The Impact of QoS Support on the End User Satisfaction in LTE Networks with Mixed Trafc

Iana Siomina and Stefan W nstedt a

Ericsson Research, Sweden, E-mail: {iana.siomina,stefan.wanstedt}@ericsson.com
Abstract How good is a best effort network for different services in multi-service LTE networks? In this paper, we try to answer the question in the context of capacity of an LTE network with users concurrently running multiple services. We present a system simulation study for two-service scenarios with VoIP and a second service represented by real-time video, mobile TV, or web surng. We present results for a best effort network and compare to those for a network with QoS provisioning. The study demonstrates that trafc differentiation and service prioritization are particularly crucial when a delay-critical service, e.g., VoIP, is in combination with a delay-insensitive intensive trafc. By prioritizing VoIP, we achieve VoIP capacity comparable to that in pure VoIP simulations at a cost of a few-percent capacity loss of the second service. A simple model for network capacity assessment for the two network types is presented to support our observations theoretically. Index terms LTE, QoS, capacity, downlink, trafc mix, VoIP, video, TCP.

I. I NTRODUCTION Long-Term Evolution (LTE) [5] is an emerging radio access network technology standardized in 3GPP [1] and evolving as an evolution of Universal Mobile Telecommunications System (UMTS). LTE is a converged all-IP network, and Quality of Service (QoS) provisioning is crucial for providing a range of IP-based services in the new generation networks. Hence an evolved 3GPP QoS concept [10] has been developed. In radio networks, QoS implies trafc differentiation and using multiple bearers (a bearer is a point-to-point communication service between two network elements) with conguration and priorities optimized to ensure sufcient service quality for each user. Network-initiated bearer establishment and networkcontrolled simplied QoS proles based on QoS Class Identiers (QCIs) are among the key elements of the evolved QoS concept. Both aim at ensuring consistent QoS and policies across different User Equipment (UE) vendors and models as well as in roaming scenarios. A QCI is a pointer to a precongured set of node specic parameters [2], e.g., scheduling weights, admission thresholds, packet discard timer, etc. A set of pre-dened QCIs is used to ensure class-based QoS. Bearer-level QoS is further detailed by the Policy Charging Enforcement Function assisted by a Subscriber Policy Register. A wide range of QoS-related functions and parameters will allow operators to adopt the level of QoS support in the network aligned with their own strategy. Furthermore, initially an operator may be primarily interested in achieving high peak rates. There even exist some scenarios in which QoS may be not needed, e.g., low trafc load (a likely case during the initial deployment phase) and mostly non-delay-critical data trafc (an older network can, for example, be used to provide voice service). Note, however, that it is not true that QoS is only needed at high network loads, irrespective of the trafc pattern. This is because services with some special QoS requirements (e.g., jitter, end-to-end delay, packet loss, buffering time, etc.) have to compete with delay-insensitive intensive best effort (BE) trafc, which may be difcult even at low loads when

