Performance Analysis of Scheduling Algorithms

Performance Analysis of Scheduling Algorithms for VoIP Services in IEEE 82.1 6e Systems Howon Lee*, Taesoo Kwon*, Dong-Ho Cho*, Geunhwi Limt and Yong Changt *Department of Electrical Engineering and Computer Science Korea Advanced Institute of Science and Technology (KAIST), Korea Email: {hwlee, tskwon8o} @comis.kaist.ac.kr, dhcho @ee.kaist.ac.kr ttelecommunication Systems Division, Samsung Electronics, Co., Ltd., Korea Email: {geunhwi.lim, yongchang} @ samsung.com Abstract- There are several scheduling algorithms for Voice over IP (VoIP) services in IEEE 82.16e systems, such as unsolicited grant service (), real-time polling service (), and extended real-time polling service (e). The e is a new scheduling algorithm for VoIP services with variable data rates and silence suppression, and this algorithm is recently proposed and accepted in the IEEE 82.16e standard. In this paper, we analyze and discuss the performance of the scheduling algorithms recommended in IEEE 82.16e systems including the e algorithm. Through the analysis of resource utilization efficiency and VoIP capacity, we show that the and algorithms have some problems, which are the waste of uplink resources in the algorithm, and additional access delay and MAC overhead due to bandwidth request process in the algorithm, to support the VoIP services. In addition, for analysis of VoIP capacity, we utilize OPNET simulation, and show that the e algorithm can support more 21% and 35% voice users compared with the and algorithms, respectively. I. INTRODUCTION T HE IEEE 82.16 standard is designed to satisfy various demands for higher capacity, higher data rate, and more advanced multimedia services to residential and small business customers [1], [2], [3], [4]. This standard has many advantages, such as rapid deployment, high speed data rate, high scalability, multimedia services, and lower maintenance and upgrade costs. Especially, the IEEE 82.16's Task Group (TG) d/e systems have proposed to make up for the weak points of 16a system, and support additional functionalities, such as mobility, Hybrid Automatic Repeat Request (HARQ), Band Adaptive Modulation Coding (AMC) scheme, and so on. In order to support multimedia services with variable requirements of qualify of service (QoS) in IEEE 82.16e systems, an efficient scheduling algorithm has to be provided. Particularly, an efficient uplink (UL) scheduling algorithm for voice services is required because voice services are delaysensitive and have an important portion in the multimedia services. As a leading technology for voice services in a packet oriented architecture, VoIP technology has been intensively investigated. There are five scheduling algorithms to support variable requirements of QoS in IEEE 82.16e systems, such as unsolicited grant service (), real-time polling service (), extended real-time polling service (e), non-realtime polling service (n), and best effort service (BE). The, and e algorithms are designed to support real-time services. So, these algorithms could be used for VoIP services. However, because the n and BE algorithms are designed to support non-realtime services, these algorithms are not suitable for VoIP services in IEEE 82.16e systems. Although IEEE 82.16a/d systems have the and algorithms for VoIP services, these algorithms are not proper as an uplink scheduling algorithm for VoIP services with variable data rates and silence suppression [5]. That is, these algorithms have some problems for the VoIP services, such as the waste of uplink resources in the algorithm and MAC overhead as well as additional access delay in the algorithm. Therefore, for efficient VoIP services, e is proposed and accepted as a new scheduling algorithm in IEEE 82.16e systems. In this paper, we analyze and discuss the performance of these scheduling algorithms (, and e) in IEEE 82.16e systems. TABLE I VOICE MODEL PARAMETERS L1 (Rate 1) 171 bits Packet size LI/2 (Rate 1/2) LI/4 (Rate 1/4) 8 bits 4 bits LI/8 (Rate 1/8) 16 bits In this paper, for performance analysis of scheduling algorithms in IEEE 82.16e systems, we assume that voice traffic follows Markov source model [6]. Definitely, we utilize Enhanced Variable Rate Codec (EVRC) with variable data rates and silence suppression. The frame duration of EVRC (TVC) is 2rms, and the voice activity factor is.43 with 29% full rate (Rate 1, pi), 4% half rate (Rate 1/2, Pi/2), 7% quarter rate (Rate 1/4, Pi/4), and 6% eighth rate (Rate 1/8, P1i8). Full/half/quarter rates are included in talk-spurt duration, and eighth rate is included in silence duration. Detailed voice model parameters are shown in Table I. The remainder of this paper is organized as follows: In Section II, we introduce and discuss several scheduling algorithms for VoIP services in IEEE 82.16e systems, such as,, and e. In Section III, we analyze resource utilization efficiency of these scheduling algorithms, and show the number of saved radio resources in the e algorithm compared -783-9392-9/6/$2. (c) 26 IEEE 1231

