Ashraf Ali Thesis Report(1)

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Funded and Supervised by:

Intracom Telecom, Hellas.

Prepared By:

Ashraf A. Ali

[email protected]

Supervised By:

Prof. Spyros Vassilaras. (AIT)

Dr. Konstantinos Ntagkounakis (Intracom Telecom)

Submitted on: April 2nd

, 2009

"Broadband Services Modeling Over

WiMAX Access Networks"

Graduation Project Report



Ashraf Ali, "Broadband Services Modeling Over WiMAX Access Networks", MSITT, Athens Greece.

2

Graduate Project

Submitted in partial fulfillment of the graduate requirements for the degree of

Master of Science in Information and Telecommunication Technologies

(MSITT) at the Athens Information Technology Center of Excellence for

Research and Graduate Education.




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Table of Contents

ACKNOWLEDGMENTS ..............................................................................................................4

ABSTRACT .................................................................................................................................5

LIST OF FIGURES AND TABLES.................................................................................................6

1. REAL-TIME SCHEDULING ALGORITHMS .............................................................................7

1.1 INTRODUCTION.......................................................................................................................7 1.2 UNSOLICITED GRANT SERVICE ..................................................................................................8

1.3 REAL TIME POLLING SERVICE .................................................................................................. 8

1.4 ENHANCED REAL-TIME POLLING SERVICE .................................................................................. 9

2. VOIP CODECS .......................................................................................................................12

2.1 G.711 CODEC.....................................................................................................................122

2.2 G.729 CODEC.......................................................................................................................13 2.3 EVRC AND AMR CODECS......................................................................................................13

2.4 G.723.1 CODEC ..................................................................................................................133

2.5 G.722.1 WIDEBAND CODEC ....................................................................................................14 2.6 ILBC (INTERNET LOW BITRATE CODEC) ……………………………………………………...…………14

3. VIDEO STREAMING OVER DATA RATE……………………………………………………...……….14

4. BASIC ETHERNET RATE.....................................................................................................145

4.1 VIRTUAL BRIDGED LOCAL AREA NETWORKS HEADER EFFECT ON BANDWIDTH ................................167

5. PAYLOAD HEADER SUPRESSION ....................................................................................... 188

6. PAYLOAD HEADER COMPRESSION .....................................................................................21

7. VOICE ACTIVITY DETECTION ........................................................................................... 233

8. ETHERNET SERVICES OVER WIMAX ................................................................................256

9. PACKET ENCAPSULATION (TOP-DOWN ANALYSIS) .........................................................277

10. DELAY ANALYSIS .............................................................................................................. 31

10.1 CODEC AND PACKETIZATION DELAY ......................................................................................31

10.2 SCHEDULING ALGORITHM ACCESS DELAY .............................................................................333

UGS access delay................................................................................................................344

rtPS access delay ................................................................................................................36 6

ertPS access delay ...............................................................................................................36 6

11. BANDWIDTH AND DELAY CALCULATOR ……………………………………………………...…..37

12. SIMULATION …………………………………………………………………………………………..…39

12.2 SIMULATION PARAMETERS AND TOPOLOGY……………………………………………….….40

12.3 SIMULATION RESULTS …………………………………………………………………….…………42

13. FUTURE WORK………………………………………………………………………………….………..47

REFERENCES ……………………………………………………………………………...………………….48




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ACKNOWLEDGMENTS

First of all, I would like to thank my supervisors; Prof. Spyros Vassilaras and Dr.

Konstantinos Ntagkounakis for their continuous support and feedback during the project

period; they never hesitated to help me. The thanks goes also to Intracom Telecom (R&D

department) for funding the project and to Broadband Wireless and Sensors Networks

(BWiSE) Research group at AIT for supporting me and treating me as a member of BWiSE

family where I have learned a lot from the supervisors and researchers.




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ABSTRACT

WiMAX, as a wireless broadband access technology, provides the end users with broadband

services similar to that provided by DSL in the traditional wired access networks. Mobile

WiMAX (802.16e) networks are expected to offer voice services to their users, if they want

to successfully compete against cellular networks. However, as WiMAX networks are all-IP

networks, Voice over IP (VoIP) is the means of offering voice services in this environment.

Moreover, the wireless access introduces additional challenges and constraints on the data

rate requirements for higher layers. This created a need for a bandwidth requirements study

for VoIP services over WiMAX which is addressed in this Thesis. To this end, a very useful

tool has been developed for calculating the Ethernet Bandwidth generated by all common

codec types to be injected in the WiMAX link. The user can select the values of several

parameters that affect the generated traffic (type of codec, headers and headers size, etc.).

The effects of Robust Header Compression, Payload Header Suppression, Voice Activity

Detection with Discontinuous Transmission and other bandwidth affecting factors can also

be studied using this tool. In addition, performance analysis of the real time services has been

performed using the ns-2 simulator for the sake of calculating the bandwidth and overhead

introduced by the WiMAX protocols. Finally, the overhead and added delay of using

different uplink scheduling algorithms was analyzed in this Thesis.




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List of Figures and Tables

Figure 1: Grant size based on voice activity ……………………………………………… (8)

Figure 2: UGS Scheduling Algorithm. …………………………………………..………… (8)

Figure 3: rtPS Scheduling Algorithm ……………………………………………………… (9)

Figure 4: ertPS Scheduling Algorithm………………………………..………………….… (10)Figure 5: Packet Transmission Delay…………………………………………………….… (10)

Figure 6: Number of saved uplink and downlink Resources……………………….……. (11)

Figure 7: Scheduling Algorithm for AMR codec…….……………………………………. (11)

Figure 8: 802.1Q header format……………………………………………………………. (17)

Figure 9: DL frame header format…………………………………………………………. (18)

Figure 10: Uplink frame Format………………………………………….………………….. (18)

Figure 11: Suppressed fields in IP, UDP and RTP………..………………………………. (19)

Figure 12: Packet Headers suppressing Algorithm. ………………….…………………… (20)

Figure 13: Ethernet Frame Format………………………………………………..………… (21)

Figure 14: PHS VS. ROHC……………………………………………………………….…… (23)

Figure 15: Voice Sample ……………………………………………………………………… (23)

Figure 16: Voice Activity Model……………………………………………………………… (24)

Figure 17: SID/Voice frames. ………………………………………………………………… (25)Figure 18: VAD effect on the overall average data rate. .………………………………… (26)

Figure 19: 802.1ad packet format……………………………………….…………………… (26)

Figure 20: Mobile WiMAX Network as DSL access Network…………………..………… (27)

Figure 21: PDU formation. …………………………………………………………………… (29)

Figure 22: WiMAX Frame Structure. ………………………………………………..……… (30)

Figure 23: Ethernet over WiMAX infrastructure. ………………………………….……… (30)

Figure 24: Top down Encapsulation………………………………………………………… (31)

Figure 25: codec delay with packing example. …………………………………………..… (33)

Figure 26: the access delay of UGS……………………………………………………..…… (35)

