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Chapter 2 System Model

2.5 Available Data Rate

For three kinds of transmitter and receiver pairs mentioned before, their SINR computation can be given by:

 The SINR from BS to MS m channel gain from BS to target MS, from BS to target RS, and from RS to target MS in the reference cell, respectively, which are determined by shadowfading pathloss, N0 is the thermal noise density, B is the set of the BSs in neighbor cells that use the same spectrum as the reference BS does, R is the set of active RSs in neighbor cells that use the same spectrum as the reference BS does, C is the concurrency set of RSs in the reference cell, and L L L are channel gain from other cell BS to reference b, r, c cell target MS, from other cell RS to reference cell target MS, and from other transmit concurrent RS to target MS in the same reference cell, respectively.

After SINR is determined, the modulation order Mj for each transmission path can be decided [17] as follows:

whereε is the bit error rate requirement. For a given value m, we can find its corresponding Mj from the table below.

m 0≤m< 4 4≤m<16 16≤m<64 64≤ m

Mj 0 4 16 64

Thus available data rate is then defined as:

1

j

R= M . (8)

Chapter 3

QoS_GTE Scheduling Scheme

As shown in Figure, the proposed QoS-guaranteed and throughput enhancement (QoS_GTE) scheduling scheme contains three parts: the transmission time based (TT_based) path selection algorithm, service order based (SO_based) resource allocation algorithm, and transmission concurrency (TC) decision algorithm.

First, TT_based path selection takes channel state information of all MSs as its input to compute the minimum transmission time for all possible paths and pick up the optimal path for each MS. Under the given path, SO_based resource allocation assigns the transmission time and bandwidth to the MSs according to the either QoS requirements or channel condition of MSs. Eventually, TC decision checks if RSs’

transmissions are able to be combined together. If the concurrency occurs, the occupied resource will be released. This released resource can be assigned to other MSs by SO_based resource allocation algorithm again. The loop between the SO_based resource allocation algorithm and TC decision algorithm will stop when there is no released resource anymore.

,0

Figure 3.1. The block diagram of QoS_GTE Scheduling Scheme

3.1 TT_based Path Selection Algorithm

Relaying technique can increase the received SINR, however, it occupies two transmission time to transmit the same traffic. BS first transmits the traffic to a certain RS then the RS forwards what it receives to the destination MS. The resource using efficiency for relaying is 1/2. From this point of view, we propose transmission time based (TT_based) path selection algorithm both to decide when the system uses relay assistant and which RS is responsible for the transmission.

Different from other path selection methods based on either distance or path-loss [4]-[5], the TT_based path selection algorithm is based on the total transmission time that a path takes, which takes the distance, the path loss, the shadow fading and the interference all into consideration. To make the system throughput high, it is necessary to transmit information within as minimal transmission time as possible.

TT_based path selection algorithm intends to find the minimum transmission time

to increase the system performance.

TT_based path selection algorithm first computes the transmission time for each possible path. Denote Rm i', the available data rate for MS_m from BS to RS_i. Rm i'', is the available data rate for MS_m from RS_i to MS_m. Then the total transmission time from BS through RS_i to MS_m tm i, is

Denote Rm,0 the available data rate through direct path, the transmission from BS to MS_m directly, so the transmission time from BS to MS_m tm o, can be

After transmission time calculating, TT_based path selection algorithm will select a suitable path which transmission time is the minimum for MS_m.

Figure shows the detail TT_based path selection algorithm process for M MSs.

There are Im∈{0,..., }i possible transmitters for MS_m, including i RSs and a BS.

After getting (i+1) different path transmission times, the path with the minimum transmission time is to be chosen. r represents the inter node index, that is, m remarking which RS is selected for the transmission to MS_m. For instance, rm =i means BS will transmit to RS_i then RS_i is going to forward the information to the MS_m. If r =m 0, it means the BS will transmit to MS directly through no RSs.

{0,..., }

Figure 3.2. The flow chart of TT_based path selection algorithm

3.2 SO_based Resource Allocation Algorithm

Since there is a tradeoff between QoS guarantee and system throughput, the proposed service order based (SO_based) resource allocation first serve MSs which are not QoS satisfied to guarantee the QoS; later, MSs with good channel conditions will be served to maximize the system throughput. The resource allocation is start after path selection.

3.2.1 Definition of Sm

In order to maintain the QoS requirement, we define a service order parameter to combine the priority and urgency together for the purpose of lowering the probability that the traffic will violate the QoS constraints. The service order parameter of MS_m at the beginning of the xth-frame, denoted by Sm(x) is defined as, xth-frame, u(x) is the urgency value of the packet to MS_m at the at the beginning of the xth-frame, p* is the maximum value of p(x) and u* is the maximum value of u(x).

