Pub Date : 2006-11-30DOI: 10.1109/IWQOS.2006.250452
Alexander Totok, V. Karamcheti
Meeting client QoS expectations proves to be a difficult task for the providers of modern Internet services, especially when Web servers experience overload conditions (steady or transient), which cause increased response times and request rejections, leading to user frustration, lowered usage of the service and reduced revenues. In this paper, we propose a server-side request scheduling mechanism that addresses these problems. Our reward-driven request prioritization (RDRP) algorithm gives higher execution priority to client Web sessions that are likely to bring more service profit (or any other application-specific reward). The method works by predicting future session structure by comparing its requests seen so far with aggregated information about recent client behavior, and using these predictions to preferentially allocate Web server resources. Our experiments using the TPC-W benchmark application and CBMG-based Web workloads, with an implementation of the RDRP techniques in the JBoss application server, show that RDRP can significantly boost reward attained by the service, while providing better QoS to clients that bring more reward
{"title":"Improving Performance of Internet Services Through Reward-Driven Request Prioritization","authors":"Alexander Totok, V. Karamcheti","doi":"10.1109/IWQOS.2006.250452","DOIUrl":"https://doi.org/10.1109/IWQOS.2006.250452","url":null,"abstract":"Meeting client QoS expectations proves to be a difficult task for the providers of modern Internet services, especially when Web servers experience overload conditions (steady or transient), which cause increased response times and request rejections, leading to user frustration, lowered usage of the service and reduced revenues. In this paper, we propose a server-side request scheduling mechanism that addresses these problems. Our reward-driven request prioritization (RDRP) algorithm gives higher execution priority to client Web sessions that are likely to bring more service profit (or any other application-specific reward). The method works by predicting future session structure by comparing its requests seen so far with aggregated information about recent client behavior, and using these predictions to preferentially allocate Web server resources. Our experiments using the TPC-W benchmark application and CBMG-based Web workloads, with an implementation of the RDRP techniques in the JBoss application server, show that RDRP can significantly boost reward attained by the service, while providing better QoS to clients that bring more reward","PeriodicalId":248938,"journal":{"name":"200614th IEEE International Workshop on Quality of Service","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-11-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126782096","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 : 2006-11-30DOI: 10.1109/IWQOS.2006.250470
Songqing Chen, Shiping Chen, Huiping Guo, Bo Shen, S. Jajodia
Massive Internet media distribution demands pro longed continuous consumption of networking and disk band widths in large capacity. Many proxy-based Internet media distribution algorithms and systems have been proposed, implemented, and evaluated to address the scalability issue. However, few of them have been used in practice, since two important issues are not satisfactorily addressed. First, existing proxy-based media distribution architectures lack an efficient media distribution control mechanism. Without protection on the Internet, content providers are hesitant to use existing fast distribution techniques. Second, little has been done to protect client privacy during client accesses. Straightforward solutions to address these two issues independently lead to conflicts. For example, to enforce distribution control, only legitimate users should be granted access rights. However, this normally discloses more information (such as which object the client is accessing) other than the client identity, which conflicts with the client's desire for privacy protection. In this paper, we propose a unified proxy-based media distribution protocol to effectively address these two problems simultaneously. We further design a set of new algorithms for cooperative proxies where our proposed scheme works practically. Simulation results show that our proposed strategy is efficient
{"title":"Efficient Proxy-Based Internet Media Distribution Control and Privacy Protection Infrastructure","authors":"Songqing Chen, Shiping Chen, Huiping Guo, Bo Shen, S. Jajodia","doi":"10.1109/IWQOS.2006.250470","DOIUrl":"https://doi.org/10.1109/IWQOS.2006.250470","url":null,"abstract":"Massive Internet media distribution demands pro longed continuous consumption of networking and disk band widths in large capacity. Many proxy-based Internet media distribution algorithms and systems have been proposed, implemented, and evaluated to address the scalability issue. However, few of them have been used in practice, since two important issues are not satisfactorily addressed. First, existing proxy-based media distribution architectures lack an efficient media distribution control mechanism. Without protection on the Internet, content providers are hesitant to use existing fast distribution techniques. Second, little has been done to protect client privacy during client accesses. Straightforward solutions to address these two issues independently lead to conflicts. For example, to enforce distribution control, only legitimate users should be granted access rights. However, this normally discloses more information (such as which object the client is accessing) other than the client identity, which conflicts with the client's desire for privacy protection. In this paper, we propose a unified proxy-based media distribution protocol to effectively address these two problems simultaneously. We further design a set of new algorithms for cooperative proxies where our proposed scheme works practically. Simulation results show that our proposed strategy is efficient","PeriodicalId":248938,"journal":{"name":"200614th IEEE International Workshop on Quality of Service","volume":"14 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-11-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128910202","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 : 2006-11-30DOI: 10.1109/IWQOS.2006.250451