Figure 27: Probability distribution function of x1…………………………………….…… (35)

Figure 28: BW and Delay calculator………………………………………………………… (38)

Figure 29: WiMAX network in PMP mode………………………………………………….. (41)

Figure 30: UGS throughput…………………………………………………………………… (43)Figure 31: nrtPS scheduling throughput…………………………………………………..... (43)

Figure 32: nrtPS BW request polling. ………………………………………………………. (44)

Figure 33: rtPS scheduling throughput……………………………………………………… (44)

Figure 34: rtPS BW polling intervals………………………………………………………… (45)

Figure 35: ertPS scheduling throughput…………………………………………..………… (45)

Figure 36: ertPS and BWREQ throughput……………………………………………..…… (46)

Figure 37: Overrall System throughput………………………………….………….…….… (46)

Table 1: G.722.1 codec types ………………………………………….…….…………….…. (14)

Table 2: Bit rate for different video scales. …………………………….….……………….. (15)

Table 3: Pure Ethernet Data Rate. ………………………………………..……………...…. (16)

Table 4: Required bandwidth for 802.1Q frames. …………………….….…………….…. (17)

Table 5: ROHC compression…………………………………………….….………………... (22)

Table 6: Exponential Voice Activity model…………………………….…………………… (24)

Table 7: SID/CN Payload size………………………………………………..………………. (24)

Table 8: Convergence Sublayer type………………………………………..……………….. (28)

Table 9: Codec Delay…………………………………………………………….……………. (32)

Table 10: Simulation Configuration parameters…………………………………….……. (40)

Table 11: Uplink and Downlink Scheduling………………………………….…………….. (41)




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1. REAL-TIME SCHEDULING ALGORITHMS

In this section an overview of uplink scheduling algorithms will be introduced, where we

will see the overhead introduced by each scheduling type and the tradeoffs among them. We

start with the scheduling algorithms since both in Fixed WiMAX standard (802.16-2004) and

Mobile extension (802.16-2005) the implementation of the uplink scheduling algorithms is

outside the scope of the standard and left for the vendor to implement it based on his needs.

So, we will introduce the scheduling types that we will use in the remaining part of our study.

1.1 Introduction

First of all we will briefly introduce the main scheduling algorithms defined in 802.16

standards. The main objective of the scheduling services is to provide minimum QoS level by

anticipating the throughput and latency needs of the uplink traffic and providing grants at the

appropriate times. UGS, rtPS, and ertPS are the real time scheduling algorithms. There are

also other non-real time scheduling services such as nrtPS, and Best effort BE services.

Those services are clearly not suitable for VoIP and other real time services. However, we

mention here the services for completeness.

In 802.16 there are many scheduling algorithms such as UGS, UGS-AD, rtPS, nrtPS, Lee's

Algorithm, and BE scheduling services. However, these algorithms have some problemssuch as waste of uplink resources, additional access delay, and MAC overhead for supporting

VoIP services with variable data rate and silence suppression. Lee's algorithm for example

does not support the variable data rate codecs such as the Enhanced Variable data Rate

Codec (EVRC) since it has only two states (on - off) where the BS assigns uplink resources

based on the SS voice state transitions. So, when certain codec uses Voice Activity Detection

(VAD) or Silence suppression (SS) the on-off state of the voice can be sent to the MAC layer

using primitives of convergence sublayer in the 802.16 systems. Thus the SS can inform the

BS to grant a bandwidth or not based on the value of certain bit in the MAC header [1].

Figure 1 shows the operation of the Base Station according to the state of the user's voice in

the SS.




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Figure 1: Grant size based on voice activity

1.2 Unsolicited Grant Service

In the Unsolicited Grant Service (UGS) the scheduler provides to the subscriber station

periodic real time fixed grants, so there is no delay at the subscriber side since it has the

needed resources all the time. The size of each grant should be enough to fit the size of the

data for the associated service with all other headers (such as the generic MAC header and

Grant management sub header). Figure 2 shows the operation of UGS for variable rate

codec.

Figure 2. UGS Scheduling Algorithm.

1.3 Real Time Polling Service (rtPS)

Another scheduling service is the real time Polling Service (rtPS). In this scheduling service

the BS assigns periodic variable size data packets at real time. So, this service requires more

overhead but considered more efficient in using the available bandwidth than UGS since it

allows the subscriber station to determine the exact needed bandwidth at every poll. The rtPS

algorithm supports variable data rates; the base station assigns resources that are sufficient

for the requesting mobile station as shown in figure 3. This approach has many advantages;

the base station only assigns the required bandwidth without wasting resources, so it transfer




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the data more efficiently than other fixed bandwidth allocation algorithms such as UGS.

Moreover, there is no need for polling in the silence period since it operates over 1/8th of the

maximum data rate. However, rtPS needs continuous polling each time the mobile station

wants to send data, this introduce MAC overhead and additional access delay. So actually the

rtPS although it supports variable data rate applications, it is considered not the best choice

for real time services such as VoIP.

Figure 3: rtPS Scheduling Algorithm

1.4 Extended Real-Time Polling Service (ertPS)

The extended real time Polling Service (ertPS) is designed as a compromise between

previous two services. So, it provides unicast grants in unsolicited manner like UGS to save

the latency of the bandwidth request. However, it all the allocations are made dynamically

similarly to rtPS service. ertPS tries to overcome all the problems in other scheduling

algorithms, such as the waste of uplink resources, the additional MAC overhead, and the

additional access delay. So it was developed especially for variable data rate with silence

suppression VoIP services. In this algorithm when the size of voice data packet decreases, the

voice user informs the BS of his voice status using grant management sub header, and the

user request the needed bandwidth using extended piggyback request (PBR) bits of the grant

management sub header. So, as a result of this process there is no waste of uplink resources

as the case in the previous algorithms. In case that the size of the voice data packet increased

the user informs the BS of his status voice information using bandwidth (BR) bits in the

bandwidth request header. So, as shown in figure 4, we only need the bandwidth request




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header only when the size of voice data increase which save the polling overhead of rtPS

algorithm. Moreover we get use of piggybacked information to ask for reducing the allocated

bandwidth from the BS.

Figure 4: ertPS Scheduling Algorithm

Figure 5: Packet Transmission Delay

The enhancement in VoIP capacity and saving the uplink resources using ertPS is

investigated in [4] which shows that ertPS can support more VoIP users within certain upper

bound access delay (which is assumed 60 ms) as shown in figure 5. The access delay is the

time needed for the packet to wait in the queue before being scheduled in the frame to be

sent, we will explain it in detail later in delay analysis section. It is clear from figure 5 that

increasing the number of VoIP users will increase the needed access delay which should not

exceed certain limit in order to guarantee certain value for real time services.




