Examensarbete LITH-ITN-KTS-EX--07/020--SE
Mobile Broadband - Service
Experience and Evaluation
Pontus Sandberg
2007-09-24
Department of Science and Technology Institutionen för teknik och naturvetenskap Linköpings universitet Linköpings universitet SE-601 74 Norrköping, Sweden 601 74 Norrköping LITH-ITN-KTS-EX--07/020--SE
Mobile Broadband - Service Experience and Evaluation Examensarbete utfört i kommunikations- och transportsystem vid Linköpings Tekniska Högskola, Campus Norrköping Pontus Sandberg
Handledare David Gundlegård Examinator Di Yuan
Norrköping 2007-09-24 Datum Avdelning, Institution Date Division, Department
Institutionen för teknik och naturvetenskap 2007-09-24
Department of Science and Technology
Språk Rapporttyp ISBN Language Report category ______Svenska/Swedish Examensarbete ISRN LITH-ITN-KTS-EX--07/020--SE x Engelska/English B-uppsats ______C-uppsats Serietitel och serienummer ISSN x D-uppsats Title of series, numbering ______
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URL för elektronisk version
Titel Title Mobile Broadband - Service Experience and Evaluation
Författare Author Pontus Sandberg
Sammanfattning Abstract Mobile broadband is becoming an alternative to fixed broadband connections for residential areas. EGDE manages Internet Radio streams up to 96kbit/s while WCDMA R99 delivers satisfactory quality for media streams up to 320 kbit/s. Internet Radio is not possible to use over GPRS with satisfactory quality. The quality assessment for Internet Radio is highly dependent on the stream bit rate and a stream of 128 kbit/s delivers high user satisfaction. For online gaming other parameters than the well known latency, jitter and packet loss affects the session quality. Shorter Counter-Strike sessions are playable over HSPA where it outperforms WCMDA R99 in both network performance and user quality assessment. World of Warcraft has more limited network requirements and is playable over both WCDMA R99 and HSPA but suffers from latency impairments when played over GPRS/EDGE. The developed objective quality prediction model for First Person Shooter (FPS) games predicts gaming quality with an RMSE value of 0.609, prediction accuracy of 91.9% and correlation between predicted and subjective MOS of 0.8803.
Nyckelord Keyword QoE, GPRS/EDGE, WCDMA, HSPA, Gaming, User Perception, Quality Model Upphovsrätt
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Copyright
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© Pontus Sandberg To Rolf and Judith Lundqvist
i ”‘If we knew what it was we were doing, it would not be called research, would it?”’
- Albert Einstein
Publication year 2007
Supervisor J¨orgenGustafsson, Ericsson AB
Examiner Associate Professor Di Yuan Supervisor David Gundleg˚ard Link¨opingsUniversitet Mobile Communications, ITN
ii Abstract
EGDE manages Internet Radio streams up to 96kbit/s while WCDMA R99 de- livers satisfactory quality for media streams up to 320 kbit/s. Internet Radio is not possible to use over GPRS with satisfactory quality. The quality assessment for Internet Radio is highly dependent on the stream bit rate and a stream of 128 kbit/s delivers high user satisfaction.
For online gaming other parameters than the well known latency, jitter and packet loss affects the session quality. High start up or map load times will cause premature game termination and poor quality grading. Shorter Counter- Strike sessions are playable over HSPA where it outperforms WCMDA R99 in both network performance and user quality assessment. World of Warcraft has more limited network requirements and is playable over both WCDMA R99 and HSPA but suffers from latency impairments when played over GPRS/EDGE.
The developed objective quality prediction model for First Person Shooter (FPS) games, like CounterStrike, shows pretty good performance, despite a small number of test persons. The model predicts online gaming quality with an RMSE value of 0.609, prediction accuracy of 91.9% and correlation between predicted and subjective MOS of 0.8803.
Keywords: QoE, GPRS/EDGE, WCDMA, HSPA, Gaming, User Perception, Quality Model
iii Abstract in Swedish
EDGE klarar att leverera Internet Radio-fl¨odenupp till bittakter av 96 kbit/s medan WCDMA R99 klarar bittakter upp till 320 kbit/s med tillfredst¨allande kvalitet. GPRS kan inte anv¨andasf¨orInternet Radio sessioner med tillfredsst¨allande kvalitet. Kvalitetensbed¨omningenav ett Internet Radio fl¨ode¨arstrikt beroende av fl¨odetsbithastighet och Internet Radio-fl¨odenp˚a128 kbit/s och h¨ogrelever- erar h¨ogkvalitet och tillfredst¨allsebland anv¨andare.
Kvalitetsbed¨omningenav online spel p˚averkars av andra parametrar ¨ande typiskt v¨alk¨andalatency, jitter och paketf¨orluster. H¨ogastart- och kart- laddningstider leder till l˚agtbetyg p˚asessionen och ett tidigt avslutande av spelet. HSPA klarar kortare spel sessioner av CounterStrike och levererar av- sev¨artmycket b¨attreprestanda och kvalitetsbed¨omning¨anf¨orWCDMA R99. World of Warcraft har betydligt mindre krav p˚aaccessn¨atverket och ¨arspel- bart ¨over b˚adeWCDMA R99 och HSPA men lider av h¨ogalatencyniv˚aerf¨or GPRS/EDGE.
Den framtagna kvalitetsprediktionsmodellen f¨orFirst Person Shooter (FPS) spel som CounterStrike visar p˚abra prestanda, trots baserad p˚aett begr¨ansat antal testpersoner. Modellen predikterar kvalitet p˚aonline spel med ett RMSE v¨ardep˚a0.609, en prediktionss¨akerhet p˚a91.9% och en korrelation p˚a0.8803 mellan predikterat och subjektivt MOS.
iv Acknowledgments
First of all I would like to thank my supervisor J¨orgenGustavsson at Ericsson Research for insightful comments and guidance during my work. I would also like to thank Gunnar Hiekkil¨afor your competent comments and honest opin- ions especially regarding the model chapter. I also appreciate the participants in my online gaming survey.
Further, thanks to Jim Olander and Jonas Nilsson for welcoming and helping me with the testing activities. You both greatly contribute to the final result of this thesis.
Finally, I would like to say thank you to my invaluable family and my lovely Helena. I am grateful for your love and the way you are supporting me in every possible way.
