Pub Date : 2003-12-01DOI: 10.1109/GLOCOM.2003.1258298
W. Pattara-Atikom, P. Krishnamurthy, S. Banerjee
It is well known that the distributed coordination function (DCF) of the IEEE 802.11 MAC protocol is not suitable for supporting multimedia and QoS-sensitive applications because of its inherent lack of QoS support and fairness. Recently, several distributed QoS mechanisms have been proposed which translate user QoS requirements into typically a single parameter of the DCF protocol. In this paper, we compare the pros and cons of the major distributed QoS mechanisms, and propose a new mechanism that provides superior performance and supports two different QoS models. The proposed mechanism is based on deficit round robin scheduling and translates the user throughput requirements into the 802.11 MAC interframe space and backoff interval parameters. We show via simulations that the proposed mechanism provides low variability of throughput and delay and has the advantage of low complexity.
{"title":"Comparison of distributed fair QoS mechanisms in wireless LANs","authors":"W. Pattara-Atikom, P. Krishnamurthy, S. Banerjee","doi":"10.1109/GLOCOM.2003.1258298","DOIUrl":"https://doi.org/10.1109/GLOCOM.2003.1258298","url":null,"abstract":"It is well known that the distributed coordination function (DCF) of the IEEE 802.11 MAC protocol is not suitable for supporting multimedia and QoS-sensitive applications because of its inherent lack of QoS support and fairness. Recently, several distributed QoS mechanisms have been proposed which translate user QoS requirements into typically a single parameter of the DCF protocol. In this paper, we compare the pros and cons of the major distributed QoS mechanisms, and propose a new mechanism that provides superior performance and supports two different QoS models. The proposed mechanism is based on deficit round robin scheduling and translates the user throughput requirements into the 802.11 MAC interframe space and backoff interval parameters. We show via simulations that the proposed mechanism provides low variability of throughput and delay and has the advantage of low complexity.","PeriodicalId":301154,"journal":{"name":"GLOBECOM '03. IEEE Global Telecommunications Conference (IEEE Cat. No.03CH37489)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129236881","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 : 2003-12-01DOI: 10.1109/GLOCOM.2003.1258983
Cheng Huang, Lihao Xu
Rate control, in conjunction with congestion control, is important and necessary to maintain both the stability of the overall network and high quality of individual data transfer flows. We study stable rate control algorithms for streaming data, based on control theory. We introduce various control rules to maintain both sending rate and receiver buffer stability. We also propose an adaptive two-state control mechanism to ensure the rate control algorithms are compatible with TCP traffics. Extensive experimental results are shown to demonstrate the effectiveness of the rate control algorithms.
{"title":"SRC: stable rate control for streaming media","authors":"Cheng Huang, Lihao Xu","doi":"10.1109/GLOCOM.2003.1258983","DOIUrl":"https://doi.org/10.1109/GLOCOM.2003.1258983","url":null,"abstract":"Rate control, in conjunction with congestion control, is important and necessary to maintain both the stability of the overall network and high quality of individual data transfer flows. We study stable rate control algorithms for streaming data, based on control theory. We introduce various control rules to maintain both sending rate and receiver buffer stability. We also propose an adaptive two-state control mechanism to ensure the rate control algorithms are compatible with TCP traffics. Extensive experimental results are shown to demonstrate the effectiveness of the rate control algorithms.","PeriodicalId":301154,"journal":{"name":"GLOBECOM '03. IEEE Global Telecommunications Conference (IEEE Cat. No.03CH37489)","volume":"12 4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129243451","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 : 2003-12-01DOI: 10.1109/GLOCOM.2003.1258203
M. Assaad, D. Zeghlache
An analytical model for the evaluation of HSDPA capacity is proposed and used to carry out a comparison of several MIMO systems. HSDPA is an evolution of the UMTS standard over the air interface to achieve higher aggregate bit rates through the introduction of adaptive modulation and coding, hybrid ARQ, fast scheduling, fast cell selection and MIMO (space time coding and Blast) techniques. The model is used to compare MIMO systems, a typically difficult task to provide some insight. The results are supported by a simulation.