the services are being run at the same UE. With BE service, the network neither provides any delivery or service quality guarantee nor prioritizes any service. This results in unspecied variable user bitrate and delivery time, depending on the current queue length, channel quality, and network load. Our objective is to investigate the performance when delivering different services over a BE bearer in terms of capacity of delay-critical services, such as Voice over Internet Protocol (VoIP), in various mixed trafc scenarios. We present an LTE performance simulation study with the focus on downlink (DL) two-service scenarios with VoIP and a second service represented by real-time video, mobile TV, or web surng. We compare per-service and combined performance in each trafc mix in a BE network to that in a network with QoS provisioning. We also present a simple model for network capacity assessment for the two network types, to support our observations theoretically. Numerous theoretical and simulation-based studies have been presented in the literature addressing various QoS aspects in cellular networks, but far fewer specically in LTE. Among them, many disregard important protocol aspects and interlayer communication, assume unrealistic trafc models or single-service scenarios. Recently, the authors of [7] presented a bottom-up approach for modeling cumulative performance degradation along protocol layers and predicting the performance of different services in single-service scenarios. The effects of QoS scheduling strategies on service performance in a trafc mix consisting of VoIP, streaming video, and Session Initiation Protocol (SIP) were studied in [14]. In [4], the authors presented a QoS-aware scheduling approach with an adaptive VoIP priority mode which aims at decreasing the negative impact of VoIP packets prioritization on the overall system throughput. Mixed trafc scheduling has also been studied for earlier UMTS networks, e.g., for HSDPA [6], [11]. II. LTE R ADIO ACCESS In the following we describe the data ow through Layer 2 and Layer 1 delivered on one or multiple bearers in the form of IP packets from the network (i.e., IP) layer and transmitted in DL. Layer 2 has three sub-layers, PDCP, RLC, and MAC. Robust Header Compression (ROHC) is optionally performed at the PDCP sub-layer. The data unit is then ciphered. The PDCP header thus carries the required information for ROHC, decompression, and deciphering. From PDCP, the protocol data unit (PDU) is delivered to RLC which performs concatenation of several PDUs from the same bearer or segmentation if needed and adds an RLC header. In MAC, PDUs from several bearers can be multiplexed, i.e., form one MAC PDU. RLC concatenation and segmentation as well as MAC multiplexing are affected by scheduling decisions, i.e., transport blocks are formed according to the amount of available resources. Finally, transport blocks are delivered from MAC to the physical layer, where a Cyclic Redundancy Check (CRC) is added. The transmission time interval (TTI), or the transmission time

978-1-4244-2644-7/08/$25.00 2008 IEEE

of a single transport block, in LTE is one sub-frame of 1 ms. The amount of data transmitted in one transport block depends on protocol and signalling overhead, selected modulation and coding scheme, and the amount of allocated resources, i.e., power and resource blocks1 (RBs). If not properly decoded from the rst attempt, multiple transmissions of the same transport block can be combined at the receiver. Orthogonal Frequency Division Multiplexing (OFDM) and Single Carrier Frequency Division Multiple Access (SC-FDMA) have been adopted as the transmission schemes for DL and uplink (UL), respectively. III. A N A PPROACH FOR T HEORETICAL C APACITY E STIMATION IN M IXED T RAFFIC S CENARIOS Next we present an approach for capacity assessment in a BE network with mixed trafc and a single queue where packets are placed in their arrival order. The approach is extended in Section III-B for a case with a separate queue for each service, where different services have different priorities. A. Without Service Prioritization Assume a user i running two services concurrently during time interval T . The two services have average packet transmission rates of f1 and f2 , respectively, and average packet sizes of s1 and s2 , respectively. Without loss of generality, we assume that f1 f2 . All packets become available for the eNodeB DL scheduler in the same order and at the same rate as they were generated. The total amount of data available for transmission during time T is thus (s1 f1 + s2 f2 )T . The amount of data that can be transmitted to user i over a radio link in one TTI depends on the scheduling decision and the selected transport format, both of which depend on available resources and channel quality. Let smax be the average i amount of data that can be transmitted to user i in a single transport block. Given for user i a set of assigned RBs, the average signal-to interference-plus-noise (SINR) ratio (may vary by RB), and the target block error rate (BLER), smax can be i found by mapping the SINR into the effective bitrate obtained from link-level simulations and subtracting the physical layer overhead after that. Note that the effective bitrate should also take into account retransmissions. We say that a VoIP user is satised with the service quality if there is no delay accumulated due to insufcient scheduling capacity of the cell. We therefore require the ratio between the data available for transmission and the scheduled data to not 1 exceed 1 , where is the maximum ratio of delayed and lost packets with which the VoIP service quality perceived by a user is still satisfactory. The number of scheduled users per TTI is limited by the number of available RBs in the frequency band (e.g., 25 in the 5 MHz band) and the total available power. There may be some other limitations, like the number of available control channels, which impose additional constraints on the network capacity. Furthermore, all users in a cell share the available resources in time and the amount of assigned resources to each user is determined by a scheduler. With a classical round robin (RR) scheduler, active users get equal time shares and in the order dened by their previous transmission time, i.e., the most recently scheduled user has the least chance to be scheduled in the next TTI. Let N be the number of users in the cell and n denote the number of users scheduled at every TTI (or the
1 A resource block is a two-dimensional unit spanning over a predened number of consecutive sub-carriers and a time slot of 0.5 s, i.e., half sub-frame.

