpFER equal to PER) Yes No

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

Open REPORT (M. SC. THESIS)

4.4 Potential advantages

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

5.2 Scheduling variables

if scheduled # 0 d ( j) = " ( j) !d + 1 if not scheduled

5.3 Scheduling algorithms for voice

Open REPORT (M. SC. THESIS)

)C ( j) & i = arg max ' $. j (I %

5.4 Web browsing scheduler

Open REPORT (M. SC. THESIS)

5.5 Retransmission power allocation for voice

C C C = s I t arg et I swp I est

Open REPORT (M. SC. THESIS)

Tx1 CQI1 CQI2 CQI3

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)

6.1 System simulator

Open REPORT (M. SC. THESIS)

Open REPORT (M. SC. THESIS)