Pub Date : 2009-05-12DOI: 10.1109/CQR.2009.5137353
Eugene S. Myakotnykh, Richard A. Thompson
The E-model is a computational tool standardized by the ITU and developed to estimate the quality of VoIP communications. The objective of this model is to determine a voice quality rating that incorporates the quality of encoding processes and the mouth-to-ear characteristics of a path along a given call. The model was originally developed for narrowband 4-kHz telephony. This paper discusses a new methodology for extending the E-model to wideband 7-kHz communications.
{"title":"Computational quality model for wideband voice-over-IP communications","authors":"Eugene S. Myakotnykh, Richard A. Thompson","doi":"10.1109/CQR.2009.5137353","DOIUrl":"https://doi.org/10.1109/CQR.2009.5137353","url":null,"abstract":"The E-model is a computational tool standardized by the ITU and developed to estimate the quality of VoIP communications. The objective of this model is to determine a voice quality rating that incorporates the quality of encoding processes and the mouth-to-ear characteristics of a path along a given call. The model was originally developed for narrowband 4-kHz telephony. This paper discusses a new methodology for extending the E-model to wideband 7-kHz communications.","PeriodicalId":186033,"journal":{"name":"2009 IEEE International Workshop Technical Committee on Communications Quality and Reliability","volume":"63 3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128019085","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 : 2009-05-12DOI: 10.1109/CQR.2009.5137355
R. Aoun, M. Gagnaire
With the fast growth of grid service demands, an efficient resource management mechanism is required. Business-oriented algorithms were shown to be adequate to resolve such a problem. Business models mainly consider the problem of computing, storage and network resources virtualization. This paper investigates the problem of resource allocation under flexible traffic in market-oriented grids. We consider a discrete time approach for resource allocation under flexible scheduled traffic in the context of layer-2 Virtual Privates Networks (VPN). We aim to exploit end-users time requirements in order to provide a fair management algorithm in terms of gain for the service provider. For that purpose, we introduce a weighted cost function enabling a service differentiation relying on time constraints disparity. We propose an exact approach based on an Integer Linear Programming (ILP) formulation to solve this problem. Simulation results outline the impact of using flexible traffic on the system performance and the gain of the service provider.
{"title":"An exact optimization tool for market-oriented grid middleware","authors":"R. Aoun, M. Gagnaire","doi":"10.1109/CQR.2009.5137355","DOIUrl":"https://doi.org/10.1109/CQR.2009.5137355","url":null,"abstract":"With the fast growth of grid service demands, an efficient resource management mechanism is required. Business-oriented algorithms were shown to be adequate to resolve such a problem. Business models mainly consider the problem of computing, storage and network resources virtualization. This paper investigates the problem of resource allocation under flexible traffic in market-oriented grids. We consider a discrete time approach for resource allocation under flexible scheduled traffic in the context of layer-2 Virtual Privates Networks (VPN). We aim to exploit end-users time requirements in order to provide a fair management algorithm in terms of gain for the service provider. For that purpose, we introduce a weighted cost function enabling a service differentiation relying on time constraints disparity. We propose an exact approach based on an Integer Linear Programming (ILP) formulation to solve this problem. Simulation results outline the impact of using flexible traffic on the system performance and the gain of the service provider.","PeriodicalId":186033,"journal":{"name":"2009 IEEE International Workshop Technical Committee on Communications Quality and Reliability","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121042025","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 : 2009-05-12DOI: 10.1109/CQR.2009.5137360
I. Kostanic, N. Mijatovic, Stephen D. Vest
A Quality of Service (QoS) assessment methodology for cellular communication networks is described. The methodology is based on the data collected through drive testing; it is focused on the end user perception of service quality and it is independent of access technologies implemented by the cellular networks. QoS assessment for both the circuit switched and packet switched side of the network is discussed. The end goal of the proposed methodology is QoS comparison between cellular networks implementing different cellular technologies. Illustrative examples based on live network measurements are used in support of the presented methodology.
{"title":"Measurement based QoS comparison of cellular communication networks","authors":"I. Kostanic, N. Mijatovic, Stephen D. Vest","doi":"10.1109/CQR.2009.5137360","DOIUrl":"https://doi.org/10.1109/CQR.2009.5137360","url":null,"abstract":"A Quality of Service (QoS) assessment methodology for cellular communication networks is described. The methodology is based on the data collected through drive testing; it is focused on the end user perception of service quality and it is independent of access technologies implemented by the cellular networks. QoS assessment for both the circuit switched and packet switched side of the network is discussed. The end goal of the proposed methodology is QoS comparison between cellular networks implementing different cellular technologies. Illustrative examples based on live network measurements are used in support of the presented methodology.","PeriodicalId":186033,"journal":{"name":"2009 IEEE International Workshop Technical Committee on Communications Quality and Reliability","volume":"89 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129040390","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 : 2009-05-12DOI: 10.1109/CQR.2009.5137354
Ling Wang, P. Verma
The notion of quality is an important factor in the design of any complex communication system. Since most communication systems are increasingly dependent on packet switching as the underlying transport mechanism, this paper explores the impact of levels of quality that a packet switched system can have on cost. Several interesting results are presented that result in the notion of an optimum level of load that a packet switched system can handle given a defined upper bound on latency that a specified percentage of packets would suffer. The results presented offer a fresh insight into the relationship between cost and quality and can be used to optimize the throughput of a packet switched system under specified levels of quality.