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Figure 6: Number of saved uplink and downlink Resources

Moreover, ertPS saves more uplink and downlink resources than the UGS and rtPS since

each resource, which is the needed bits per frame per user, using the same codec type is less

than the other algorithms. Moreover, the saving in the uplink and downlink resourcesincreases as the number of VoIP user increase. Figure 6 shows the saved resources saved in

ertPS compared to UGS and rtPS over different number of users.

Figure 7: Scheduling Algorithm for AMR codec

One of the promising VoIP codecs that was considered by IEEE 802.16 task group m as a

strong candidate for the 802.16 m systems is the Adaptive Multi-rate codec (AMR) [3]. This

codec generate 95-477 bits speech frames every 20 ms, and generate Silence Insertion

Descriptor (SID) frame every 160 ms of the silent period. It is obvious that this type of

codecs can not work with the Lee's algorithm since it causes a waste of uplink resources.

Moreover, since the allocated BW in the silent period of Lee's algorithm is too small, it will

not be able to send the SID frame during the silent period. On the other hand there is no

problem in the rtPS algorithm since the SID frame can request the needed bandwidth from

the BS by explicit request. So, a new algorithm was proposed in [3] that is designed for AMR




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codec. The operation of the algorithm is described in figure 7. in silent period the base

station stops the allocation of periodic grant period, then in case of a talk activity the SS

informs the BS about the voice activity and the needed grant size using Bandwidth-Request-

and-Uplink-Sleep-Control (BRUSC) header, then the BS allocates the grant periodically

every 20 ms and in case of data rate change it can adapt easily to the new rate by sending a

new bandwidth request. On the other hand, in case of a silent period the SS send a request to

the BS using the BR field in order to get a grant to send the SID during the silent period

whenever it is received. As shown in figure 7 the wasted resources are minimized.

2. VoIP CODECS

In order to determine precisely the bandwidth requirements of the VoIP services, we need to

investigate the different Codec types that are being used in nowadays networks, and based on

the studied specification we will be able to exploit the differences and trade ofs among

different codecs; such as needed bandwidth, Codec delay, Quality of voice, and the added

overhead. First of all we will start with brief discussions of most used VoIP codecs and

analyze them then we will make a comparative study among them.

It is important to note that since VoIP is expected to be the killer application of Broadband

access networks, knowing the bandwidth to support the largest number of user within

acceptable QoS guarantees is of a great importance. Hence, we investigated the bandwidth

requirements of different VoIP codecs.

2.1 G.711 Codec

The G.711 is a voice codec that is widely used in different applications [7]. The sampling

rate is 8 KHz and the samples are partitioned into 10 ms frames with bit rate of 64 Kbps

which present a very good voice quality, however in large scale usage may cause a

bottleneck in the network parts that cannot adapt to the large bandwidth requirements. So, in

order to enhance the performance of this algorithm in packet based multimedia applications

such as VoIP, a modified version of the base algorithm was proposed in G.711 Appendix II

[8]. The algorithm use Discontinuous Transmission (DTX), Voice activity Detection (VAD),

and Comfort Noise Generation (CNG) Algorithms in order to reduce the bit rate in a packet

based applications and hence increase the bandwidth efficiency. Table II.1/G.711 in [8]




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shows the bandwidth saving can reach to 39% reduced bandwidth compared to the basic

G.711 Algorithm.

2.2 G.729 Codec

G.729 operates on speech frames of 10 ms corresponding to 80 samples at a sampling rate of

8000 samples per second with 8 KHz sample rate, but it needs 5 ms look ahead delay which

makes the total algorithmic delay 15 ms [5]. Annex A of the standard (G.729 a) has the same

main features of G.729 but considered as a reduced complexity version. It has been

developed for multimedia simultaneous voice and data applications. However a silence

suppression scheme is developed as Annex b of the standard (G.729 b), it defines a voice

activity detector (VAD) and comfort noise generator (CNG) for G.729 in order to reduce the

transmission rate during the silence period. So, G.729b has been heavily used in VoIP

applications for a long time.

2.3 EVRC and AMR Codecs

There is also other type of codecs that has different bit rates. EVRC and AMR are examples

of such codec types. As described in the previous section the Adaptive Multi-rate (AMR)

speech coded is one of the strong candidates for 802.16 m [3]. The AMR codec consists of

eight source codecs with bit-rates of 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbps.

the coder operates over 20 ms frames where in each frame, 95, 103, 118, 134, 148, 159, 204

or 244 bits are produced, corresponding to a bit-rate of 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.2

or 12.2 kbit/s. On the other hand, EVRC changes the data rate from the full rate to 1/8th of

the full rate based on the voice activity. EVRC compresses each 20 milliseconds of 8000 Hz,

16-bit sampled speech input into output frames in one of the three different sizes: Rate 1 (171

bits), Rate 1/2 (80 bits), or Rate 1/8 (16 bits).

2.4 G.723.1 Codec

G.723.1 is an example of low rate coders; this coder has two bit rates associated with it 5.3

and 6.3 kbps it can switch between the two rates at the frames boundary The frame size is 30

ms with look ahead of 7.5 ms, resulting in a total algorithmic delay of 37.5 ms. The encoder

operates on blocks (frames) of 240 samples each. That is equal to 30 ms at an 8-kHz




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sampling rate. Each frame holds 189 bits in case of 6.3 Kbps and 158 bits in case of 5.3 Kbps

[18].

2.5 G.722.1 Wideband Codec

G.722.1 digital wideband coder algorithm provides an audio bandwidth of 50 Hz to 7 kHz,

operating at a bit rate of 24 kbps or 32 kbps. It operates on 20-ms frames (320 samples) of

audio with look-ahead buffer size of 20 ms. The frame size is 480 bits for 24 Kbps and 640

bits for 32 Kbps. G.722.1 Annex C [19] the sampling rate doubles from 16 to 32 kHz, and it

supports three data rates as shown in table 1.

Table 1: G.722.1 codec types.

2.6 iLBC (internet Low Bitrate Codec)

iLBC is suitable for robust voice communication over IP. The codec is designed for narrow

band speech and results in a payload bit rate of 13.33 kbps with an encoding frame length of

30 ms and 15.20 kbps with an encoding length of 20 ms. iLBC codec enables graceful speech

quality degradation in the case of lost frames, which occurs in connection with lost or

delayed IP packets.

3. VIDEO STREAMING OVER DATA RATE

In this section we will show the bandwidth need of video streaming applications. However,

in the following sections we will focus on only VoIP as example of real time service. In

video streaming services, we need to provide a flexible bitstream adaptation for multimedia

services. Hence, a new category of video coding called Scalable Video Coding (SVC) is used




16

Where TS is the Total Frame Size, RTP is RTP header, IP is IP header, UDP is UDP header,

Eth. is Ethernet Header, CRC is CRC header, SS is Voice Sample Size, NoS is Number of

Samples per Frame, DR is Data Rate, and PPS is Packets Per Second.

Table 3: Pure Ethernet Data Rate.