Dot line: Assigned resources Solid line : Used resources Rate 1 Dot line: Assigned resources Solid line: Used resources Rate 1 Rtate 112 Rate 112 * * Rate 114 Rate 1/4 lrate 1/8 Rate 118 Time OFF Data rate inerement Data rate OFF decrement Data rate increment Data rate deerement Fig. 1. Operation of algorithm Fig. 2. with the and algorithms. Also, in Section IV, we analyze VoIP capacity based on the packet transmission delay of these algorithms in IEEE 82.16e systems through OPNET simulation. Finally, in Section V, we make conclusions. data more efficiently than the algorithm. However, this bandwidth request process always causes MAC overhead and additional access delay. Hence, the algorithm has larger MAC overhead and access delay than the and e algorithms. In Fig. 2, the dot line and the solid line show the amount of assigned uplink resources by the BS and the amount of used uplink resources by the voice user, respectively. In this algorithm, since the user requests exact amount of uplink resources for transmitting his voice packet, the dot line and the solid line are nearly the same. To avoid the polling process in case of silence duration, we assume that minimum polling size is the size of voice packet generated by the minimum data rate of the voice codec. Thus, there is no polling process in the silence duration of the users. In the algorithm, the user can use the piggyback requests of the grant management subheader for VoIP services. But, since VoIP services are delay-sensitive and the usage of piggyback requests is not proper method for supporting service-flows that have periodic property, the use of piggyback requests is not a desirable method for VoIP services. By precise negotiation of the polling period in the initialization process, the use of piggyback requests may be avoided for VoIP services. SCHEDULING ALGORITHMS FOR VoIP SERVICES IN IEEE 82.16E SYSTEMS A. Algorithm The algorithm is designed to support real-time service flows that generate fixed-size data packets periodically [3]. The base station (BS) periodically assigns fixed-size grants to the voice user. These fixed-size grants are sufficient to send voice data packets. The grant size and grant period are negotiated in the initialization process of the voice session. Thus, this algorithm can minimize MAC overhead and uplink access delay caused by the bandwidth request process of the user to send voice packets. However, this algorithm has only a small capacity for VoIP services. Generally, voice users do not always have voice packets to send, because they have periods of silence [7]. In this algorithm, since the BS always assigns fixed-size grants that are sufficient to send voice packets to the voice user, it causes a waste of uplink resources, as shown in Fig. 1. In this figure, a dot line and a solid line show the amount of assigned uplink resources by the BS and the amount of used uplink resources by the user, respectively. The blank regions represent the waste of uplink resources. In summary, in case of the algorithm, since the BS always allocates the same amount of uplink resources to each user regardless of his voice status, it causes the waste of uplink resources. II. B. Algorithm The algorithm is designed to support real-time service flows that generate variable size data packets periodically [3]. The BS assigns uplink resources that are sufficient for unicast bandwidth requests to the voice user. This period is negotiated in the initialization process of the voice sessions. Generally, this process is called a bandwidth request process, or polling process. Because this algorithm always uses a bandwidth request process for suitable size grants, it transports Operation of algorithm C. e Algorithm The e algorithm is designed to support real-time service flows that generate variable size data packets on a periodic basis, such as VoIP services with silence suppression [4]. This algorithm is recently proposed and accepted in IEEE 82.16e standard. In order not only to reduce MAC overhead and access delay of the algorithm, but also to prevent the waste of uplink resources of the algorithm, the e algorithm assigns uplink resources according to the status of the voice users without MAC overhead. Here, we describe the detailed operation of the e algorithm as follows. Firstly, the voice user informs the BS of his voice status information using Grant Management subheader in case that the size of a voice data packet is decreased [4]. The user requests the bandwidth for sending the voice packets using 1232