{"title":"On the capacity of HSDPA","authors":"M. Assaad, D. Zeghlache","doi":"10.1109/GLOCOM.2003.1258203","DOIUrl":"https://doi.org/10.1109/GLOCOM.2003.1258203","url":null,"abstract":"An analytical model for the evaluation of HSDPA capacity is proposed and used to carry out a comparison of several MIMO systems. HSDPA is an evolution of the UMTS standard over the air interface to achieve higher aggregate bit rates through the introduction of adaptive modulation and coding, hybrid ARQ, fast scheduling, fast cell selection and MIMO (space time coding and Blast) techniques. The model is used to compare MIMO systems, a typically difficult task to provide some insight. The results are supported by a simulation.","PeriodicalId":301154,"journal":{"name":"GLOBECOM '03. IEEE Global Telecommunications Conference (IEEE Cat. No.03CH37489)","volume":"41 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123890422","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 : 2003-12-01DOI: 10.1109/GLOCOM.2003.1258312
F. Alesiani, A. Tarable
Sstarting from recent results on MIMO channels, we design a new transceiver for a code division multiaccess scenario with multiantenna transmitter and receiver. The proposed structure is based on a combination of trellis modulation, turbo codes and differential unitary space-time modulation.
{"title":"Differential space-time CDMA with turbo decoding","authors":"F. Alesiani, A. Tarable","doi":"10.1109/GLOCOM.2003.1258312","DOIUrl":"https://doi.org/10.1109/GLOCOM.2003.1258312","url":null,"abstract":"Sstarting from recent results on MIMO channels, we design a new transceiver for a code division multiaccess scenario with multiantenna transmitter and receiver. The proposed structure is based on a combination of trellis modulation, turbo codes and differential unitary space-time modulation.","PeriodicalId":301154,"journal":{"name":"GLOBECOM '03. IEEE Global Telecommunications Conference (IEEE Cat. No.03CH37489)","volume":"17 7","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"120901612","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 : 2003-12-01DOI: 10.1109/GLOCOM.2003.1258191
Xia Lei, Youxi Tang, Shaoqian Li, Yingtao Li
The paper proposes a new method, minimum clipping power loss scheme (MCPLS), to control the clipping noise based on the relationship between the power of clipped portion and the clipping noise power of OFDM signals. First, the transmitter generates multi-route signals that carry the same information then estimates the clipping noise power of every route under the same clipping threshold. The transmitter then selects the signal with the lowest clipping noise power to clip and transmits the signal. The simulation results proved that this method can mitigate clipping noise and improve the performance of the system. The signal-to-noise ratio is gained 8 dB to achieve 10/sub -3/ symbol-error-rate when the number of subcarriers is 256, the modulation mode is 16-QAM, the clipping threshold is 1.5 and the redundancy of system is 0.59%. The floor of symbol-error-rate is reduced from 10/sub -3/ to 3/spl middot/10/sub -5/. The arithmetic achieves better effect if the threshold is lower.
{"title":"A minimum clipping power loss scheme for mitigating the clipping noise in OFDM","authors":"Xia Lei, Youxi Tang, Shaoqian Li, Yingtao Li","doi":"10.1109/GLOCOM.2003.1258191","DOIUrl":"https://doi.org/10.1109/GLOCOM.2003.1258191","url":null,"abstract":"The paper proposes a new method, minimum clipping power loss scheme (MCPLS), to control the clipping noise based on the relationship between the power of clipped portion and the clipping noise power of OFDM signals. First, the transmitter generates multi-route signals that carry the same information then estimates the clipping noise power of every route under the same clipping threshold. The transmitter then selects the signal with the lowest clipping noise power to clip and transmits the signal. The simulation results proved that this method can mitigate clipping noise and improve the performance of the system. The signal-to-noise ratio is gained 8 dB to achieve 10/sub -3/ symbol-error-rate when the number of subcarriers is 256, the modulation mode is 16-QAM, the clipping threshold is 1.5 and the redundancy of system is 0.59%. The floor of symbol-error-rate is reduced from 10/sub -3/ to 3/spl middot/10/sub -5/. The arithmetic achieves better effect if the threshold is lower.","PeriodicalId":301154,"journal":{"name":"GLOBECOM '03. IEEE Global Telecommunications Conference (IEEE Cat. No.03CH37489)","volume":"9 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121231059","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 : 2003-12-01DOI: 10.1109/GLOCOM.2003.1258480
H. Seba, A. Bouabdallah, N. Badache
This paper considers the problem of fault-tolerance and built-in robustness in key agreement for dynamic peer groups. A fault-tolerant key establishment protocol is developed by extending the group Diffie-Hellman key agreement protocol to support asynchronous settings and faulty participants. The protocol uses recent results on failure detection in asynchronous distributed systems. Simulation results show that the key agreement protocol augmented with failure detection increases significantly the number of group members that participate in the computation of the group key while introducing a low message overhead.