Codec Sample

Time

(ms)

Frame

Size

(bits)

PPS IP

Header

(bits)

RTP

Header

(bits)

UDP

Header

(bits)

Ethernet

Header+ CRC

(bits)

Total

FrameSize

(bits)

Total

RateKbps

G.711 10 640 100 160 96 64 144 1104 110.40

G.729 10 80 100 160 96 64 144 544 54.40

G.723.1

5.3 Kbps 30 158 33 160 96 64 144 622 20.52

G.723.1

6.3 Kbps

30 189 33 160 96 64 144 653 21.54

G.722.1

24 Kbps 20 480 50 160 96 64 144 944 47.20

G.722.1

32 Kbps 20 640 50 160 96 64 144 1104 55.20

G.722.1C

48

Kbps*

20 960 50 160 96 64 144 1424 71.20

EVRC

Rate1 20 172 50 160 96 64 144 636 31.80

EVRC

Rate1/2 20 80 50 160 96 64 144 544 27.20

EVRC

Rate1/8 20 16 50 160 96 64 144 480 24

AMR

4.75

Kbps

20 95 50 160 96 64 144 559 27.95

AMR

12.2

Kbps

20 244 50 160 96 64 144 708 35.40




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4.1 Virtual Bridged Local Area Networks header effect on Bandwidth

With 802.1Q standard the formation of virtual LANs inside the network is allowed and thus

the inter-communication between different VLANs is also possible using three layer switches

or routers. However, this implies that we need extra headers to be added to the Ethernet

headers in order to support VLAN tagging process. Figure 8 shows the frame format of the

802.1Q standard where it shows that 32 bit extra header is added. (TPID, PCP, CFI, VID).

[23]

Figure 8: 802.1Q header format

The Tag Protocol Identifier (TPID) is a 16 bit field that is always set to 0x8100 in order to

identify the 802.1Q frame. The Priority Code Point (PCP) is a three bit field that is used to

priorities the real time traffic. The Canonical Format Indicator (CFI) is one bit field that

indicate whether the MAC address format is canonical or not. And finally, the Virtual LAN

Identifier (VID) is a twelve bit field that indicates the Virtual LAN number the packet

belongs to.

Table 4 shows the new bandwidth requirements after counting the additional 802.1Q header.

The increase in the bandwidth required ranges from 2.2% to 6.6% based on the codec used.

Table 4: Required bandwidth for 802.1Q frames.

Codec G.711 G.729

G.723.1

5.3

G.723.1

6.3

G.722.1

24

G.722.1

32

G.722.1

C

48

EVRC

Rate1

EVRC

Rate1/2

EVRC

Rate1/8

AMR

4.75

AMR

12.2

Total

Size

1136 576 654 685 976 1136 1456 668 576 512 591 740

Total

Rate

113.6 57.6 21.5 22.6 48.8 56.8 72.8 33.4 28.8 25.6 29.55 37

Percent

change

2.8% 5.8% 5.1% 4.9% 3.3% 2.8% 2.2% 5.0% 5.8% 6.6% 5.7% 4.5%




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5. PAYLOAD HEADER SUPPRESSION

Wireless resources are considered scarce. Hence, we need to save the resources using header

compression and suppression since there is redundancy in information and we need to save

the bandwidth to increase the capacity of the system. But, before discussing the suppression

and compression, we will first study the overhead at the MAC and Physical layer in order to

have idea about the wasted bandwidth due to the added overhead.

Figure 9: DL frame header format

The overheads in the downlink and uplink frames are shown in figure 9 and figure 10. As

shown in figure 9, there are many overheads. In the physical layer there are Preamble, FCH,

Broadcast Messages, and Padding. And in the MAC layer there are MAC header, MAC

subheaders (optional), and CRC. On the other hand figure 10 shows the overheads such as

the contention slots, and MAC headers at the UL physical PDU.

Figure 10: Uplink frame Format

As discussed previously, the need for saving the bandwidth and reducing the overhead

becomes very important. So, next we will investigate the methods proposed for this purpose

in VoIP. In Payload Header Suppression (PHS) the redundant parts due to the higher layers

in the payload header of MAC SDU are suppressed and then returned at the receiver side.

Those headers are RTP, UDP, and IPv6 headers. Which adds 60 -120 bytes of overhead. It is

important to note that only the constant parts of the headers within the same flow are




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suppressed, the suppressed fields in RTP, UDP, and IPv6 are shown in figure 11. The entire

shaded field can be suppressed, so as shown in [13] we can reduce the 60 byte headers to 15

bytes using PHS in the maximum case and in average to 30 bytes.

Figure 11: Suppressed fields in IP, UDP and RTP

When the PHS is enabled, each MAC SDU is prefixed with Payload Header Suppression

Index (PHSI) which references the Payload Header Suppression Field (PHSF) [11]. The

process of suppressing the PDU header is shown in figure 12.




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Figure 12: Packet Headers suppressing Algorithm.

Since the PHS rule determine the fields that have to be suppressed and this rule differs from

one packet to another; hence the suppression and thus the affect on the total packet size will

differ from one packet to another, Figure 13 shows the all the headers in the Ethernet PDU.

We assume that Ethernet Header (14 bytes) is suppressed; 37 bytes are suppressed from IP




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field in case that IPv6 is used as indicated by figure 11. And 8 bytes are suppressed in case

IPv4 is used which represent the source and destination addresses. Moreover, in the UDP

header 4 bytes are suppressed as indicated in figure 11 that represent the source and

destination port addresses. And from the RTP header we can suppress the SSRC identifier (4

bytes) as shown in figure 11.

Based on the above assumptions we can reduce the total header (Ethernet, IP, UDP, RTP,

CRC) from 58 bytes to 30 bytes in case we use IPv4. On the other hand, we reduce the total

header from 78 to 59 in case of IPv6. However, we add only one byte as a Payload Header

Suppression Index (PHSI).

Figure 13: Ethernet Frame Format

So the Total Size of the Ethernet Frame is given as follows:

( ). * SB+PHSIPHS

TS RTP IP UDP Eth CRC SS NoS = + + + + + − ………….. (3)

6. PAYLOAD HEADER COMPRESSION

Another way to save bandwidth and reduce the overhead is using header compression. The

CS may use one of compression algorithms to compress the header and decompress again at

the receiver side. There are many header compression protocols, Robust Header

Compression (ROHC) protocol is one of the best algorithms that outperform other protocols

since it has the ability to tolerate high loss rate in wireless connections [14]. The protocol has

different values of compression efficiency based on the operation mode. There are three

modes of operation; the Unidirectional, the Bidirectional Optimistic, and the Bidirectional

Reliable mode.

