28th NATIONAL RADIO SCIENCE CONFERENCE (NRSC 2011)

April 26-28, 2011, National Telecommunication Institute, Egypt

VoIP Capacity Estimation In Mobile WiMAX Networks

Nawal A.El-jishaw/1, Mohamed Zahra#2, M Ebrahi #3, Mostafa M El-gamala#4. lComputer and control Dep., Faculty of Electronic Eng. Elmenofia Uni. Menouf, Egypt.

2,3Comm. Dep., Faculty of Eng., AI-Azhar Uni. Cairo, Egypt. 4Telecom Egypt, Tanta, Egypt.

ABSTRACT

The IEEE 802.16 standard which has emerged as a broadband wireless access technology, promises to deliver high data rates over large areas to a large number of users in the near future. We present a simple analytical method for VoIP capacity estimation in IEEE 802.16e mobile WiMAX networks. Various overheads that impact the capacity are explained and methods to reduce these overheads are also presented. The analysis process helps explain various features of mobile WiMAX. It is shown that proper use of overhead reducing mechanisms and proper scheduling can make an order of magnitude difference in performance. The paper gives the maximum number of voice sessions using multiple VoIP codecs. Via simulation using ns2, the results of the analytical calculations will be validated using simple scheduler.

Keywords: WiMAX, VoIP

I. Introduction

The widespread use of IP-based technologies resulted in the vision of converged networks that promises cost­

efficiency by supporting voice, video, and data on a single network. This gave rise to the popularity ofVoIP,

which provide efficient voice delivery over packet-switched networks by better resource utilization as compared

to traditional circuit-switched mode. WiMAX (Worldwide Interoperability for Microwave Access), is a cell-based

technology aimed at providing last-mile wireless broadband access at a cheaper cost. The core of Wi MAX

technology is specified by the IEEE 802.16 standard that provides specifications for the Medium Access Control

(MAC) and Physical (PRY) layers as shown in Fig. I.

Data link layer { Common part sublayer

Security sub layer

Convergence sublayer

Physical layer OFDM I

OFDMA

Figure 1- OSI layers related to WiMAX.

In this paper we present a simple analytical method of calculate the maximum number of VoIP users in a Mobile WiMAX system for different voice codecs. The input parameters can be easily changed allowing service providers to examine the effect of parameters changes and to study the sensitivity to various parameters. We explain all the factors that affect the WiMAX VoIP capacity. Also the paper presents WiMAX network simulation to validate the analytical calculations.

The rest of the paper is organized as follows. Section III presents an overview about the WiMAX physical layer. Introduction about the WiMAX MAC layer will be presented in section IV. Section V will present the PRY layer and MAC layer overhead analysis. Section VI provides VoIP capacity estimation over WiMAX network. Section VII simulate WiMAX network using ns2 to validate the analytical calculations. The paper will be concluded in section VIII.

28th NATIONAL RADIO SCIENCE CONFERENCE (NRSC 2011)

April 26-28, 2011, National Telecommunication Institute, Egypt

II. Mobile WiMAX PHY layer

Today, almost all upcoming broadband access technologies including Mobile WiMAX and its competitors use OFDMA. WiMAX allows almost any available spectrum width to be used. Allowed channel bandwidths vary from 1.25 MHz to 28 MHz. The channel is divided into many equally spaced subcarriers. Some of which are used for data transmission while others are reserved for monitoring the quality of the channel (pilot subcarriers). The remaining subcarriers used as guard subcarriers and DC subcarrier. The data and pilot subcarriers are modulated using one of several available modulation and coding schemes (MCS).

The WiMAX profiles use only Time Division Duplexing (TDD) which the transmission consists of frames as

shown in Fig. 2. The downlink (DL) subframe and uplink (UL) subframe are separated by a transmit to transmit

gap (TTG) and receive to transmit gap (RTG). The frames are shown in two dimensions with frequency along the

vertical axis and time along the horizontal axis [1].