{"title":"Increasing the robustness of initial key agreement using failure detectors","authors":"H. Seba, A. Bouabdallah, N. Badache","doi":"10.1109/GLOCOM.2003.1258480","DOIUrl":"https://doi.org/10.1109/GLOCOM.2003.1258480","url":null,"abstract":"This paper considers the problem of fault-tolerance and built-in robustness in key agreement for dynamic peer groups. A fault-tolerant key establishment protocol is developed by extending the group Diffie-Hellman key agreement protocol to support asynchronous settings and faulty participants. The protocol uses recent results on failure detection in asynchronous distributed systems. Simulation results show that the key agreement protocol augmented with failure detection increases significantly the number of group members that participate in the computation of the group key while introducing a low message overhead.","PeriodicalId":301154,"journal":{"name":"GLOBECOM '03. IEEE Global Telecommunications Conference (IEEE Cat. No.03CH37489)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121409535","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 : 2003-12-01DOI: 10.1109/GLOCOM.2003.1258592
M. Yuksel, E. Erkip
We examine a network consisting of one source, one destination and two amplifying and forwarding relays and consider a scenario in which destination and relays can have various processing limitations. For all possible diversity combining schemes at the relays and at the destination, we find diversity order results analytically and confirm our findings through numerical calculations of bit error rate (BER) versus signal-to-noise-ratio (SNR) curves. We compare our results with direct transmission, well known transmit diversity methods and traditional multihop transmission and conclude that diversity reception in multihop networks provides the lowest error rate.
{"title":"Diversity in relaying protocols with amplify and forward","authors":"M. Yuksel, E. Erkip","doi":"10.1109/GLOCOM.2003.1258592","DOIUrl":"https://doi.org/10.1109/GLOCOM.2003.1258592","url":null,"abstract":"We examine a network consisting of one source, one destination and two amplifying and forwarding relays and consider a scenario in which destination and relays can have various processing limitations. For all possible diversity combining schemes at the relays and at the destination, we find diversity order results analytically and confirm our findings through numerical calculations of bit error rate (BER) versus signal-to-noise-ratio (SNR) curves. We compare our results with direct transmission, well known transmit diversity methods and traditional multihop transmission and conclude that diversity reception in multihop networks provides the lowest error rate.","PeriodicalId":301154,"journal":{"name":"GLOBECOM '03. IEEE Global Telecommunications Conference (IEEE Cat. No.03CH37489)","volume":"20 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116514244","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 : 2003-12-01DOI: 10.1109/GLOCOM.2003.1258285
S. Elayoubi, T. Chahed, G. Hébuterne
In this paper, we focus on the capacity of a multicell UMTS system. We first determine an upper bound on the other cell interference and obtain novel expressions for the SIR and powers for both the uplink and the downlink. The former is an asynchronous CDMA system, often using one of three types of receivers : matched filter, minimum mean-square error (MMSE) and decorrelator. In the latter, the SIR depends on the distance between the user and the base station. We show that the inter-cell interference alters the capacity of the system for all kinds of receivers, though the effective bandwidth expressions remain identical to the ones obtained in the single-cell case. As an application, we show how to use this analytical model to develop a new efficient connection admission control (CAC) algorithm.
