Pub Date : 2008-12-01DOI: 10.1109/ATNAC.2008.4783296
I. Palma-Lázgare, J. A. Delgado-Penín
The high-data rate links in broadband wireless communications are being essential for a constant growth in the tough environment of radio transmissions, and orthogonal frequency division multiplexing (OFDM) can deal with these circumstances. Coded OFDM (COFDM) research in wireless communications is a concept of the well-use spectrum for robust high-data rate transmissions, and its regulation in the IEEE and ITU may have a profitable contribution in our high altitude platform (HAP) study. Moreover, HAP station (HAPS) based systems are now taking part in the world of wireless technologies to carry on with the anywhere and anytime wireless network service considerations. Due to all last considerations, HAP-channel modelling and COFDM-HAPS' performance evaluations conform our study. For our system representation the HAP-based system and ground users are considered as fixed terminals. Herein, our stratospheric channel modelling is approximated to real transmission conditions via the experimental land mobile satellite (LMS) record adoption. Performance results by means of BER vs SNR simulations are plotted and show that our proposal can overcome the wireless link effects of frequency selectivity and multipath fading; our results can offer an alternative idea of an efficient and robust solution for distribution of broadband wireless communications.
{"title":"Fixed Broadband Wireless Access based on HAPS using COFDM Schemes: Channel Modelling and Performance Evaluation","authors":"I. Palma-Lázgare, J. A. Delgado-Penín","doi":"10.1109/ATNAC.2008.4783296","DOIUrl":"https://doi.org/10.1109/ATNAC.2008.4783296","url":null,"abstract":"The high-data rate links in broadband wireless communications are being essential for a constant growth in the tough environment of radio transmissions, and orthogonal frequency division multiplexing (OFDM) can deal with these circumstances. Coded OFDM (COFDM) research in wireless communications is a concept of the well-use spectrum for robust high-data rate transmissions, and its regulation in the IEEE and ITU may have a profitable contribution in our high altitude platform (HAP) study. Moreover, HAP station (HAPS) based systems are now taking part in the world of wireless technologies to carry on with the anywhere and anytime wireless network service considerations. Due to all last considerations, HAP-channel modelling and COFDM-HAPS' performance evaluations conform our study. For our system representation the HAP-based system and ground users are considered as fixed terminals. Herein, our stratospheric channel modelling is approximated to real transmission conditions via the experimental land mobile satellite (LMS) record adoption. Performance results by means of BER vs SNR simulations are plotted and show that our proposal can overcome the wireless link effects of frequency selectivity and multipath fading; our results can offer an alternative idea of an efficient and robust solution for distribution of broadband wireless communications.","PeriodicalId":143803,"journal":{"name":"2008 Australasian Telecommunication Networks and Applications Conference","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125995817","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2008-12-01DOI: 10.1109/ATNAC.2008.4783309
C. Kikkert, O. Kenny
This paper describes the Digital Signal Processing (DSP) techniques required to ensure that a Ka band Satellite Beacon Receiver/Radiometer remains locked to the satellite beacon and provide a measure of both the sky noise and the attenuation caused by rain to satellite signals. To enable the receiver to stay locked to satellite beacon signals with received power levels between -110 dBm and -170 dBm, a receiver with an ultra low phase noise, an absence of spurious signals and using advanced Digital Signal Processing (DSP) techniques is required.