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Figure 2 - A sample OFDMA TDD frame structure

Subtracting the guard subcarriers and the DC subcarrier from the total subcarriers gives the set of used subcarriers. For both the uplink and downlink, these subcarriers are allocated as pilot subcarriers and data subcarriers according to one or another of the defmed OFDMA permutation modes. Two families of permutation modes exists; diversity and contiguous. The most common diversity permutation mode is Partial Usage of the SubChannel (PUSC). A subchannel is the minimum transmission unit in an OFDMA frequency dimension. A slot in the OFDMA PHY has both a time and subchannel dimension. A slot is the minimum possible data allocation unit in the 802.16 standard. The slot defmition for downlink PUSC is 1 subchannel x 2 OFDMA symbols, and Uplink PUSC is 1 subchannel x 3 OFDMA symbols [2]. Table 1 shows summery about the physical mobile WiMAX parameters. From table 1, note that WiMAX system profile support a frame size of 5 ms only, thus 47 symbols allow for 1.6 symbol times for TTG + RTG. The frame size includes 1 symbol for preamble [3].

Table 2 lists the number of bytes per slot for various MCS values. For each MCS, the number of bytes is equal to (#bits per symbols x Coding Rate x 48 data subcarriers and symbols per slot / 8 bits). For 10 MHz channel and 2:1 DL:UL ratio, the DL symbol equal to 28. Thus the DL subframe will consist of 14*30 slot. Table 3 lists the total number of DL slots and UL slots for different DL:UL ratios.

28th NATIONAL RADIO SCIENCE CONFERENCE (NRSC 2011)

April 26-28, 2011, National Telecommunication Institute, Egypt

Table 1 - OFDMA Parameters for Mobile WiMAX.

Parameters Value set 1 Value set 2

System Channel Bandwidth (MHz) 10 5

Sampling Frequency (Fp in MHz) 11.2 5.6

FFT Size (NFFT ) 1024 512

Sub-Carrier Frequency Spacing 10.9375 kHz 10.9375 kHz

Useful Symbol Time (Tb = lit) 91.4 !IS 91.4 J.lS

Guard Time (Tg =Tb/8) 11.4 !IS 11.4 J.ls

OFDMA Symbol Duration (Ts =Tb + Tg) 102.9 J.lS 102.9 J.ls

Frame duration 5 ms 5 ms

Number of OFDMA Symbols 47 47

Null Subcarriers 184 92

Pilot Subcarriers 120 60

DLPUSC Data Subcarriers 720 360

Sub-channels 30 15

Null Subcarriers 184 92

Pilot Subcarriers 280 140

ULPUSC Pilot Subcarriers 560 280

Sub-channels 35 17

Table 2 - Number of bytes per slot for various Mes.

MCS Bit/symbol Coding rate Byte/slot

QPSK1I2 2 0.5 6

QPSKJ/4 2 0.75 9

16QAM1I2 4 0.5 12

16QAM3/4 4 0.75 18

64QAM2/3 6 0.67 24

64QAMJ/4 6 0.75 27

28th NATIONAL RADIO SCIENCE CONFERENCE (NRSC 2011)

April 26-28, 2011, National Telecommunication Institute, Egypt

Table 3 - Number of slots for the different DL: UL ratios.

� 1024 512

DL UL DL UL

1:1 345 233 172 113

2:1 420 175 210 85

3:1 510 105 255 51

III. Mobile WiMAX MAC layer

The MAC layer of WiMAX is comprised of three sublayers service-specific convergence sub layer, common part sub layer, and privacy sub layer. The common part sub layer is the core functional layer which provides bandwidth and establishes and maintains connections. Moreover, as the WiMAX MAC provides a connection­oriented service to the subscriber stations (SS), the common part sub layer also provides a connection identifier (CID) to identify which connection the MPDU is servicing. The common part sub layer defines five QoS classes, Unsolicited Grant Service (UGS), real-time Polling Service (rtPS), enhanced real time Polling Service (ertPS), non real-time Polling Service (nrtPS), and Best Effort (BE). UGS support constant bit rate (CBR) applications such as VoIP without silence suppression. rtPS support variable bit rate applications like video transmission, ertPS support voice with silence suppression [4]. The following analysis assume that the voice use UGS

scheduling service.

IV. Overhead analysis

In DL subframe, overhead consist of preamble, FCH, DLMAP and UL-MAP as shown in Fig. 2. The MAP entries consist of fixed part and variable part and these entries can result in a significant amount of overhead. WiMAX Forum recommends using compressed MAP, which reduces the DL-MAP entry overhead to 11 bytes including 4 bytes for Cyclic Redundancy Check (CRC). The fixed UL-MAP is 6 bytes long with an optional 4-byte CRC. With a repetition code of 4 and QPSKYz, both fixed DLMAP and UL-MAP take up 16 slots.