The basic concept of ROHC is to find the redundancies in packet headers. IP, UDP, and RTP

headers all have many redundancies within the same packet (intra-redundancy) and between

different packets within the same service flow (inter-redundancy). Moreover, the changes in

many fields are incremental between consecutive packets. So we can efficiently suppress and




22

compress many fields in the headers [24]. The compression algorithm of ROHC is very

complicated and as mentioned previously the compression ratio and efficiency depends on

redundancy among packets. However in [24] it was found that compression ratio for

IPv4/UDP/RTP headers is 2.5% to 5% which means that the 40 bytes header is compressed

to 1~2 bytes. For simplicity we will assume that the size of the header after compression is 2

bytes [25]. On the other hand Table 5 shows that using the high efficiency compression

RTP/UDP/IPv6 headers were reduced from 60 bytes to 4 bytes. So, generally specking

ROHC compresses the header to less than 10% of the original size.

Table 5. ROHC compression

The Saved Bytes using ROHC is:

( ) ROHC SB IP UDP RTP H = + + − ………………………………………………… (4)

Where SB is the saved bytes after using ROHC and ROHC H

is the new header size in bytes,

which depends on the version of IP used.

And the Total size of the Ethernet frame is given as follows:

( ). * SB ROHC

TS RTP IP UDP Eth CRC SS NoS = + + + + + − ……………………… (5)

Based on the previous discussion, we compared between ROHC, PHS, and the basic Ethernet

Bitrate. As shown in figure 14 there is a considerable saving in bandwidth in PHS. But the

improvement added by ROHC is small. Hence, we prefer using PHS due to its computational

simplicity.




23

Figure 14. PHS VS. ROHC

7. VOICE ACTIVITY DETECTION

The Voice Activity Detection (VAD) or Silence Suppression (SS) is a technique used in

many codecs in order to save the VoIP bandwidth. So, VoIP frames will not be generated

continuously for each user, since there are silent periods and talk periods as shown in the

voice sample in figure 15.

Figure 15. Voice Sample

It was found statistically the mean duration of both ON period and OFF periods follow the

exponential distribution parameters given in table 6.

0

20

40

60

80

100

120

G . 7

1 1

G . 7

2 9

G . 7

2 3 . 1

5 . 3

K b p s

G . 7

2 3 . 1

6 . 3

K b p s

G . 7

2 2 . 1

2 4 K b p s

G . 7

2 2 . 1

3 2 K b p s

G . 7

2 2 . 1

C

4 8 K b p s

E V R C

R a t e 1

E V R C

R a t e 1 / 2

E V R C

R a t e 1 / 8

A M R 4 . 7

5

K b p s

A M R 1 2 . 2

K b p s

B i t R a t e ( K

b p s )

Default Ethernet Bit Rate PHS Bit Rate ROHC Bit Rate




24

Table 6: Exponential Voice Activity model

Each source will alternate between OFF-state and ON-state for certain period as shown in

figure 16.

Figure 16: Voice Activity Model

It is important to note that during silence periods, the Codec send Silence Insertion

Descriptor (SID) in order to send the information of the background noise to the codec at the

receiver side. The SID frame compresses information for sounds that are audible for the

human ear perception. The payload size for different types of codecs is shown in table 7. It is

important to note that this is only the Payload size without packing. So we have to consider

also IP/UDP/RTP headers and the link layer added header for the final packet size.

Moreover, the SID frames are not sent successively back to back. It is usually sent either

based on considerable change in the background noise or uniformly.

Table 7: SID/CN Payload size

CodecG.71

1

G.72

9

G.723.1

5.3

G.723.1

6.3

EVRC

Rate1

EVRC

Rate1/2

EVRC

Rate1/8

AMR

4.75

AMR

12.2

SID/CN

Payload 8* 16 32 32 16 16 16 40 40




25

Actually, the size of SID payload is very small. However after adding the headers of

Ethernet/IP/UDP/RTP the size will be considerable and must be included in the calculations

of average bit rate over both the ON and OFF periods as shown in figure 17.

Figure 17. SID/Voice frames.

The voice rate , SID rate, and average total rate is given as follows:

……………………….. (6)

……………..………… (7)

……………...………… (8)

WhereV

R ,v

L ,SID

R ,SID

L , Avg

R ,η , and DTX

F are the Voice rate, length of the voice sample,

SID rate, length of SID frame, average total rate, voice activity factor, and discontinuous

transmission ratio.

We calculated the average data rate of several Codecs assuming the voice activity follows

exponential distribution and the SID frames are sent uniformly ( DTX F =50%). The result is

shown in figure 18.

. . .(1 )

vV

SID

SID

Avg V SID DTX

L R

T

L R

T

R R R F η η

=

=

= + −




26

Figure 18. VAD effect on the overall average data rate.

8. ETHERNET SERVICES OVER WiMAX

WiMAX networks need to support transparent Ethernet, so we need to add extensions to

support the interfacing between the wired Ethernet-based DSL and the WiMAX. IEEE

802.1ad-2005 is one of the standards that offer the capabilities of IEEE 802.1Q virtual

bridged LANs to a number of customers with no need for alignment across the customers.

The format of the standard is shown in figure 19.

Figure 19: 802.1ad packet format

As shown in figure 19, the standard adds another layer of addressing to 802.1Q. The

Customer VID (C-VID) and the Service Provider VID (S-VID) both are used to address

traffic between the end customer and service providers. This is called VLAN stacking or Q-

in-Q which is the most widely used in Ethernet Carriers in service provider networks

especially in DSL networks. The different modes of Q-in-Q networks are mentioned in [26].

It is important to know that the Mobile WiMAX Networks can support the Ethernet services

and/or the IP services together as we will see in the following sections. WiMAX network can

emulate the behavior of DSL networks in terms of connectivity between customer interface

(T interface) and the service Provider Interface (V interface). The idea is shown in figure 20.




27

Figure 20: Mobile WiMAX Network as DSL access Network

9. PACKET ENCAPSULATION (TOP-DOWN ANALYSIS)

The VoIP frames generated by the codec are encapsulated at the MAC layer. There are two

layers in the MAC layer; the Convergence Sublayer (CS), the MAC Common Part Sublayer,

and the MAC Security Sublayer. The CS receives data packets named Service Data Units

(SDU) from the higher layer. This layer acts as an adaptation layer that performs certain

operations, such as header compression, on SDUs based on the operating protocol in the

higher layer. Table 8 shows the different higher layer protocol convergence sublayers that are

supported in WiMAX. However, only the IP and Ethernet (802.3) CS are implemented in

WiMAX Forum.




28

Table 8: Convergence Sublayer types

One of the optional implementations of the CS is to perform Payload Header Suppression

(PHS) on SDUs; this will save the bandwidth by removing redundant data after merging

several SDUs together. So, this operation is used in bandwidth sensitive services such as

VoIP. The suppressed part of the header depends on the associated service. For example, in

VoIP applications the repetitive part of the header includes source IP, destination IP, and the

length indicator.

On the other hand, the Common Part Sublayer works independently from the higher layers. It

receives the SDUs from the higher layers and assembles or fragments the SDUs depending

on their size to make MAC Payload Data Units (PDU) after adding the MAC header. The

PDUs are scheduled in the frame as a Downlink or Uplink Burst which shares a common

quality of service among different PDUs and considered the basic payload field in the

WiMAX frame. Figure 21 clarifies the idea.




