Pub Date : 2001-11-25DOI: 10.1109/GLOCOM.2001.966170
Yun-Wen Chen, Ren-Hung Hwang, Ying-Dar Lin
Recently, QoS routing has been studied intensively. In QoS routing, an essential issue is routing granularity. Most of the research adopts per-flow granularity in the forwarding table. Some research advocates per-source-destination pair granularity in the forwarding table with route pinning. The flow based approach has finer granularity, thus is more efficient in traffic engineering and resource utilization. However, the computation overhead and storage overhead are also higher. On the other hand, source-destination based granularity is more efficient on packet processing and forwarding, but has higher blocking probability. We propose the concept of forwarding with routing marks. With a limited number of routing marks, the proposed routing algorithm reduces the forwarding complexity and storage overhead significantly while yielding very competitive performance in terms of fractional reward loss.
{"title":"Multipath QoS routing with bandwidth guarantee","authors":"Yun-Wen Chen, Ren-Hung Hwang, Ying-Dar Lin","doi":"10.1109/GLOCOM.2001.966170","DOIUrl":"https://doi.org/10.1109/GLOCOM.2001.966170","url":null,"abstract":"Recently, QoS routing has been studied intensively. In QoS routing, an essential issue is routing granularity. Most of the research adopts per-flow granularity in the forwarding table. Some research advocates per-source-destination pair granularity in the forwarding table with route pinning. The flow based approach has finer granularity, thus is more efficient in traffic engineering and resource utilization. However, the computation overhead and storage overhead are also higher. On the other hand, source-destination based granularity is more efficient on packet processing and forwarding, but has higher blocking probability. We propose the concept of forwarding with routing marks. With a limited number of routing marks, the proposed routing algorithm reduces the forwarding complexity and storage overhead significantly while yielding very competitive performance in terms of fractional reward loss.","PeriodicalId":346622,"journal":{"name":"GLOBECOM'01. IEEE Global Telecommunications Conference (Cat. No.01CH37270)","volume":"203 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-11-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124551816","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 : 2001-11-25DOI: 10.1109/GLOCOM.2001.966250
S. Manvi, P. Venkataram
The allocation of bandwidth for multimedia traffic poses a technical challenge due to bursty and isochronous nature of applications. We propose a mobile agent based approach for bandwidth allocation in multimedia communication. It is based on the the network congestion monitored by the agents at the clients. The mobile agent hosted by the server will allocate bandwidth online to the applications within the requested range at regular intervals. Also, it keeps the aggregated bandwidth below the link bandwidth of server and clients. The approach reduces the network control traffic used in traditional online bandwidth allocation policies. The scheme is simulated and its performance is evaluated in terms of several parameters, like, bandwidth utilization, application rejection, agent response time, and agent migrations. The simulation results show that the use of agents increase flexibility and efficiency in end-to-end bandwidth allocation and operate asynchronously. The flexibility in using this technology is that, the allocation policies can be changed, customized and implemented easily.
{"title":"Mobile agent based online bandwidth allocation scheme for multimedia communication","authors":"S. Manvi, P. Venkataram","doi":"10.1109/GLOCOM.2001.966250","DOIUrl":"https://doi.org/10.1109/GLOCOM.2001.966250","url":null,"abstract":"The allocation of bandwidth for multimedia traffic poses a technical challenge due to bursty and isochronous nature of applications. We propose a mobile agent based approach for bandwidth allocation in multimedia communication. It is based on the the network congestion monitored by the agents at the clients. The mobile agent hosted by the server will allocate bandwidth online to the applications within the requested range at regular intervals. Also, it keeps the aggregated bandwidth below the link bandwidth of server and clients. The approach reduces the network control traffic used in traditional online bandwidth allocation policies. The scheme is simulated and its performance is evaluated in terms of several parameters, like, bandwidth utilization, application rejection, agent response time, and agent migrations. The simulation results show that the use of agents increase flexibility and efficiency in end-to-end bandwidth allocation and operate asynchronously. The flexibility in using this technology is that, the allocation policies can be changed, customized and implemented easily.","PeriodicalId":346622,"journal":{"name":"GLOBECOM'01. IEEE Global Telecommunications Conference (Cat. No.01CH37270)","volume":"23 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-11-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129427389","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 : 2001-11-25DOI: 10.1109/GLOCOM.2001.965085
Mei Yang, S. Zheng, D. Verchère
The ubiquity of IP has led to IP-over-WDM as the core architecture for the next-generation optical Internet. Optical burst switching (OBS) has been proposed to be a competitive switching technology for DWDM networks. The data channel scheduling algorithm is one of the major challenges in OBS. The same-service-to-all model of the current Internet is inadequate for the diverse quality of service expectations of Internet applications and users. Differentiated service (DiffServ) was proposed to provide a scalable and manageable architecture for service differentiation in IP networks. This paper proposes a scheduling algorithm based on an existing LAUC-VF algorithm to support DiffServ and takes advantage of MPLS. Simulation results demonstrate that this algorithm has better QoS performance than the existing LAUC-VF algorithm.