Pontus Sandberg, Lule˚a,2007
v Abbreviations
2G 2nd generation of mobile networks 3G 3rd generation of mobile networks 3GPP 3rd generation partnership project ACK Acknowledgment ARQ Automatic Repeat-reQuest BER Bit Error Rate BLER BLock Error Rate BSC Base Station Controller BSS Base Station System CDMA Code Division Multiple Access CS Circuit-Switched DL DownLink DSL Digital Subscriber Line EDGE Enhanced Data rates for GSM Evolution EGPRS Enhanced GPRS ETSI European Telecommunications Standards Institute EUL Enhanced Uplink FDMA Frequency Division Multiple Access FPS First Person Shooter GGSN Gateway GPRS Support Node GPRS General Packet Radio Service GSM Global System for Mobile communications HSDPA High Speed Downlink Packet Access HSPA High Speed Packet Access HSUPA High Speed Uplink Packet Access IP Internet Protocol ISDN Integrated Services Digital Network ITU International Telecommunication Union kbit/s Kilobit per second kHz kilo Hertz KPI Key Performance Indicator LAN Local Area Network LTE Long Term Evolution MB MegaByte Mbit/s Megabit per second Mcps Mega chips per second
vi MHz Mega Hertz MIMO Multiple Input Multiple Output MMORPG Massive Multiplayer Online Role Playing Game MS Mobile Station NSS Network Switching System OSS Operation and Support Subsystem PLMN Public Land Mobile Network PSDN Public Switched Data Network PSTN Public Switched Telephone Network QAM Quadrature Amplitude Modulation QoE Quality of Experience QoS Quality of Service RAN Radio Access Network RNC Radio Network Controller RPG Role Playing Game RTS Real Time Strategy RTT Round Trip Time SF Spreading Factor SGSN Serving GPRS Support Node SIM Subscriber Identity Module SIR Signal to Interference Ratio TCP Transmission Control Protocol TDMA Time Division Multiple Access TP ThroughPut TS TimeSlot TTI Transmission Time Interval UDP User Datagram Protocol UE User Equipment UL UpLink UMTS Universal Mobile Telecommunication System WCDMA Wide Code Division Multiple Access VoIP Voice over IP
vii Contents
1 Introduction 1 1.1 Related Work ...... 1 1.2 Objective ...... 1 1.3 Work Description ...... 2 1.4 Chapter outline ...... 2
2 Cellular Radio Networks 3 2.1 Technical Approach ...... 3 2.1.1 Cellular Networks ...... 4 2.1.2 Handovers ...... 4 2.1.3 Multi-path Fading ...... 5 2.1.4 Path Loss ...... 5 2.2 Multiple Access Schemes ...... 5 2.2.1 Time Division Multiple Access ...... 5 2.2.2 Frequency Division Multiple Access ...... 6 2.2.3 Code Division Multiple Access ...... 6 2.2.4 Othogonal Frequency Division Multiplexing ...... 7 2.3 GSM ...... 7 2.3.1 GSM System Architechture ...... 7 2.4 GPRS ...... 9 2.4.1 EDGE ...... 10 2.5 UMTS ...... 11 2.5.1 UMTS QoS Classes ...... 11 2.6 WCDMA R99 ...... 12 2.6.1 Power Control ...... 12 2.7 HSPA ...... 13 2.7.1 HSDPA ...... 14 2.7.2 EUL ...... 14 2.8 LTE ...... 14
3 Quality of Experience Parameters 15 3.1 QoS vs. QoE ...... 15 3.2 User-Driven Performance Evaluation ...... 16 3.3 Key Performance Indicators ...... 17 3.3.1 Network delay ...... 19 3.3.2 Latency ...... 19 3.3.3 Jitter ...... 19 3.3.4 Packet Loss ...... 19
viii CONTENTS CONTENTS
3.3.5 Throughput ...... 19 3.4 Service Performance Requirements ...... 20 3.4.1 Voice ...... 20 3.4.2 Videophone ...... 20 3.4.3 Online Gaming ...... 21
4 Service Evaluation Cases 22 4.1 Internet Radio ...... 22 4.1.1 Audio Coding ...... 24 4.2 Online Gaming ...... 25 4.2.1 FPS ...... 25 4.2.2 RTS ...... 26 4.2.3 MMORPG ...... 26 4.2.4 Game Phases ...... 27 4.2.5 Traffic types ...... 27 4.3 Test setup ...... 28 4.3.1 Hardware ...... 29 4.3.2 Network load ...... 29 4.3.3 Software ...... 30 4.4 Data Post-Processing ...... 31 4.4.1 CounterStrike Gaming Experience ...... 31
5 Test Results and Analysis 32 5.1 Internet Radio ...... 32 5.1.1 Service Functionality ...... 34 5.2 Online Gaming, FPS: CounterStrike ...... 36 5.2.1 Service Functionality ...... 37 5.3 Online Gaming, MMORPG: World of Warcraft ...... 39 5.3.1 Service Functionality ...... 40
6 Online Gaming Quality Model: FPS 42 6.1 Principal Components Analysis ...... 42 6.2 Quality Model ...... 46 6.2.1 Performance Calculation Methods ...... 48 6.2.2 Model Performance ...... 50 6.3 Extended Quality Model ...... 52
7 Conclusions 54
ix Chapter 1
Introduction
Mobile broadband is becoming an alternative to fixed broadband connections for residential areas. High speed radio access, such as HSPA with bit rates up to 14 Mbit/s, enables the mobile terminals and systems to be used for Internet ser- vices traditionally accessed over DSL or other fixed access. However, it is not obvious that all services normally used with fixed broadband can be used over mobile broadband with equal quality.
The service characteristics put different requirements on the Internet con- nection. For example, delay (or latency) is most important for online gaming, while throughput (or high bit rate) is most important for file sharing. The combination of service requirements and a certain radio access will result in a perceived quality of the service for a user. Can mobile broadband substitute a broadband connection with no perceived diffrence or does it exist service specific limitations?
1.1 Related Work
Previously performed studies in this subject have been done at Swinburne Uni- versity [1, 2] where focus have been on game traffic analysis and service as- sessment for fixed network impairments. Two Master Thesis’s [3, 4] have been performed at Ericsson where characteristics and functionality of online games have been evaluated in a test radio environment. Previously only limited online gaming service evaluation in an operators network have been done, as this thesis contains.
1.2 Objective
This master thesis evaluates how well broadband services work in different ra- dio networks and environments, for example GPRS/EDGE, WCDMA R99 and HSPA. For each service this thesis describes, identification of key parameters that influences service quality and their limitations. One important issue is to evaluate how a degradation in the radio environment affects’s the user percep- tion. The services that has been tested and evaluated are: Internet Radio and two network based online games. The two games are CounterStrike and World
1 1.3. WORK DESCRIPTION of Warcraft, and originates from two separate game genres. Finally, this thesis describes an objective quality prediction model for online gaming.
1.3 Work Description
The work have been be divided into the following steps: 1. A literature survey have been conducted to gain insight to cellular radio networks and service evaluation 2. The testing activities have been defined together with an introduction to the testing equipment 3. Testing execution for chosen services together with a smaller group of test persons in a subjective test.
4. Data analysis of test results, i.e. service evaluation over various radio accesses and identification of the most important service performance pa- rameters 5. The analysis work serves as input to a quality prediction model for online gaming 6. Finally, analysis and conclusions of service experience over mobile net- works and online gaming quality model performance
1.4 Chapter outline
The structure of this thesis is laid out as follows: Chapter 2 gives an intro- duction to Cellular Radio Networks and a technical description of multiple ac- cess schemes and techniques. Chapter 3 describes Quality of Experience, Key Performance Indicators and Service Evaluation. This is followed by chapter 4 which outlines the services who are tested and the testing equipment. Chapter 5 provide results and some initial analysis from the testing activities. Section 6 describes an online gaming quality model for estimating the quality of an online gaming session and a proposed future extension of this model. Finally, chapter 7 contains conclusions from the test results chapter and future work in the field of study.
2 Chapter 2
Cellular Radio Networks
Technologies as the Universal Mobile Telecommunications Systems (UMTS) and High Speed Packet Access (HSPA) are cornerstones in todays and future commu- nications. They are the continuation of the first digital communication system, the Global System for Mobile Communications (GSM). The first step toward Third Generation networks (3G) were done by introducing data capabilities to GSM through the General Packet Radio Service (GPRS). UMTS and HSPA provides the contingency for new services as video telephony, web browsing, file downloads, video streaming and other multimedia services over an air-interface. The mobile wireless systems are currently evolving and there is a steady demand for higher data rates and improved Quality of Service (QoS). This chapter will introduce the reader to Cellular Radio Networks and also take a glimpse at future wireless networks like Long Term Evolution (LTE). Further additional reading on mobile communications systems can be found in [5] and [6].
