Extended-rtPS Algorithm for VoIP Services in IEEE systems

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1 Extended-rtPS Algorithm for VoIP Services in IEEE systems Howon Lee, Taesoo Kwon and Dong-Ho Cho Department of Electrical Engineering and Computer Science Korea Advanced Institute of Science and Technology (KAIST) 373-1, Guseong-dong, Yuseong-gu, Daejeon, Korea Telephone: , Fax: { hwlee, Abstract There are several scheduling algorithms for Voice over IP (VoIP) services in IEEE systems, such as unsolicited grant service (UGS), real-time polling service (rtps), UGS with Activity Detection (UGS-AD), and Lee s algorithm using Grant-Me bit of the generic MAC header. However, these algorithms have some problems of a waste of uplink resources, additional access delay, and MAC overhead for supporting VoIP services with variable data rates and silence suppression. To solve these problems, we propose a novel uplink scheduling algorithm (Extended-rtPS) for the VoIP services in IEEE systems. Through the performance analysis and simulation results of resource utilization, VoIP capacity, total throughput, and packet transmission delay, we show that our proposed algorithm can solve the problems of the conventional algorithms, and has the best performance among these algorithms. In addition, with simulation results of packet transmission delay, we prove that our proposed algorithm can support more 74%, 24%, and 9% voice users compared with the UGS, rtps, and UGS-AD (Lee s) algorithms, respectively. I. INTRODUCTI THE IEEE standard is designed to satisfy various demands for higher capacity, higher data rate, and more advanced multimedia services [1], [2]. 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 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 (Band-AMC) scheme, and so on. There are four uplink scheduling algorithms to support variable requirements of QoS in IEEE systems, such as UGS, rtps, non-rtps (nrtps), and best effort service (BE). However, these algorithms are not suitable for VoIP services with variable data rates and silence suppression in IEEE systems. In addition, there are some previously proposed algorithms for the VoIP services, such as UGS-AD in the DOCSIS system, and Lee s algorithm [3]- [5]. But, these algorithms also have problems for supporting VoIP services with variable data rates and silence suppression. That is, these algorithms cannot This research was supported in part by University IT Research Center Project. support voice codecs with variable data rates well, such as enhanced variable rate codec (EVRC) [6]. The UGS-AD and Lee s algorithms can follow voice codec with only two data rates (on-off). Therefore, we propose a novel uplink scheduling algorithm for the VoIP services, ertps. This algorithm is recently proposed and adopted in IEEE e standard by us [7]. On the contrary, although the UGS-AD algorithm is also proposed, this algorithm is rejected in this standard [2], [4]. In this paper, for performance analysis of conventional algorithms and our proposed algorithm (UGS, rtps, UGS-AD, Lee s, ertps), we utilize EVRC with variable data rates and silence suppression [6]. The frame duration of EVRC (T vc ) is 20ms, and the voice activity factor is with 29% full rate (, p 1 ), 4% half rate (/2, p 2 ), 7% quarter rate (/4, p 3 ), and 60% eighth rate (/8, p 4 ). Also, the sizes of voice packets in each rate, (L 1, L 2, L 3, L 4 ) are 171, 80, 40 and 16 bits. Full, half and quarter rates are included in talk-spurt (on) duration, and eighth rate is included in silence (off) duration. When using a voice codec with a voice activity detector (VAD) or silence detector (SD), the voice user can know his voice status in the higher layer. This higher layer information can be known in the MAC layer by using primitives of Convergence Sublayer (CS layer) in IEEE e systems [2], [8]. The remainder of this paper is organized as follows: In Section II, we introduce and discuss conventional scheduling algorithms, such as the UGS, rtps, UGS-AD, Lee s, nrtps, and BE algorithms. In Section III, we propose a novel scheduling algorithm to solve the problems of the conventional algorithms. Also, in Section IV and V, we analyze and simulate the performance of resource utilization, VoIP capacity, throughput, and packet transmission delay of the conventional and proposed algorithms. Finally, in Section VI, we make conclusions. II. CVENTIAL UPLINK SCHEDULING ALGORITHMS A. UGS Algorithm /06/$20.00 (c) 2006 IEEE The UGS algorithm is designed to support real-time service flows that generate fixed-size data packets periodically [1]. In this algorithm, the base station (BS) periodically assigns fixedsize grants to the voice user, and these fixed-size grants are sufficient to send voice data packets generated by the maximum