{"title":"The notion of cost and quality in packet switched networks: An abstract approach","authors":"Ling Wang, P. Verma","doi":"10.1109/CQR.2009.5137354","DOIUrl":"https://doi.org/10.1109/CQR.2009.5137354","url":null,"abstract":"The notion of quality is an important factor in the design of any complex communication system. Since most communication systems are increasingly dependent on packet switching as the underlying transport mechanism, this paper explores the impact of levels of quality that a packet switched system can have on cost. Several interesting results are presented that result in the notion of an optimum level of load that a packet switched system can handle given a defined upper bound on latency that a specified percentage of packets would suffer. The results presented offer a fresh insight into the relationship between cost and quality and can be used to optimize the throughput of a packet switched system under specified levels of quality.","PeriodicalId":186033,"journal":{"name":"2009 IEEE International Workshop Technical Committee on Communications Quality and Reliability","volume":"39 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134552115","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 : 2009-05-12DOI: 10.1109/CQR.2009.5137357
Yuta Yamamoto, M. Yamamoto
In ad hoc wireless networks, end-to-end TCP is well-known that it can obtain very low throughput with increase of the number of hops. One promising way to resolve this technical problem of TCP throughput degradation is proxy-based TCP. In proxy-based TCP, an end-to-end TCP session is divided by (an) intermediate proxy host(s), which leads to reduction of hop-count of each divided session and improvement of TCP throughput. However, we show that proxy-based TCP surprisingly does not bring performance improvement and obtain lower throughput than conventional end-to-end TCP in a realistic situation of limited buffer space in proxy host(s). The reason for this degradation is interruption of segment transmission caused by wireless channel capture around the proxy host. In this paper, for resolving this problem, we propose intentional window flow control which prevents wireless channel capture around the proxy host. In our proposed intentional window flow control method, the proxy host and the destination host return advertised window size which is intentionally set to small value almost independently from its vacant buffer space. Performance evaluation results show that our proposed intentional flow control with small advertised window size prevents wireless channel capture around the proxy host and significantly improves proxy-based TCP performance.
{"title":"Intentional window flow control for proxy-based TCP in ad hoc networks","authors":"Yuta Yamamoto, M. Yamamoto","doi":"10.1109/CQR.2009.5137357","DOIUrl":"https://doi.org/10.1109/CQR.2009.5137357","url":null,"abstract":"In ad hoc wireless networks, end-to-end TCP is well-known that it can obtain very low throughput with increase of the number of hops. One promising way to resolve this technical problem of TCP throughput degradation is proxy-based TCP. In proxy-based TCP, an end-to-end TCP session is divided by (an) intermediate proxy host(s), which leads to reduction of hop-count of each divided session and improvement of TCP throughput. However, we show that proxy-based TCP surprisingly does not bring performance improvement and obtain lower throughput than conventional end-to-end TCP in a realistic situation of limited buffer space in proxy host(s). The reason for this degradation is interruption of segment transmission caused by wireless channel capture around the proxy host. In this paper, for resolving this problem, we propose intentional window flow control which prevents wireless channel capture around the proxy host. In our proposed intentional window flow control method, the proxy host and the destination host return advertised window size which is intentionally set to small value almost independently from its vacant buffer space. Performance evaluation results show that our proposed intentional flow control with small advertised window size prevents wireless channel capture around the proxy host and significantly improves proxy-based TCP performance.","PeriodicalId":186033,"journal":{"name":"2009 IEEE International Workshop Technical Committee on Communications Quality and Reliability","volume":"27 35","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114047383","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 : 2009-05-12DOI: 10.1109/CQR.2009.5137358
Sanghyun Chi, B. Womack
Clients have still hesitated to switch conventional phone service with voice over IP networks (VoIP) service because VoIP service providers are not successful in providing consistent quality during a call. The uncertainness of IP networks, the legacy of packet-switched networks, makes it hard to predict service quality and demands real-time based monitoring. In this paper, we propose a prediction voice quality metric to monitor the quality of VoIP service. Based on a learning machine, the proposed metric nonlinearly weighs network parameters to estimate speech quality. Finally, performance analysis shows that the proposed metric achieves the high prediction accuracy.