2.1 Technical Approach
In radio network communication there are two kinds of radio bearers, Circuit- Switched (CS) and Packet-Switched (PS). When a CS connection is established network capacity is reserved during the entire session, even though sometimes no data is transmitted. Existing networks as PSTN and ISDN are typical CS networks. Billing is here based on the session time.
The other network is the PS network. No capacity is reserved and data is routed through the network in small pieces, called packets or frames. The Inter- net is a typical PS network and billing is based on received/transmitted data. Although both CS and PS exist in todays telecommunication there is an obvious convergence toward an all-IP packed-based network. Higher data-rate demands and lower maintenace costs are the two main reasons for this IP convergence. [5]
Definition: A mobile station (MS) is the user equipment used to access the cellular radio network, for example; a mobile phone, a PDA or a Laptop.
3 2.1. TECHNICAL APPROACH
2.1.1 Cellular Networks Cellular radio is a technique to increase the capacity of mobile radio services. The use of multiple low-power transmitters in a cellular network puts limits on the transmission range. The geographical area is then divided into smaller areas, called cells, each served by one or more antennas located on a base station. The location of a base station is called a site. A base station who uses an antenna that radiates in in all directions is called an omnidirectional. By introducing directional antennas cells can be further divided. This method is called sectori- sation. Figure 2.1 shows three sites (left) and on directional antenna (right).
Definition: Uplink transmission is defined as traffic from the mobile station to the network and downlink transmission as the opposite.
Adjacent cells in GSM are assigned different frequencies to avoid crosstalk or interference. In WCDMA systems this is done by using diffrent othogonal codes. However, cells that are sufficiently spatial separated can reuse the same frequencies or codes to increase the system capability. A cellular radio network is often visualised by a simplified model using a hexagonal pattern. In practice, due to variations in signal propagations conditions and local hot-spots a precise hexagonal pattern is not used.
Figure 2.1: Three omnidirectional cell sites (left) where one antenna covers one whole cell. To the right a sectorized cell with three directed antennas.
2.1.2 Handovers Cellular systems are based on mobility. Mobile stations are able to move while connected to the network. When a user crosses a cell border it will result in a handover between the two cells. The connection will be switched from the sub- channel in the first cell to an available subchannel in the new cell. In case of no free channels the connection will be dropped. By measuring the broadcast chan- nel of nearby base stations, a handover is made when the neighboring broadcast channel is higher than the current one. There are three types of handovers:
• Hard handover; A mobile station is switched between two base stations, at any time instance the MS isconnected to only one of them. • Soft handover; A mobile station is connected to two, or more cells at the same time.
4 2.2. MULTIPLE ACCESS SCHEMES
• Softer handover; Like soft handover, buth all cells belog to the same base station.
2.1.3 Multi-path Fading A fundamental characteristic of many radio channels is that the strength of the received signal fluctuates in a random way. This phenomenon is called fading. A mobile station is often not in the line of sight from a base station, meaning several bouncing rays of the signal will reach the receiver. Each ray causes a phase shift and because of movement of obstacles, transmitter and receiver signal power will vary. Multi-path fading is often referred to as fast fading and is often modeled as a Rayleigh distribution. Another fading phenomenon is shadow fading. It is caused by series of obstacles that causes signal attenuation.
2.1.4 Path Loss Due to the distance between the transmitter and the receiver the signal suffers from attenuation, referred to as path loss. The detailed components are Pr receiver transmit power, Ps sender transmit power, Gs sender antenna gain and Gr is the receiver antenna gain and are signal dependent. A general path loss model is formulated as:
Pr = PsGsGrGpath (2.1) where d is the distance between sender and receiver, d0 reference distance, λ0 carrier frequency and α the path loss exponent (normally in the range from 2 to 5 according to measurements). Depending on radio environment, a variety of wireless propagation models can be used. The most known ones are the Okumura-Hata and the Walfisch-Ikegami propagation models. They take into account: frequency, base- and mobile station antenna height, area type and distance and are derived from a combination of empirical and deterministic research. [7]
2.2 Multiple Access Schemes
In a cellular radio network multiple users are able to communicate at the same time, using the same physical resource (i.e. the air interface) to receive and transmit information. In wireless communication there are several different multiple access schemes to divide the air interface between users.
2.2.1 Time Division Multiple Access
Time Division Multiple Access (TDMA) divides a time period Tf into a number of non overlapping subintervals, each with the duration Tf /N. Each subinterval, or timeslot can be assigned to a user. To increase the capacity several timeslots can be assigned to one user, called multi-slot operation. Synchronization and guard-bands in the time domain is needed to prevent interference between users.
5 2.2. MULTIPLE ACCESS SCHEMES
Figure 2.2: Multiple access schemes.
2.2.2 Frequency Division Multiple Access Another approach is Frequency Division Multiple Access (FDMA) where the available frequency spectrum is used to create a number, N, of non-overlapping subchannels. These channels are separated by guard-bands. A combination be- tween TDMA and FDMA schemes is possible and is referred to as F/TDMA. It can be used with fixed channel assignment or by changing channel with a specific pattern, called frequency hopping number.
In FDMA and TDMA systems a channel is assigned to a user for the duration of a session. When the traffic is of bursty nature, as for speech with periods of silence, these silent periods carries no information. A part of the system capacity is being wasted. Systems limited by frequencies or in the latter case timeslots is called hard capacity systems.
2.2.3 Code Division Multiple Access Code Division Multiple Access (CDMA) overcomes the problem of capacity waste by allowing users to communicate in the same time- and frequency do- main but separated by codes. CDMA systems are interference limited, meaning every new user contributes to the interference level. By multiplying the signal with a code, the signal will be spread over a wide frequency, i.e. the power per Hertz will be low. At the receiving side the same code will be applied to all incoming signals and the wanted signal will be despread and decoded while the other signal components will appear as noise. When the interference level is so high that the receiver barley can detect the wanted signal, no more users is allowed to communicate in the cell. [8]
In CDMA systems there are many users who communicates in the same frequency- and time domain. Mobiles that are on different distances from the
6 2.3. GSM base station transmits with a certain individual transmission power. If a mobile near the base station transmits with too high power it will make other signals with lower power undistinguishable. On the other hand, if a station far from the base station transmits with to little power, the signal may never reach the base station. This problem is called the near-far effect and is dealt with by power control, where the channel is estimated before transmitting. This approach is used in current 3G systems such as UMTS where frequency bands are 5MHz wide. The access method is called Wideband CDMA (WCDMA). Figure 2.2 shows TDMA, FDMA and CDMA schemes. [9], [10]
2.2.4 Othogonal Frequency Division Multiplexing Orthogonal Frequency Division Multiplexing (OFDM) uses a large number of closely-spaced orthogonal sub-carriers. The idea is to split a high-rate data stream into a number of low-rate streams to be transmitted in parallel. Multi path delay and interference is decreased when a lower data rate is transmitted on each sub-carrier. OFDM can be viewed as many simultaneously slowly trans- mitted narrow band signals instead of just one fast wideband signal. OFDM is chosen by the 3GPP for its Long Term Evolution Project (LTE) described in section 2.8.
2.3 GSM
The digital GSM-system replaced the old analog cellular phone system and started a new era in telecommunications. GSM represents the second genera- tion networks, often referred to as 2G. GSM specifications were settled by the European Telecommunications Standards Institute (ETSI). Although GSM was standardized in Europe; it became, and still is, world-wide utilized. GSM uses TDMA to divide the available bandwidth and to serve as many users as possi- ble. The spectrum is divided into uplink and downlink bands and the channels are separated with a 200 kHz guard band, which gives 125 carrier frequencies in the 900 MHz band.
