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Evaluating Active Buffer Management for HSDPA Multi-flow Service using OPNET Suleiman Y. Yerima and Khalid Al-Begain, Faculty of Advanced Technology

Abstract—High Speed Downlink Packet Access (HSDPA) refers to enhancements to UMTS (Universal Mobile Telecommunications System) cellular networks to provide higher capacity for new packet switched services. End user communication sessions with mixed diverse traffic flows such as voice, video and data are expected to characterise certain multimedia services on HSDPA. Thus, new approaches to network and radio resource management are required to improve operational efficiency and subscriber experience. We propose active buffer management in HSDPA base stations in tandem with conventional resource management as a possible solution for service quality enhancement in end user sessions with such multi-flow traffic. Hence in this paper we study the performance of a proposed active buffer management scheme termed active Time-Space Priority (TSP) by comparing with First Come First Served (FCFS) and the Drop-Tail TimeSpace Priority (DT-TSP) as reference, via simulation on HSDPA network model developed using OPNET modeler. The multi-flow scenario considered is a real-time packet voice call with simultaneous non real-time file download to the same user. The results show that active TSP achieves better throughput for the non-real-time flow whilst guaranteeing lower latency and jitter for the real-time voice traffic compared to FCFS. The outcome of the experiment underscores the significance of active HSDPA base station buffer management to end user multi-flow traffic performance. Index Terms— Active buffer management, High Speed Downlink Packet Access, OPNET, Time-Space Priority

I. INTRODUCTION

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N the past few years, UMTS cellular networks have been deployed on a large scale. UMTS was designed to support a variety of services with peak data rates of up to 2 Mb/s in indoor or small-cell outdoor environments and wide-area coverage rates of up to 384 kb/s. In 2005, technical improvements to UMTS radio access network (UTRAN) downlink, collectively termed High Speed High Speed Downlink Packet Access (HSDPA) were introduced to support growing demand for broadband services. HSDPA provides peak data rates of up to 14 Mb/s, lower transmission latency, improved Quality of Service, and enhanced spectral efficiency for UMTS downlink traffic [1].

Manuscript received February 21, 2008. This work was supported in part by the ORS Award Scheme and the Faculty of Advanced Tech. Studentship. S. Y. Yerima is with the Faculty of Advanced Technology, Department of Computing and Mathematical Sciences, University of Glamorgan, Pontypridd, Mid Glamorgan, CF37 1DL, United Kingdom, Phone: +44 (0)1443 48 3612; e-mail: syerima@ glam.ac.uk). K. Al-Begain, is the Head of Integrated Communications Research Centre (ICRC), University of Glamorgan . (e-mail: kbegain@ glam.ac.uk).

HSDPA was initially envisaged to support asymmetric data services such as internet browsing. But with the growing popularity of packet voice and multimedia services, coupled with the availability of advanced receivers, sessions with multiple flows of for example voice, video, and data to a single end user is a possible scenario on HSDPA. Regardless of traffic profile, the subscriber’s main concern is good quality service. On the other hand operators want to achieve efficient network resource utilization. Meeting both requirements impose additional challenges on network and radio resource management to support multi-flow services. In HSDPA, Packet Scheduling (PS) is a new radio resource management functionality included in the base station (Node B). Its function is to allocate transmission slots to users in a HSDPA cell according to some specified algorithm. Numerous schemes have been proposed in the literature, some of which are adaptations of well known scheduling algorithms. Majority of HSDPA PS algorithms are designed to take into account inter-user service class differentiation in Quality of Service (QoS) based scheduling. Some, however, like the solution proposed in [2], take into account intra-user traffic differentiation and are applicable to scheduling end users with multi-flow traffic. Packet scheduling in the Node B necessitates buffering of data packets. This provides opportunity to apply buffer management, which is crucial to multi-flow sessions because the diverse flows usually have different QoS requirements. Majority of existing HSDPA Packet Schedulers are designed for single flow traffic per user. Different from the work in [2], ours focuses on integrating active buffer management into HSDPA network and radio resource management, to improve intra-user multi-flow QoS. We propose an active buffer management scheme based on the Time-Space Priority (TSP) queuing as a solution for HSDPA intra-user multi-flow sessions. The advantage of TSP is that it provides not only intra-user flow differentiation, but also prioritized queuing with delay and loss differentiation. This allows for joint optimization of the QoS requirements of real-time and non-real-time flows. In this paper the impact of active buffer management on HSDPA multi-flow sessions is evaluated by comparative analysis of the active TSP scheme with non buffer-managed First Come First Serve (FCFS) queuing, and also with Droptail (DT-TSP), a non active TSP scheme. The experiments were performed on a HSDPA simulation model developed using OPNET. The multi-flow scenario considered was a real-time packet voice call with simultaneous non real-time file download to an end user. The organization of the rest of the paper is as follows.