number of control channels). Then, the maximum amount of data that can be transmitted to user i during time T is given by T + dmax max , si N n (1)

where dmax is the maximum scheduling delay, and is the TTI length. The user satisfaction requirement thus reads (s1 f1 + s2 f2 ) T N 1 n , max ) smax (T + d 1 i or (f1 + f2 ) T N 1 n , (T + dmax ) 1
smax

(2)

(3)

where = s2 and = is1 . Recall that (3) is dened for s1 a single user (user i), i.e., N is the maximum cell load that the VoIP satisfaction criteria admits for user i. VoIP capacity of the cell can be dened as the maximum cell load (the number of users in the cell) under which at least 95 % of VoIP users are satised. In a scenario where all users run the same trafc mix the capacity is dened as the number of users per cell that satises (3) for the fth percentile of the smax -values of all users in the cell. When the number of allocated RBs is the same for all users in a TTI, the fth percentile of the smax -values corresponds to the fth percentile of the SINR values. Figure 1 shows the relation between , , and N when (3) is a strict equality and the other parameters are set as follows: f1 = 50 fps, f2 = 15 fps, = 0.02, dmax = 0.23 s, n = 4, = 0.001 s, and T = 30 s. For a known trafc mix, we can also compute , e.g., 30 in a scenario with 12.2 kbps VoIP and 110 kbps video. With this setting, the capacity of 15 users per cell can only be achieved if the fth percentile of the -values is at least 1.95, i.e., the worst user out of 95 % best should be able to transmit at every occasion the amount of data equal to 1.95 VoIP packets in average. As previously mentioned, depends on SINR and to some extent, i.e., within a TTI, can also be controlled by the scheduler. Observe from Figure 1 that the absolute capacity gain of such control is much larger for small , i.e., when the trafc loads of both services do not differ too much. Another observation is that a large difference in the amount of generated trafc among the services has a strongly negative impact on the capacity of delay-critical services, e.g., even though the model does not take into account TCP retransmissions, we can expect that TCP trafc, especially coming from a xed network, can easily block VoIP packets and thus result in low VoIP capacity.

200 Number of users per cell 150 100 50 0 0 10 20 30 0 0.5 1 1.5 2 2.5

Fig. 1.

Maximum cell load for a satised VoIP user.

B. With Service Prioritization The model presented in Section III-A assumes a single-queue scenario without trafc differentiation. If both trafc types have the same delay budget, then ideally they have also the same

capacity (in reality, there may be some difference due to, for example, adaptive transmission rate). With multiple bearers it is, however, possible to differentiate among different trafc types and apply different scheduling priorities. Adopting model (3), we assume that if the rst service has a relative priority of > 1 1 over the second then we can disregard of the data trafc from the second service when computing capacity of the rst one, i.e., the amount of trafc to be scheduled is f1 + f2 . This means we increase capacity of the rst service at a cost of some capacity loss of the second service. Note that + models a special case corresponding to absolute priority of the rst service. When computing the capacity of the second service, the amount of trafc to be scheduled is f1 + f2 , where T . The relative capacity gain of the rst service and the relative capacity loss of the second service compared to the BE case can be computed respectively as f1 + f2 f1 + f2 and f1 + f2 . f1 + f2

TABLE I N ETWORK CONFIGURATION AND PARAMETER SETTING Bandwidth Inter-site distance Number of sites / cells per site Antenna type Number of RBs in the band Maximum DL power Simulation time interval DL assignment channels (PDCCH) Channel quality report interval Channel quality estimation error ROHC User mobility VoIP trafc Voice codec Encoding / decoding delay Voice activity factor SID frames during inactivity period Maximum end-to-end delay Maximum delayed and lost frames Related RTP and RTCP Real-time video trafc Video codec Resolution Average bitrate / frame rate Encoding / decoding delay Playout frequency Related RTP and RTCP Mobile TV trafc Video codec Resolution Average bitrate / frame rate Request message size Buffer length Maximum pre-buffering time Maximum total re-buffering time Playout frequency Related RTP and RTCP Web surng trafc HTTP request size Web page size Number of web page downloads 400 Byte 100 kByte 1 H.264, Baseline prole CIF (352 288) 242 kbps / 25 fps 400 Byte 1s 10 s 10 s 60 Hz Yes, on the same bearer H.264, CBR QCIF (176 144) 110 kbps / 15 fps 0.04 s / 0.04 s 60 Hz Yes, on the same bearer AMR 12.2 kbps 0.015 s / 0.005 s 0.5 Yes 0.23 s 2% Yes, on the same bearer 5 MHz 500 m 7/3 Directional, SIMO 25 20 W 30 s 4 0.1 s Log. normal, 3 dB std. dev. Yes 3 km/h in average