Why we consider both priority and urgency is that if only priority value is considered, low priority traffics are backlogged as long as there are traffics with priority higher than them even though these low priority traffics have waited for a long time or are going to violate the constraints. In order to avoid the unfair condition above, we advocate the low priority traffic deserves being served when it is in an emergency condition. We add normalized priority value and normalized urgency value together to decide which traffic should first be served impartially.

It has been mentioned that there are four traffic types considered. Each traffic type is given an integer priority value p(x). The voice traffic is assigned with the highest priority and its priority value is four. The second higher one is the video streaming traffic and its priority value is three. The third one is the HTTP traffic and its priority value is two. The last one is FTP of best effort traffic and its priority value is one.

Assume that each MS can request only one type of traffic service at each time.

However, it would be different among MSs, which have experienced various delay,

even these MSs have same priority value. The urgency value u(x) is brought out. The higher urgency value u(x) of a MS implies that this MS’s packet is having higher possibility to violate the QoS constraints. The u(x) plays an important role in QoS_guaranteed issue and u (x) is defined as,

( ) ( ) ( )

u x =l x +t x , (12)

The l(x) is the urgency value for a packet contributed by the remaining bits which are not transmitted yet at the beginning of the xth –frame, which is definition as:

( ) L Cx 1

l x L

= , (13)

The L is the size of the packet and Cx-1 is the total bits that have been transmitted at the end of the (x-1)th–frame. The t(x) is the urgency value for a packet contributed by the delay time at the beginning of the xth –frame.

In the following, the t(x) of real-time traffic is designed. D* is QoS maximum delay tolerance requirement in the unit for frame. We separate the D* into U* levels.

t(x) is set to be one when the delay of the packet at the xth –frame, d(x), is not greater than D* /2, which implies this packet is the less urgent one. Then we divide the other D* /2 range equally into (U*-1) parts and the value of t(x) is set to be two, three… U*, respectively. The packet with t(x)=U* is approaching the delay upper bound D* and is so urgent to be served immediately. Notice that the method above only works when the delay constraint D* is larger than the level of t (x), denoted as U*. This concept can be expressed in mathematic modeling as below.

( )

fraction of a frame. Since our minimum unit in scheduling is one frame, a fraction of a frame will never occur. For this reason we modified our mechanism when D* is not larger than U*. This concept can be expressed in mathematic modeling as below.

( )

There are total D* +1 levels in t (x) and the difference of delay constraint between any two adjacent levels is one frame, the minimum scheduling unit.

For non real-time traffic, the QoS constraint is the minimum required transmission rate R*. If a packet with size L is transmitted at the rate R*, the transmission time for

where N is the number of frame per second. In other words, D* is the maximum transmission time for a packet with size L. After having D*, all the process of defining t(x) is the same as real-time traffic.

3.2.2 Resource Allocation

For the purpose of QoS guarantee, the MSs can be served first when their Sm(x) are larger than service order threshold Sth. Also, the service order of these high Sm(x) MSs is according to the Sm(x) in descending way, which means that a MS with the higher Sm(x) is able to get what it needs earlier than the one with lower Sm(x). So Sth is a service order lower bound to make sure that the QoS is maintained as possibly as it can. If there is remaining resource after the first step, best first resource allocation is adopted, instead of using Sm(x) to decide the next served MS. Residual free resource is allocated to other MSs who can get the maximum transmitted bits in that channel to enhance the total throughput.

As in Figure, SO_based resource allocation algorithm computes Sm(x) for MS_m in user_set M, denote M is the set of MSs to be served. Therefore the MS which Sm(x) is greater than Sth will be served at first. Later, if there is remaining resource, best first allocation makes use of the remaining resource as well as possible.

m th

if S ≥S Sm

*

* arg max{ m}

m M

m S

=

\ { *}

M =M m

Figure 3.3. The flow chart of SO_based resource allocation algorithm.

3.3 TC Decision Algorithm

Since using relaying may cause resource inefficiency, how to reuses the resource is becoming a critical topic to enhance the system throughput. The TC decision algorithm aims to find which RSs can transmit at the same time using the same resource. Therefore, the original occupied resource can be reserved to other transmitters so that using resource efficiency can be improved, which compensates the drawback of using relaying.