2 Section II describes HSDPA , while Section III explains the compared schemes. Modelling of the test scenarios is detailed in Section IV, while results are shown and discussed in Section V. Lastly, conclusions are drawn in Section VI.

air interface to the receiver is fast because a short Transmission Time Interval (TTI) of 2ms is used instead of 10ms used in UMTS for RNC packet scheduling.

II. BASIC FEATURES OF HSDPA

Several buffer management schemes appear in the literature [ 3] but not all are applicable to multi-flow traffic in HSDPA. For our purposes, we can classify buffer management into QoS enabled and non QoS enabled in terms of queuing discipline and also into active and nonactive in terms of terms input traffic regulation capability.

III. BUFFER MANAGEMENT SCHEMES HSDPA utilizes a shared channel (HS-DSCH) to transmit data to the User Equipments (UE) over the downlink of a HSDPA enhanced UMTS cell. It provides better end-user experience, with shorter connection and response times. In addition, three to five fold sector throughput can be achieved, resulting in significantly more data users in a cell. A simplified UMTS/HSDPA architecture is shown in Fig. 1. It consists of three interacting domains; Core Network (CN), UMTS Terrestrial Radio Access Network (UTRAN) and the UE i.e. the receiver. The Core Network is responsible for switching, transit and routing of user traffic. UTRAN provides the air interface access for the receiver and handles all radio related functionalities. UTRAN consists of a Radio Network Controller (RNC) and base station or Node B. The main features of HSDPA are Adaptive Modulation and Coding (AMC), Fast Hybrid-ARQ, and fast Packet Scheduling. These features all reside within the Node B. User Equipments

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Fig. 1. Simplified diagram of HSDPA enabled UMTS Radio Access Network (UTRAN) topology showing the main entities

A. Adaptive Modulation and Coding The basic principle of AMC is to change the modulation and coding scheme for data transmission in accordance with the variations in channel conditions of the UE. The basic unit of data transmission is known as a Transport Block and the AMC scheme selected corresponds to a Transport Format. The Transport Block Size varies with the selected AMC scheme. When receiver has good channel conditions, AMC allows for larger Transport Block Size. Channel conditions are estimated by the receiver using a Channel Quality Indicator which is feedback via the uplink control channel. AMC uses the Channel Quality Indicator for Transport Format selection. B. Hybrid Automatic Repeat Request(HARQ) The HARQ entity in HSDPA is responsible for retransmission of data that the receiver is unable to decode due to transmission errors. HARQ operates in the physical layer of the Node B allowing for fast re-transmission of erroneous packets. C. Packet Scheduling This is a radio resource sharing functionality performed in the Node B. Scheduling of data packet transmission over the