29

Figure 21: PDU formation.

As shown in figure 21 the MAC header is added to every PDU. There are two types of

headers; the generic WiMAX header, and the bandwidth request header. The generic PDU is

used to carry data, such as VoIP frames, and the MAC layer signaling messages. The size of

the generic MAC header is 6 bytes. On the other hand the bandwidth request PDU is used by

the subscriber to request certain amount of bandwidth from the base station, but the

bandwidth request PDU does not carry CRC or payload. It is important to note that there are

also other sub headers that follow the generic MAC header and used for specific purposes.

After adding the MAC header, the PDUs are passed to the scheduler which uses a scheduling

algorithms to allocate the needed bandwidth over the physical resources based on the quality

of service requirements, channel state and fairness. The scheduler determines the optimum

usage of the physical resources on frame by frame bases in order to meet the channel

coherence time. VoIP PDUs for example need minimum amount of bandwidth based on the

voice encoding standard being used. Hence the scheduler must ensure that this minimum BW

is satisfied for all allocated VoIP users within the WiMAX frame. The frame structure is

shown in figure 22. Moreover, the scheduler must be able to select the sub carriers for each

user to satisfy the highest throughput over good channel conditions. However, it is important

to note that there is no specific scheduling procedure to be used, and it was left for the

manufacturer to implement the scheduling approach. But we can say that there are two broad

scheduling categories in WiMAX; (1) Algorithms that minimize the total transmission power

with a constraint on user data rate, and (2) Algorithms that maximize the total data rate with a

constraint on the total transmit power. So there is trade off between the maximum total data




30

rate and the total transmitted power. However, most WiMAX algorithms fall in the second

category such as: Maximum Sum Rate Algorithm, Maximum Fairness Algorithm,

Proportional Rate Constraints Algorithm, and Proportional Fairness Scheduling.

Figure 22: WiMAX Frame Structure.

As mentioned previously, WiMAX forum also supports Ethernet frame Convergence

Sublayer. We need Ethernet over WiMAX in order to offer Ethernet transparent connectivity

over wireless access network, for example in areas where there is no wired infrastructure as

shown in figure 23. Where the service provider operates over Ethernet network and the

Ethernet packets are encapsulated over WiMAX network.

Figure 23: Ethernet over WiMAX infrastructure.

It is important to note that the support of Ethernet does not require a new network

architecture since it is smoothly supported into the Mobile WiMAX network architecture and

can operate in parallel with IP services in the same network.

Figure 24 shows the general layered hierarchy starting from the voice encoding until the

802.16 frame formation process [20].




31

Figure 24: Top Down Encapsulation

10. DELAY ANALYSIS

In this section we will analyze the delay of VoIP from the subscriber station to the core

network. It is described in ITU-T G.114 [15] that the mouth to ear delay for real time

services such as VoIP should not exceed 150 ms in order not to be significantly affected.

However, delays that exceed 400 ms are not acceptable for real time services. Next in this

section we will describe all delay components that contribute to the total mouth to ear delay,

starting from the codec and packetization delay, scheduling delay, access and transmitting

delay, network delay, and receiver side delay

10.1 Codec and Packetization Delay

The ITU-T G.114 standard describes the codec delay. The codec wait for certain speech

sample time before processing and compressing in a frame, the frame will not be generated

until all the voice samples are collected by the encoder. This time differs from one codec to

another. Moreover, some codecs, as described in previous sections, has a look a head time to

improve the compression efficiency. So, the codec delay is equal to the frame size plus the




32

look ahead delay (if exist). We ignore the delays due to encapsulation and adding headers to

pass the frame to the MAC layer since it is very small and negligible. Table 9 shows the

sample and look ahead delay for each codec.

Table 9: Codec Delay.

Codec Type Sample Time Look ahead time

G.711 10 0

G.729 10 5

G.723.1 5.3 Kbps 30 7.5

G.723.1 6.3 Kbps 30 7.5

G.722.1 24 Kbps 20 20

G.722.1 32 Kbps 20 20

G.722.1C 48 Kbps 20 20

EVRC Rate1 20 10

EVRC Rate1/2 20 10

EVRC Rate1/8 20 10

AMR 4.75 Kbps 20 5

AMR 12.2 Kbps 20 5

It is important to know in 802.16 the convergence sublayer (CS) may pack multiple frames

from the higher layers (i.e. VoIP SDUs) in one PDU. If this is the case, then we have to wait

an additional frame time before generating the PDU to the physical layer. The idea is

explained in [15] but for IP networks, I use the same figure 25 in the reference to clarify the

idea.




33

Figure 25: codec delay with packing example.

In figure 25 the CS pack two frames in each PDU. In this case the delay increase by one

frame time compared to the normal case where one frame is packed in every PDU.

10.2 Scheduling Algorithm Access Delay

After studying the codec and packetization delay, we are now ready to analyze the

scheduling algorithm delay. We will analyze the downlink scheduling and uplink scheduling

delay.

In downlink, the BS scheduler looks into each peer node and tries to schedule data from its

connections to the downlink sub-frame. The service of connections is performed in a round

robin way. As long as there is free space in the downlink sub-frame, the scheduler will assign

a data burst to the connection. Once the free space in the downlink sub-frame is over, the

scheduler will remember where it has stopped, and in the next frame it will continue to assign

bursts from there. So, as long as there is enough space for the data in the downlink sub frame,

the scheduler will send it immediately [20].

In uplink, the BS scheduler will first reserve space for contention slots. The information

about the bandwidth needs is stored in the peer node objects in the MAC layer. When

scheduling uplink transmissions, the scheduler will look at this information from

connections, and will use a round robin procedure as for the downlink part. During this

process, it will also add 1 OFDM symbol (= short preamble) between bursts. The SS

scheduler is responsible for taking the data transmission opportunities allocated by the BS

and for assigning them to the appropriate incoming connections. If bandwidth is assigned to a

given connection, the SS scheduler will also fill the bursts with the data from its outgoing

connections. If the SS has not an outstanding request, the SS scheduler looks at all outgoing




34

connections and generates bandwidth requests, sent in the contention phase. Each request is

of the aggregated type [20].

The access delay depends on the scheduling service. We will analyze mainly three real time

scheduling services; UGS, rtPS, and ertPS.

UGS access delay

In UGS Algorithm, the base station will provide the subscriber station with fixed sized grants

that are negotiated during connection establishment phase. So, the subscriber station can

always send the data to the base station. Hence, the access delay for this scheduler is the

same as the MAC frame duration. However, it is important to note that this applies as long as

the number of subscriber station is within the maximum number of stations the UGS can

support. Otherwise, an additional queuing delay will be added.

The access delay is given by the equation 9; where the average time is calculated rather than

the max time in the worst case.