Bandwidth request header Dot linte: Assigned resources Solid line: Used resources Rate I by the minimum data rate of the voice codec. Hence, there is also no polling process in the silence duration of the voice users. Grant Management subheader Rate 11 Time 1F Data rate increment Fig. 3. Data rate decrement Operation of e algorithm III. ANALYSIS OF RESOURCE UTILIZATION EFFICIENCY A. algorithm By using voice codec parameters in Table I and the size of generic MAC header (Lh, 6bytes) of IEEE 82.16e systems [4], the average amount of assigned uplink resources of the algorithm (R,9,gs) can be calculated by R = (Li + Lh) x (Pt + P1/2 + P1/4 + P1/8) 219 bits fr./user. (1) In this algorithm, the BS assigns R (219 bitslframe) to each voice user. In this paper, 'fr.' and 'frame' mean MAC frame in IEEE 82.16e systems. Since the BS always assigns fixed-size grants according to maximum data rate of EVRC to the voice users regardless of their voice status, the algorithm always allocates relatively a lot of uplink resources to the users compared with the and e algorithms. extended PBR (PiggyBack Request) bits of Grant Management subheader. In the e algorithm, to distinguish these extended PBR bits with general PBR bits, the user sets the MSB of PBR bits to 1. In this case, the BS assigns uplink resources according to the requested size periodically, until B. algorithm the voice user requests another size of the bandwidth. The average amount of assigned uplink resources of the Secondly, the voice user informs the BS of his voice status information using Bandwidth request header in case algorithm (R,tps) can be obtained as that the size of a voice data packet is increased [4]. The user Rrtps {(Ll + Lh) x p} + {(LI/2 + Lh) X P/2} requests the bandwidth for sending the voice packets using + {(Ll/4 + Lh)x P1/4} + {(Ll/8 + Lh)Ox Pl/} BR (Bandwidth Request) bits of Bandwidth request header. In the same way as the case of the data rate decrement, to + {(L1/8 + Lbh) x (P1 + P1/2 + Pi/4)} distinguish these BR bits with general BR bits, the user sets = 138.79 bits/fr./user. (2) the MSB of BR bits to 1. In this case, the BS assigns uplink resources according to the requested size periodically, until In equation (2), Lbh is the size of bandwidth request header the user requests another size of the bandwidth. The BS shall in IEEE 82.16e systems, 6 bytes [4]. In case of the provide the first bandwidth allocation to the next MAC frame algorithm, we assume a minimum polling size (L1/8 + Lbh) after this bandwidth request process. The second bandwidth as the size of voice packet generated by the minimum data allocation is done after the bandwidth allocation interval of rate of the voice codec for the users. However, if we assume the service flow based on the time which the BS allocated the minimum polling size as the size of the bandwidth request the bandwidth that used for the bandwidth request process, as header (Lbh), the polling process would be also executed in the silence duration of the voice user. Then, it could cause a waste shown in Fig. 3. In summary, in case of the VoIP services using this algo- of uplink resources and additional access delay in the silence rithm, the BS recognizes Grant Management subheader and duration of the user. So, it is not preferable. Consequently, in Bandwidth request header especially. In this algorithm, if the this algorithm, the BS assigns R,tps (138.79 bitslframe) to user requests the bandwidth for sending the voice packets, then each voice user. the BS shall change its polling size according to the bandwidth size requested by the user, and keeps its changed polling size C. e algorithm until the user sends another requests. In other words, the BS Similar to the and algorithms, the average amount may not change its polling size without any requests from the of assigned uplink resources of the e algorithm (R,,tps) voice users. Using this e algorithm, the BS can obtain can be represented as better data transport efficiency compared with the and algorithms. Rertps = {(L1 + Lh) x Pi} + {(L1/2 + Lh) X Pi/2} Fig. 3 shows the operation of the e algorithm when the + {(Ll/4 + Lh) x Pl/4} + {(Ll/8 + Lh) x Pl/8} voice user uses EVRC with variable data rates and silence = 113.19 bitslfr./user. (3) suppression [6]. In case that the voice data rate is decreased, since the user uses remained uplink resources assigned to In the e algorithm, the BS assigns Rertps (113.19 him for transmitting Grant Management subheader, there is no bits f rame) to each voice user. Contrary to the and waste of uplink resources. In this algorithm, we also assume algorithms, since the e algorithm can follow all data rates that minimum polling size is the size of voice packet generated of each user without MAC overhead, this algorithm has the 1233

