Muayad S. Al-Janabi, C. Tsimenidis, B. Sharif, S. L. Goff
This paper presents a joint adaptive modulation and coding (AMC) scheme, which exploits the coherence bandwidth of the wireless channel to divide the transmitted frame into independent sub-channels that correspond to the channel coherence bandwidth. This strategy is implemented for orthogonal frequency division multiplexing (OFDM) systems with channel state information (CSI) feedback. Low density parity check (LDPC) codes are utilized for encoding by employing signal-to noise ratio (SNR) dependent coding rates, as well as distinct modulation schemes to achieve adaptivity to time-varying channel conditions. The performance of the proposed system was tested on Rayleigh fading channels that exhibit frequency coherent bands. Numerical results obtained via simulation demonstrate that the throughput of the proposed system and bit error rate (BER) performance are better than previously suggested approaches.
{"title":"Adaptive MCS Selection in OFDM Systems Based on Channel Frequency Coherence","authors":"Muayad S. Al-Janabi, C. Tsimenidis, B. Sharif, S. L. Goff","doi":"10.1109/AICT.2009.37","DOIUrl":"https://doi.org/10.1109/AICT.2009.37","url":null,"abstract":"This paper presents a joint adaptive modulation and coding (AMC) scheme, which exploits the coherence bandwidth of the wireless channel to divide the transmitted frame into independent sub-channels that correspond to the channel coherence bandwidth. This strategy is implemented for orthogonal frequency division multiplexing (OFDM) systems with channel state information (CSI) feedback. Low density parity check (LDPC) codes are utilized for encoding by employing signal-to noise ratio (SNR) dependent coding rates, as well as distinct modulation schemes to achieve adaptivity to time-varying channel conditions. The performance of the proposed system was tested on Rayleigh fading channels that exhibit frequency coherent bands. Numerical results obtained via simulation demonstrate that the throughput of the proposed system and bit error rate (BER) performance are better than previously suggested approaches.","PeriodicalId":409336,"journal":{"name":"2009 Fifth Advanced International Conference on Telecommunications","volume":"90 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122279734","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}
Buffering capability is added within each routing node of the modified Data Vortex network with the 4-ary routing scheme. In two different cases, either with single packet or with two packets routing within the node, the effect of buffering capability is studied in such networks. The results have confirmed that with additional hardware cost in buffering, the performance can be enhanced as that in the original binary routing Data Vortex networks. While buffering based on single packet routing allows for modest performance improvement, buffering based on two packets routing allows for much better latency and throughput performance compared with buffer-less networks, and such improvement is also more significant than the performance enhancement in binary decoding Data Vortex networks with the same buffering capabilities.
{"title":"Effect of Buffering Capability in Data Vortex Networks Based on 4-ary Routing Scheme","authors":"Qimin Yang","doi":"10.1109/AICT.2009.76","DOIUrl":"https://doi.org/10.1109/AICT.2009.76","url":null,"abstract":"Buffering capability is added within each routing node of the modified Data Vortex network with the 4-ary routing scheme. In two different cases, either with single packet or with two packets routing within the node, the effect of buffering capability is studied in such networks. The results have confirmed that with additional hardware cost in buffering, the performance can be enhanced as that in the original binary routing Data Vortex networks. While buffering based on single packet routing allows for modest performance improvement, buffering based on two packets routing allows for much better latency and throughput performance compared with buffer-less networks, and such improvement is also more significant than the performance enhancement in binary decoding Data Vortex networks with the same buffering capabilities.","PeriodicalId":409336,"journal":{"name":"2009 Fifth Advanced International Conference on Telecommunications","volume":"35 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116664464","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}
Self similar behavior of the aggregate traffic is a well known issue in the networking area. In this paper, we first study the self similarity of the empirically aggregated VoIP traffic in a heterogeneous wireless network testbed environment and then model it using Fractional Gaussian Noise (fGn). The heterogeneity of the environment is provided by exploiting different wireless technologies in backbone and access networks. The backbone of the testbed is IEEE 802.16d WiMAX whereas the access network is IEEE 802.11b WiFi mesh architecture. We evaluate the self similarity in terms of throughput and packet inter-arrival time using empirically captured VoIP calls generated by softphones in the laboratory. We prove the collected data’s self similar characteristics with stochastic analysis using autocorrelation functions. We implement three well-known time domain estimators to obtain Hurst values for both metrics. We also suggest the Fractional Gaussian Noise (fGn) Model for the empirically aggregated VoIP data. The self similarity analysis and modeling performed in this work will motivate new design issues on the quality of service frameworks and resources allocation mechanisms such as buffers in wireless heterogeneous networks.