Observe that even with small the gain of the rst service capacity can be much larger than the capacity loss of the second service. In our VoIP+video example with = 1.95 for the fth percentile, VoIP priority of = 2.25 allows us to double VoIP capacity at a cost of 11 % video capacity loss. IV. A S IMULATION S TUDY In this section, we present a DL network study conducted by means of simulations for three trafc mixes and compare the observations with our theoretical ndings. An advanced dynamic system-level simulator with detailed protocol implementation has been used to conduct the study for a network with a regular hexagonal layout. The simulator adopts a wrap-around technique to resolve the interference problem in border cells and ensure uniform cell load for homogeneously distributed trafc. Some important parameters are detailed in Table I. A. Scenarios and Scheduling Description Two scenarios have been considered. In the rst scenario, all user trafc goes into the same queue and is further not distinguished by the scheduler. In the second scenario, two queues with different priorities are used for user trafc. The queue with a higher priority is intended for VoIP trafc, whilst user trafc generated by the second service goes into the second queue. The scheduling algorithm in both scenarios is RR, i.e., all users get the same time share for their transmissions and the user with the highest priority is scheduled rst. The servicespecic priorities in the second scenario affect the nal user priority in the way that the maximum of the service priorities among active contexts is added to the user priority. As a result, a user with VoIP data in the queue has a higher priority than the one having only data from the second service, and a user with VoIP data only has the same priority as a user under the same conditions with VoIP data and the other data. In both scenarios, a user with a pending retransmission has the highest priority, which is a xed value not depending on the transmitted data. The maximum number of scheduled users per TTI is limited by the number of control channels. Furthermore, we experiment with two bearer congurations, tcp-optimized and voip-optimized. The latter is only used for VoIP and video bearers in the second scenario. Voip-optimized bearers imply RLC unacknowledged mode (RLC UM), i.e., disable ARQ retransmissions, and allow for maximum 10 and 8 HARQ transmission attempts for DL and UL, respectively. For tcp-optimized bearers, the acknowledged RLC mode (RLC AM)

is used in combination with at most 9 and 7 HARQ transmission attempts for DL and UL, respectively. Also, we use a slightly shorter T1-timer for triggering out-of-order delivery for voipoptimized bearers (0.04 s) compared to tcp-optimized (0.06 s). B. Trafc Models The following three trafc mixes have been considered, mix1: VoIP & real-time video, mix2: VoIP & mobile TV, mix3: VoIP & web surng. One trafc mix per user is assumed in every simulation and it is the same for all users. All users remain in the system throughout the simulation, i.e. 30 s. The initial user distribution is random uniform. A simple handover model is used to support user mobility. Each session comprises two participants, one is always located on the network side (behind the Internet). Packets generated by VoIP, video, and mobile TV services are encapsulated into Real-time Transport Protocol (RTP) packets with 12 Byte headers. Each RTP stream is accompanied by an RTP Control Protocol (RTCP) stream of control packets (RFC3550). When delivered from the network layer, Robust Header Compression (ROHC) is applied in PDCP in order to reduce header overhead. For each header type, a predened ROHC prole is selected: ROHC-RTP (RFC3095), ROHC-UDP (RFC3095), ROHC-TCP (RFC4996), or ROHC-IP (RFC3843).