Hui Wang, D. Chiu, John C.S. Lui
The connectivity between millions of nodes on the Internet is provided by the interconnection of many ISPs' networks. These ISPs, in their decisions to peer with each other, define a set of transit relationships. These transit relationships are the primary factors that dictate how traffic flows through the Internet. BGP-based inter-domain routing that implements these transit relationships can be considered economically efficient. The advent of peer-to-peer (P2P) applications and overlay networks, however, changes the rules by providing traffic routing favoring the applications' needs. This can lead to reduced economic efficiency and upset the ISPs' business model. In this paper, we propose simple models to represent P2P traffic demand, peering and routing in a market place of two competing ISPs to illustrate this tussle of the Internet. Based on these models, we also propose and investigate alternative peering and provisioning strategies available to the ISPs and analyze their effectiveness
{"title":"Modeling the Peering and Routing Tussle between ISPs and P2P Applications","authors":"Hui Wang, D. Chiu, John C.S. Lui","doi":"10.1109/IWQOS.2006.250451","DOIUrl":"https://doi.org/10.1109/IWQOS.2006.250451","url":null,"abstract":"The connectivity between millions of nodes on the Internet is provided by the interconnection of many ISPs' networks. These ISPs, in their decisions to peer with each other, define a set of transit relationships. These transit relationships are the primary factors that dictate how traffic flows through the Internet. BGP-based inter-domain routing that implements these transit relationships can be considered economically efficient. The advent of peer-to-peer (P2P) applications and overlay networks, however, changes the rules by providing traffic routing favoring the applications' needs. This can lead to reduced economic efficiency and upset the ISPs' business model. In this paper, we propose simple models to represent P2P traffic demand, peering and routing in a market place of two competing ISPs to illustrate this tussle of the Internet. Based on these models, we also propose and investigate alternative peering and provisioning strategies available to the ISPs and analyze their effectiveness","PeriodicalId":248938,"journal":{"name":"200614th IEEE International Workshop on Quality of Service","volume":"45 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-11-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122877527","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 : 2006-11-30DOI: 10.1109/IWQOS.2006.250473
A. Tam, D. Chiu, John C.S. Lui, Y. Tay
Admission control has been shown to be a preferred alternative to TCP-friendly congestion control for inelastic flows in heterogeneous networks shared by elastic and inelastic traffic. However, it is possible for an inelastic flow to adopt different level of aggressiveness in implementing the admission control. How these different levels of aggressiveness affect the system performance remains an open issue. In this paper, we evaluate a full spectrum of (abstract) admission control algorithms in terms of their aggressiveness towards elastic flows. A totally aggressive version would admit an inelastic flow even if this means elastic flows' fair bandwidth share is reduced to close to zero. In the other extreme, a TCP-friendly version would only admit an inelastic flow if its desired rate is no higher than what the elastic flows will receive after its arrival. We show that the performance of inelastic flows is asymptotically insensitive to their aggressiveness without strong assumptions about flow file size or holding time distributions. This makes a strong case for adopting a less aggressive, yet TCP-friendly admission control in a heterogeneous network. Extensive simulations are carried out to validate the performance, stability and asymptotic behavior the the proposed TCP-friendly admission control policy
{"title":"A Case for TCP-Friendly Admission Control","authors":"A. Tam, D. Chiu, John C.S. Lui, Y. Tay","doi":"10.1109/IWQOS.2006.250473","DOIUrl":"https://doi.org/10.1109/IWQOS.2006.250473","url":null,"abstract":"Admission control has been shown to be a preferred alternative to TCP-friendly congestion control for inelastic flows in heterogeneous networks shared by elastic and inelastic traffic. However, it is possible for an inelastic flow to adopt different level of aggressiveness in implementing the admission control. How these different levels of aggressiveness affect the system performance remains an open issue. In this paper, we evaluate a full spectrum of (abstract) admission control algorithms in terms of their aggressiveness towards elastic flows. A totally aggressive version would admit an inelastic flow even if this means elastic flows' fair bandwidth share is reduced to close to zero. In the other extreme, a TCP-friendly version would only admit an inelastic flow if its desired rate is no higher than what the elastic flows will receive after its arrival. We show that the performance of inelastic flows is asymptotically insensitive to their aggressiveness without strong assumptions about flow file size or holding time distributions. This makes a strong case for adopting a less aggressive, yet TCP-friendly admission control in a heterogeneous network. Extensive simulations are carried out to validate the performance, stability and asymptotic behavior the the proposed TCP-friendly admission control policy","PeriodicalId":248938,"journal":{"name":"200614th IEEE International Workshop on Quality of Service","volume":"45 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-11-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122488860","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 : 2006-11-30DOI: 10.1109/IWQOS.2006.250472