{"title":"Self Similarity Analysis and Modeling of VoIP Traffic under Wireless Heterogeneous Network Environment","authors":"B. Canberk, S. Oktug","doi":"10.1109/AICT.2009.19","DOIUrl":"https://doi.org/10.1109/AICT.2009.19","url":null,"abstract":"Self similar behavior of the aggregate traffic is a well known issue in the networking area. In this paper, we first study the self similarity of the empirically aggregated VoIP traffic in a heterogeneous wireless network testbed environment and then model it using Fractional Gaussian Noise (fGn). The heterogeneity of the environment is provided by exploiting different wireless technologies in backbone and access networks. The backbone of the testbed is IEEE 802.16d WiMAX whereas the access network is IEEE 802.11b WiFi mesh architecture. We evaluate the self similarity in terms of throughput and packet inter-arrival time using empirically captured VoIP calls generated by softphones in the laboratory. We prove the collected data’s self similar characteristics with stochastic analysis using autocorrelation functions. We implement three well-known time domain estimators to obtain Hurst values for both metrics. We also suggest the Fractional Gaussian Noise (fGn) Model for the empirically aggregated VoIP data. The self similarity analysis and modeling performed in this work will motivate new design issues on the quality of service frameworks and resources allocation mechanisms such as buffers in wireless heterogeneous networks.","PeriodicalId":409336,"journal":{"name":"2009 Fifth Advanced International Conference on Telecommunications","volume":"83 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129139962","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}
In the paper we introduce a group members arrangement as a new parameter for analyzing multicast routing algorithms finding multicast trees. We also propose a new multicast routing algorithm without constraints. The objective of STA (Switched Trees Algorithm) is to minimize the total cost of the multicast tree using a modification of the classical Prim's algorithm (Pruned Prim's Heuristic) and the SPT (Shortest Path Tree) algorithm that constructs a shortest path tree between a source and each multicast node.
{"title":"Analysis of the Influence of Group Members Arrangement on the Multicast Tree Cost","authors":"M. Piechowiak, M. Stasiak, P. Zwierzykowski","doi":"10.1109/AICT.2009.77","DOIUrl":"https://doi.org/10.1109/AICT.2009.77","url":null,"abstract":"In the paper we introduce a group members arrangement as a new parameter for analyzing multicast routing algorithms finding multicast trees. We also propose a new multicast routing algorithm without constraints. The objective of STA (Switched Trees Algorithm) is to minimize the total cost of the multicast tree using a modification of the classical Prim's algorithm (Pruned Prim's Heuristic) and the SPT (Shortest Path Tree) algorithm that constructs a shortest path tree between a source and each multicast node.","PeriodicalId":409336,"journal":{"name":"2009 Fifth Advanced International Conference on Telecommunications","volume":"58 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133781911","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}
The Measurement-based Admission Control is used to achieve the required Quality of Service. The Measurement-based Admission Control mechanism provides significant functionality for integrated service guaranties. This paper presents the model of the traffic analyzer for real-time applications. It consists of two cross-dependent sub-modules: traffic measurement and traffic estimator. The model presented in the paper is characterized by low system overheads for real-time traffic parameters estimation.