2 Resource Resource /2 /2 /4 /8 /4 /8 Fig. 1. Operation of UGS algorithm Fig. 2. Operation of rtps algorithm data rate of EVRC. 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 with the same size, because they have variable data rates and silence suppression [6]. In this algorithm, since the BS always assigns fixed-size grants that are sufficient to send voice packets generated by the maximum data rate of EVRC, it causes a waste of many 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 other words, in the UGS 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 a lot of uplink resources. But, because VoIP services are delay-sensitive, the usage 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. C. UGS-AD Algorithm The UGS-AD algorithm is designed to support real-time service flows that generate fixed size data packets on a semiperiodic basis [3], [4]. This is a combined algorithm of the UGS and rtps algorithms. This algorithm has two scheduling modes (UGS, rtps), and can switch these modes according to the status of voice users. If the VoIP services were initiated, this algorithm firstly starts as the rtps mode. In case of the rtps mode, if the voice user requests bandwidth size as zero byte, the BS maintains its mode. However, if the user requests another bandwidth size (non-zero bytes), the BS has to switch B. rtps Algorithm its mode to the UGS. In case of the UGS mode, the BS The rtps algorithm is designed to support real-time service operates by contraries. flows that generate variable size data packets periodically [1]. As shown in Fig. 3, in the UGS-AD algorithm, since the BS The BS assigns uplink resources that are sufficient for unicast cannot follow the half and quarter rates of EVRC, the waste bandwidth requests to the voice user. Generally, this process of uplink resources would occur. In case of the eight rate, this is called a bandwidth request process, or polling process. Because this algorithm always uses a bandwidth request process rtps. When a data rate of the user is changed from the quarter waste is not caused, because the BS switches its mode to the for suitable size grants, it transports data more efficiently rate to the eighth rate, for notifying his status, the user utilizes than the UGS algorithm. However, this bandwidth request grant management subheader with a bandwidth request of zero process always causes MAC overhead and additional access byte [2]. This process does not cause MAC overhead, since delay. Hence, the rtps algorithm has larger MAC overhead and the user uses remained uplink resources. In case that the data access delay than the UGS, UGS-AD, Lee s and our proposed rate of the voice user is increased to the full rate, he should algorithms. transmit bandwidth request header with a bandwidth request In Fig. 2, since the user requests exact amount of uplink of non-zero bytes [2]. In this case, at first, the BS allocates resources for transmitting his voice packets, the dot line and enough uplink resources to the user for sending first delayed the solid line are nearly the same. In this paper, to avoid voice packet and second voice packet generated by the full rate the polling process in a silence duration, we assume that of EVRC. Then, the BS periodically assigns uplink resources minimum polling size is the size of voice packet generated by according to the general operation of the UGS mode. the minimum data rate of the voice codec, except the UGS The UGS-AD algorithm can solve the problems caused by algorithm. Thus, there is no polling process in the silence the UGS algorithm (waste of uplink resources) and the rtps duration of the users. algorithm (MAC overhead and access delay), in case that the In the rtps algorithm, the user can use the piggyback voice users use voice codecs with only two data rates (onoff). However, in case of EVRC with variable data rates and requests of the grant management subheader for VoIP services. 2061

3 Resource UGS mode rtps mode Resource /2 /4 /8 /2 Grant Management subheader /4 /8 Fig. 4. Operation of Lee s algorithm voice state is on. Thus, the voice user cannot send the voice packet using this grant size. In this case, the user can send Fig. 3. Operation of UGS-AD algorithm this voice packet by using piggyback requests of the grant management subheader or bandwidth requests of bandwidth request header, as shown in Fig. 4. silence suppression, this algorithm cannot solve the problems perfectly. In this case, the waste of uplink resources occurs in E. nrtps and BE algorithms talk-spurt (on) duration of the voice users. The nrtps and BE algorithms are designed to support nonrealtime service flows, such as HyperText Transfer Protocol D. Lee s Algorithm (HTTP) and File Transfer Protocol (FTP) [1], [2]. In other Similar to the UGS-AD algorithm, Lee s algorithm partially words, these algorithms are not proper to use for VoIP services can solve the problems of the UGS and rtps algorithms, in with variable data rates and silence suppression. Thus, in this that the BS basically assigns uplink resources to voice users by paper, we did not consider these algorithms, since we are considering only