H. Lundqvist, G. Karlsson
This paper investigates the possibility to differentiate services by using endpoint traffic controls, while sharing a single best-effort network The starting point is a scheme based on a combination of probe-based admission control in the end-systems for streaming traffic and error correction to isolate different flows. There are inherent limitations in pure host-based control mechanisms and we therefore investigate the improvement that can be achieved when explicit congestion notification is used in routers. In particular we investigate the sensitivity to different active queue management mechanisms and parameter settings. Simulations show that a high ECN marking rate helps to differentiate between flows, but it is important to make sure that the chosen AQM works well for the chosen parameters. This study also shows that a main benefit of ECN is to enable the extension of the scheme to wireless environments
{"title":"Host-Based Service Differentiation with Congestion Feedback","authors":"H. Lundqvist, G. Karlsson","doi":"10.1109/IWQOS.2006.250472","DOIUrl":"https://doi.org/10.1109/IWQOS.2006.250472","url":null,"abstract":"This paper investigates the possibility to differentiate services by using endpoint traffic controls, while sharing a single best-effort network The starting point is a scheme based on a combination of probe-based admission control in the end-systems for streaming traffic and error correction to isolate different flows. There are inherent limitations in pure host-based control mechanisms and we therefore investigate the improvement that can be achieved when explicit congestion notification is used in routers. In particular we investigate the sensitivity to different active queue management mechanisms and parameter settings. Simulations show that a high ECN marking rate helps to differentiate between flows, but it is important to make sure that the chosen AQM works well for the chosen parameters. This study also shows that a main benefit of ECN is to enable the extension of the scheme to wireless environments","PeriodicalId":248938,"journal":{"name":"200614th IEEE International Workshop on Quality of Service","volume":"66 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-11-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121261247","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 : 2006-11-30DOI: 10.1109/IWQOS.2006.250465
Camelia Al-Najjar, A. Reddy
The emerging popularity and interest in voice-over-IP (VoIP) has been accompanied by customer concerns about voice quality over these networks. The lack of an appropriate real-time capable infrastructure in packet networks along with the threats of denial-of service (DoS) attacks can deteriorate the service that these voice calls receive. Traditionally, each voice call employs its own end-to-end forward-error-correction (FEC) mechanisms. In this paper, we show that when VoIP calls are aggregated over a network link or path, the provider can employ a suitable linear-time encoding for the aggregated voice traffic, resulting in considerable quality improvement with little redundancy. We show that it is possible to achieve rates closer to link capacity as more calls are combined with very small output loss rates even in the presence of significant packet loss rates in the network. The advantages of the proposed scheme exceed similar or other techniques applied to individual voice calls
{"title":"A Service Provider's Approach for Improving Performance of Aggregate Voice-over-IP Traffic","authors":"Camelia Al-Najjar, A. Reddy","doi":"10.1109/IWQOS.2006.250465","DOIUrl":"https://doi.org/10.1109/IWQOS.2006.250465","url":null,"abstract":"The emerging popularity and interest in voice-over-IP (VoIP) has been accompanied by customer concerns about voice quality over these networks. The lack of an appropriate real-time capable infrastructure in packet networks along with the threats of denial-of service (DoS) attacks can deteriorate the service that these voice calls receive. Traditionally, each voice call employs its own end-to-end forward-error-correction (FEC) mechanisms. In this paper, we show that when VoIP calls are aggregated over a network link or path, the provider can employ a suitable linear-time encoding for the aggregated voice traffic, resulting in considerable quality improvement with little redundancy. We show that it is possible to achieve rates closer to link capacity as more calls are combined with very small output loss rates even in the presence of significant packet loss rates in the network. The advantages of the proposed scheme exceed similar or other techniques applied to individual voice calls","PeriodicalId":248938,"journal":{"name":"200614th IEEE International Workshop on Quality of Service","volume":"22 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-11-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133418835","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 : 2006-11-30DOI: 10.1109/IWQOS.2006.250480
Mea Wang, Baochun Li