………………………..(9)

Wheretx_ugsT , DLT ,

ttgT ,

ULT , 1 x , and 2 x , are the UGS transmission delay, Downlink sub frame

duration, Transmission transition gap, Uplink sub frame duration, random variable that

represent the arrival of voice frames from higher layer as multiple of slot duration, random

variable represent multiples of slot duration in the uplink interval.

From the above equation we note that x1 ranges from 0 slots until the integer value of

maximum number of slots in the downlink sub frame. Assuming using Partial Usage Sub

Carrier (PUSC) in both the uplink and downlink intervals, the slot duration in the downlink

sub frame equals three symbols and two symbols in the uplink interval. Figure 26 shows an

example of WiMAX super frame with total period of 5 ms. So, in this example the duration

of downlink sub frame equivalent to 8 slots and 10 slots in the uplink sub frame.

tx_ugs DL 1 2

ULDL1 2

T = [T .2 ] [ .3 ]

TTwhere x {0, }, and x {0, }

2. 3.

sym ttg sym

sym sym

x T T x T

T T

− + +

∈ ∈




35

Figure 26: The access delay of UGS

Where 1( ) f x and similarly 2( ) f x is represented as shown in figure 27 as a uniform

distribution function. Hence, the probability of frame arrival is uniform over the downlink

and uplink periods.

Figure 27 Probability distribution function of x1

Finally, we calculate the Average transmission duration by taking the expectation of equation

9 as follows:

( ) ( )tx_ugs tx_ugs DL 1 2

tx_ugs DL 1 2

T ( .) T [T .2 ] [ .3 ]

T ( .) [T .2 ] [ .3 ]

sym ttg sym

sym sym ttg

Avg E E x T T x T

Avg E x T x T T

= = − + +

= − + +

………….…. (9)

Assuming uniform probability distribution function, we get the following:

DLtx_ugs

TT ( .)

2 2

ULttg

T Avg T = + + ………………………………………….….. (10)




36

rtPS access delay

This algorithm, as mentioned previously, is used to support real-time service flows that

generate variable size data packets periodically. Hence, the BS assigns uplink resources that

are sufficient for unicast bandwidth requests to the voice user. This period is negotiated in theinitialization process of the voice sessions. Generally, this process is called a bandwidth

request process, or polling process. This bandwidth request process always causes MAC

overhead and additional access delay. Hence, the rtPS algorithm has larger MAC overhead

and access delay than the UGS and ertPS algorithms.

The transmission delay over rtPS scheduling is given in the following equation:

……………………………………(11)T tx_rtPS = T mf + T tx_ugs

Which means that the transmitter has to wait one MAC frame time for polling BS and then

he transmit his data in the uplink field similar to the UGS case. Thus the difference between

the UGS and the rtPS delays is the MAC frame time (Tmf ).

ertPS access delay

This 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. This algorithm is

recently proposed and accepted in IEEE 802.16e standard. In order not only to reduce MAC

overhead and access delay of the rtPS algorithm, but also to prevent the waste of uplink

resources of the UGS algorithm, the ertPS algorithm assigns uplink resources according to

the status of the voice users without MAC overhead. The user requests the bandwidth for

sending the voice packets using extended PBR (Piggyback Request) bits of Grant

Management sub header. In the ertPS 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 the voice user requests another

size of the bandwidth.




37

So, [22] shows that the access delay for this algorithm follows the same equation for that of

UGS since it uses piggybacking algorithm that affect the overhead not the access time. So

ertPS access time is given be the following equation:

……………………………(12)T

tx_ertPS

= T

tx_ugs

11. BANDWIDTH AND DELAY CALCULATOR

Based on all the previous bandwidth and delay analysis, we developed a flexible bandwidth

and subscriber to core network delay calculator. Using this tool, which is shown in figure 28,

the user can do the following:

1) Set the header values (such as selecting between IPv4 and IPv6) adding the 802.1Q

header.

2) Select one of the fourteen supported Codecs and select the number of samples per

frame (which affect the total delay and data rate).

3) Moreover, the user can see the parameters associated with selected codec type such as

number of packets per second, the frame duration and size.

4) Select the operating Convergence Sublayer service (either Ethernet over WiMAX or

IP over WiMAX)

5) Select one of the compression and suppression techniques (PHS, ROHC, or none) and

based on the selected value display the saved bytes after using suppression or

compression.

6) Select whether Voice Activity detection is used or not, and set all the associated

parameters with it such as: Voice Activity factor, and discontinuous transmission

ratio. And based on that display the SID frame size and the average number of saved

bits using VAD.

7) Select whether the headers needed to provide DSL like services over WiMAX are

included or not.

8) Display the MAX and Average final required data rate based on the selected values

described above.




38

9) Set the WiMAX super frame duration and set the percent of the Downlink interval to

the total frame duration.

10) Display the look a head delay, Sample time, Processing delay and total Codec delay

based on the selected codec type.

11) Display the scheduling delay for all real time scheduling types as discussed before.

12) Finally, Display the total subscriber to core network delay.

Figure 28 BW and Delay calculator




39

12. SIMULATION

We used the network simulator ( NS2) in order to study the uplink scheduling algorithms and

its performance. Section 11.1 will present a brief overview of NS2, Section 11.2 will present

the simulation environment and topology, and finally some results of the scheduling

algorithms overhead will be presented in section 11.3.

12.1 Network Simulator ( NS2)

NS (version 2) is an object-oriented, discrete event driven network simulator developed at

UC Berkely and written in C++ and OTcl (Tcl script language with Object-oriented

extensions). NS2 implements network protocols such as TCP and UDP, traffic source

behaviour such as FTP, Telnet, Web, CBR and VBR, router queue management mechanism

such as Drop Tail, RED and CBQ, routing algorithms such as Dijkstra, and more. NS also

implements multicasting and some of the MAC layer protocols for LAN simulations.

NS-2 includes a tool for viewing the simulation results, called NAM. NAM is a Tcl/TK

based animation tool for viewing network simulation traces and real world packet trace data.

The first step to use NAM is to produce the trace file. The trace file should contain topology

information, e.g., nodes, links, as well as packet traces. Usually, the trace file is generated by

NS. During an ns simulation, user can produce topology configurations, layout information,

and packet traces using tracing events in ns. When the trace file is generated, it is ready to be

animated by NAM. Upon start-up, NAM will read the trace file, create topology, pop up a

window, do layout if necessary, and then pause at the time of the first packet in the trace file.

Through its user interface, NAM provides control over many aspects of animation

The official NS2 module does not support WiMAX MAC. In order to support WiMAX over

NS2 we used the module developed by Networks & Distributed Systems Laboratory (NDSL)

centre, which is part of the Chang Gung University in Taiwan. The module supports Point to

Multipoint MAC mode (PMP) which is needed in our case for the connectivity between the

base station and surrounding subscriber stations. The module does not support mobility,

hence the modulation and coding schemes does not change based on the distance changing

between the subscriber and base station.