TABLE II AVERAGE AMOUNT OF ASSIGNED UPLINK RESOURCES 3 - Ave. amount of assigned uplink res. / fr. / user 219 bitslfr.luser 138.79 bits1fr.1user 113.19 bits1fr.1user e, 25- W Number of saved uplink resources W Number of saved downlink resources rips a) n 25 CO 15 I smallest value (R,,tps) among these scheduling algorithms in IEEE 82.16e systems. We can show the average amount of assigned uplink resources of each algorithm in Table II. Also, in Table III, we can obtain the average amount of downlink (DL) and uplink saved resources in the e algorithm against the and algorithms. For example, when N users use VoIP services in one MAC frame, the e algorithm can save {(R - Rertps) x N} bits and {(Rrtps - Rertps) x N} bits of uplink resources compared with the and algorithms, respectively. When N = 2, the e algorithm can save 2116.2 bits (264.525 bytes) and 512 bits (64 bytes) of uplink resources, respectively. Besides, compared with the algorithm, the and e algorithms do not experience polling process for bandwidth requests of the voice users in talk-spurt (on) duration. Hence, these algorithms can save downlink resources for sending UL-MAP messages. The general size of UL-MAP message is 36 bits (4.5 bytes) in IEEE 82.16e systems [4]. When M users use only VoIP services in one MAC frame, the and e algorithms can save (36 x M) bits of downlink resources compared with the algorithm. If M = 2, these algorithms can save 72 bits (9 bytes) of downlink resources. Especially, since UL-MAP messages use very robust burst profile (QPSK modulation and 1/12 coding) compared with general data packets, a lot of downlink resources would be saved in the and e E 5 U. n 1 2 3 4 Number of VolP users Fig. 4. Total number of saved resources in e algorithm compared with and algorithms vs. number of VoIP users are transmitted by QPSK 1/2, and QPSK 1/12, respectively. Through this figure, we can show that a lot of downlink resources are wasted in the algorithm. Although the number of assigned uplink resources in the algorithm is larger than that of the algorithm, since a great number of downlink resources is wasted in the algorithm for transmitting of UL-MAP messages, the total number of wasted resources in the algorithm is larger than that of the algorithm. In case of the e algorithm, the wastes of downlink and uplink resources in the and algorithms do not occur. Therefore, the e algorithm can save a lot of uplink and downlink resources compared with the and algorithms, and these saved resources would be utilized for other unicast, multicast and broadcast services. algorithms. IV. ANALYSIS OF VoIP CAPACITY Fig. 4 shows the total number of saved resources in the erps algorithm compared with the and algorithms. Generally, VoIP capacity is restricted by packet transmission To obtain this figure, we assume an OFDMA system, and delay bound and radio resource saturation. In this paper, we define that one basic resource unit consists of 48 subcarriers. analyze the VoIP capacity of IEEE 82.16e systems with Also, we assume that voice packets and UL-MAP messages packet transmission delay of MAC SDUs (Service Data Units) for the,, and e algorithms. To obtain simulation results, we used an OPNET simulator TABLE III and assumed an OFDMA system. This system has various AVERAGE AMOUNT OF SAVED UPLINK RESOURCES IN THE ERTPS MCS levels, such as QPSK, 1/12, QPSK 1/8, QPSK 1/4, QPSK ALGORITHM COMPARED WITH THE AND RTPS ALGORITHMS 1/2, 16QAM 1/2, 16QAM 3/4, 64QAM 2/3, and 64QAM 5/6. To solve intercell interference problem, that is a major Ave. amount of saved UL res. / fr. / user] of all OFDMA systems, for cell-boundary voice problem Comp. Alg. no. users Downlink (bits/fr.) Uplink (bits/fr.) we utilized repetition scheme in case of 1/4, 1/8 and users, - 1 158.1 1/12 coding. By using an average SINR value of each voice - 2 2116.2 the BS decides the MCS level of each user. The voice user, - 3 3174.3 users are uniformly distributed in seven cells, and experience - 4 4232.4 interference from 1st-tier and 2nd-tier cells. In addition, we - 1 36 256 used an ITU-R pedestrian model for path-loss and frequency 512-2 72 selective Rayleigh fading. We assume that a MAC frame of - 3 18 768 IEEE 82.16e OFDMA systems consists of 36 symbols (Time 4 144 124 Domain) and 124 subcarriers (Frequency Domain), and the - 1234