With network coding, intermediate nodes between the source and the receivers of an end-to-end communication session are not only capable of relaying and replicating data messages, but also of coding incoming messages to produce coded outgoing ones. Recent studies have shown that network coding is beneficial for peer-to-peer content distribution, since if eliminates the need for content reconciliation, and is highly resilient to peer failures. In this paper, we present our recent experiences with a highly optimized and high-performance C++ implementation of randomized network coding at the application layer. We present our observations based on an extensive series of experiments, draw conclusions from a wide range of scenarios, and are more cautious and less optimistic as compared to previous studies
{"title":"How Practical is Network Coding?","authors":"Mea Wang, Baochun Li","doi":"10.1109/IWQOS.2006.250480","DOIUrl":"https://doi.org/10.1109/IWQOS.2006.250480","url":null,"abstract":"With network coding, intermediate nodes between the source and the receivers of an end-to-end communication session are not only capable of relaying and replicating data messages, but also of coding incoming messages to produce coded outgoing ones. Recent studies have shown that network coding is beneficial for peer-to-peer content distribution, since if eliminates the need for content reconciliation, and is highly resilient to peer failures. In this paper, we present our recent experiences with a highly optimized and high-performance C++ implementation of randomized network coding at the application layer. We present our observations based on an extensive series of experiments, draw conclusions from a wide range of scenarios, and are more cautious and less optimistic as compared to previous studies","PeriodicalId":248938,"journal":{"name":"200614th IEEE International Workshop on Quality of Service","volume":"214 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-11-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133837079","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 : 2006-11-30DOI: 10.1109/IWQOS.2006.250447
Chuan Wu, Baochun Li
It is critical to monitor the performance and "health" of large-scale peer-to-peer applications. As an example, operators of peer-to-peer live streaming applications may be interested in observing performance bottlenecks, peer failures, and network topologies. In most cases, such observations are used to diagnose potential problems in the protocol design, to troubleshoot network outage, or to improve the Quality of Service of the peer-to-peer network in general. They are not time sensitive in nature, as delayed observations up to minutes or even hours are still valuable. However, such historical and delay-tolerant observations should include measurements of peers that have already failed or departed, as peer dynamics significantly affect the health of peer-to-peer applications. Such a delay-tolerant observation of peer-to-peer applications over a historical period of time is referred to as a diagnosis. In this paper, we present Echelon, a time-insensitive way to construct the diagnosis of a large-scale peer-to-peer application. Replacing the traditional wisdom of logging servers, we leverage the power of network coding to collect application-specific measurements on each peer, and disseminate them to other peers in a coded form. Over time, measurements of departed peers can still be recovered, simply by probing a small subset of peers in the network. Simulation studies have shown that Echelon is highly configurable, bandwidth efficient, and extremely tolerant of peer dynamics, thanks to the advantages of randomized network coding
{"title":"Echelon: Peer-to-Peer Network Diagnosis with Network Coding","authors":"Chuan Wu, Baochun Li","doi":"10.1109/IWQOS.2006.250447","DOIUrl":"https://doi.org/10.1109/IWQOS.2006.250447","url":null,"abstract":"It is critical to monitor the performance and \"health\" of large-scale peer-to-peer applications. As an example, operators of peer-to-peer live streaming applications may be interested in observing performance bottlenecks, peer failures, and network topologies. In most cases, such observations are used to diagnose potential problems in the protocol design, to troubleshoot network outage, or to improve the Quality of Service of the peer-to-peer network in general. They are not time sensitive in nature, as delayed observations up to minutes or even hours are still valuable. However, such historical and delay-tolerant observations should include measurements of peers that have already failed or departed, as peer dynamics significantly affect the health of peer-to-peer applications. Such a delay-tolerant observation of peer-to-peer applications over a historical period of time is referred to as a diagnosis. In this paper, we present Echelon, a time-insensitive way to construct the diagnosis of a large-scale peer-to-peer application. Replacing the traditional wisdom of logging servers, we leverage the power of network coding to collect application-specific measurements on each peer, and disseminate them to other peers in a coded form. Over time, measurements of departed peers can still be recovered, simply by probing a small subset of peers in the network. Simulation studies have shown that Echelon is highly configurable, bandwidth efficient, and extremely tolerant of peer dynamics, thanks to the advantages of randomized network coding","PeriodicalId":248938,"journal":{"name":"200614th IEEE International Workshop on Quality of Service","volume":"11 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-11-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126700133","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 : 2006-11-30DOI: 10.1109/IWQOS.2006.250474
Mahmoud Elhaddad, R. Melhem, T. Znati