After the SO_based resource allocation algorithm, which RSs are active in the frame has already been decided. Denote I={1,…,i} is the set of active RSs. i is the index of RS. We have to build a concurrency candidate set Ai directing at each active RSi. Any active RSa whose angle separated from RSi is not less than 120∘is included into Ai. In other words, RSa is the concurrency candidate of RSi. For an example, in Figure(a), assume all RSs are active so the concurrency candidates for RS1 are RS5, RS6, RS7, RS8 and RS9, that is, A1={5,6,7,8,9}. Notice that there are at most five concurrency candidates for RSi.

After candidate selection, picking up the exact concurrency RSs from the candidate set is followed. If Ai includes the two candidate RSs which are 120∘separated from RSi, the set α120 is includes these two RSs and these two RSs are chosen as concurrency RSs to transmit with RSi at the same time. If α120= , φ which means the two 120∘RSs are not active at the same time, there is only one candidate RSa in Ai with maximum traffic load ∆ selected as concurrency RS a a i for RSi. As the example illustrated above, RS5 and RS9 are both in Ai so they are concurrency RSs for RS1. The transmit time T for the transmit concurrency RSs is decided by the maximum transmission time of transmit concurrency RSs.

120° 120°

(a)

Downlink frame

Time Sub-channel

TB TRa

RS_

RS_a RS_iRS_k j BS ...

TR

TRi TRj

(b)

(c)

Figure 3.4.Transmission concurrency example. (a) candidate selection (b) an OFDMA frame before concurrency (c) an OFDMA frame

after concurrency

Figure(b) shows an OFDMA frame before concurrency and suppose RSa, RSi, and RSj can transmit concurrently. After transmission concurrency, as Figure(c), RSa, RSi, and RSj are combined together to transmit, which makes original-occupied resources released. The transmission time T is set to be TRa since TRa is the maximum transmission time among TRa, TRi and TRj. The released resources can be utilized again and the assignment of these resources is controlled by SO_based resource allocation algorithm. The detail flow chart of TC decision algorithm is described in Figure. The loop between the SO_based resource allocation algorithm and the TC decision algorithm will never stop until there is no resource released after the TC.

if Ai

if

α

120 =

φ

if I =φ

max{ }

i

i a

a A

a =

120

ai

\ { }i I =I a

Figure 3.5. The flow chart of TC decision algorithm.

Chapter 4

Simulation Results and Discussions

In the simulations, the WiMAX relay-assisted network is set to be compatible with IEEE 802.16 standard. System parameters are listed in Table 4.1, and scalable parameters in physical layer are modified according to the suggested values in [18].

The OFDMA system is based on 5 MHz bandwidth and the frame duration is 5 ms.

The number of sub-carrier is equal to the FFT size, 512, but only 384 sub-carriers are used for data transmission, while the others are used for pilot channel or guard channel. The cell size is set to be 1500 m and the RS coverage is 500 m. BS power is 25 W and each RS power is 5 W and equal power is supposed for each sub-channel.

Table 4.1. System Parameters

Sampling frequency spacing 11.16 kHz

OFDMA symbol duration 100.8 µs

Useful symbol time 89.6 µs

Guard time 11.2 µs

Number of data subcarriers 384

Number of subchannels 24

Number of data subcarriers per subchannel 16 Number of OFDMA symbol for downlink

transmission per frame 24

Thermal noise density (N0) -174 dBm/Hz

Power of BS 25 W

Power of RS 5 W

4.1 Manhattan Street Like Environment

A Manhattan street scenario is considered. The width of each street is 30 meters.

The separation of two adjacent parallel streets is 200 meters. The probability of being LOS and NLOS are given as:

( )

The models of path loss and shadowing are defined differently with respect to the transmitter and receiver pair, BS to MS, BS to RS, and RS to MS, which has already mentioned in (1), (2), (3) in Section 2.4. The corresponding path loss parameters for BS to MS channel, BS to RS channel, and RS to MS channel corresponding path loss parameters are listed in Table 4.2 and Table 4.3. The MS position is distributed uniformly and randomly. By snap-shot method, the MS position will be re-generated after every 10 seconds so that channels also vary after each 10 seconds. Channels are assumed to be fixed within 10 seconds.

Table 4.2. Path-loss and shadow fading parameters Path loss

Table 4.3. Path-loss parameters of channel from RS to MS Path-loss

4.2 Traffic Models and QoS Requirements

The QoS requirements and detailed parameters for the four traffic models are given in this section. Each voice user is modeled as an ON-OFF model, in which the length of ON period and OFF period follows the exponential distribution with mean 1 second and 1.35 seconds, respectively [14]. During ON period, the mean data rate is 12.2 kbps. The video streaming traffic consists of a sequence of video frames which are emitted regularly with interval 100 ms. the video traffic model parameters are listed in Table 4.4.4, where the source data rate is 64 kbps.