on-off transitions of them. In this algorithm, focused on realtime VoIP services. because the voice user has to inform the BS of his voice state transitions, it requires a method for relaying his voice status III. PROPOSED ALGORITHM information. Therefore, since the conventional generic MAC To solve the problems of the UGS, rtps, UGS-AD, and header of IEEE systems has two reserved bits for other Lee s algorithms, such as the waste of uplink resources, MAC additive operations, Lee s algorithm uses one reserved bit in overhead, and additional access delay, we propose a novel this header for a method to inform the BS of the user s voice uplink scheduling algorithm for VoIP services with variable state transitions [1], [2]. This one reserved bit is defined as a data rates and silence suppression in IEEE systems. Grant-Me (GM) bit in this algorithm. Basically, in order to solve these problems, our proposed When the voice state of the user is on, he sets the algorithm allocates uplink resources according to all status of GM bit to 1, otherwise he sets the GM bit to 0. The the voice users without MAC overhead. Here, we describe the voice user can inform the BS of his voice state transitions detailed operation of our proposed algorithm. effectively without MAC overhead, because it conveys the Firstly, the voice user informs the BS of his voice status voice status information using the conventional generic MAC information using grant management subheader in case that the header. Detailed operation of this algorithm according to the size of a voice data packet is decreased [2]. The user requests GMbitisasfollows. the bandwidth for sending the voice packets using extended If the GM bit is 0 : The BS assigns the minimum grant PBR (piggyback request) bits of grant management subheader. size to the voice user. This minimum size is sufficient in case Since the user uses remained uplink resources assigned to that the user informs the BS of his voice state transitions. him, there is no waste of uplink resources. In our proposed When the GM bit is changed, from 1 into 0, the BS once algorithm, to distinguish these extended PBR bits with general assigns maximum grant size to the user whose voice state is PBR bits, the user sets the MSB of PBR bits to 1. In this case, off, as shown in Fig. 4. Hence, it causes a little waste of the BS assigns uplink resources according to the requested size uplink resources, which could be negligible. periodically, until the voice user requests another size of the If the GM bit is 1 : The BS assigns the maximum grant bandwidth. size to the voice user. That size is sufficient to send voice data Secondly, the voice user informs the BS of his voice packets. In case that the GM bit is changed, from 0 into 1, status information using bandwidth request header in case that the BS once assigns minimum grant size to the user whose the size of a voice data packet is increased [2]. The user 2062

4 Resource Grant Management subheader /2 /4 /8 Fig. 5. Operation of proposed algorithm requests the bandwidth for sending the voice packets using BR (bandwidth request) bits of bandwidth request header. In the same way as the case of the data rate decrement, to distinguish these BR bits with general BR bits, the user sets the MSB of BR bits to 1. Then, the BS also assigns uplink resources according to the requested size periodically, until the user requests another size of the bandwidth. In case of the data rate increment, the BS shall provide the first bandwidth allocation to the next MAC frame after this bandwidth request process. The second bandwidth allocation is done after the bandwidth allocation based on the time which the BS allocated the bandwidth used for the bandwidth request process, as shown in Fig. 5. In summary, in case of VoIP services with variable data rates and silence suppression using this algorithm, the BS recognizes grant management subheader and bandwidth request header especially. In this algorithm, if the user requests the bandwidth for sending the voice packets, then the BS shall change its polling size according to the bandwidth size requested by the user, and keeps its changed polling size until the user sends another requests. In other words, the BS may not change its polling size without any requests from the voice users. Since the proposed algorithm can follow all data rates of the voice users, the BS can obtain better data transport efficiency compared with the UGS, rtps, UGS-AD and Lee s algorithms. IV. PERFORMANCE ANALYSIS A. Resource Utilization In case that the total number of voice users is N Tot,the numbers of users in 1, 1/2, 1/4, and 1/8 rates (N 1, N 2, N 3, N 4 ) are obtained by N 1 = N Tot p 1, N 2 = N Tot p 2, N 3 = N Tot p 3, and N 4 = N Tot p 4.ByusingN 1, N 2, N 3, and N 4, we can calculate the numbers of required uplink resources in the UGS, rtps, UGS-AD, Lee s, and our proposed algorithms. The numbers of required uplink resources in the UGS and rtps