In Europe GMS frequencies are also available at 450, 1800 and 1900 MHz. Each carrier utilizes one TDMA-frame and this frame consists of eight timeslots (TS). The duration of one frame is 4,615 ms, giving each TS 0,577 ms. The Guard Period (GP) is an empty space separating the carriers during transmis- sion. Like other cellular networks GSM allows roaming services, meaning that users can use their phones outside of their home network.
2.3.1 GSM System Architechture The GSM network consists of the Base Station System (BSS) and the Network Switching System (NSS). The BSS carries out all the radio functions and consists of the following units: • Base Transceiver Station (BTS) communicates radio signals with the user equipment or the Mobile Station (MS). • Base Station Controller (BSC) which manages and controls the radio func- tion.
7 2.3. GSM
4.615 ms 0 1 2 3 4 5 6 7
TB Data Training Data TB GP
3 57 1 26 1 57 3 8.25 156.25 bits
0.577 ms
Figure 2.3: A basic TDMA frame with eight timeslots. The tail-bits (TB) are used as start/stop sequence for a TS and the training sequence is used by the equalizer to create a channel model.
The MS can be a hand held device or installed in a vehicle. A MS has to be personalized, i.e. associated with a given subscription. In the GSM system this is done with the Subscriber Identity Module, or SIM. Without a SIM, a user can’t access the network, except for emergency traffic. The NSS includes the following functional units: • Mobile Services Switching Centre (MSC), responsible for switching and su- pervision functions. The MSC also serves as a gateway from the PSTN and other networks and serves a number of Base Stations Controllers (BSC). • The Home Location Register (HLR) contains user information and au- thentication parameters. • The Visitor Location Register (VLR) contains user information relating where the user currently is located. This means when a user is called, only a particular location area have to be paged, not the entire network. • In the Authentication Centre (AuC) HLR is provided with authentication and ciphering keys for security reasons.
• The last unit in the NSS is the Equipment Identity Register (EIR). It is connected with the MSC via a signaling link and checks the validity of the used equipment. Network management is realized by Operation and Support Subsystem (OSS) and is connected to both NSS and BSS. OSS realizes functions as performance management, network administration and traffic measurements. [11], [12]
In PSTN speech is digitalized at a rate of 64kbit/s. The GSM uses another, highly optimized approach to make more efficient use of the available spec- trum/bandwidth. Speech is coded at 12.2 kbit/s that uses a full rate channel.
8 2.4. GPRS
PSDN ISDN IP PSTN PLMN
NSS
VLR GGSN HLR MSC OSS AuC SGSN EIR
BSS BSC
BTS
MS
Figure 2.4: GSM and GPRS architechture.
Speech quality is ensured by additional bits for error detection and correction. Coding sequences at 12.2 kbit/s involves speech blocks of 20 ms and consists of 260 bits. [13]
2.4 GPRS
General Packet Radio Service (GPRS) is a packet-based IP connectivity solution and was the start of Mobile Internet with applications as WAP, iMode, MMS and so forth. GPRS is an extension to the GSM architecture and not an entirely new system, additional GPRS nodes are presented in figure 2.4. GPRS is, together with EDGE, described as 2,5G. By allowing timeslots previously used for circuit-switched telephony to be filled with data packets enabled new network capabilities with slightly modified network architecture. The network can adjust capacity between voice and data functions to allow more data traffic when voice traffic is low and vice versa. The GPRS elements to be added in the GSM architechture are: • Serving GPRS Support Node (SGSN) directs/receives packed data to/from the BSC and authenticates and tracks the location of the MS. • The Gateway GPRS Support Node (GGSN) acts as a proxy toward ex- ternal IP-networks and manages dynamic IP addresses to the MS. GPRS offers a connection between voice and data services, meaning while in a data session; users can accept an incoming voice call. GPRS uses four different coding schemes, CS1 to CS4 depending on the radio environment. Close to the BTS a less robust and thereby faster coding scheme can be used. Throughput
9 2.4. GPRS
GPRS TP/TS (kbit/s) EDGE TP/TS (kbit/s) CS1 8 MCS1 8,8 CS2 12 MCS2 11,2 CS3 14,4 MCS3 14,8 CS4 20 MCS4 17,6 MCS5 22,4 MCS6 29,6 MCS7 44,8 MCS8 54,4 MCS9 59,2
Table 2.1: Modulation and coding schemes for GPRS and EDGE per timeslot varies from 8 to 20 kbit/s depending on coding scheme. If addi- tional slots are available they can be assigned to a user creating a multi slot transmission and boost the data rate, and typical GPRS rates is around 40 - 50 kbit/s. An assignment of 3-4 TS per MS is most often used for data traffic. [5]
2.4.1 EDGE Enhanced Data rates for GSM Evolution (EDGE) is an enhancement to the packet switched GPRS. EDGE is a method to increase the data rates on a GSM radio link by adding new modulation and channel coding techniques. Through the use of existing GPRS standards EDGE can deliver significant improvements, this implies that EDGE is an add-on to existing GPRS networks and cannot work alone. To achieve higher data rates EDGE has the contingency to utilize more modulation methods (MCS 1-8, Table 2.1), each optimized to the existing radio environment. Theoretically EDGE can provide 59,2 kbit/s in each if the eight timeslots, adding up to a peak network rate of 473,6 kbit/s if eight times- lots are used. Todays devices can occupy up to four timeslots which results in a peak data rate of about 200 kbit/s. [5], [14]
Before radio blocks can be sent over a radio interface, the transmitter has to address the packets with a unique identification number. In GPRS the packets are numbered from 1 to 128, with a sliding window of 64 packets. After the transmission of a packet sequence the transmitter receives information if some packets were defected and to be resent. This erroneously decoded packet may have the same identification number as a new packet in the queue. Because the addressing window only have 64 addresses the protocol between the terminal and the network might stall and the whole frame have to be retransmitted. In EDGE the identification numbers have been increased to 2048 and the address- ing window to 1024. This minimizes the risk for stalling and prevents unwanted throughput decrease.
EDGE uses link adaptation which is a technique that measures the quality of the radio link and based on this knowledge it chooses among the available modulations schemes. Measurements are done on each burst instead of every 240ms in the GPRS system. This results in highly accurate measurements and enables to a quick reaction when the radio environment changes. [14]
10 2.5. UMTS
2.5 UMTS
UMTS is one of the third-generation-mobile technologies (3G) and employs Wideband CDMA as radio-access technology (WCDMA), described in chapter 2.6. The primary benefits for UMTS are higher user bit rates, better spectral efficiency and simultaneous transmission of voice and data, lower infrastructure costs and a migration towards an all-packet-based network. Compared to other mobile networks, like GSM, UMTS provides different QoS classes for different kinds of traffic described below. The UMTS network achitechture consists of three parts namely; the mobile station, the radio access network UTRAN and the core network seen in figure 2.5.
Core Network
Iu
UTRAN RNC Iur RNC
Iub Node B
Node B Node B UE Uu
Figure 2.5: UMTS network architechture
2.5.1 UMTS QoS Classes UMTS employs four fundamental Quality of Service (QoS) traffic classes for packet-switched data. This QoS architecture divides applications and services depending on, for example, how delay-sensitive the traffic is. These four classes are:
• Conversional - real time data as voice, video telephony and online and interactive games. Very delay sensitive. • Streaming - Streaming of multimedia, moderately delay sensitive. • Interactive - Web browsing. Preserve data integrity, some delay accepted.
• Background - Downloads of email or files. Preserve data integrity, delay may be seconds or even minutes. A conversional service as real-time conversation is restricted by human percep- tion of maximum end-to-end delay. Evaluations have shown that an unaccept- able quality is reached when the end-to-end delay exceeds 400 ms.