A. The First Come First Serve (FCFS) Scheme FCFS is the queuing of packets in the buffer queue as they arrive. Access to transmission is scheduled by the Packet Scheduler on a first come first serve basis. With the multiple flows FCFS cannot prioritize transmission. We choose this as a reference non-QoS enabled scheme for comparison. B. The Time Space Priority(TSP) Scheme The Time-Space priority buffer management scheme is a QoS enabled, priority queuing that combines time priority and space priority schemes with a (fixed or variable) threshold to control the QoS parameters (loss, delay, and jitter) of diverse flows within a ‘multimedia’ stream. The basic idea of Time-Space Priority concept [4] is that realtime flows, such as Video or Voice packets, are given transmission priority because of their stringent delay requirements; while non real-time flows, such as email or file downloads, have buffer space priority to minimize loss. Thus unlike many priority queuing schemes, TSP provides both delay and loss differentiation. This concept is illustrated in Fig. 2. Arriving real-time packets are queued in front of the non real-time packets for priority transmission on the shared channel. The number of real-time packets admitted into the buffer queue is restricted to take advantage of loss tolerance so that more buffer space is allocated to the non-real time flow to minimize the dropping of non real time packets due to full buffer. The non-active TSP scheme drops arriving non-real time packets at the tail of the queue when the queue is full and is thus called Drop-Tail TSP (DT-TSP). DT-TSP is QoS enabled but non-active buffer management scheme because of lack of control over arriving packet rates. VoIP + Data Source

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Fig. 2. HSDPA UTRAN model with Time-Space Priority buffer management scheme.

C. The Active Time Space Priority(TSP) Scheme This is an extension of the TSP scheme with active queue management to control the arrival of packets to the base station buffer. A possible extension uses two additional thresholds as shown in Fig. 3. The additional thresholds L and H are used to manage the queue size and prevent buffer

3 overflow by regulating the data units’ arrival rate. The scheme, which shall be referred to as active TSP in this paper, is illustrated in Fig. 3. Non-real time packet arrivals is controlled by L and H. Real-time data units still receive precedence queuing like in TSP with unregulated arrival in HSDPA Unacknowledged Mode. In this scheme, feedback to the RNC is via NBAP signalling [5] allocating grants to RNC to control packet unit arrival rate. When the average queue occupancy exceeds L, the rate is reduced by a factor k, but when it exceeds H, zero grants are allocated. The space between H and N absorbs bursty instantaneous arrivals. In all our experiments k is set to 0.5. Radio Network Controller

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Radio link simulation included path loss and shadowing models with transmit powers and AMC schemes setting as assumed in [8]. Number of H-SDSCH codes is assumed to be 5, while CQI feedback latency was set to 6ms. Four HARQ processes were used in the HARQ manager, while round robin scheduling was employed in the packet scheduler. Other parameters include, external network + Core Network delay =70ms, RNC to Node B propagation delay = 20ms, MAC-d protocol data unit (PDU) size = 320 bits. MAC-d PDU is obtained from IP packet segmentation in the Radio Network Controller. Buffer scheme parameters include: R=10 PDU, N = 150 PDU for DT-TSP; R=10 PDU, L=30 PDU, H=70 PDU, N = 150 PDU for active TSP; while N=150 PDU for FCFS. Performance metrics include: (a) UTRAN VoIP (PDU) delay: defined as the delay experienced in seconds by the VoIP PDU within the UTRAN i.e. RNC queuing delay + Iub latency + Node B queuing delay. (b) End-to-End FTP throughput: defined as the number of bits received per second in the receiver for the non-real-time file download using FTP over TCP.

Fig. 3. HSDPA UTRAN model with Active Time-Space Priority buffer management scheme.

V. RESULTS AND DISCUSSION OF RESULTS

IV. HSDPA MODELLING AND SIMULATION

Figures 4 to 6 show results of the end-to-end FTP throughput measurements for a test receiver 120s multi-flow session for each scheme. The instantaneous throughput is plotted against the session time for 1, 5, 10, 20 and 30 users.

This section briefly describes the OPNET HSDPA model used for the investigation. It also outlines the experimental parameters and measured performance metrics.