{"title":"Predicting the quality of voice over IP networks","authors":"Sanghyun Chi, B. Womack","doi":"10.1109/CQR.2009.5137358","DOIUrl":"https://doi.org/10.1109/CQR.2009.5137358","url":null,"abstract":"Clients have still hesitated to switch conventional phone service with voice over IP networks (VoIP) service because VoIP service providers are not successful in providing consistent quality during a call. The uncertainness of IP networks, the legacy of packet-switched networks, makes it hard to predict service quality and demands real-time based monitoring. In this paper, we propose a prediction voice quality metric to monitor the quality of VoIP service. Based on a learning machine, the proposed metric nonlinearly weighs network parameters to estimate speech quality. Finally, performance analysis shows that the proposed metric achieves the high prediction accuracy.","PeriodicalId":186033,"journal":{"name":"2009 IEEE International Workshop Technical Committee on Communications Quality and Reliability","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133443662","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 : 2009-05-12DOI: 10.1109/CQR.2009.5137359
T. Kikkawa, T. Miyata, K. Yamaoka
We construct an ALM (application layer multicast) tree that achieves the maximum bandwidth by using underlay traffic information. Since this problem was previously shown to be NP-hard, no polynomial-time algorithms for maximum-bandwidth ALM tree construction exist. In our work, we assume that the underlay network has a tree network topology, and we show a polynomial-time algorithm that constructs a maximum-bandwidth ALM tree on such a tree network topology. We propose an ALM tree construction algorithm that achieves bandwidth b and apply this algorithm with various bandwidth values b. We show that it can make a maximum-bandwidth ALM tree in polynomial time.
{"title":"Proof of optimal algorithm for maximum-bandwidth ALM tree construction","authors":"T. Kikkawa, T. Miyata, K. Yamaoka","doi":"10.1109/CQR.2009.5137359","DOIUrl":"https://doi.org/10.1109/CQR.2009.5137359","url":null,"abstract":"We construct an ALM (application layer multicast) tree that achieves the maximum bandwidth by using underlay traffic information. Since this problem was previously shown to be NP-hard, no polynomial-time algorithms for maximum-bandwidth ALM tree construction exist. In our work, we assume that the underlay network has a tree network topology, and we show a polynomial-time algorithm that constructs a maximum-bandwidth ALM tree on such a tree network topology. We propose an ALM tree construction algorithm that achieves bandwidth b and apply this algorithm with various bandwidth values b. We show that it can make a maximum-bandwidth ALM tree in polynomial time.","PeriodicalId":186033,"journal":{"name":"2009 IEEE International Workshop Technical Committee on Communications Quality and Reliability","volume":"92 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117334811","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 : 2009-05-12DOI: 10.1109/CQR.2009.5137352
A. Asthana, J. Olivieri
As the industry moves to more mature software processes (e.g., CMMI) there is increased need to adopt more rigorous, sophisticated (i.e., quantitative) metrics. While quantitative product readiness criteria are often used for business cases and related areas, software readiness is often assessed more subjectively & qualitatively. Quite often there is no explicit linkage to original performance and reliability requirements for the software. The criteria are primarily process-oriented (versus product oriented) and/or subjective. Such an approach to deciding software readiness increases the risk of poor field performance and unhappy customers. Unfortunately, creating meaningful and useful quantitative in-process metrics for software development has been notoriously difficult.
{"title":"Quantifying software reliability and readiness","authors":"A. Asthana, J. Olivieri","doi":"10.1109/CQR.2009.5137352","DOIUrl":"https://doi.org/10.1109/CQR.2009.5137352","url":null,"abstract":"As the industry moves to more mature software processes (e.g., CMMI) there is increased need to adopt more rigorous, sophisticated (i.e., quantitative) metrics. While quantitative product readiness criteria are often used for business cases and related areas, software readiness is often assessed more subjectively & qualitatively. Quite often there is no explicit linkage to original performance and reliability requirements for the software. The criteria are primarily process-oriented (versus product oriented) and/or subjective. Such an approach to deciding software readiness increases the risk of poor field performance and unhappy customers. Unfortunately, creating meaningful and useful quantitative in-process metrics for software development has been notoriously difficult.","PeriodicalId":186033,"journal":{"name":"2009 IEEE International Workshop Technical Committee on Communications Quality and Reliability","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125595161","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 : 2009-05-12DOI: 10.1109/CQR.2009.5137356
Y. Fukushima, T. Murase, R. Fujimaki, S. Hirose, T. Yokohira
One promising approach for large-scale simultaneous events (e.g., DDoS attacks and worm epidemics) is to use a multi-stage change-point detection scheme. The scheme adopts two-stage detection. In the first stage, local detectors (LDs), which are deployed on each monitored subnet, detects a change point in a monitored metric such as outgoing traffic rate. If an LD detects a change-point, it sends an alert to global detector (GD). In the second stage, GD checks whether the proportion of LDs that send alerts simultaneously is greater than or equal to a threshold value. If so, it judges that large-scale simultaneous events are occurring. In previous studies for the multi-stage change-point detection scheme, it is assumed that weight of each alert is identical. Under this assumption, false-positive rate of the scheme tends to be high when some LDs sends false-positive alerts frequently. In this paper, we weight alerts based on false-positive rate of each LD in order to decrease false-positive rate of the multi-stage change-point detection scheme. In our scheme, GD infers false-positive rate of each LD and gives lower weight to LDs with higher false-positive rate. Simulation results show that our proposed scheme can achieve lower false-positive rate than the scheme without alert weighting under the constraint that detection rate must be 1.0.