{"title":"Real-Time Traffic Analyzer for Measurement-Based Admission Control","authors":"M. Kulikovs, E. Petersons","doi":"10.1109/AICT.2009.18","DOIUrl":"https://doi.org/10.1109/AICT.2009.18","url":null,"abstract":"The Measurement-based Admission Control is used to achieve the required Quality of Service. The Measurement-based Admission Control mechanism provides significant functionality for integrated service guaranties. This paper presents the model of the traffic analyzer for real-time applications. It consists of two cross-dependent sub-modules: traffic measurement and traffic estimator. The model presented in the paper is characterized by low system overheads for real-time traffic parameters estimation.","PeriodicalId":409336,"journal":{"name":"2009 Fifth Advanced International Conference on Telecommunications","volume":"49 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133765349","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}
In a VoIP (Voice over Internet Protocol) telephone system, we face the significant challenge of providing a teleconference feature that can support a large-scale teleconference without using excessive network bandwidth. This paper examines a new, bandwidth-efficient way of implementing a real-time VoIP teleconference system, and we present measurements of the improvements that we gained by refining our original concept. Our newly refined teleconference technique provides all of the features that previous teleconference systems provide, but our new approach consumes considerably less data bandwidth than previous systems require. The new system allows a network with a given capacity to accommodate nearly double the number of conference participants that an existing system would allow.
在VoIP (Internet Protocol Voice over Internet Protocol)电话系统中，我们面临的一个重大挑战是如何在不使用过多网络带宽的情况下提供支持大规模电话会议的电话会议功能。本文研究了一种新的、带宽高效的实现实时VoIP电话会议系统的方法，并提出了我们通过改进原始概念而获得的改进措施。我们最新改进的电话会议技术提供了以前的电话会议系统提供的所有功能，但是我们的新方法比以前的系统所需的数据带宽消耗少得多。新系统使具有一定容量的网络能够容纳的与会者人数几乎是现有系统所能容纳的两倍。
{"title":"Quantifying Improvements from Refinements in a VoIP Teleconference System","authors":"Teck-Kuen Chua, D. Pheanis","doi":"10.1109/AICT.2009.15","DOIUrl":"https://doi.org/10.1109/AICT.2009.15","url":null,"abstract":"In a VoIP (Voice over Internet Protocol) telephone system, we face the significant challenge of providing a teleconference feature that can support a large-scale teleconference without using excessive network bandwidth. This paper examines a new, bandwidth-efficient way of implementing a real-time VoIP teleconference system, and we present measurements of the improvements that we gained by refining our original concept. Our newly refined teleconference technique provides all of the features that previous teleconference systems provide, but our new approach consumes considerably less data bandwidth than previous systems require. The new system allows a network with a given capacity to accommodate nearly double the number of conference participants that an existing system would allow.","PeriodicalId":409336,"journal":{"name":"2009 Fifth Advanced International Conference on Telecommunications","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122187132","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}
The quality of VoIP communication relies significantly on the network that transports voice packets because this network does not usually guarantee available bandwidth, delay, and loss that are critical for real-time voice traffic. The solution proposed here is to manage a voice-over-IP stream dynamically, changing encoding parameters as needed to assure quality. The paper proposes an adaptive-rate control algorithm that establishes interaction between a VoIP sender and a receiver, and manages voice quality in real-time. Simulations demonstrate that the system provides better average communications quality than traditional fixed-rate VoIP.
{"title":"Adaptive Speech Quality Management in Voice-over-IP Communications","authors":"Eugene S. Myakotnykh, Richard A. Thompson","doi":"10.1109/AICT.2009.17","DOIUrl":"https://doi.org/10.1109/AICT.2009.17","url":null,"abstract":"The quality of VoIP communication relies significantly on the network that transports voice packets because this network does not usually guarantee available bandwidth, delay, and loss that are critical for real-time voice traffic. The solution proposed here is to manage a voice-over-IP stream dynamically, changing encoding parameters as needed to assure quality. The paper proposes an adaptive-rate control algorithm that establishes interaction between a VoIP sender and a receiver, and manages voice quality in real-time. Simulations demonstrate that the system provides better average communications quality than traditional fixed-rate VoIP.","PeriodicalId":409336,"journal":{"name":"2009 Fifth Advanced International Conference on Telecommunications","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129618382","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}
The paper suggests a software architecture to improve the accessibility to information from texts. This software combines techniques like: Image Processing, Optical Character Recognition, Machine Translation, Text Analyze and Text to Speech. The application uses a scanner or a web cam as an image input device, recognizes the text by using OCR, enables text translation by using Google’s machine translation implementation, interpret the text as a future development and reads the text by using TTS technology. In this way the user can put the text information source into a scanner or under a web cam and can hear the text translated and interpreted, if required. A functional prototype is presented and conclusions are issued.