The variable part of DL-MAP consists of one entry per bursts and requires 60 bits per entry. Similarly, the variable part of UL-MAP consists of one entry per bursts and requires 52 bits per entry. These are all repeated 4 times and use only QPSKYz MCS. It should be pointed out that repetition consists of repeating slots (and not bytes). Thus, both DL and UL MAPs entries also take up 16 slots each per burst. The UL subframe also has fixed and variable parts. Ranging and contention are in the fixed portion. Their size is defined by the network administrator.

The other fixed portion is channel quality indication (CQI) and acknowledgements (ACK). These regions are also defmed by the network administrator. Obviously, more fixed portions are allocated; less number of slots is available for the user workloads. In our analysis, we allocated three OFDM symbol columns for all fixed regions. Each UL burst begins with a UL preamble. One OFDM symbol is used for short preamble and two for long preamble. We allocate one slot for the UL preamble. Each MAC PDU has at least 6-bytes of MAC header and a variable length payload consisting of a number of optional subheaders, data, and an optional 4- byte CRe. The optional subheaders include fragmentation, packing, and mesh. Each of these is 2 bytes long [1][3].

V. VoIP capacity estimation over mobile WiMAX network

The VoIP codec must be determined to calculate the VoIP capacity over WiMAX network. Table 4 lists the codec parameters for common codecs and mean opinion score (MOS) for each one [6].

28th NATIONAL RADIO SCIENCE CONFERENCE (NRSC 2011)

April 26-28, 2011, National Telecommunication Institute, Egypt

Table 4 - VoIP codec parameters.

Voice payload Voice payload

Codec size (byte) size (ms) MOS

G711 160 20 4.1

G729 20 20 3.92

G723 20 30 3.8

G726 80 20 3.85

ILBC 38 20 -

GSM 37 22.5 -

The following equations calculate the maximum number ofVoIP users in mobile WiMAX networks.

Vr = (CI+Ho)/Ct (1)

Where Vr is the voice codec rate plus the higher layer overhead, CI is codec payload length in byte, Ho is the higher layer overhead, and Ct is codec payload length in ms.

Higher layer overhead (Ho) can be 1 byte if Robust Header Compression (ROHC) is used which is an algorithm to compress the higher layer headers [7]. Ho can be three byte if Packet Header Suppression (PHS) is used, which is a WiMAX option to not transmit the fixed headers. Ho can be the normal headers which is RTPIUDP/IP headers (12/8/20 byte). According to table 4 and the different Ho values, the values of Vr will be calculated for the different voice Codecs.

Nsd = (((Vr * Fs)+Mh)/ M)+Mvh (2)

Where Nsd is the number of slot needed by voice packet in downlink direction, Fs is the frame size which is 5 ms (table 1), Mh is the MAC header length which is 6 byte, M is the number of bytes per slot in each MCS (table 2), and Mvh is the number of MAP slots per voice user. By using equation (2) the number of slots required for VoIP packet will be calculated for every MCS.

Ndl = (Ds-Mf)/ Nsd (3)

Where Ndl is the maximum number of VoIP sessions in downlink direction, Ds is the number of downlink

slots (table 3), and Mf is the fixed number of slots in MAP message. By using the different values of Nsd in

equation (3), the maximum number of VoIP sessions in downlink direction will be calculated for every VoIP

codec and different higher layer overhead.

Nsu = (((Vr * Fs) + Mh)/ M) + P (4)

Where Nsu is the number of slots needed by VoIP packet in uplink direction, P is the number of slots for

uplink preamble.

Nul =Us/ Nsu (5)

28th NATIONAL RADIO SCIENCE CONFERENCE (NRSC 2011)

April 26-28, 2011, National Telecommunication Institute, Egypt

Where Nul is the maximum number of VoIP sessions in uplink direction, and Us is the number of uplink slots

(table 3).

N = min[Nul,Ndl] (6)

Where N is the effective number ofVoIP sessions in mobile WiMAX networks.