{"title":"Digital Signal Processing for a Ka Band Satellite Beacon Receiver / Radiometer","authors":"C. Kikkert, O. Kenny","doi":"10.1109/ATNAC.2008.4783309","DOIUrl":"https://doi.org/10.1109/ATNAC.2008.4783309","url":null,"abstract":"This paper describes the Digital Signal Processing (DSP) techniques required to ensure that a Ka band Satellite Beacon Receiver/Radiometer remains locked to the satellite beacon and provide a measure of both the sky noise and the attenuation caused by rain to satellite signals. To enable the receiver to stay locked to satellite beacon signals with received power levels between -110 dBm and -170 dBm, a receiver with an ultra low phase noise, an absence of spurious signals and using advanced Digital Signal Processing (DSP) techniques is required.","PeriodicalId":143803,"journal":{"name":"2008 Australasian Telecommunication Networks and Applications Conference","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126011728","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2008-12-01DOI: 10.1109/ATNAC.2008.4783325
Jeong Han Jeong, Moohong Lee, Byungjik Keum, Jungkeun Kim, Y. S. Shim, Hwang-Soo Lee
To decode a broadcasting signal such as a T-DMB signal using a software baseband receiver running on a digital signal processor (DSP), real-time input data buffering is important. A time offset of each received frame, which is caused by a difference in the sampling frequency between the transmitter and the receiver, makes input buffer management difficult, eventually resulting in the performance deterioration of the receiver. This work proposes an input data buffering scheme based on a ring buffer for a T-DMB software baseband receiver running on a DSP. The time offset of each received frame is estimated by a time synchronization block using a phase reference symbol and is used by a buffer controller to control the ring buffer so that the receiver on the DSP always reads valid data for data decoding. The validity of the proposed scheme is confirmed by showing that the ring buffer never goes into an overflow state when buffering the input data with a time-varying time offset. Thus, the specified receiver performance is guaranteed over time.
{"title":"A Real-time Input Data Buffering Scheme Based on Time Synchronization for a T-DMB Software Baseband Receiver","authors":"Jeong Han Jeong, Moohong Lee, Byungjik Keum, Jungkeun Kim, Y. S. Shim, Hwang-Soo Lee","doi":"10.1109/ATNAC.2008.4783325","DOIUrl":"https://doi.org/10.1109/ATNAC.2008.4783325","url":null,"abstract":"To decode a broadcasting signal such as a T-DMB signal using a software baseband receiver running on a digital signal processor (DSP), real-time input data buffering is important. A time offset of each received frame, which is caused by a difference in the sampling frequency between the transmitter and the receiver, makes input buffer management difficult, eventually resulting in the performance deterioration of the receiver. This work proposes an input data buffering scheme based on a ring buffer for a T-DMB software baseband receiver running on a DSP. The time offset of each received frame is estimated by a time synchronization block using a phase reference symbol and is used by a buffer controller to control the ring buffer so that the receiver on the DSP always reads valid data for data decoding. The validity of the proposed scheme is confirmed by showing that the ring buffer never goes into an overflow state when buffering the input data with a time-varying time offset. Thus, the specified receiver performance is guaranteed over time.","PeriodicalId":143803,"journal":{"name":"2008 Australasian Telecommunication Networks and Applications Conference","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127397732","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2008-12-01DOI: 10.1109/ATNAC.2008.4783357
Sureshkumar V. Subramanian, R. Dutta
In recent years, transmission of packets over Internet has been a real alternative to the traditional Public Switched Telephone Networks (PSTN). The Internet protocol (IP) offers more flexibility in the design and implementation of features and services. The session initiation protocol (SIP) is a commonly adopted signaling protocol for Voice over IP (VoIP) by many telecommunication industries. Since the signaling system of PSTN, signaling system number 7 (SS7) was designed for high reliability, but IP works on the "best effort" basis which motivated us to study the performance characteristics of SIP control plane. In this paper, we studied an M/M/1 performance model of the SIP proxy server, showed its limitations, and designed an alternative M/D/1 performance model that enhances the SIP proxy server performance.
{"title":"Comparative Study of M/M/1 and M/D/1 Models of a SIP Proxy Server","authors":"Sureshkumar V. Subramanian, R. Dutta","doi":"10.1109/ATNAC.2008.4783357","DOIUrl":"https://doi.org/10.1109/ATNAC.2008.4783357","url":null,"abstract":"In recent years, transmission of packets over Internet has been a real alternative to the traditional Public Switched Telephone Networks (PSTN). The Internet protocol (IP) offers more flexibility in the design and implementation of features and services. The session initiation protocol (SIP) is a commonly adopted signaling protocol for Voice over IP (VoIP) by many telecommunication industries. Since the signaling system of PSTN, signaling system number 7 (SS7) was designed for high reliability, but IP works on the \"best effort\" basis which motivated us to study the performance characteristics of SIP control plane. In this paper, we studied an M/M/1 performance model of the SIP proxy server, showed its limitations, and designed an alternative M/D/1 performance model that enhances the SIP proxy server performance.","PeriodicalId":143803,"journal":{"name":"2008 Australasian Telecommunication Networks and Applications Conference","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134356383","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2008-12-01DOI: 10.1109/ATNAC.2008.4783297
Jian Cao, M. Gregory
In this paper, we evaluate the performance of Voice over Internet Protocol (VOIP) services that use different compression and decompression (CODEC) schemes, over a hybrid network that includes a Universal Mobile Telecommunications System (UMTS) network segment. We focus on the VoIP transmission end-to-end delay. We found that different CODECs provide very different results depending on the hybrid network. The research found that for VoIP services to operate over a hybrid network including a UMTS network segment, with an end-to-end delay comparable to that of circuit switched voice service, there is a requirement for careful comparison and design on choosing the CODEC scheme.