{"title":"Software Architecture for Better Text-Based Infromation Accessibility","authors":"V. Topac, V. Stoicu-Tivadar","doi":"10.1109/AICT.2009.41","DOIUrl":"https://doi.org/10.1109/AICT.2009.41","url":null,"abstract":"The paper suggests a software architecture to improve the accessibility to information from texts. This software combines techniques like: Image Processing, Optical Character Recognition, Machine Translation, Text Analyze and Text to Speech. The application uses a scanner or a web cam as an image input device, recognizes the text by using OCR, enables text translation by using Google’s machine translation implementation, interpret the text as a future development and reads the text by using TTS technology. In this way the user can put the text information source into a scanner or under a web cam and can hear the text translated and interpreted, if required. A functional prototype is presented and conclusions are issued.","PeriodicalId":409336,"journal":{"name":"2009 Fifth Advanced International Conference on Telecommunications","volume":"154 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127280970","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}
As a system for guaranteeing Quality of Service of continuous media by traffic control, there exists Integrated Services using Resource ReSerVation Protocol that guarantees bandwidth for object traffics by executing bandwidth control. This system, however, restricts bandwidth for other traffics through the session of object traffics because of its static bandwidth control. Then, to solve the problem, we studied and designed a dynamic traffic control system that can not only guarantee Quality of Service of continuous media perfectly but also generate excess bandwidth for other traffics by executing bandwidth control dynamically. Furthermore, we evaluated the performance of the system by experiment for confirming the effectiveness of it. In this paper, we describe the design of the system and the evaluation results.
{"title":"Design and Evaluation of a Dynamic Traffic Control System for Continuous Media","authors":"M. Yabe, Naoto Yoshizawa, Tomoya Wakaume","doi":"10.1109/AICT.2009.62","DOIUrl":"https://doi.org/10.1109/AICT.2009.62","url":null,"abstract":"As a system for guaranteeing Quality of Service of continuous media by traffic control, there exists Integrated Services using Resource ReSerVation Protocol that guarantees bandwidth for object traffics by executing bandwidth control. This system, however, restricts bandwidth for other traffics through the session of object traffics because of its static bandwidth control. Then, to solve the problem, we studied and designed a dynamic traffic control system that can not only guarantee Quality of Service of continuous media perfectly but also generate excess bandwidth for other traffics by executing bandwidth control dynamically. Furthermore, we evaluated the performance of the system by experiment for confirming the effectiveness of it. In this paper, we describe the design of the system and the evaluation results.","PeriodicalId":409336,"journal":{"name":"2009 Fifth Advanced International Conference on Telecommunications","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129452296","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}
Multimedia flows transport over multiple IP domains requires quality of services (QoS) guarantees. In multi-domain contexts, the end-to-end QoS assurance needs cooperation of several resource management entities. Cooperation of the domains but also preservation of each domain’s autonomy are both required. This paper presents results on functional validation of a multiple domain overlay-developed management system, for QoS enabled network services. The testbed described is an instance of complex system architecture, targeting the transport and delivery of multimedia streams over multi-domain IP networks and heterogeneous access networks.
{"title":"Overlay Management of Network Services for Multimedia Flows Transport over Multiple Domains - Testbed Validation","authors":"E. Borcoci, R. Lupu, S. Obreja","doi":"10.1109/AICT.2009.29","DOIUrl":"https://doi.org/10.1109/AICT.2009.29","url":null,"abstract":"Multimedia flows transport over multiple IP domains requires quality of services (QoS) guarantees. In multi-domain contexts, the end-to-end QoS assurance needs cooperation of several resource management entities. Cooperation of the domains but also preservation of each domain’s autonomy are both required. This paper presents results on functional validation of a multiple domain overlay-developed management system, for QoS enabled network services. The testbed described is an instance of complex system architecture, targeting the transport and delivery of multimedia streams over multi-domain IP networks and heterogeneous access networks.","PeriodicalId":409336,"journal":{"name":"2009 Fifth Advanced International Conference on Telecommunications","volume":"360 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-24","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122775871","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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