{"title":"A QoS supporting scheduling algorithm for optical burst switching DWDM networks","authors":"Mei Yang, S. Zheng, D. Verchère","doi":"10.1109/GLOCOM.2001.965085","DOIUrl":"https://doi.org/10.1109/GLOCOM.2001.965085","url":null,"abstract":"The ubiquity of IP has led to IP-over-WDM as the core architecture for the next-generation optical Internet. Optical burst switching (OBS) has been proposed to be a competitive switching technology for DWDM networks. The data channel scheduling algorithm is one of the major challenges in OBS. The same-service-to-all model of the current Internet is inadequate for the diverse quality of service expectations of Internet applications and users. Differentiated service (DiffServ) was proposed to provide a scalable and manageable architecture for service differentiation in IP networks. This paper proposes a scheduling algorithm based on an existing LAUC-VF algorithm to support DiffServ and takes advantage of MPLS. Simulation results demonstrate that this algorithm has better QoS performance than the existing LAUC-VF algorithm.","PeriodicalId":346622,"journal":{"name":"GLOBECOM'01. IEEE Global Telecommunications Conference (Cat. No.01CH37270)","volume":"30 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-11-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129678856","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 : 2001-11-25DOI: 10.1109/GLOCOM.2001.965525
Zigang Yang, Ben Lu, Xiaodong Wang
We consider the design of a blind optimal multiuser receiver for space-time block coded (STBC) multi-carrier code-division multiple-access (MC-CDMA) systems in unknown frequency-selective fading channels. A novel blind Bayesian multiuser detector is derived for joint estimation of unknown fading channel and symbols. Such a detector is based on the Bayesian inference of all unknown quantities. The Gibbs sampler, a Markov chain Monte Carlo (MCMC) method, is then used for Bayesian computation. Moreover, being soft-input and soft-output, the proposed Bayesian multiuser detector fits well into the turbo receiver framework and it exchanges the extrinsic information with the MAP channel decoder to successively refine its performance. Finally, its performance over random generated slow and fast frequency-selective fading channel is demonstrated through simulation examples.
{"title":"Blind Bayesian multiuser receiver for space-time coded MC-CDMA system over frequency-selective fading channel","authors":"Zigang Yang, Ben Lu, Xiaodong Wang","doi":"10.1109/GLOCOM.2001.965525","DOIUrl":"https://doi.org/10.1109/GLOCOM.2001.965525","url":null,"abstract":"We consider the design of a blind optimal multiuser receiver for space-time block coded (STBC) multi-carrier code-division multiple-access (MC-CDMA) systems in unknown frequency-selective fading channels. A novel blind Bayesian multiuser detector is derived for joint estimation of unknown fading channel and symbols. Such a detector is based on the Bayesian inference of all unknown quantities. The Gibbs sampler, a Markov chain Monte Carlo (MCMC) method, is then used for Bayesian computation. Moreover, being soft-input and soft-output, the proposed Bayesian multiuser detector fits well into the turbo receiver framework and it exchanges the extrinsic information with the MAP channel decoder to successively refine its performance. Finally, its performance over random generated slow and fast frequency-selective fading channel is demonstrated through simulation examples.","PeriodicalId":346622,"journal":{"name":"GLOBECOM'01. IEEE Global Telecommunications Conference (Cat. No.01CH37270)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-11-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130377666","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 : 2001-11-25DOI: 10.1109/GLOCOM.2001.965649
M. Uysal, C. Georghiades
Most current space-time block codes are designed based on an orthogonality principle. By relaxing the orthogonality requirement, it is possible to construct new codes with higher throughput efficiency. In this paper, new non-orthogonal space-time block codes for three transmit antennas are proposed, achieving throughput rates larger than those of currently known orthogonal designs. The proposed codes are found through a code search based on the rank criterion, determinant criterion and rank distribution. Performance results through Monte Carlo simulations demonstrate that there is a trade-off between the error-rate, diversity order and throughput rate.