{"title":"Performance Evaluation of VoIP Services using Different CODECs over a UMTS Network","authors":"Jian Cao, M. Gregory","doi":"10.1109/ATNAC.2008.4783297","DOIUrl":"https://doi.org/10.1109/ATNAC.2008.4783297","url":null,"abstract":"In this paper, we evaluate the performance of Voice over Internet Protocol (VOIP) services that use different compression and decompression (CODEC) schemes, over a hybrid network that includes a Universal Mobile Telecommunications System (UMTS) network segment. We focus on the VoIP transmission end-to-end delay. We found that different CODECs provide very different results depending on the hybrid network. The research found that for VoIP services to operate over a hybrid network including a UMTS network segment, with an end-to-end delay comparable to that of circuit switched voice service, there is a requirement for careful comparison and design on choosing the CODEC scheme.","PeriodicalId":143803,"journal":{"name":"2008 Australasian Telecommunication Networks and Applications Conference","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129258910","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2008-12-01DOI: 10.1109/ATNAC.2008.4783287
J. Arnold, Olaf Maennel, A. Flavel, Jeremy McMahon, M. Roughan
Newly announced IP addresses (from previously unused IP blocks) are often unreachable. It is common for network operators to filter out address space which is known to be unallocated ("bogon" addresses). However, as allocated address space changes over time, these bogons might become legitimately announced prefixes. Unfortunately, some ISPs still do not configure their bogon filters via lists published by the Regional Internet Registries (RIRs). Instead, they choose to manually configure filters. Therefore it would be desirable to test whether filters block legitimate address space before it is allocated to ISPs and/or end users. Previous work has presented a methodology that aims at detecting such wrongly configured filters, so that ISPs can be contacted and asked to update their filters. This paper extends the methodology by providing a more formal algorithm for finding such filters, and the paper quantitatively assesses the performance of this methodology.
{"title":"Quantitative Analysis of Incorrectly-Configured Bogon-Filter Detection","authors":"J. Arnold, Olaf Maennel, A. Flavel, Jeremy McMahon, M. Roughan","doi":"10.1109/ATNAC.2008.4783287","DOIUrl":"https://doi.org/10.1109/ATNAC.2008.4783287","url":null,"abstract":"Newly announced IP addresses (from previously unused IP blocks) are often unreachable. It is common for network operators to filter out address space which is known to be unallocated (\"bogon\" addresses). However, as allocated address space changes over time, these bogons might become legitimately announced prefixes. Unfortunately, some ISPs still do not configure their bogon filters via lists published by the Regional Internet Registries (RIRs). Instead, they choose to manually configure filters. Therefore it would be desirable to test whether filters block legitimate address space before it is allocated to ISPs and/or end users. Previous work has presented a methodology that aims at detecting such wrongly configured filters, so that ISPs can be contacted and asked to update their filters. This paper extends the methodology by providing a more formal algorithm for finding such filters, and the paper quantitatively assesses the performance of this methodology.","PeriodicalId":143803,"journal":{"name":"2008 Australasian Telecommunication Networks and Applications Conference","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125613415","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2008-12-01DOI: 10.1109/ATNAC.2008.4783331
A. Naghshegar, A. Darehshoorzadeh, A. Dana, M. Dehghan
The topology of wireless multihop ad hoc networks can be controlled by varying the transmission power of each node. Topology control is the problem of changing node's transmission power in ad hoc networks so it maintains a specified network topology while minimizing energy consumption and increasing life time. In this paper, we changed the criteria of choosing neighbors in neighbor-based topology control XTC over AODV routing for mobile ad hoc networks and evaluated the effect of them with different parameters.