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B. Simulation Parameters and performance metrics In the experiments a test user equipment was connected to the HSDPA UTRAN through which multi-flow traffic was received in a simulated 120s voice conversation and file download session. VoIP packets were being received while file download was taking place using FTP over TCP. The set up represents a single HSDPA cell.

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A. HSDPA OPNET Model OPNET [6] is a powerful tool for modelling communication networks and distributed systems. It allows for the performance analyses of modelled systems via simulation. More importantly, it incorporates GUI tools for all phases of a study including model design, development, data collection and data analysis. For this study, a custom HSDPA simulation model library was designed and implemented. This was because current OPNET version 14.0 does not include a HSDPA model library. Our OPNET HSDPA library includes a multi-flow traffic source node, a UTRAN node model, a receiver node model, bidirectional link models and Packet formats. The multi-flow source node includes a full implementation of VoIP ON/OFF source with same parameters from [7], and a customizable FTP source with TCP Reno implementation. The multi-flow node transmits VoIP and FTP packets via the UTRAN node to the receiver node which has peer TCP module where end to end throughput can be calculated. The UTRAN node includes RNC and Node B modules. In the Node B module MAC-hs queues, AMC, HARQ, and Packet Scheduling are all implemented. The receiver node contains modules implementing HARQ, packet re-assembly queues and TCP.

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Fig. 5. End-to-end Throughput at test UE for DT-TSP Vs session time. Instantaneous throughput with 5, 10, 20 and 30 users are also shown.

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Fig. 7. Average end-to-end Throughput at test UE for Vs number of users in cell. Results given for FCFS, DT-TSP and active TSP.

For the test receiver under observation in the simulated HSDPA cell, The impact of FCFS scheme on the FTP throughput is shown in Fig. 4. The test UE is assumed to be located 200m away from the base station. With only the test UE in the cell, FTP throughput is observed to peak to almost 240 kbps and remains steady. When additional users share the channel significant drops in throughput measured in the test UE is observed. Also, with multiple users, rather than a steady peak throughput, we observe variation of the throughput of the test UE. Two phenomena in the base station buffer account for this. Increased waiting times of the non-real-time FTP Protocol Data Units (PDU) in the FCFS queue and the loss of FTP PDUs due to buffer overflow. Buffer overflow leads to retransmission from the RNC. Both events increases the FTP packets’ Round Trip Time consequently degrading TCP performance resulting in lower and unsteady end-toend throughput. The same experiment is repeated with DT-TSP implementation in the base station buffer and the results are shown in Fig. 5. Test UE throughput degradation with more users also occurs. Recall that DT-TSP applies priority queuing by allowing VoIP packets, if present, to receive priority transmission. The space priority mechanism allows more space for non-real-time PDUs to minimise loss but with the drop tail (DT) policy when the buffer overflows, PDU losses still occur, leading to retransmissions. The PDU losses coupled with the delay due to lower priority queuing