1) VoIP: A VoIP session starts in the beginning of the simulation with a random offset uniformly sampled from [0 s, 0.2 s]. The conversation between two users is continuous and has an activity factor of 0.5. The talk spurt lengths are drawn from an exponential distribution. During silence periods (discontinuous transmission, DTX), Silence Descriptor (SID) frames are transmitted periodically (every 160 ms) . 2) Real-Time Video: Real-time (RT) video trafc is represented by full duplex conversational video with frame sizes read from a realistic video trace. Each video session starts with a random offset uniformly distributed in [0 s, 0.3 s]. 3) Mobile TV: Unlike with RT video when a client just passively receives frames from a server, mobile TV is a streaming application where the trafc is initiated by a client request transmitted to the server which then starts transmitting a mobile TV stream. The clients have a playout buffer. The initial buffering time in mobile TV should be small to avoid long delays when switching channels. An MTV music clip trace has been used to generate the mobile TV stream. Except for a larger bitrate, the trafc also has a larger variation in frame sizes compared to RT video because of I-frames inserted every 25th frame. A client initiates a session in the beginning of the simulation with a random offset in [0 s, 0.3 s] and buffers frames until the buffer is lled. Rebuffering occurs when the buffer is empty. The session lasts for 30 s, but it can be stopped when either the maximum prebuffering or the total re-buffering times is exceeded. 4) Web Surng: Every user sends a single Hypertext Transfer Protocol (HTTP) request initiated randomly within the rst 25 s of the simulation and downloads a single web page. Note that unlike other discussed trafc types, this is TCP trafc. C. Service Quality and Capacity Metrics There is no standard denitions of user satisfaction for multimedia applications[12]. It is, however, common to measure user satisfaction of, for example, a VoIP user by the number of delayed and lost frames [3]. For conversational video users, using the same metric can still be justied by the requirement of synchronization with the voice service. The approach we adopted uses a model that sets a relationship between bitrate, delay/loss rate, and subjective video quality [8], [9]. Given the minimum required Mean Opinion Score (MOS) and a video bitrate, the model gives us the maximum allowed loss/delay rate. For web users, throughput is the most critical metric. For mobile TV, we adopt the Video Streaming Quality Index (VSQI) model [13], from which the quality of a video stream can be computed by Qf inal = (1QBuf Red )(1QP lRed )(1QBrRed )(Qmax 1)+1, where QBuf Red is the quality reduction factor due to buffering effects, QP lRed is the reduction factor due to packet loss, QBrRed is the quality reduction due to the total bitrate, and Qmax is the maximum quality for QCIF videos. The rst parameter is a polynomial function of the initial buffering percentage relative to the clip length, re-buffering frequency, and total re-buffering time. QP lRed is a power function of packet loss, and QBrRed is an exponential function of the average bitrate. All the tted functions have been parameterized by means of the root mean square error regression for different codecs, including H.264 which is used on our simulations. Below we detail the user satisfaction criteria for the four service types. A user is satised with the VoIP service if the experienced ratio of lost and end-to-end delayed frames is at most 2 %.

A user is satised with the video service if the MOS-score for the video session is at least 3.0. A user is satised with mobile TV streaming if the MOSscore given by the VSQI model is at least 3.0. A user is satised with a TCP download if the experienced throughput is at least 300 kbps. The cell capacity is dened for each service separately as the maximum number of users in the cell among which at least 95 % are satised with service in question. We also compute combined capacity requiring each satised user to be satised with all the services at the same time. Note that all presented results are relevant for relative comparisons.