{"title":"Dynamic Topology Control Scheme in MANETs for AODV Routing","authors":"A. Naghshegar, A. Darehshoorzadeh, A. Dana, M. Dehghan","doi":"10.1109/ATNAC.2008.4783331","DOIUrl":"https://doi.org/10.1109/ATNAC.2008.4783331","url":null,"abstract":"The topology of wireless multihop ad hoc networks can be controlled by varying the transmission power of each node. Topology control is the problem of changing node's transmission power in ad hoc networks so it maintains a specified network topology while minimizing energy consumption and increasing life time. In this paper, we changed the criteria of choosing neighbors in neighbor-based topology control XTC over AODV routing for mobile ad hoc networks and evaluated the effect of them with different parameters.","PeriodicalId":143803,"journal":{"name":"2008 Australasian Telecommunication Networks and Applications Conference","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124921008","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2008-12-01DOI: 10.1109/ATNAC.2008.4783348
I. Ahmad, J. Kamruzzaman, D. Habibi, Farzana Islam
Resource sharing between book-ahead (BA) and instantaneous request (IR) reservation often results in high preemption rate of on-going IR calls. High IR call preemption rate causes interruption to service continuity which is considered as detrimental in a QoS-enabled network. A number of call admission control models have been proposed in literature to reduce the preemption rate of on-going IR calls. Many of these models use a tuning parameter to achieve certain level of preemption rate. This paper presents an artificial neural network (ANN) model to dynamically control the preemption rate of on-going calls in a QoS-enabled network. The model maps network traffic parameters and desired level of preemption rate into appropriate tuning parameter. Once trained, this model can be used to automatically estimate the tuning parameter value necessary to achieve the desired level of preemption rate. Simulation results show that the preemption rate attained by the model closely matches with the target rate.
{"title":"An Intelligent Model to Control Preemption Rate of Instantaneous Request Calls in Networks with Book-Ahead Reservation","authors":"I. Ahmad, J. Kamruzzaman, D. Habibi, Farzana Islam","doi":"10.1109/ATNAC.2008.4783348","DOIUrl":"https://doi.org/10.1109/ATNAC.2008.4783348","url":null,"abstract":"Resource sharing between book-ahead (BA) and instantaneous request (IR) reservation often results in high preemption rate of on-going IR calls. High IR call preemption rate causes interruption to service continuity which is considered as detrimental in a QoS-enabled network. A number of call admission control models have been proposed in literature to reduce the preemption rate of on-going IR calls. Many of these models use a tuning parameter to achieve certain level of preemption rate. This paper presents an artificial neural network (ANN) model to dynamically control the preemption rate of on-going calls in a QoS-enabled network. The model maps network traffic parameters and desired level of preemption rate into appropriate tuning parameter. Once trained, this model can be used to automatically estimate the tuning parameter value necessary to achieve the desired level of preemption rate. Simulation results show that the preemption rate attained by the model closely matches with the target rate.","PeriodicalId":143803,"journal":{"name":"2008 Australasian Telecommunication Networks and Applications Conference","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125512919","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2008-12-01DOI: 10.1109/ATNAC.2008.4783356
P. Simoens, P. Praet, B. Vankeirsbilck, J. De Wachter, L. Deboosere, F. De Turck, B. Dhoedt, P. Demeester
In a thin client computing architecture, application processing is delegated to a remote server rather than running the application locally. User input is forwarded to the server, and the rendered images are relayed through a dedicated remote display protocol to the user's device. Existing remote display protocols have been successfully optimized for applications with only minor and low-frequent screen updates, such as a spreadsheet or a text editor. However, they are not designed to cope with the fine-grained and complex color patterns of multimedia applications, leading to high bandwidth requirements and an irresponsive user interface. In this article, a hybrid remote display protocol approach is presented. The existing Remote FrameBuffer protocol of Virtual Network Computing (VNC-RFB) protocol is leveraged with a video streaming mode to transport the rendered images of multimedia applications to the client. Dependent on the amount of motion in the images to be presented, the images are relayed to the client either through the VNC-RFB protocol or through video streaming in the H.264 format. The architecture of this hybrid image renderer is presented and the implementation is detailed. Furthermore, the decision heuristic to switch between the VNC-RFB and the streaming mode is discussed. Experimental results clearly show the advantage of the hybrid approach in terms of client CPU and bandwidth requirements.