11 2.6. WCDMA R99
2.6 WCDMA R99
WCDMA R99 technology has emerged as the most widely adopted third gener- ation interface and its standardization is done by the 3GPP (the 3rd Generation Partnership Project). In release 99 theoretical downlink data rate is just over 2Mbit/s and uplink pratical 384 kbit/s. WCDMA R99 uses a chip rate of 3,84 Mcps which lead to a carrier bandwidth of approximately 5 MHz. WCDMA R99 spreads user information over a wide bandwidth by multiplying it with spreading codes, this keeps interference low and supports very high bit rates. It also supports Bandwidth on Demand (BoD) where a highly variable user data rate is supported. Each user allocates frames of 10 ms duration and the data capacity among the users can change from frame to frame. Finally WCDMA R99 is backwards compatible with existing GSM networks; therefore handovers between the two networks is possible.
2.6.1 Power Control A fast closed loop is the solution to power control in WCDMA R99. The base station and the MS performs frequent estimations (1500 times per second) of the received Signal-to-Interference Ratio (SIR), both in the up- and downlink. This SIR is then compared with the target SIR and the mobile station transmit power is then regulated. Power control regulates the target SIR according to the needs of the individual radio link and aims for a constant quality. This quality is often defined as a specific bit error rate (BER) or block error rate (BLER). The fact that battery lifetime of mobile stations is limited, saving power resources is crucial. [13], [9]
12 2.7. HSPA
2.7 HSPA
High Speed Packet Access is a WCDMA upgrade and with theoretical data rates of up to 14 Mbit/s in the downlink and up to 5 Mbit/s in the uplink. HSPA boosts system performance and improves the user experience by providing a reduced latency and increased data rates. HSPA makes mobile broadband a reality by improving service experience for web access, FTP, VoIP and streaming services. By the following means can HSPA increase the system capacity: • High speed shared channel transmission, shared both in code and time domain.
• A shorter Transmission Time Interval (TTI), reduces round-trip time and improves link adaptation. • Fast scheduling, prioritizes users with good channel conditions. • Fast retransmission and higher modulation schemes.
• Fast hybrid Automatic Repeat reQuest (ARQ). A mobile station can re- quest for a rapid retransmission, improves system performance and en- hances robustness.
4000 HSPA TP downlink HSPA TP uplink 3500
3000
2500
2000 kbit/s
1500
1000
500
0 0 50 100 150 200 250 time [s]
Figure 2.6: FTP upload and download over HSPA with EUL, HSPA reference speedtest performed at UTN
HSPA is a combination between High Speed Downlink Packet Access (HS- DPA) and High Speed Uplink Packet Access (HSUPA), also known as Enhanced Uplink (EUL). Below are basic principles of HSDPA and EUL described.
13 2.8. LTE
2.7.1 HSDPA HSDPA is based on a shared channel transmission, meaning the channel codes and the transmission power in a cell is seen as a common resource. This results in more efficient use of available resources. In 3GPP Rel 5, HSDPA adds a new transport channel to WCDMA, the High Speed Shared Channel (HS-DSCH). In future HSPA evolved, additional downlink 2x2 MIMO (Multiple Input Multiple Output) is supported. By using two antennas and 64 QAM modulation scheme the peak data rate can be doubled, enabling a data rate of 42 Mbit/s. Operators are currently deploying HSDPA networks with downlink data rates of up to 7.2 Mbit/s.
2.7.2 EUL Enhanced Uplink (EUL) is defined in 3GPP Rel 6 as a new transport channel, called Enhanced Channel (E-DCH) and is dedicated to only one user during a session. EUL improves uplink performance by introducing features as multi code transmission, fast hybrid ARQ, improved scheduling and shorter TTI, as short as 2 ms compared to previous 10 ms. When EUL together with HSDPA is fully deployed operators have the freedom to move traditional voice traffic over to an IP-based platform. Hence new innovative voice and data applications are enabled in the packet domain. [15], [16]
2.8 LTE
3GPP:s Long Term Evolution (LTE) is the continued improvement of the UMTS network to ensure future competitiveness among other emerging radio technolo- gies. LTE aims to be an extensive evolution of today’s 3G and a precursor to upcoming 4G networks. The overall intent of LTE is to provide an extremely high-performance radio access with downlink data rates of 100 Mbit/s and 50 Mbit/s in the uplink. Planned to be released in 2009, a number of specific goals for the LTE project have already been set up.
LTE is going to be an ”All-IP” network, based on the Internet IP core protocol. LTE will also support high user mobility and increased QoS for packet services; this means for example traditional circuit-switched telephony will be replaced by Voice over IP (VoIP). LTE also facilitates usage of advanced antenna technologies, such as MIMO concepts to boost system performance even further. [5], [17]
14 Chapter 3
Quality of Experience Parameters
A wide range of applications as e-mail, file downloads, instant messaging, video and voice streaming and on-line gaming are widely used over a traditional fixed Internet accesses. With the introduction of mobile broadband raises the oppor- tunity of using these applications over a radio interface. But there are uncer- tainties of how well traditional broadband services works when connected over a radio access network and which parameters that influence the service quality the most. This chapter will introduce the reader to Quality of Experience (QoE) and evaluation of important service parameters. Additional reading can be found in [18] and [19].
3.1 QoS vs. QoE
Mobile data services are growing very rapidly. Improved terminals, new appli- cations and networks with higher data rates contribute to this higher demand. Technology centered measurements like data rate, delay and bit error rate re- veals little of the true user experience of a session. This type of measurements are often called the QoS of the network. With the growth of mobile services it has become important for operators and developers to measure the QoE both of the network and the services provided. QoE could be defined as ”‘the overall acceptability of an application or service, as perceived subjectively by the end- user”’.
The QoS approach is technical and often described as a hard measurement. QoE on the other hand, delivers a framework of how a user perceives the usabil- ity and quality of a service, i.e. user satisfaction [19]. As human experience is open minded and dynamic is has become important to recognize user perception quality.
15 3.2. USER-DRIVEN PERFORMANCE EVALUATION
Quality of Experience
Network coverage, Objective factors End-to-end service offers and as user network QoS expectations or level of support requirements
Underlying network infrastructure
Figure 3.1: The user can’t see the underlying network infrastructure, but ”feels” the overall service quality which highly equals to Quality of Experience [19]
3.2 User-Driven Performance Evaluation
A major challenge for wireless IP-based networks is to provide adequate QoS for different types of services. This requires a detailed knowledge of the perfor- mance requirements for the particular services and applications. The starting point for deriving these performance requirements must be the user. The mea- suring of hard QoS parameters is not enough for knowing user satisfaction. High throughput is meaningless when the end-user is unsatisfied with the service qual- ity. Figure 3.1 presents a typical sketch of network quality, from a users’ point of view. It is possible to gain a greater use of new networks if subjective service evaluation can be performed when designing the networks. This gain includes a larger scope of available services that can be supported and accessed through the network with sufficient user satisfaction.
A typical user is not concerned with how a particular service is implemented. However, the user is interested in comparing the service over a wireless access with the fixed broadband access. User driven performance evaluation should be expressed by parameters that: • Take into account all aspects of the service from the user’s point of view • Focus on user-perceivable effects, rather than their causes within the net- work
• Independent of the specific network architecture or technology • Can be objectively or subjectively measured at the service access point • Can be easily related to network performance parameters, for data analysis [18]
16 3.3. KEY PERFORMANCE INDICATORS
Figure 3.2: Collection of network KPI:s can be made in all of the network architechture nodes or in the user equipment
3.3 Key Performance Indicators
QoS figures from a network can be collected either at system level or in the user equipment. At system level the data is collected from log files, sniffing tools or counters in network nodes or base stations. The other side is at the user equipment where network measurement software tools as protocol- or link sniffing are used during fixed- or mobile networks tests. Assembled data from user equipment or station logs serves as input to Key Performance Indicator (KPI) computations, where typical KPI:s are presented in table 3.3 and 3.3. A Network Management System (NMS) at a higher network level can also be used for KPI collection.