A simple matlab code is used to calculate the effective numbers of VoIP users with different codecs, DL:UL ratios, and higher layer overhead. The results show that the same VoIP capacity obtained using ROHC and PHS, but when use normal higher layer headers (RTPIUDP/IP) the VoIP capacity decreased by a factor of 0.03 to 0.5 depending on the used codec and MCS. Also there is no difference in maximum VoIP users between the three header types in 64-QAM2/3 and 64-QAM3/4. Table 5 shows the VoIP SS capacity for different DL:UL ratio assuming G723 VoIP codec according to the above equations. From the table it's shown that the best DL:UL ratio which give the maximum VoIP capacity is 3: 1.

Table 5 -VoIP SS capacity for different DL:UL ratio

DL:UL 1:1 2:1 3:1

MCS QPSK1I2 64QAM3/4 QPSK1I2 64QAM3/4 QPSK1I2 64QAM3/4

VoIP capacity 18 19 22 23 27 29

We present two scenarios. The fIrst one assume an error-free channel while the second extends the results to a case in which different users have different error rates due to channel condition.

1- First scenario

Given the VoIP codec characteristics and the overheads discussed so far, it is straightforward to compute the VoIP system capacity for any given codecs using the above formulas. Fig. 3 shows the VoIP capacity for different VoIP codecs and different MCS's assuming 3:1 DL:UL. The fIgure shows that the best VoIP codec which give the maximum capacity for all MCS's is G723. The maximum VoIP capacity per sector is 29 SS. This small capacity is due to the simple scheduler in the BS which schedules every SS in the same WiMAX frame. Thus increase the overhead in WiMAX frame comparing to VoIP payload. Note that in [1] the VoIP capacity is 23 because the paper choose DL:UL equal to 2:1 which decrease the capacity.

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Figure 3 -VoIP capacity for different VoIP codecs and different MCS's.

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28th NATIONAL RADIO SCIENCE CONFERENCE (NRSC 2011)

April 26-28, 2011, National Telecommunication Institute, Egypt

To increase the VoIP capacity enhanced scheduler can be used to aggregate the VoIP payload segments and send it once every predetermined number of WiMAX frame. This is to avoid the large number of overhead associated with every frame. This enhanced scheduler introduce delay to the VoIP traffic, but this delay chosen so that it is less than the maximum value allowed for VoIP traffic. Fig. 4 shows the VoIP capacity using enhanced scheduler which achieves the maximum delay around 60 ms. The figure shows that using enhanced scheduler the VoIP capacity improved by a factor of 5 to 10 depending upon the used codec and MCS; for example VoIP capacity using QPSK1I2 equal to 240. Note that in [I] the VoIP capacity is 192 because the paper choose DL:UL equal to 2:1 which decrease the capacity.

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2- Second scenario

SI G711 C1G723 t;:;lG726 EI G729 �ILBC !;;JGSM

The MCS is determined by the quality of the channel. In this section, we present a capacity analysis assuming a mix of channels with varying quality resulting in different levels of MCS for different users. Table 6 shows MCS distribution assumptions. The assumptions belong to the DL direction in an urban environment and are extrapolated from different measurement experiments [5].

Table 6 - MCS distribution assumptions.

MCS Weight

Fade 5%

QPSK Yz 2.5%

QPSK% 2.5%

16-QAM Yz 5%

16-QAM% 5%

64-QAM2/3 40%

64-QAM% 40%

28th NATIONAL RADIO SCIENCE CONFERENCE (NRSC 2011)

April 26-28, 2011, National Telecommunication Institute, Egypt

The average number of bytes per symbol (Mav) can be calculated by the following equation:

Mav=LWi*Mi (7)

Where W is the MCS weight, M is the number of bytes per symbol in each MCS, and i is the MCS. The VoIP capacity in imperfect channel scenario can be obtained by recalculate equation (2) and equation (4) using Mav instead of M Table 7 shows the maximum VoIP capacity in the imperfect channel using the simpler scheduler and enhanced scheduler.

Table 7 - Maximum VoIP capacity using simple scheduler and enhanced scheduler.