{"title":"On the capacity of multi-cell UMTS","authors":"S. Elayoubi, T. Chahed, G. Hébuterne","doi":"10.1109/GLOCOM.2003.1258285","DOIUrl":"https://doi.org/10.1109/GLOCOM.2003.1258285","url":null,"abstract":"In this paper, we focus on the capacity of a multicell UMTS system. We first determine an upper bound on the other cell interference and obtain novel expressions for the SIR and powers for both the uplink and the downlink. The former is an asynchronous CDMA system, often using one of three types of receivers : matched filter, minimum mean-square error (MMSE) and decorrelator. In the latter, the SIR depends on the distance between the user and the base station. We show that the inter-cell interference alters the capacity of the system for all kinds of receivers, though the effective bandwidth expressions remain identical to the ones obtained in the single-cell case. As an application, we show how to use this analytical model to develop a new efficient connection admission control (CAC) algorithm.","PeriodicalId":301154,"journal":{"name":"GLOBECOM '03. IEEE Global Telecommunications Conference (IEEE Cat. No.03CH37489)","volume":"9 ","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"113993490","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 : 2003-12-01DOI: 10.1109/GLOCOM.2003.1258225
D. Pong, T. Moors
This paper proposes an admission control algorithm that enables the upcoming IEEE 802.11e contention based enhanced distributed channel access (EDCA) to provide quantitative bandwidth guarantees for wireless local area networks (WLANs), rather than a relative prioritized service. The algorithm estimates the throughput that flows would achieve if a new flow with certain parameters was admitted, and so indicates whether such a new flow can be admitted while preserving the quality of service (QoS) of existing flows. The algorithm deals with the EDCA parameters of minimum contention window size and transmission opportunity duration, and indicates what values should be used for different flows. Simulation results confirm the accuracy of the throughput estimates and the effectiveness of the admission control algorithm.
{"title":"Call admission control for IEEE 802.11 contention access mechanism","authors":"D. Pong, T. Moors","doi":"10.1109/GLOCOM.2003.1258225","DOIUrl":"https://doi.org/10.1109/GLOCOM.2003.1258225","url":null,"abstract":"This paper proposes an admission control algorithm that enables the upcoming IEEE 802.11e contention based enhanced distributed channel access (EDCA) to provide quantitative bandwidth guarantees for wireless local area networks (WLANs), rather than a relative prioritized service. The algorithm estimates the throughput that flows would achieve if a new flow with certain parameters was admitted, and so indicates whether such a new flow can be admitted while preserving the quality of service (QoS) of existing flows. The algorithm deals with the EDCA parameters of minimum contention window size and transmission opportunity duration, and indicates what values should be used for different flows. Simulation results confirm the accuracy of the throughput estimates and the effectiveness of the admission control algorithm.","PeriodicalId":301154,"journal":{"name":"GLOBECOM '03. IEEE Global Telecommunications Conference (IEEE Cat. No.03CH37489)","volume":"45 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114852869","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 : 2003-12-01DOI: 10.1109/GLOCOM.2003.1258652
Dong Chen, Tamio Saito
Joint detection (JD) is a key technology in 3/sup rd/ generation mobile communication systems like UTRA-TDD; it can be used to eliminate the intracell multiple access interference (MAI) and intersymbol interference (ISI). However, high complexity blocks its commercial application. The bottleneck can be generalized into one problem, to calculate the matrix inversion. A new method is proposed to reduce the complexity of the joint detection algorithm, and the simulation results indicate that the performance is acceptable in practical applications. The algorithm complexity is reduced 18 times, based on simulation duration, compared with a traditional Cholesky method. Even if compared with approximate Cholesky decomposition, the complexity is reduced by 6 percent, based on the complex multiplication times.
{"title":"A new method to reduce the complexity of joint detection algorithm","authors":"Dong Chen, Tamio Saito","doi":"10.1109/GLOCOM.2003.1258652","DOIUrl":"https://doi.org/10.1109/GLOCOM.2003.1258652","url":null,"abstract":"Joint detection (JD) is a key technology in 3/sup rd/ generation mobile communication systems like UTRA-TDD; it can be used to eliminate the intracell multiple access interference (MAI) and intersymbol interference (ISI). However, high complexity blocks its commercial application. The bottleneck can be generalized into one problem, to calculate the matrix inversion. A new method is proposed to reduce the complexity of the joint detection algorithm, and the simulation results indicate that the performance is acceptable in practical applications. The algorithm complexity is reduced 18 times, based on simulation duration, compared with a traditional Cholesky method. Even if compared with approximate Cholesky decomposition, the complexity is reduced by 6 percent, based on the complex multiplication times.","PeriodicalId":301154,"journal":{"name":"GLOBECOM '03. IEEE Global Telecommunications Conference (IEEE Cat. No.03CH37489)","volume":"51 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2003-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128016456","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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