We consider the problem of packet scheduling in a network with small router buffers. The objective is to provide a statistical bound on the worst-case packet loss rate for a traffic aggregate (connection) routed along any network path, given maximum permissible link utilization (load). This problem is argued to be of interest in networks providing statistical loss-rate guarantees to ingress-egress connections with fixed bandwidth demands. We introduce a scheduling algorithm for networks using per packet transmission reservation. Reservations allow loss guarantees at the aggregate level to hold for individual flows within the aggregate. The algorithm employs randomization and traffic regulation at the ingress, and batch local scheduling at the links. It ensures that a large fraction of packets from each connection are consistently subject to small loss probability at every link. These packets are therefore likely to survive long paths. To obtain the desired loss-rate bound, we analyze the performance of the algorithm under global routing and bandwidth allocation scenarios that maximize the loss rate of a connection routed along an arbitrary network path. We compare the bound to that obtained using the scheduling algorithm that combines the FCFS service discipline and the drop-tail policy. We find that the proposed algorithm significantly improves the constraints on link utilization and path length necessary to achieve strong loss-rate guarantees
{"title":"Supporting Loss Guarantees in Buffer-Limited Networks","authors":"Mahmoud Elhaddad, R. Melhem, T. Znati","doi":"10.1109/IWQOS.2006.250474","DOIUrl":"https://doi.org/10.1109/IWQOS.2006.250474","url":null,"abstract":"We consider the problem of packet scheduling in a network with small router buffers. The objective is to provide a statistical bound on the worst-case packet loss rate for a traffic aggregate (connection) routed along any network path, given maximum permissible link utilization (load). This problem is argued to be of interest in networks providing statistical loss-rate guarantees to ingress-egress connections with fixed bandwidth demands. We introduce a scheduling algorithm for networks using per packet transmission reservation. Reservations allow loss guarantees at the aggregate level to hold for individual flows within the aggregate. The algorithm employs randomization and traffic regulation at the ingress, and batch local scheduling at the links. It ensures that a large fraction of packets from each connection are consistently subject to small loss probability at every link. These packets are therefore likely to survive long paths. To obtain the desired loss-rate bound, we analyze the performance of the algorithm under global routing and bandwidth allocation scenarios that maximize the loss rate of a connection routed along an arbitrary network path. We compare the bound to that obtained using the scheduling algorithm that combines the FCFS service discipline and the drop-tail policy. We find that the proposed algorithm significantly improves the constraints on link utilization and path length necessary to achieve strong loss-rate guarantees","PeriodicalId":248938,"journal":{"name":"200614th IEEE International Workshop on Quality of Service","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-11-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122722511","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 : 2006-11-30DOI: 10.1109/IWQOS.2006.250462
H. Munoz, Ignacio Más Ivars, G. Karlsson
This paper presents a distributed admission control for the distributed coordination function of IEEE 802.11 wireless LANs that limits the risk of congestion collapse due to a high arrival rate of flows. This control scheme requires no modification to the current distributed coordination function; it works by performing a short, non-disturbing probe that estimates the MAC service time. The flow is admitted if the estimate is below a threshold. We show how the threshold may be adjusted dynamically to maintain an average packet loss rate below a configurable limit. We show via extensive simulations that the admission control avoids congestion due to flow arrivals and that it maintains the loss probability below the given threshold regardless of the offered load or number of stations. Our simulations also analyze the main drawback of our scheme: a reduction in the link utilization. The admission control efficiently protects cells from overload and it may offer soft QoS guarantees to multimedia flows without the need for scheduling or polling mechanisms in IEEE 802.11
{"title":"Overload Protection for IEEE 802.11 Cells","authors":"H. Munoz, Ignacio Más Ivars, G. Karlsson","doi":"10.1109/IWQOS.2006.250462","DOIUrl":"https://doi.org/10.1109/IWQOS.2006.250462","url":null,"abstract":"This paper presents a distributed admission control for the distributed coordination function of IEEE 802.11 wireless LANs that limits the risk of congestion collapse due to a high arrival rate of flows. This control scheme requires no modification to the current distributed coordination function; it works by performing a short, non-disturbing probe that estimates the MAC service time. The flow is admitted if the estimate is below a threshold. We show how the threshold may be adjusted dynamically to maintain an average packet loss rate below a configurable limit. We show via extensive simulations that the admission control avoids congestion due to flow arrivals and that it maintains the loss probability below the given threshold regardless of the offered load or number of stations. Our simulations also analyze the main drawback of our scheme: a reduction in the link utilization. The admission control efficiently protects cells from overload and it may offer soft QoS guarantees to multimedia flows without the need for scheduling or polling mechanisms in IEEE 802.11","PeriodicalId":248938,"journal":{"name":"200614th IEEE International Workshop on Quality of Service","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-11-30","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121114711","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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