D. Simulation Results Table II presents results for the single-queue (BE network) and two-queue (QoS network) scenarios. For each scenario, we show per-service capacity and the combined capacity for each trafc mix. As expected, both services in each of the rst two mixes in the BE network have similar capacity the result of a single queue and comparable delay budgets. The small differences are mainly due to variable packet sizes and fast fading. Large I-frames in mix2 further degrade VoIP performance. In mix3, VoIP heavily suffers from web surng, emphasizing the importance of trafc differentiation and QoSaware scheduling. Throughput-based satisfaction criterion for web trafc explains the difference in per-service capacity in mix3. The results for the two-queue scenario show that by prioritizing VoIP trafc we can gain a lot in VoIP capacity at a cost of very small capacity loss of the second service. The VoIP capacity gain is 105 %, 258 %, and 710 % for mix1, mix2, and mix3, respectively, i.e., it increases with trafc intensity of the second service. The combined capacity increases in mix2 and mix3, i.e., only if it has been limited by VoIP in the BE network. In Figure 2 we show user satisfaction as a function of the average cell load in each trafc mix. Per-service user satisfaction is depicted by solid and dashed lines, and the dotted lines denote combined user satisfaction and the 95 % capacity threshold. Per-service capacity is further compared to the maximum achievable, i.e., obtained in a single-service BE network (REF). The reference results are shown in Table III. We note that video and web trafc capacities in Table II are close to the reference results. The gap is larger for mobile TV. VoIP capacity in the QoS network is also much below the reference capacity. The latter is mainly explained by the scheduler design. Recall that if
TABLE II R ELATIVE CELL CAPACITY IN BE AND Q O S NETWORKS Trafc mix mix1 mix2 mix3 Service VoIP Video VoIP Mobile TV VoIP Web surng BE network QoS network

Per-service Combined Per-service Combined 14.56 14.26 6.36 6.61 2.05 6.5 14.25 6.34 2.05 152.85 14.07 164.29 6.53 145.66 6.6 14.07 6.53 6.6

TABLE III R EFERENCE CAPACITY IN SINGLE - SERVICE SCENARIOS IN A BE NETWORK Service VoIP Video Relative capacity 231.0 16.8 Service Mobile TV Web download Relative capacity 10.19 7.43

1 0.98

0.98 Satisfied users share Satisfied users share

0.96 0.94 0.92 0.9 0.88 0.86 0.84 0.82 14.5 15 15.5 0.8 4.5 BE: VoIP BE: Mobile TV BE: VoIP & Mobile TV QoS: VoIP QoS: Mobile TV QoS: VoIP & Mobile TV 5 5.5 6 6.5 Relative cell load 7 7.5 Satisfied users share

0.9

0.96

0.8

0.94 BE: VoIP BE: Video BE: VoIP & Video QoS: VoIP QoS: Video QoS: VoIP & Video 12 12.5 13 13.5 14 Relative cell load

0.7 BE: VoIP BE: Web BE: VoIP & Web QoS: VoIP QoS: Web QoS: VoIP & Web 2 3 4 5 Relative cell load 6 7 8

0.92

0.6

0.9

0.5 1

0.88 11.5

(a) Mix1 Fig. 2.

2250 2000 1750 User throughput, [kbps] 1500
mean 95th percentile

(b) Mix2 Per-service and combined user satisfaction as a function of the average cell load.