{"title":"Accuracy improvement of multi-stage change-point detection scheme by weighting alerts based on false-positive rate","authors":"Y. Fukushima, T. Murase, R. Fujimaki, S. Hirose, T. Yokohira","doi":"10.1109/CQR.2009.5137356","DOIUrl":"https://doi.org/10.1109/CQR.2009.5137356","url":null,"abstract":"One promising approach for large-scale simultaneous events (e.g., DDoS attacks and worm epidemics) is to use a multi-stage change-point detection scheme. The scheme adopts two-stage detection. In the first stage, local detectors (LDs), which are deployed on each monitored subnet, detects a change point in a monitored metric such as outgoing traffic rate. If an LD detects a change-point, it sends an alert to global detector (GD). In the second stage, GD checks whether the proportion of LDs that send alerts simultaneously is greater than or equal to a threshold value. If so, it judges that large-scale simultaneous events are occurring. In previous studies for the multi-stage change-point detection scheme, it is assumed that weight of each alert is identical. Under this assumption, false-positive rate of the scheme tends to be high when some LDs sends false-positive alerts frequently. In this paper, we weight alerts based on false-positive rate of each LD in order to decrease false-positive rate of the multi-stage change-point detection scheme. In our scheme, GD infers false-positive rate of each LD and gives lower weight to LDs with higher false-positive rate. Simulation results show that our proposed scheme can achieve lower false-positive rate than the scheme without alert weighting under the constraint that detection rate must be 1.0.","PeriodicalId":186033,"journal":{"name":"2009 IEEE International Workshop Technical Committee on Communications Quality and Reliability","volume":"89 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2009-05-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132307099","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 : 2008-09-25DOI: 10.1109/CQR.2009.5137351
Kazumine Ogura, Tomoki Fujikawa, Su Zhou, J. Katto
This paper presents TCP-Fusion supporting RTT (Round Trip Time) fairness in addition to throughput efficiency and friendliness to TCP-Reno. When multiple TCP flows having different RTT values compete, more bandwidth is unfairly allocated to the flow having smaller RTT. This means that a user with longer RTT may not be able to obtain sufficient bandwidth by the current methods. On the other hand, recent studies on the TCP congestion control to achieve RTT fairness and throughput efficiency are evolving actively. An example for RTT fairness is TCP-Libra and an example for throughput efficiency is Hybrid TCP congestion control. This paper focuses on Hybrid TCP (exploiting residual link capacity when TCP-Reno drops its rate) and improves its RTT fairness by incorporating the ideas of TCP-Libra (congestion window increase in proportion to square of RTT) to its loss mode. Experiments are carried out to validate the proposed method and much better performances in RTT fairness and throughput are provided against conventional methods.
{"title":"Improvement of RTT-fairness in Hybrid TCP congestion control","authors":"Kazumine Ogura, Tomoki Fujikawa, Su Zhou, J. Katto","doi":"10.1109/CQR.2009.5137351","DOIUrl":"https://doi.org/10.1109/CQR.2009.5137351","url":null,"abstract":"This paper presents TCP-Fusion supporting RTT (Round Trip Time) fairness in addition to throughput efficiency and friendliness to TCP-Reno. When multiple TCP flows having different RTT values compete, more bandwidth is unfairly allocated to the flow having smaller RTT. This means that a user with longer RTT may not be able to obtain sufficient bandwidth by the current methods. On the other hand, recent studies on the TCP congestion control to achieve RTT fairness and throughput efficiency are evolving actively. An example for RTT fairness is TCP-Libra and an example for throughput efficiency is Hybrid TCP congestion control. This paper focuses on Hybrid TCP (exploiting residual link capacity when TCP-Reno drops its rate) and improves its RTT fairness by incorporating the ideas of TCP-Libra (congestion window increase in proportion to square of RTT) to its loss mode. Experiments are carried out to validate the proposed method and much better performances in RTT fairness and throughput are provided against conventional methods.","PeriodicalId":186033,"journal":{"name":"2009 IEEE International Workshop Technical Committee on Communications Quality and Reliability","volume":"30 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2008-09-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131958900","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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