Table 4.4. Video streaming traffic model parameters [15]

Parameters Value Parameters Value

T 100 ms M s 250 bytes

N 8 αi 1.2

αs 1.2 K i 2.5 ms

K s 40 bytes M i 12.5 ms

Parameters for HTTP traffic model are given in Table 4.5 [15]. Note that each HTTP packet shall be smaller than 1500 bytes. As to FTP traffic, the size of each file is modeled as truncated lognormal distribution with mean 2 MB, standard deviation 0.722 MB, and a maximum value 5 MB. The arrival interval between two sequential files is exponentially distributed with mean 180 seconds. The parameters of FTP are defined in Table 4.6 [15].

Table 4.5. HTTP traffic model parameters Parameters Value Parameters Value

σm 25032 bytes M e 2M bytes

µm 10710 bytes αNe 1.1

m s 50 bytes K Ne 2

M s 2M bytes MNe 53

σe 126168 bytes λr 0.033

µe 7758 bytes λP 7.69

m e 50 bytes L 1500

Table 4.6. FTP traffic model parameters Parameters Value

σf 0.722M bytes µ f 2M bytes M f 5M bytes

λr 0.006

Table 4.7 lists the QoS requirements of each traffic types and the he transmission time for a frame is 5 ms. In addition, the minimum required transmission rate of HTTP users (in non-real-time services) is slightly larger than the arrival rate of HTTP traffic in page download. This means transmission rate for download of a web page is guaranteed and hence the response time of a web browser is small.

Table 4.7. The QoS requirements of each traffic type

Traffic Type Requirement Value

required BER 10-3

maximum delay tolerance 40 ms (8 frames) voice

(real-time)

maximum allowable dropping ratio 1%

required BER 10-4

maximum delay tolerance 10 ms (2 frames) video

(real-time)

maximum allowable dropping ratio 1%

required BER 10-6

HTTP

(non-real-time) minimum required transmission rate

100 kbps (500 bits per frame)

FTP

(best-effort) required BER 10-6

4.3 Performance Evaluation

In two-hop relay network, it is proven that the least maximum path loss (LMP) relaying node selection scheme [5] has the highest probability in selecting the good channel, which is based on a route that has the lowest bottleneck (in terms of path loss). Let PLn1, PLn2 denote the path loss in dB associated with the first and the second hop, respectively, along the nth route. Then the selected route, rs is determined as follows:

( )

arg min max pathloss of the first hop , pathloss of the second hop .

s n RSs

r =  

We make some modification on LMP scheme in the simulations. That is, after choosing a two hop route rs, it has to compare its bottleneck path loss again with the direct path pathloss. The route with lower bottleneck path loss is finally chosen.

In the simulations, the number of user is composed of four service traffic types with equal percentage. We define the traffic load as the ratio of the total average rate

of all users over the system maximum transmission rate. The maximum transmission rate is achieved when Q users are multiplexed for each sub-channel and the highest modulation order is used for all users. It is equal to 11.74 Mbps in the simulation environment of this thesis. Note that the average arrival rate of voice, video, HTTP, and FTP users is equal to 5.2 Kbps, 64 Kbps, 14.5 Kbps, and 88.9 Kbps, respectively.

Thus, the traffic load varies from 0.15 to 0.9 as the number of users varies from 40 to 240.

For fair comparison, the scheduling algorithm of each relay scheme is SO_based resource allocation algorithm. The level of t(x), denoted as U*, is equal to 6. The threshold value Sth is set to be the average of maximum and minimum Sm of HTTP traffic (Sth=1.0714). Although HTTP is the third priority traffic, HTTP should also own a service right to guarantee the QoS as possibly as it can when HTTP is facing some kind of time emergency. This is why we set Sth as the average of maximum and minimum Sm of HTTP traffic.

4.3.1 Performance Evaluation on Relay schemes

In this simulation, twelve RSs are surrounding the BS uniformly in a circle and RSs are at the cell radius of three fourths. While the desired site location of three-fourth cell radius is not exactly at an intersection, the RS would be placed on the

In this simulation, twelve RSs are surrounding the BS uniformly in a circle and RSs are at the cell radius of three fourths. While the desired site location of three-fourth cell radius is not exactly at an intersection, the RS would be placed on the

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