algorithm (R UGS, R rtps )are R UGS = R 1 N i. (1) 2063 Fig. 6. Number of required uplink resources vs. number of users R rtps = R 4 N i + R i N i. (2) Here, R i is the number of required uplink resources for transmitting voice packets generated by each data rate of EVRC (L i ). In case of the UGS-AD and Lee s algorithms, although the operations of these algorithms are not the same, the numbers of required uplink resources (R UGS AD, R Lee ) are exactly the same. So, in this paper, we analyze and discuss only UGS-AD algorithm. R UGS AD is given as R UGS AD = R 1 N i + R 4 N 4. = R Lee. (3) Also, the number of required uplink resource in our proposed algorithm (R PRD ) can be represented by R PRD = R i N i. (4) When we use QPSK 1/2 for transmitting voice data packets, R 1, R 2, R 3, and R 4 are 6, 4, 3, and 2 resource units. For performance analysis, we assume an OFDMA system, and one resource unit consists of 48 OFDM subcarriers. In addition, we assume that the size of generic MAC header is 6 bytes, and the RTP/UDP/IP headers are compressed by robust header compression (ROHC) [2], [9]. The sizes of uncompressed and compressed RTP/UDP/IP header are 40 bytes and 3 bytes. In case that N Tot = 20, N 1, N 2, N 3, and N 4 are 6, 1, 1, and 12, respectively. Hence, R UGS, R rtps, R UGS AD, and R PRD are 120, 83, 72, and 67. We can show that our proposed algorithm has the smallest the number of required uplink resources compared with the UGS, rtps, UGS-AD, Lee s algorithms. Fig. 6 shows the number of required uplink resources against the number of voice users. In case that the voice users use another voice codec that has only on-off transitions, by equations 3 and 4, we can show

5 Fig. 7. Maximum number of users vs. MCS levels that the numbers of required uplink resources in the UGS-AD (Lee s) and our proposed algorithms are nearly the same. B. VoIP Capacity by Analysis of Resource Utilization By using R UGS, R rtps, R UGS AD, and R PRD, we can calculate the maximum supportable number of voice users in each algorithm (N M UGS, N M rtps, N M UGS AD, N M PRD ). With the total number of uplink resources (R Tot = 140), N M UGS = Tvc N M UGS AD = Tvc R Tot RTot R UGS, N M rtps = Tvc R UGS AD, and N M PRD = Tvc RTot RTot R rtp S, R PRD. In these equations, is a MAC frame duration in IEEE systems, 5ms, and T vc is a frame duration of EVRC, 20ms. In case of QPSK 1/2, N M UGS, N M rtps, N M UGS AD, and N M PRD are 93, 131, 149, and 164. We can also show that our proposed algorithm can support the largest number of voice users compared with the UGS, rtps, UGS-AD, Lee s algorithms. Fig. 7 shows the maximum number of voice users (VoIP capacity) against several Modulation and Coding Scheme (MCS) levels. C. Throughput With the sizes of generic MAC header (L MH ) and compressed RTP/UDP/IP header (L UH ), we can calculate downlink throughput of each algorithm (S UGS, S rtps, S UGS AD, S Lee, S PRD ). S op = T vc s.t. {(L MH + L UH + L i ) N i } N i <N M op. (5) In the UGS, rtps, UGS-AD, Lee s, and proposed algorithms, op in equation 5 is the same as UGS, rtps, UGS-AD, Lee, and PRD D. Packet Transmission Delay Except the rtps algorithm, the packet transmission delays of the UGS, UGS-AD, Lee s, and our proposed algorithms (T tx UGS, T tx UGS AD, T tx Lee, T tx PRD ) are represented as T tx op = T dsf + T ttg + T q op. (6) T dsf and T ttg are the durations of the downlink subframe and the transmit transition gap (TTG) in IEEE systems. T dsf and T ttg are constants and T q op is queuing delay of each algorithm. Also, by using the durations of the uplink subframe T usf and the receive transition gap (RTG) (T rtg ), and queuing delay of the rtps algorithm (T q rtps ), the packet transmission delay of the rtps algorithm (T tx rtps ) can be given as T tx rtps = p i (T dsf + T usf + T ttg + T rtg ) + T dsf + T ttg + T q rtps = p i + T dsf + T ttg + T q rtps. (7) In equation 7, similar to T dsf and T ttg, T usf and T rtg are also constants. And, T dsf +T usf +T ttg +T rtg = =5ms. Since the rtps algorithm experiences the polling process in full, half and quarter rates, an addition delay would be generated in this algorithm. In this analysis, we assume that the polling process causes the delay of one MAC frame duration. However, in case that many voice users are activated, the polling process can cause the delay of more MAC frame durations. Although, the rtps algorithm has additional delay terms compared with any other algorithm, we must not conclude that the rtps algorithm has the largest packet transmission delay among these algorithms, because a dominant factor of packet transmission delay is a queuing delay of each algorithm. T q op is mainly determined by RTot R op and N Tot. The relationship of these elements is as follows. T q op R Tot, N Tot. (8) R op In Section V, through simulation results, we prove that these relationships are correct. V. SIMULATI RESULTS For simulation results, we assume the IEEE OFDMA system. A MAC frame consists of 36 symbols (time domain) and 1024 subcarriers (frequency domain). In this simulation, voice packets are sent by using QPSK 1/2. So, one basic resource unit is the same as six bytes. Also, we assume that durations of full, half, quarter and eighth rates are 290, 40, 70 and 600ms, respectively. Other simulation parameters are the same as we assume in Section IV. Fig. 8 shows the total throughputs of the conventional and proposed algorithms. Through this figure, we can show that the values of throughput saturation points for the UGS, rtps, UGS-AD (Lee s) and our proposed algorithms are 93, 131, 149 and 164, respectively, and the throughput of our proposed