{"title":"New space-time block codes for high throughput efficiency","authors":"M. Uysal, C. Georghiades","doi":"10.1109/GLOCOM.2001.965649","DOIUrl":"https://doi.org/10.1109/GLOCOM.2001.965649","url":null,"abstract":"Most current space-time block codes are designed based on an orthogonality principle. By relaxing the orthogonality requirement, it is possible to construct new codes with higher throughput efficiency. In this paper, new non-orthogonal space-time block codes for three transmit antennas are proposed, achieving throughput rates larger than those of currently known orthogonal designs. The proposed codes are found through a code search based on the rank criterion, determinant criterion and rank distribution. Performance results through Monte Carlo simulations demonstrate that there is a trade-off between the error-rate, diversity order and throughput rate.","PeriodicalId":346622,"journal":{"name":"GLOBECOM'01. IEEE Global Telecommunications Conference (Cat. No.01CH37270)","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-11-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126791447","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 : 2001-11-25DOI: 10.1109/GLOCOM.2001.965565
J. Campello, D. Modha
Campello, Modha and Rajagopalan (see Proc. Int. Conf. Communications (ICC), Helsinki, Finland, 2001) proposed a simple-to-implement heuristic, namely, bit-filling, for constructing high rate and high girth LDPC codes. In the present work, we extend bit-filling, and demonstrate that the extended algorithm produces better codes, that is, codes with higher rate/girth and good bit error rate performance. We demonstrate the positive effect of girth on bit error rate performance.
{"title":"Extended bit-filling and LDPC code design","authors":"J. Campello, D. Modha","doi":"10.1109/GLOCOM.2001.965565","DOIUrl":"https://doi.org/10.1109/GLOCOM.2001.965565","url":null,"abstract":"Campello, Modha and Rajagopalan (see Proc. Int. Conf. Communications (ICC), Helsinki, Finland, 2001) proposed a simple-to-implement heuristic, namely, bit-filling, for constructing high rate and high girth LDPC codes. In the present work, we extend bit-filling, and demonstrate that the extended algorithm produces better codes, that is, codes with higher rate/girth and good bit error rate performance. We demonstrate the positive effect of girth on bit error rate performance.","PeriodicalId":346622,"journal":{"name":"GLOBECOM'01. IEEE Global Telecommunications Conference (Cat. No.01CH37270)","volume":"55 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-11-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129105541","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 : 2001-11-25DOI: 10.1109/GLOCOM.2001.966293
X. Wei, C.Y. Yang, S. Mao
Three different kinds of time delay estimation methods, the traditional correlation receiver, the method based on subspace algorithm and the estimator using the extended Kalman filter, are analyzed. The performance of these methods is compared in the multipath environment. The factors that affect their performance are studied. The advantages and disadvantages of each method are concluded.
{"title":"The performance of several time delay estimation methods in multipath channel","authors":"X. Wei, C.Y. Yang, S. Mao","doi":"10.1109/GLOCOM.2001.966293","DOIUrl":"https://doi.org/10.1109/GLOCOM.2001.966293","url":null,"abstract":"Three different kinds of time delay estimation methods, the traditional correlation receiver, the method based on subspace algorithm and the estimator using the extended Kalman filter, are analyzed. The performance of these methods is compared in the multipath environment. The factors that affect their performance are studied. The advantages and disadvantages of each method are concluded.","PeriodicalId":346622,"journal":{"name":"GLOBECOM'01. IEEE Global Telecommunications Conference (Cat. No.01CH37270)","volume":"3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-11-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123821491","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 : 2001-11-25DOI: 10.1109/GLOCOM.2001.965727
C. Lamy, O. Pothier
Previously, joint source channel techniques mostly focused on systems using fixed-length coding, even though variable-length coding (VLC) is widely used, particularly in video coding. Typically, VLC bit streams are made channel-robust through packetization and standard forward-error correction (FEC). However, when the channel conditions are fairly mild, FEC can reveal itself bandwidth-inefficient. A variable-rate extension of joint source channel decoding could thus potentially replace FEC under mild conditions or, for noisier channels, could be used together with FEC to ameliorate the coding rate, extending in both cases the range of situations under which the bit stream is adequately protected. We propose two reduced-complexity VLC soft-input decoding techniques, as well as a comparison with existing algorithms. Experimental results of a new proposed VLC decoding algorithm show very good performance and low complexity.