To be able to rate QoE of the users the relationship between QoS KPI and QoE KPIs has to be identified. KPI measurements are possible for both PS and CS services, although this section only considers PS based services. [18]
17 3.3. KEY PERFORMANCE INDICATORS
Quality of Experience is often based on human perception expressed with words like excellent, good, fair, poor and bad, instead of just in metrics. A combination of these two is also possible. There is a methodology where QoS performance parameters are mapped onto user perceptible QoE targets. This is a very challenging task and hard to do with acceptable accuracy.
User expectation comes in place when identifying QoE KPI:s and can be grouped into two main categories, namely reliability and comfort. Another classification of QoE KPI:s are: accessibility, retainability and integrity. Table 3.1 and 3.2 outlines reliability and comfort when perceived user experience is evaluated.
Service availability Local and global coverage - Seamless usage Service accessibility Success rate of user connections Service access time Delay in setup time - Experienced response time by user Continuity of service The retainability of a service connection and its performance over time - Duration of a session or a call - Blocking and downtime of the system
Table 3.1: Reliability, availability, accessibility and maintenance KPI:s
Quality of Session Application layer packet loss Ratio of achieved and demanded bit rate Bearer stability, bit rate variation Average throughput (kbit/s) Average latency (ms) Average delay variation (ms) Ease of use Ease of service accessibility Level of support Quick user support
Table 3.2: Service comfort KPI:s [19]
18 3.3. KEY PERFORMANCE INDICATORS
Figure 3.3: Latency measurement from the user equipment to the server where the latency equals T1 + T2
3.3.1 Network delay Delay is a very important measurement and can be described in many ways, for example the time it takes to set up a service or to receive information during a service session. There are three types of delay, namely delay introduced from the user equipment, end node delay (i.e. server delay) and network delay, which further can be divided into core network delay and access network delay.
3.3.2 Latency Latency is measured from the time a request (e.g. a single packet) leaves the user to the time the response (e.g. an ACK) arrives back at the user from the serving entity, figure 3.3. It can also be measured in a one-way fashion, e.g. the time from the sending node to the receiving node. Latency is often refered to as round-trip time (RRT). [3]
3.3.3 Jitter Jitter or delay variation, is an important parameter in packet switched networks as the Internet. The packet inter-arrival times can differ due to queuing delays in nodes along the route from the source to thedestination. Jitter at the application layer is removed by using buffers at the receiving end. In UDP based services delayed packets will be discarded causing periods with no data at the application level.
3.3.4 Packet Loss Another service parameter is packet loss, sometimes referred to as information loss and is caused by lost packets. It is obvious that packet loss has a direct impact on the perceived quality for the end user. If TCP (Transmission Control Protocol) is being used, lost packets are resent, but this is not the case when UDP (User Datagram Protocol) is used. [3]
3.3.5 Throughput Throughput, or data rate in the up- or downlink is the traffic to or from the user equipment. The data rate between two end points during a session is important when evaluating the service. To fully utilize a service, the data rate has to reach the minimum level of the service requirements, and with limited variations. A smaller variation in data rate is handled by input buffers in the application at
19 3.4. SERVICE PERFORMANCE REQUIREMENTS
Internet Radio Online Gaming Audio Codec Game start-up time Stream bit rate Map load time Buffering time Mean latency Re-buffering time Kills/Deaths Inter-buffer times Jitter Packet loss
Table 3.3: Service specific parameters for Internet Radio and online gaming the user equipment. Packet loss, jitter, latency and throughput are the four main parameters that influences users’ service perception the most.
3.4 Service Performance Requirements
3GPP together with the International Telecommunications Union (ITU) have stated requirements between communication entities for different types of mul- timedia services. These parameters represent commonly accepted values for voice, video and data services as real-time games and are presented in table 3.4, for Internet Radio and online gaming is service parameters presented in table 3.3.5. Both QoS parameters QoE for a service are necessary when evaluating the service quality.
3.4.1 Voice The human ear is sensitive to quality variations and conversational voice require- ments are heavily influenced by one-way delay and jitter. In the span between 150 and 400 ms a voice conversation can proceed with acceptable quality, al- though a value below 150 ms is preferable. Emerging IP-based services as VoIP will have to meet the quality and performance levels of the established PSTN, presumably with even better performance to reach market and user satisfaction. A typical figure for VoIP communications is packet loss is below 1%. Compared to CS networks where frame loss is typical below 2% with an end-to-end delay below 250 ms.
3.4.2 Videophone Videophone means a full-duplex communication where the traffic consists of both voice and video. In fact, the same requirements for voice will apply and the added video service must be within certain limits to provide sufficient quality. Due to human eye perception the videphone packet loss limit is around 1%.
20 3.4. SERVICE PERFORMANCE REQUIREMENTS
3.4.3 Online Gaming Online games represent the gaming genre when the game is played over a net- work connection usually over the Internet or just over a Local Access Network (LAN). With the growt of broadband accesses the demand on online games have been increasing and previously single player games have implemented network support. The market produces a variety of network games and they all have different network requirements for their functionality.
Medium Application Data rate Latency Audio Voice service e.g. 4-25 kbit/s < 150 ms preferred Skype < 400 ms limit Video Videophone/Webcam 32-384 kbit/s < 150 ms preferred < 400 ms limit < 100 ms lip-synch Online gaming FPS e.g. < 60 kbit/s < 50 ms preferred CounterStrike < 150 limit RTS e.g. < 80 kbit/s < 150 ms preferred World of Warcraft < 450 limit
Table 3.4: 3GPP framework requirements for network based on voice and mul- timedia services. [20]
21 Chapter 4
Service Evaluation Cases
This chapter will introduce the services which have been evaluated in this the- sis. A user connected to the Internet over a fixed broadband access is nowadays using a range of network based services. Wireless radio access networks as GPRS/EDGE, WCDMA R99 or HSPA opens the possibility to extend the use of Internet services. This chapter describes the services and their specific char- acteristics, and chapter 5 contains the test evaluation.
Measurements of radio network performance include methods to supervise system and service quality and to find reasons for low performance. Tradition- ally focus has been on speech and video quality for services like normal voice calls, video streaming and video telephony. Previously performed multimedia evaluations has got limited focus on online gaming, although it employs mil- lions of simultaneous users every day. Therefore this service evaluation puts more weight in the online gaming area.
Evaluation of Internet Radio was done by one user, namely the author him- self, and the online gaming evaluation was made both by the author and a testing group of nine persons.
4.1 Internet Radio
Internet Radio is nowadays a popular service where listeners can tune in their favorite radio channel, located anywhere in the world with Internet broadcast- ing. There are channels dedicated to (almost) all existing music styles and some concentrated on news coverage and stand-up comedy. Internet Radio is not synonymous with file downloading because the media is streamed from a server and not stored at the user terminal.
Radio stations are chosen from a portal and a variety of bit rates are avail- able, which gives the listener a hint of expected audio quality. One popular portal that hosts a range of Internet Radio stations is www.shoutcast.com with thousands of simultaneous listeners. An Internet Radio session is initialized when the user tunes-in a station, followed by a music or information flow. [21], [22]
22 4.1. INTERNET RADIO
Figure 4.1: An Internet Radio stream modeled as a queuing system.