6 Fig. 8. Total throughput vs. number of users Fig. 9. Packet transmission delay vs. number of users algorithm is much larger than that of any other algorithms. The value of saturation point in each algorithm is the same with the maximum number of voice users in each algorithm in Fig. 7 The saturation of the throughput in each algorithm is generated due to resource limitation. In case of our proposed algorithm, because the number of required uplink resources is the smallest among these algorithms, the throughput is saturated later compared with any other algorithm. In addition, since we did not assume aggregations of voice packets in each data rate, the UGS algorithm has the worst performance in this simulation. However, if we assume the aggregations of voice packets, the throughputs of the UGS, UGS-AD, and Lee s algorithms could be somewhat increased. As shown in Fig. 9, we can show that the values of restricted points for the UGS, rtps, UGS-AD (Lee s) and our proposed algorithms are 101, 142, 162 and 176, respectively, and the values of the restricted points for packet transmission delay are larger than that of total throughput of each algorithm. This is because delay bound for packet transmissions is not 0 ms, but 60 ms. In this simulation, in the consideration of maximum end-to-end delay bound (285 ms by ITU-T), backbone delay, packet processing delay, and handset playback buffer delay, we assume that delay bound of packet transmission is 60ms [10]. Similar to the performance of throughput for each algorithm, in case of our proposed algorithm, the maximum supportable number of users is the largest in IEEE systems. We can show that our proposed algorithm can support more 74%, 24%, and 9% voice users compared with the UGS, rtps, and UGS-AD (Lee s) algorithms, respectively. VI. CCLUSIS In this paper, we have proposed a novel uplink scheduling algorithm (ertps) for VoIP services with variable data rates and silence suppression in IEEE systems. In spite of the fact that there are some scheduling algorithms for the VoIP services, such as the UGS, rtps, UGS-AD, and Lee s algorithm, these algorithms have some problems of the waste of uplink resources, MAC overhead and additional access 2065 delay owing to polling process. Especially, in case of the UGS- AD and Lee s algorithms, although these algorithms partially can solve the problems of the UGS and rtps algorithms, they are not perfect solutions for supporting VoIP services with variable data rates and silence suppression. However, our proposed algorithm can perfectly solve these problems of the conventional algorithms. Through the performance analysis and simulation results of resource utilization, VoIP capacity, throughput, and packet transmission delay, we have shown that our proposed algorithm has the best performance among these algorithms in IEEE systems. In particular, our proposed algorithm can support more 74%, 24%, and 9% voice users compared with the UGS, rtps, and UGS-AD (Lee s) algorithms, respectively. We can say that our proposed algorithm is the best algorithm for VoIP services with variable data rates and silence suppression in IEEE systems. REFERENCES [1] IEEE , IEEE Standard for Local and Metropolitan Area Networks Part 16: Air Interface for Fixed Broadband Wireless Access Systems, Jun. 24, [2] IEEE e/D , 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, Aug., [3] Data-Over-Cable Service Interface Spec., DOCSIS 2.0, Radio Frequency Interface Spec. - CM-SP-RFIv2.0-I , Aug. 4, 2004 [4] IEEE C802.16e-04/503 UGS with Activity Detection for e, Nov. 16, [5] Howon Lee, T. Kwon, D. Cho, An Enhanced Uplink Scheduling Algorithm Based on Voice Activity for VoIP Services in IEEE d/e System, IEEE Communications Letters, pp , Aug [6] TIA/EIA/IS-127, Enhanced variable rate codec, speech service option 3 for wideband spread spectrum digital systems, [7] IEEE C802.16e-04/522r3 Extended rtps for VoIP services, Nov. 16, [8] IEEE C802.16e-04/523r1 Header compression specific Convergence Sublayer, Nov. 16, [9] C. Bormann et al., RObust Header Compression(ROHC): Framework and Four Profiles: RTP, UDP, ESP, and Uncompressed, IETF Network Working Group, RFC 3095, Jul [10] ITU-T Recommendation G.114, One-way transmission time, May, 2003.

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