40

12.2 Simulation Parameters and Topology

Using the NDSL module described before, we have simulated a small network of one base

station surrounded by several subscriber stations. The parameters of the simulation are shown

in Table 10. We used 5 ms WiMAX frames with 3:1 downlink to uplink duration. It is

important to note that since the NDSL module used does not support mobility, the

modulation schemes are not adaptive and fixed over the simulation time. In other words, the

modulation and coding does not change when the distance between the subscriber and base

station changes. Moreover the nodes are fixed in positions and hence the transmission power

is fixed regardless of the position from the base station.

Table 10: Simulation Configuration parameters.

Total number of subcarriers 2048

Frame duration 5 ms

RTG duration 0.02941 ms

TTG duration 0.02941 ms

OFDMA symbol duration 0.10084 ms

DL subframe 37 OFDMA symbols and 60 subchannels

UL subframe 12 OFDMA symbols and 96 subchannels

Utilized DL/UL modulation types {QPSK ½, QPSK ¾, 16-QAM ½, 16-

QAM ¾, 64-QAM 2/3, 64-QAM ¾

Number of base stations 1

Number of subscriber stations 5

Simulations Area 500 m * 500 m

Simulation time 10 s




41

The Topology of the network is shown in figure 29.

Figure 29. WiMAX network in PMP mode

The uplink and downlink traffic fro each node is described in Table 11. We used the uplink

scheduling algorithms described before for both the uplink and downlink directions.

Table 11: Uplink and Downlink Scheduling

Node Downlink Uplink

Subscriber Station 1 rtPS UGS

Subscriber Station 2 UGS ertPS(1)

Subscriber Station 3 ertPS rtPS

Subscriber Station 4 nrtPS nrtPS

Subscriber Station 5 UGS ertPS(2)

(1)

Simulate G.711 VoIP traffic with exponential ON and OFF periods with VAD

(2)

Simulate G.729 VoIP traffic with exponential ON and OFF periods with VAD




42

There are applications running over the scheduling algorithms described in the table above.

For example we modified the code of the exponential traffic model in NS2 to simulate a

VoIP application with exponential ON and OFF periods as discussed before. Moreover we

set the type of the packets to ertPS in order to be scheduled over the predefined ertPS

scheduler class in the NDSL module. Based on the packet size, rate, the silence interval and

talk interval, which are the parameters that characterize the exponential traffic in NS2, we

can simulate any VoIP codec that implement VAD with any silence and activity periods

distribution. We named the newly defined traffic model as “VoIP_traffic” and can be used

inside the TCL script by referring to the TCL class “Application/traffic/VoIP_traffic”

For the UGS scheduler we run Constant Bit Rate traffic (CBR) over it which is

characterized in NS2 by the rate and packet size. For the rtPS we used a variable bit rate

traffic that changes the size of the sent packet from 200 bytes till 1000 bytes with constant

generation or inter-arrival time. And finally, for the nrtPS scheduler, we used a traffic model

that generates large epochs of data over regular time interval. All the traffic models will be

shown in the next subsection.

12.3 Simulation results

Based on the previously defined and used traffic models and schedulers we have simulated

the network for 10 seconds and based on the generated trace files with the help of analyzing

tool called Trace Graph v 2.02 [28] we got some performance metrics; such as the throughput

of the scheduling service and the overhead of the bandwidth request headers introduced by

each scheduling type.

For example the throughput of the UGS scheduling service is shown in figure 30. As

expected, it is constant and we did not find any polling overhead represented by bandwidth

request messages. Similarly, the throughput of nrtPS scheduling as described before and the

bandwidth request polling intervals are shown in figures 31 and 32 respectively. It is clear

that the BW polling occurs whenever there is nrtPS data to be sent by the subscriber station

(the overhead is represented as number of sent BW request messages over time).




43

Figure 30. UGS throughput

Figure 31. nrtPS scheduling throughput




44

Figure 32. nrtPS BW request polling.

Figures 33 and 34 show the throughput and BW request overhead for the rtPS scheduling

service. The ertPS throughput in figure 33 is averaged over 0.1 s and we note that due to the

continuous change in the generated rate from the subscriber station, the polling of bandwidth

request messages is continuous as described in the rtPS scheduling services section.

Figure 33. rtPS scheduling throughput




45

Figure 34. rtPS BW polling intervals

The throughput of the ertPS scheduling service and the Bandwidth request headers are shown

in figures 35. Both values are averaged over 1 second in order to show increase in BW

requirements over considerable amount of time instead of showing the fast changes. The

ertPS traffic simulates a VoIP application with exponential ON and OFF time periods as

shown in figure 36. Here we assume that there is no traffic during the silence period for

simplicity. As noticed in the figure 36 there are six periods of BW request overhead increase.

This increase corresponds to the data rate increase of the VoIP traffic since we mentioned in

the ertPS scheduling section that we only send BW request header whenever the rate of the

service increases.

Figure 35. ertPS scheduling throughput




46

Figure 36. ertPS and BWREQ throughput

Finally, Figure 37 shows the total and uplink and downlink throughput of the system for all

the scheduling services used during the simulation time. We notice that although we used the

same scheduling services in both the uplink and downlink directions, the downlink traffic is

more stable and offer service to the subscribers in steady behavior. On the other hand the

uplink traffic throughput based on the changes of the traffic needs (depends on the service

behavior) and the allocated bandwidth (depends on the scheduling service and BS).

Figure 37. Over all System throughput




47

13. FUTURE WORK

Based on the results presented in the previous section, we find a gap between the throughput

of the sent data by the subscriber station and that received from the base station. So, we need

to mind the gap as much as we can. This means that a need for a new uplink scheduling

algorithm which maximize the throughput and minimize the polling overhead with minimum

access delay is needed. However, developing such algorithm is not an easy task especially

that it tries to overcome all the shortcoming of already exist uplink scheduling algorithms.

Moreover, so far we have used a module that supports only fixed WiMAX topology where

the positions of the nodes are fixed and hence the modulation and coding schemes are

predefined regardless the change in the node positions or channel response. This is not the

ideal case where the main target of the 802.16e standard is to support mobility for end users.

And by mobility we don’t only mean moving nodes but also a dynamic topology that

supports handoff mechanisms between the subscriber stations and different base stations.

Hence, we are looking for testing different communication scenarios for end to end

connectivity over dynamic WiMAX communication system. For such dynamic topology, we

intend to calculate the overall system capacity in terms of number of users per cell.




48

REFERENCES

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[2] Howon Lee, Taesoo Kwon and Dong-Ho Cho "Extended-rtPS Algorithm for VoIP

Services in IEEE 802.16 Systems". IEEE Communications Conference. Volume 5. 2006

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49

[10] Jeffery G. Andrews, Arunabha Ghosh, and Rias Muhamed, “Fundamentals of

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[23] IEEE Standard “802.1Q IEEE Standard for Local and metropolitan area networks

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[27] Hung-Hui Juan, Hsiang-Chun Huang, ChingYao Huang, and Tihao Chiang." Scalable

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Ashraf Ali Thesis Report(1)

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