Since Internet Radio uses a buffer for the incoming data stream and is TCP- based, playback will continue until the buffer is empty and no packets are lost. Quality impairment is therefore gaps causing silence periods when the applica- tion is rebuffering. A condition with an empty buffer is the result of insufficient bandwidth or large amount of TCP retransmissions. Retransmissions can be triggered by bad radio environment interference by nearby cells, or bad propa- gation environment.
The buffer can be modeled as a queuing system with a traffic arrival rate, λ, buffer size, n and traffic departure rate, µ (play back stream rate). Serving discipline of the queue applies first come first served, i.e. in order of arrival. In the stationary case when the buffer is full, traffic arrival rate equals traffic departure rate (i.e. audio playback rate) regardless if the network can deliver higher data rates.
Inter arrival mean equals 1/λ and service mean is 1/µ. The parameter ρ is defined as traffic intensity and it is required that µ ρ = < 1 (4.1) λ since, otherwise, there will be an buffer overflow. [23]
Depending on TCP-connections and due to the nature of Internet Radio, quality degradation effects are silent periods when the application buffer runs empty. The duration, frequency and time between these gaps effect the ser- vice perception together with codec and stream bit rate of the audio session. Additional measurements as tune-in times and tune-in failure rates can also be monitored and can be seen as service quality parameters. The distance between the traffic arrival and playback curve corresponds to the amount of data in the client buffer and if the playback curve reaches the traffic arrival curve the media stream will be interrupted, figure 4.2. [24]
23 4.1. INTERNET RADIO
Traffic arrival
n io at er en Network delay Buffer g fic
Sequence number Sequence f ra T k ac yb la P
Time
Figure 4.2: Playback of an Internet Radio stream.
4.1.1 Audio Coding Audio coding-decoding system called codec constitutes the most critical element in the quality of a digital radio system. Due to the nature of the human audi- tory system and the psycho acoustical model, some audio components will not be perceived by the ear. Therefore codecs aim to code information that the ear will hear while maintaining high audio quality.
Some of todays’ available codecs are the MP2, MP3, AAC, AAC+ and HDC codecs. The MP2 and MP3 codecs were the first ones to compress audio bit rates to useful levels. The AAC/AAC+ codecs provide a far more efficient com- pression and outperforms both MP2 and MP3. Different encoding algorithms do have ”‘sweet spots”’ where they work best. At bit-rates much larger than this target bit-rate the audio quality improves only very slowly with bit-rate, at much lower bit-rates the quality decreases very fast. MP3 at 128 kbit/s provides equivalent AAC quality at 96 kbit/s where 192 kbit/s is needed for the MP2 codec. For most people an MP3 codec at 128 kbit/s constitute does sufficient quality with minor or no audio impairment. Internet Radio stations coded with the MP3 codec been utilized in all the test cases. Table 4.1 outlines a comparision of audio bit rates fot the MP3 codec. [25]
24 4.2. ONLINE GAMING
32 kbit/s Similar to analog AM quality 96 kbit/s Similar to analog FM quality 128 - 160 kbit/s Decent quality, difference can sometimes be obvious 128 - 192 kbit/s Often used for MP3 music files 192 kbit/s Very good quality 224 - 320 kbit/s High, near transparent, very good quality 1411 kbit/s WAV sound format of Compact Disc Digital Audio
Table 4.1: Comparsion of audio bit rates for the MP3 codec. [26]
4.2 Online Gaming
The growth and penetration of broadband access networks to the home has fueled the growth of online games played over the Internet. Gaming servers reports that the number of online players is steadily growing. There are four main network parameters that affect a user’s perception of a multiplayer online game: latency, jitter, packet loss and throughput.
Game genres are divided by how the user interacts with the game and how the game world is being viewed. Two factors, interaction and perspective, helps the classification that determines latency impacts on online games. Gamers can also be classified. Those who spend a lot of their leisure time and money into gaming, are often called Hardcore gamers. They have higher demands on their network access and user equipment. Leisure gamers are people who spend less time and money into their hobby, and their demands are not as high as the hardcore gamers.
The online gaming part of this thesis considers two games, CounterStrike and World of Warcraft.
4.2.1 FPS The first person shooter (FPS) genre, with games as CounterStrike (CS), Quake and Unreal has become extremely popular in recent years. The game is viewed through the eyes of an avatar in the game itself, creating a first person perspec- tive. An avatar is the player’s physical representation as a three-dimensional model. FPS games are highly interactive and require quick hand-eye coordi- nation in moving the cross-hair towards an opponent, forcing players to make split-second decisions.
CS is played online and consists of many users in a highly dynamic environ- ment. Players are grouped together in teams (called clans) and the goal is to complete a specific task, for example to defeat the opposite team. Communi- cation between the team members is done via an Internet Relay Chat (IRC), a many-to-many communication protocol or a voice service implemented inside the program. FPS relies on fast user responses and is therefore very sensitive for delays. It is today’s most network demanding game regarding latency, jitter and packet loss. The amount of data being transmitted depends on number of players active in each session.
25 4.2. ONLINE GAMING
It is obvious that the game requirements are differing depending on the game and player expectations. A very experienced player may experience gam- ing impairments more annoying than a less experienced player. Nevertheless, performance limits for different games can also vary during game play. In FPS games periods of close combat puts high requirements in low delay while in other periods, for example during movement, the same delay impairment is not as crucial. [27]
4.2.2 RTS Real Time Strategy (RTS) requires slightly lower user response time and the game play is less reaction based. RTS perspective is often from an overview position, giving the player freedom to supervise large areas at the same time. Players interact with a handful of other players during a game session and their contribution to the game is often greater than in a FPS game. Players manages their resources and struggles to harvest goods, build buildings and train armys. Finally players challenge their opponents in battle. RTS games results in longer playing sessions and leads to an unwillingness to leave during the game session. Example of RTS games are Command and Conquer, Star Craft and the precursor game Dune II.
4.2.3 MMORPG Massive Multiplayer Online Role Playing Games (MMORPG) like World of Warcraft (WoW) and Guild Wars, is a multi-user online game where players interact with one another in a virtual world. A subscription is needed to play and the game consists of highly variable environments with thousands of play- ers online at the same time. An avatar is created when first entering the game and the character can belong to different character classes and races. In the game the player receives quests from computer-controlled players (also called Non Player Characters) and receives experience, money and items as rewards for completed quests. [28]
Players can join groups and complete quests together. Conversation is done via a built in chat or through an external voice program. Gained experience increases the players skills and opens up new levels with increased difficulty. MMORPG games have compared to RTS games both tighter delay restrictions and higher data rates. This is needed due to the higher number of concurrently online players. But the MMORPG is played at a slower pace compared to a FPS game. [20], [29]
26 4.2. ONLINE GAMING
4.2.4 Game Phases Most multiplayer online games utilizes a client-server architechture where game- state information is shared among connected users. Game state information is supposed to be as close to real-time as possible and player updates are sent in regular fashion to the server. Player updates are actions as walking or shooting. Generated network traffic alters between roughly four phases during game-play. These phases affects player interaction, although the duration and frequency of these phases varies. Depending on the game played most online games consist of:
• Setup - In this phase, the game hosts is waiting for other player to join the game. Players are also choosing game parameters as team, maps, character outfit and other game specific options. A typical player often makes only a few game setting, resulting in limited network traffic from the client to the game server. This phase is not latency sensitive but depending on high throughput. • Synchronization - The synchronization phase is just before the actually game starts. Here the choices made are by the players in the setup phase synchronized exchanged between game host and active players. Game specific information like as downloads of maps, are also done in this phase. Users don’t interact with the game and this phase is often characterized by high bit rates to get the game to started as soon as possible. Latency is not an issue here. • Game play - During this phase the game is played and player interactions are communicated to the game server. A FPS combat game communicates the movement of an avatar and firing of a weapon towards other avatars. The game play phase has generally moderately bit rates but exchanges small packets of information in a frequent manner aiming to keep latency at a minimum. During this phase latency, jitter and packet loss are ma- jor performance degrading parameters and effects player actions and the perceived gaming experience. • Transition - When a user moves between for example two worlds new information as maps or attributes have to be loaded. During this phase network traffic is limited and the user is not active and due to most of the information is extracted from a local disk. [27], [2]
4.2.5 Traffic types The network traffic varies depending on which phase (as described above) the player currently is in. During the setup and synchronization phases where in- formation loss is crucial TCP is used for most of the traffic. During game play when player updates is sent to the server most traffic flows over UDP, where in- formation loss is resulting in sudden game freezing or other impairments which reduces the game experience.