Codec G711 G723 G726 G729 ILBC GSM

VoIP Capacity

(Simple scheduler) 12 26 17 24 24 23

VoIP Capacity

(Enhanced scheduler) 152 312 211 296 296 282

VI. Simulation results

NS2 is used for the simulation [8]. The WiMAX module is developed by WiMAX forum. Table 8 lists the WiMAX network parameters used in simulation. Fig. 5 shows the voice throughput for voice SS. It is observed that the voice throughput for voice SS has small value in the fIrst 2 seconds, and then it is fIxed at 5.3 Kb/s. This is because of the collision between the different SS's requesting services from the BS. After reserving slots for every SS there isn't service request so that there isn't collision, thus there isn't loss due to collision as shown in fIg. 6. From fIg. 5 and fIg 6 it is clear that the UGS service is the best choice for voice without silence suppression, this is because the fIxed throughout reserved for each voice session. The maximum voice SS numbers obtained from simulation nearly coincides with the analytical results above.

Table 8 - WiMAX network confIguration used in simulation.

Physical layer OFDMA

Frame size 5ms

B andwidth lOMHz

Permutation mode puse

DL:UL 3:1

VoiceCodec G723

Cyclic prefix 118

MCS 64-QAM3/4

28th NATIONAL RADIO SCIENCE CONFERENCE (NRSC 2011)

April 26-28, 2011, National Telecommunication Institute, Egypt

6

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1

0 1 2 3 4 5 6 7 8 9 10

Time (sec.)

Figure 5 - Voice Throughput

8 7 6

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::: 4 0 ...J 3 2 1 0

1 2 3 4 5 6 7 8 9 10 Time (sec.)

Figure 6 - Voice loss rate due to collision

Table 9 lists the voice fragments per WiMAX frame, the maximum number ofVoIP SS, and the average voice delay. The table shows that when send the complete voice frame in one WiMAX frame, the average delay is 4.6 ms and the maximum voice station numbers are 10 which is very small number. When fragment the voice frame per 6 WiMAX frame, the delay increases to 29.4 ms and the SS number increase to 27 stations which is close to analytical calculations using the simple scheduler.

Table 9 - Average delay and jitter using UGS service.

Voice fragments per Station numbers Delay (ms) WiMAXframe

1 10 4.6

2 16 9.6

6 27 29.4

VII. Conclusion

In this paper, a study is devoted to compute the VoIP capacity of a mobile WiMAX network and account for various overheads. The study showed that the best VoIP codec from the capacity point of view is G723. Also, the best DL:UL ratio is 3:1.

28th NATIONAL RADIO SCIENCE CONFERENCE (NRSC 2011)

April 26-28, 2011, National Telecommunication Institute, Egypt

Proper scheduling can change the capacity by an order of magnitude. The users should be scheduled so that their number of bursts is minimized while still meeting their delay constraint. This reduce the overhead significantly particularly for small packet traffic such as VoIP. The paper showed that the maximum number of VoIP users in mobile WiMAX network is about 29 user using simple scheduler and 330 users using enhanced scheduler.

References

[1] Chakchai So-In, Raj Jain, Abdel-Karim Al Tamimi, "Capacity Estimation of IEEE 802.l6e Mobile WiMAX Networks", Journal of Computer Systems, Networks, and Communications, Vol. 1, No. 1, April 2010.

[2] Loutfi Nuaymi , "WiMAX: Technology for Broadband Wireless Access", John Wiley & Sons. Section 5-4, 2007.

[3] WiMAXTM System Evaluation Methodology ,WiMAX Forum, Version 2.1, pp 198, July 7, 2008. [4] Shamik Sengupta, Mainak Chatterjee, and Samrat Ganguly. "Improving Quality ofVoIP Streams over

WiMAX". IEEE transactions on computers, vol. 57, no. 2, February 2008, pp 145 -156. [5] Amir Masoud, "Capacity and cell range estimation for multi traffic users in mobile WiMAX ", M.Sc thesis,

University college of Boras, September 2008, pp 26. [6] "VoIP cedecs," On line at:

http://www.cisco.comlenlUS/tech/tk698/technologies _tech _ note09186a0080094ae2.shtml , last visit, March 2010.

[7] Esa Pin, Jarno Pinola, Frerk Fitzek and Kostas Pentikousis, " ROHC and Aggregated VoIP over Fixed WiMAX: An Empirical Evaluation", IEEE Xplore. p.p 1141-1146. November 2008.

[8] Network Simulator - ns2: http://www.isLeduinsnamlns.