and access to transmission adversely affect TCP performance. which is why for 5, 10 and 20 users we observe that FCFS gives better FTP throughput. DT-TSP PDU dropping rate can be improved significantly with active queue management providing rate control. To study the impact of this on the end-to-end FTP throughput especially with multiple user scheduling, we repeated the experiment replacing DT-TSP scheme with the active TSP scheme. The results are depicted in Fig. 6. Note that with only the test UE in the cell the observed throughput was lower than that of FCFS or DT-TSP but with 5 users we notice slightly better throughput. From Fig. 7 we see that with 10 users active TSP enables 70% increase in average throughput compared to FCFS and 170% increase compared to DT-TSP. Similarly, with 20 users almost similar average throughput performance gain is achieved. Notice also from Fig. 6 that instantaneous throughput peaks and stays almost steady indicating much better TCP performance. This is also consistent with the observation that no non-real-time PDU losses occurred in the base station due to buffer overflow during all the experiments with the active TSP. Next we discuss the VoIP performance in the multi-flow session for the three schemes. Figure 8 shows FCFS results. As number of users increase, UTRAN PDU delay and the delay variation (jitter) in the test UE increases. Notice that the minimum UTRAN delay is 20ms which is the RNC to Base Station propagation delay assumed in the simulation. Clearly, the Packet Scheduling delay increases with number of users. The UTRAN delay is the sum of the RNCbase station propagation delay (20ms) and the queuing delay in the base station. Hence the Packet scheduler delay is equal to the queuing delay in the base station. This means that peak scheduler delays can be estimated from the graphs. For example, for 5 users with FCFS, peak scheduler delay estimate is 130ms (150ms – 20ms). Similarly for 10 users peak scheduler delay is approximately 270ms. Assuming that a Packet Scheduler delay budget of 80ms to 150 ms is acceptable [7], for VoIP end-to-end QoS (of 250ms one way delay) to be guaranteed. It becomes clear that FCFS will be unable to support the VoIP QoS requirements of the test UE with up to 5 users present in the cell under the given scenario because most VoIP PDU delays are close to violating the packet scheduling deadline. With 10 users it is even worse since the 270 ms scheduling delay is already above the 250 ms end-to-end QoS constrain. A discard timer can be used to drop VoIP PDUs likely to exceed the scheduling delay budget, but the adverse effect is increased loss of VoIP PDUs which would degrade voice quality. Moreover, observe from Fig. 10 that the average UTRAN PDU delay for 20 and 30 users using FCFS is about 120ms and 180ms respectively which will render a discard time useless. DT-TSP and active TSP show identical VoIP performance in all experiments so only one of the graphs is shown. From Fig. 9, minimum UTRAN delay is also 20ms i.e. equal to the RNC to Node B propagation delay. Also, with only the test UE in the cell, the maximum UTRAN delay is 22ms. This is because the maximum scheduling delay is equal to the HSDPA Transmission Time Interval of 2ms since only one user is being scheduled. When more users are added to the cell the scheduling delay increases but for all the cases observed, the peak UTRAN delay was within 60ms.

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Fig. 8. FCFS VoIP UTRAN delay in test UE during multi-flow session.

download session is simulated using OPNET modelling of detailed UTRAN functionality. Comparison between DTTSP, FCFS and the active buffer managed TSP is made. The VoIP delay in the UTRAN and end-to-end- FTP throughput is observed for a test receiver including cases where multiple users share the HSDPA channel with round robin scheduling. The experiments shows that packet losses due to base station overflow degrades TCP throughput and this can be mitigated with active an active buffer managed scheme like active TSP. Furthermore, with stringent end-to-end delay bound specified for real-time VoIP, FCFS is not a suitable option for multi-flow traffic with real-time flow despite its better non-real-time FTP throughput over DT-TSP. From the study we conclude that for multi-flow sessions with real-time and non-real-time flows, active buffer management with a scheme like TSP that ensures minimal non-real-time packet dropping while keeping real-time delay within acceptable bounds, is crucial to end-to-end QoS performance. Further work will extend the simulation to include advanced Packet Schedulers to study the impact on active TSP performance. ACKNOWLEDGEMENT The authors would like to thank OPNET for modelling tool support through their OPNET University Program. REFERENCES [1] [2]

Fig. 9. DT-TSP VoIP UTRAN delay in test UE during multi-flow session. Very similar results are obtained for active TSP in all cases. [3]

Also, from Fig. 10, all average UTRAN delays are below 60ms. This means that VoIP QoS can be satisfied for the test UE in all cases because the observed peak and average delays are well within the given packet scheduler’s delay budget. Thus, discard timer will not be required for DT-TSP and active TSP in the investigated scenarios. VI. CONCLUDING REMARKS This paper examines the impact of active base station buffer management in HSDPA for scenarios where multiple diverse flows exist for a single end user. A VoIP and data

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