Most games are designed with low bit rate requirements, instead sending frequent but small update packets to the server. It reduces the effects of sudden
27 4.3. TEST SETUP
CounterStrike − Downlink throughput − WCDMA R99 400
SYNCHRONIZATION 350
300
250
SETUP TRANSITION GAMEPLAY 200 kbit/s
150
100
50
0 11:45:45 11:46:19 11:46:52 11:47:26 11:48:00 11:48:33 11:49:07 11:49:40 time [HH:MM:SS]
Figure 4.3: Throughput downlink for the gamephases in CounterStrike mea- sured with TEMS Investigation. packet loss and can in some cases be handled by repair techniques. One popular loss repair technique is called ”Dead reckoning”. It is a process that estimates one’s current position based upon previously determined positions. More de- tailed information can be founf in [3]. Impairments during game play are often referred to as ”lags” and some common problems are listed below. [4].
• Game freezing, the screen freezes for a random time because of a major packet loss and is often followed by a sudden update, called warping. • Warping, an update from the client to the server, or vice versa, is lost. This loss results in a hasty re-location of the user to the last known client position. • Delayed responses, due to high network latency, a user can suffer from a delayed response. A user action will not take place as the user intended.
4.3 Test setup
The coming sections presents an outline of the test setup and how the test- ing activities were performed. The evaluation cases for each service have been performed using a test protocol with a set of variables for each test case. The perceived user experiences were graded with an quality scale ranging from ”bad” to ”excellent”. A Mean Opinion Score (MOS) was the calculated. MOS pro- vides a numerical indication of the perceived quality of received media after compression and/or transmission. The MOS is expressed as a single number in the range 1 to 5, where 1 is lowest perceived quality (bad), and 5 is the highest perceived quality (excellent). Table 4.2 presents the MOS scale. Finally
28 4.3. TEST SETUP the perceived user experiences were commented in more vivid terms to evaluate session.
MOS Quality Impairment 5 Excellent Imperceptible 4 Good Perceptible but not annoying 3 Fair Slightly annoying 2 Poor Annoying 1 Bad Very annoying
Table 4.2: Mean Opinion Score.
4.3.1 Hardware Testing activities were made in both an operators network (Telia) and at Erics- son’s test networks in Stockholm (STN) and Ursviken (UTN). A Sierra Wireless AIR-Card 850 with possibilities to connect to GPRS/EDGE, WCDMA R99 and HSDPA networks was used for the operator network tests. HSPA tests were done by using a EUL capable Qualcomm TM7200 mobile at the test networks in STN and UTN. The GPRS/EDGE and WCDMA R99 tests were performed at a variety of places in the area around Pors¨on,Lule˚a.To evoke varying (and sometimes bad) radio conditions drive tests were performed for GPRS/EDGE, WCDMA and HSPA networks. Broadband reference tests were performed at the office over a fixed broadband connection.
Table 4.3 lists the specifications of the laptop being used. Due to latency and throughput limitations in GPRS/EGDE access networks, these two radio access techniques are only used for Internet Radio evaluation and not considered when evaluating FPS online games.
Equipment Specification Brand Dell Latitude D610 CPU Intel Pentium M 1,6GHz Video card Intel Graphics Media Accelerator 128MB RAM memory 1GB RAM Hard drive 60GB - 5400 rpm Operating system Windows XP Professional SP2 Screen 14.1 inch TFT active matrix 1400 x 1050
Table 4.3: Specifications of the user terminal.
4.3.2 Network load The network load in WCDMA R99 networks in an around Lule˚ais at a consis- tently low level and the tests have not been affected by concurrent traffic. At STN and UTN experiments performed were with no other, or limited concurrent network traffic.
29 4.3. TEST SETUP
4.3.3 Software Logging software running in parallel with the service being evaluated enabled intense RLC- and IP-level statistics for test analysis. These statistics served as input for later performance analysis of the measured parameters. In partic- ular Ericsson’s optimization tool TEMS Investigation was used for logging of radio network statistics in GPRS/EDGE, WCMDA R99 and HSPA networks. Further, Wireshark (former Ethereal) packet sniffer was used for logging of IP traffic statistics. For measuring latency times PingPlotter, a simple ping tool was used. PingPlotter uses ICMP (Internet Control Message Protocol) packets for measuring latency times in the network, which generates some limitied extra network traffic. Figure 4.4 outlines data collection and analysis.
Core Network
User Equipment Application
Wireshark TEMS PingPlotter
logs
PERL
Matlab
Figure 4.4: Sequence picture for data collection and analysis
30 4.4. DATA POST-PROCESSING
4.4 Data Post-Processing
Three different log files were created from Wireshark, TEMS Investigation and the PingPlotter software. Log files were parsed with perl script and .m files were created for later analysis in Matlab. One important issue was to analyze graphs in system time to be able to corrolate time events between different log files. Figure 4.4 outlines information extraction flow.
4.4.1 CounterStrike Gaming Experience Nine players with different gaming background and skill level played the game CounterStrike. The players first estimated their playing skill and amount of playing hours and were then divided into three separate groups: ”‘novice”’, ”‘intermediate” and ”‘skilled”’. Each playing session lasted 10 minutes. Due to changing playing conditions on game servers the goal was to play as similar sessions as possible for all the players and test cases. The two utilized servers are listed in table 4.4.
Game map is the visual appearence where the playing session was performed. At game servers there are a a range of maps available. The map was first played over a broadband access as a reference; afterwards the same map was played over a WCDMA R99 connection. Each session lasted 10 minutes. The goal was to kill as many of the other players as possible, and at the same time avoid to get killed by the other players. If the player itself got killed, he could re- enter the game after a short while and continue playing. The quality grading scale in table 4.2 were explained to facilitate the service session grading before session started. Additional grading was with a the Stay/Leave scale. The scale faciliates the willingness to stay or leave the game due to the perceived quality, where grade 1 means that the user rather quits playing than to continue the session. Each playing session ended with an interview to evaluate the players gaming experience. The players the answered a questionnaire on the perceived service experience where the questions are listed in Appendix I together with raw data from the tests.
Server IP-addr Map SWE-BiH #1 213.114.234.192 de dust2 cz CZ1.2 #2 211.98.162.100 de dust2 cz
Table 4.4: Utilized CounterStrike servers and gaming map where test sessions were performed.
31 Chapter 5
Test Results and Analysis
This chapter presents test results from the evaluated services Internet Radio and online gaming (CounterStrike and World of Warcraft). Each test section is concluded with a short service evaluation. Additional conclutions are found in chapter 7.
5.1 Internet Radio