{"title":"Reduced complexity maximum a posteriori decoding of variable-length codes","authors":"C. Lamy, O. Pothier","doi":"10.1109/GLOCOM.2001.965727","DOIUrl":"https://doi.org/10.1109/GLOCOM.2001.965727","url":null,"abstract":"Previously, joint source channel techniques mostly focused on systems using fixed-length coding, even though variable-length coding (VLC) is widely used, particularly in video coding. Typically, VLC bit streams are made channel-robust through packetization and standard forward-error correction (FEC). However, when the channel conditions are fairly mild, FEC can reveal itself bandwidth-inefficient. A variable-rate extension of joint source channel decoding could thus potentially replace FEC under mild conditions or, for noisier channels, could be used together with FEC to ameliorate the coding rate, extending in both cases the range of situations under which the bit stream is adequately protected. We propose two reduced-complexity VLC soft-input decoding techniques, as well as a comparison with existing algorithms. Experimental results of a new proposed VLC decoding algorithm show very good performance and low complexity.","PeriodicalId":346622,"journal":{"name":"GLOBECOM'01. IEEE Global Telecommunications Conference (Cat. No.01CH37270)","volume":"107 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-11-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123156594","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 : 2001-11-25DOI: 10.1109/GLOCOM.2001.966231
Mandis Beigi, D. Verma
In a quality of service (QoS) enabled network, the Internet service providers (ISPs) need to define the network requirements for the customers by defining a set of policies. Every time a new customer gets added or removed or when a customer's resource requirements change, the administrator needs to modify the policies installed on the routers/servers in the network. However, these modifications will affect the traffic flowing in the network and in turn might take away some of the existing customers' resources. Therefore, there is a need to predict how changing the policies will affect the performance of the existing traffic flows in the network. We present a mechanism for predicting whether adding, removing or changing a policy will degrade the performance of the traffic flows belonging to the previous customers. The network administrator can use this information to decide whether such modifications to the policies are desired.
{"title":"Network prediction in a policy-based IP network","authors":"Mandis Beigi, D. Verma","doi":"10.1109/GLOCOM.2001.966231","DOIUrl":"https://doi.org/10.1109/GLOCOM.2001.966231","url":null,"abstract":"In a quality of service (QoS) enabled network, the Internet service providers (ISPs) need to define the network requirements for the customers by defining a set of policies. Every time a new customer gets added or removed or when a customer's resource requirements change, the administrator needs to modify the policies installed on the routers/servers in the network. However, these modifications will affect the traffic flowing in the network and in turn might take away some of the existing customers' resources. Therefore, there is a need to predict how changing the policies will affect the performance of the existing traffic flows in the network. We present a mechanism for predicting whether adding, removing or changing a policy will degrade the performance of the traffic flows belonging to the previous customers. The network administrator can use this information to decide whether such modifications to the policies are desired.","PeriodicalId":346622,"journal":{"name":"GLOBECOM'01. IEEE Global Telecommunications Conference (Cat. No.01CH37270)","volume":"28 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-11-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114540767","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 : 2001-11-25DOI: 10.1109/GLOCOM.2001.965192
Liang Xu, Xuemin Shen, J. Mark
We consider rate and power adaptations for homogeneous data traffic in the uplink of direct-sequence code-division multiple-access (DS-CDMA) cellular systems, where the transmission power and the transmission rate of each data user are adapted in accordance to the random activities of users and varying data traffic. The performances in terms of data throughput, average packet delay and average power consumption, are analyzed and compared between adaptive rate and power control schemes. In the rate-adaptive system, the received power of each data user is fixed while the transmission rate is dynamically adjusted to maintain a target bit energy-to-equivalent noise spectral density ratio (E/sub b//N/sub e/). On the other hand, in the power-adaptive system, the transmission rate is fixed and the received power is adapted to maintain the target E/sub b//N/sub e/. Analytical results show that the rate-adaptive scheme provides a significant power gain and lower average packet delay over the power-adaptive scheme for the data users to achieve the same throughput.
{"title":"Performance analysis of adaptive rate and power control for data service in DS-CDMA systems","authors":"Liang Xu, Xuemin Shen, J. Mark","doi":"10.1109/GLOCOM.2001.965192","DOIUrl":"https://doi.org/10.1109/GLOCOM.2001.965192","url":null,"abstract":"We consider rate and power adaptations for homogeneous data traffic in the uplink of direct-sequence code-division multiple-access (DS-CDMA) cellular systems, where the transmission power and the transmission rate of each data user are adapted in accordance to the random activities of users and varying data traffic. The performances in terms of data throughput, average packet delay and average power consumption, are analyzed and compared between adaptive rate and power control schemes. In the rate-adaptive system, the received power of each data user is fixed while the transmission rate is dynamically adjusted to maintain a target bit energy-to-equivalent noise spectral density ratio (E/sub b//N/sub e/). On the other hand, in the power-adaptive system, the transmission rate is fixed and the received power is adapted to maintain the target E/sub b//N/sub e/. Analytical results show that the rate-adaptive scheme provides a significant power gain and lower average packet delay over the power-adaptive scheme for the data users to achieve the same throughput.","PeriodicalId":346622,"journal":{"name":"GLOBECOM'01. IEEE Global Telecommunications Conference (Cat. No.01CH37270)","volume":"20 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2001-11-25","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116513284","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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