{"title":"Design and implementation of a hybrid remote display protocol to optimize multimedia experience on thin client devices","authors":"P. Simoens, P. Praet, B. Vankeirsbilck, J. De Wachter, L. Deboosere, F. De Turck, B. Dhoedt, P. Demeester","doi":"10.1109/ATNAC.2008.4783356","DOIUrl":"https://doi.org/10.1109/ATNAC.2008.4783356","url":null,"abstract":"In a thin client computing architecture, application processing is delegated to a remote server rather than running the application locally. User input is forwarded to the server, and the rendered images are relayed through a dedicated remote display protocol to the user's device. Existing remote display protocols have been successfully optimized for applications with only minor and low-frequent screen updates, such as a spreadsheet or a text editor. However, they are not designed to cope with the fine-grained and complex color patterns of multimedia applications, leading to high bandwidth requirements and an irresponsive user interface. In this article, a hybrid remote display protocol approach is presented. The existing Remote FrameBuffer protocol of Virtual Network Computing (VNC-RFB) protocol is leveraged with a video streaming mode to transport the rendered images of multimedia applications to the client. Dependent on the amount of motion in the images to be presented, the images are relayed to the client either through the VNC-RFB protocol or through video streaming in the H.264 format. The architecture of this hybrid image renderer is presented and the implementation is detailed. Furthermore, the decision heuristic to switch between the VNC-RFB and the streaming mode is discussed. Experimental results clearly show the advantage of the hybrid approach in terms of client CPU and bandwidth requirements.","PeriodicalId":143803,"journal":{"name":"2008 Australasian Telecommunication Networks and Applications Conference","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131763822","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2008-12-01DOI: 10.1109/ATNAC.2008.4783343
K. Zen, D. Habibi, I. Ahmad
In the IEEE 802.15.4 medium access control (MAC) protocol for wireless sensor networks, a sensor node needs to associate with a coordinator before it starts sending or receiving data. The sensor node will mostly choose the nearest coordinator to associate with. However, this method is not suitable for a constantly moving sensor node because it will end up switching coordinators too often due to short connectivity time. The IEEE 802.15.4 has a simplistic and inadequate method of choosing a coordinator in this context. In this paper, we introduce a method to increase the mobile sensor node connectivity time with its co-ordinator in IEEE 802.15.4 beacon-enabled mode. Our method is based on the timestamp of the beacons received from the nearby coordinators and filtering weak beacon signals. By choosing the coordinator which has sent the most recent received beacon with good signal quality, we increase the moving node connectivity time with the coordinator. Our technique results in significant improvement by reducing the number of times the moving node switches coordinators. This increases the throughput and reduces the wasted power in frequent associations.
{"title":"Improving Mobile Sensor Connectivity Time in the IEEE 802.15.4 Networks","authors":"K. Zen, D. Habibi, I. Ahmad","doi":"10.1109/ATNAC.2008.4783343","DOIUrl":"https://doi.org/10.1109/ATNAC.2008.4783343","url":null,"abstract":"In the IEEE 802.15.4 medium access control (MAC) protocol for wireless sensor networks, a sensor node needs to associate with a coordinator before it starts sending or receiving data. The sensor node will mostly choose the nearest coordinator to associate with. However, this method is not suitable for a constantly moving sensor node because it will end up switching coordinators too often due to short connectivity time. The IEEE 802.15.4 has a simplistic and inadequate method of choosing a coordinator in this context. In this paper, we introduce a method to increase the mobile sensor node connectivity time with its co-ordinator in IEEE 802.15.4 beacon-enabled mode. Our method is based on the timestamp of the beacons received from the nearby coordinators and filtering weak beacon signals. By choosing the coordinator which has sent the most recent received beacon with good signal quality, we increase the moving node connectivity time with the coordinator. Our technique results in significant improvement by reducing the number of times the moving node switches coordinators. This increases the throughput and reduces the wasted power in frequent associations.","PeriodicalId":143803,"journal":{"name":"2008 Australasian Telecommunication Networks and Applications Conference","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2008-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123378574","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
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