O CD cn CU vo o U E U, c O cl.3.28.26.24.22.2.18.16.14.12.1.8.6.4.2. - -*-DL Delay -*-UL Delay -A- UL Delay -v- UL e Delay l e U(-S / rtps *@./p --- -- 2 4 8' 6 Number of VolP users Fig. 5. Packet transmission delay of MAC SDUs vs. '-1- number VoIP users r ratio of downlink and uplink subframe length is 2:1. Thus, total numbers of downlink and uplink resources are 384 and 14, respectively. One resource unit consists of 48 subcarriers and downlink and uplink resources are scheduled by roundrobin (RR) scheduler. In this simulation, if the number of voice users is increased, since the number of VoIP packets would be increased, the interference level of all users is augmented. Then, the average SINR values of the voice users are lowered and the voice users would use more robust MCS level that used before. Hence, the saturation of downlink and uplink resources occurs, and the transmission delay of voice packets is rapidly increased. In general, the maximum end-to-end delay bound of voice packet is assumed to 285ms by ITU-T [8]. Thus, in this paper, we assume that the delay bound for packet transmission of MAC SDUs is 6ms considering backbone delay, packet processing delay, handset playback buffer delay, and so on. As shown in Fig. 5, the maximum supportable numbers of voice users for the, and e algorithms are 68, 76 and 92, respectively. Since the e algorithm utilizes downlink and uplink resources more efficiently than any other algorithm in IEEE 82.16e systems, the packet transmission delay is the most slowly increased among these scheduling algorithms. In case of the algorithm, although the waste of uplink resources is larger than that of the algorithm, since the scheduler aggregates the voice packets within the inter-grant time, the VoIP capacity of the algorithm adaptively increases with system load. So, the VoIP capacity of the algorithm is larger than that of the algorithm. In summary, the e algorithm can support more 21% and 35% voice users compared with the and algorithms, respectively. We can show that the e algorithm has largest VoIP capacity in IEEE 82.16e systems. these scheduling algorithms, we used EVRC with variable data rates and silence suppression. Also, we have explained that the and algorithms have some problems for supporting the VoIP services, such as the waste of uplink resources in the algorithm, and additional access delay as well as MAC overhead owing to bandwidth request process in the algorithm, and shown that the e algorithm can solve these problems of the and algorithms. By using the analysis of resource utilization efficiency, we have demonstrated that the e algorithm can save a lot of downlink and uplink resources compared with the and algorithms. In addition, with the simulation results of packet transmission delay for voice packets, we have shown that the,, e algorithms can support 68, 76, 92 voice users, respectively. Consequently, through the performance analysis of resource utilization efficiency and VoIP capacity, we have proven that the e algorithm has the best resource utilization efficiency and the largest VoIP capacity among the various scheduling algorithms in IEEE 82.16e systems. The e algorithm could be used efficiently in any wireless communication systems that support VoIP services with variable data rates and silence suppression. REFERENCES [1] IEEE 82.16-21, "IEEE Standard for Local and Metropolitan Area Networks Part 16: Air Interface for Fixed Broadband Wireless Access Systems," Apr. 8, 22. [2] IEEE 82.16a-23, "IEEE Standard for Local and Metropolitan Area Networks PPart 16: Air Interface for Fixed Broadband Wireless Access Systems - Amendment 2: Medium Access Control Modifications and Additional Physical Layer Specifications for 2-11 GHz," Apr. 1, 23. [3] IEEE 82.16-REVd/D5-24, "IEEE Standard for Local and Metropolitan Area Networks - Part 16: Air Interface for Fixed Broadband Wireless Access Systems," May. 13, 24. [4] IEEE 82.16e/D8-25, "IEEE Standard for Local and Metropolitan Area Networks - Part 16: Air Interface for Fixed and Mobile Broadband Wireless Access Systems - Amendment for Physical and Medium Access Control Layers for Combined Fixed and Mobile Operation in Licensed Bands," May 2, 25. [5] Howon Lee, Taesoo Kwon, Dong-Ho Cho, "An Enhanced Uplink Scheduling Algorithm Based on Voice Activity for VoIP Services in IEEE 82.16d/e System," IEEE Communications Letters, pp. 216-218, Aug. 25. [6] TIA/EIA/IS-127, "Enhanced variable rate codec, speech service option 3 for wideband spread spectrum digital systems," 1996. [7] P.T. Brady, "A Model for Generating ON-OFF Speech Patterns in TwoWay Conversations," Bell Syst. Technology Journal, vol. 48, no. 7, pp. 2445-2472, Sept., 1969. [8] ITU-T Recommendation G.1 14, "One-way transmission time," May 23. V. CONCLUSIONS In this paper, we have analyzed and discussed the performance of scheduling algorithms in IEEE 82.16e systems, such as,, and e. To analyze performance of 1235