(c) Mix3

BE QoS REF

retransmissions, high power and RB utilization in some cells and at the same time small average number of scheduled users (below 1.25 in all scenarios and trafc mixes at the capacity limit of the second service). V. C ONCLUSIONS Service differentiation and prioritization of delay-critical trafc are important not only at high loads but at any load when multiple services are concurrently run at a user terminal which is a likely scenario in LTE networks. This is especially important when a delay-critical service, e.g., VoIP, is in combination with a delay-insensitive intensive trafc, web surng or TCP download. We have shown theoretically and by simulations that prioritization of such a service as VoIP typically does not cause large quality degradation of other services due to small VoIP packet sizes but allows more efcient radio resource utilization. Consequently, in the QoS scenario we could allow more (VoIP) users in the system together with the mixed trafc users. R EFERENCES
[1] 3GPP TS 36-Series, http://www.3gpp.org/. [2] 3GPP TS 23.107, Quality of Service (QoS) concept and architecture. [3] 3GPP TR 26.975, Performance characterization of the Adaptive MultiRate (AMR) speech codec. [4] S. Choi, K. Jun, Y. Shin, S. Kang, B. Choi. MAC scheduling scheme for VoIP trafc service in 3G LTE, in Proc. of the 66th IEEE Vehicular Technology Conference (VTC2007-Fall), Sep. 2007. [5] E. Dahlman, S. Parkvall, J. Sk ld, and P. Beming. 3G Evolution: HSPA o and LTE for Mobile Broadband. Academic Press, 2007. [6] M. Ericson and S. W nstedt. Mixed trafc HSDPA scheduling Impact a on VoIP capacity, in Proc. of the 65th IEEE Vehicular Technology Conference (VTC2007-Spring), pp. 12821286, Apr. 2007. [7] G. G mez, J. Poncela Gonz lez, M. C. Aguayo-Torres, J. F. Paris, J. T. o a Entrambasaguas. QoS modeling for performance evaluation over evolved 3G networks, in Proc. of the 3rd ACM Workshop on QoS and Security for Wireless and Mobile Networks (Q2SWinet), Oct. 2007. [8] J. Gustafsson. Study of multimedia streaming packet losses and interruptions, in preparations for a parametric opinion model for multimedia streaming services, Contr. 29 to ITU-T Study Gr. 12 meeting, Jan. 2007. [9] F. Hultin. Congestion notication and rate-adaptation for real-time services in all-IP radio networks, Masters Thesis, Lule Un. of Technology, 2007. a [10] R. Ludwig, H. Ekstr m, P. Willars, and N. Lundin. An evolved 3GPP o QoS concept, in Proc. of the 63rd IEEE Vehicular Technology Conference (VTC2006-Spring), pp. 388392, May 2006. [11] K. I. Pedersen, T. E. Kolding, and P. Mogensen. QoS Considerations for HSDPA and Performance Results for Different Services, in Proc. of 64th IEEE Vehicular Technology Conference (VTC2006-Fall), Sep. 2006. [12] J. Saliba, A. Beresford, M. Ivanovich, and P. Fitzpatrick. User-perceived quality of service in wireless data networks, Personal and Ubiquitous Computing, 9(6), Nov. 2005. [13] X. Tan, J. Gustafsson, G. Heikkil . Perceived video streaming quality a under initial and rebuffering degradations, in Proc. of the 5th Intl. Conference on Measurement of Audio and Video Quality in Networks (MESAQIN 2006), May 2006. [14] M. Wernersson, S. W nstedt, and P. Synnergren. Effects of QoS schedula ing strategies on performance of mixed services over LTE, in Proc. of the 18th Intl. Symposium on Personal, Indoor and Mobile Radio Communications (PIMRC 07), Sep. 2007.

1250 1000 750 500 250 0 1 3 5 7 9 Relative cell load 11 13

5th percentile

Fig. 3.

TCP throughput in BE and QoS networks vs. REF network.

there are no pending retransmissions, users with pending VoIP packets have absolute priority, which does not mean though that only VoIP trafc is scheduled. Also, resegmentation of transport blocks that need to be retransmitted may result in that some transport blocks do not carry VoIP packets and thus steal resources from VoIP trafc. This is more likely to happen in mix3 which results in the lowest VoIP capacity in the QoS network. Furthermore, for the same network, VoIP capacity is larger in mix2 than in mix1, which is explained by that most of the mobile TV frames (except I-frames) are smaller than those from video and cause less disturbance to prioritized VoIP. For web trafc, we also present the average throughput and the 5th and the 95th percentile throughput as functions of the average cell load in the BE and QoS networks as well as in the REF network. Not surprising, the BE network is the best for users with good channel conditions. These users experience the largest throughput degradation when VoIP is prioritized over other trafc types. The loss ranges from zero to 15 % for some loads. QoS reduces the average throughput by at most 10 % at high loads. Users with bad channel conditions are almost not affected by QoS, except for the lowest load when the QoS network is noticeably better. In general, our observations on the throughput degradation with prioritized VoIP are in line with our conclusions in Section III-B. We have also computed the fth percentile of the user average -values in mix1 and found it comparable to that computed in our example in Section III-A (1.76 vs. 1.95 for 15 users per cell). The ratio between the VoIP capacity gain and the second services capacity loss is signicantly larger than that computed in Section III-B, even though the results are not directly comparable because of scheduling. One reason for the large capacity ratio is the conservative channel quality measurements ltering and link adaptation that limit high-bitrate trafc capacity in the interference-limited environment and also make it less sensitive to prioritization of low-bitrate trafc. Another reason can be seen from the low average number of