In this article we present a simulation-based comparison of one of the best-known multicast congestion control schemes—TCP-friendly Multicast Congestion Control (TFMCC)—against our proposed Adaptive Smooth Multicast Protocol (ASMP). ASMP consists of a single-rate multicast congestion control mechanism which takes advantage of the RTCP Sender (SR) and Receiver Reports (RR) in order to adjust the sender's transmission rate in respect of the network conditions. The innovation in ASMP lays in the ‘smooth’ transmission rate, which is TCP-friendly and prevents oscillations. We use an integrated simulation environment named Multi-Evalvid-RA for the evaluation of the two congestion control schemes. Multi-Evalvid-RA provides all the necessary tools to perform simulation studies and assess video quality by using both network-centric metrics along with video quality measurements. Performance evaluation results show that ASMP is a very efficient solution for rate-adaptive multimedia applications and a serious competitor to well-known TFMCC. Copyright © 2012 John Wiley & Sons, Ltd.

Uncontrolled video transmission without any flow/congestion control mechanisms should be avoided as it leads to poor-quality service due to packet losses. Therefore, moderate transmission rates that satisfy the video application requirements are preferable to avoid packet losses. Otherwise, error resilience mechanisms should be applied to recover from losses, especially from I-frames that cannot be recovered by the video decoders. More efficient video encoding techniques along with error resilience and higher access capacity are the key elements for increasing video quality.

Next-Generation Network (NGN) is a critical scenario in terms of network management because of its network dimension, its number of users and its heterogeneity. Since the introduction of the Simple Network Management Protocol (SNMP) at the beginning of the 1990s, much effort has been devoted to the development of new network management technologies. Both the Desktop Management Task Force (DMTF) and the Internet Engineering Task Force (IETF) have developed different network and system management protocols, such as Common Open Policy Service, Web-Based Enterprise Management, Network Configuration and even adapted other protocols, such as Diameter and Web Services. A network management technology with poor scalability could compromise NGN management and ultimately NGN network behaviour. This paper analyses the network overhead of several management technologies developed by the DMTF and IETF, and goes on to compare their results with the usage of SNMP. Furthermore, some deployment recommendations are proposed for performance optimization in NGNs. Copyright © 2012 John Wiley & Sons, Ltd.

This paper analyses the network overhead of several management technologies developed by the DMTF and IETF, and goes on to compare their results with the usage of SNMP. Furthermore, some deployment recommendations are proposed for performance optimization in NGNs.

Wavelength-routed Generalized Multi-Protocol Label Switching (GMPLS) networks use Resource reSerVation Protocol—Traffic Engineering (RSVP-TE) as signaling protocol to set up and tear down lightpaths. RSVP-TE uses a soft-state control mechanism to manage lightpaths. In the soft-state control mechanism, each node sets a timer for each control state and resets the timer with refresh messages to maintain the state. When the timer expires due to losses of refresh messages, the control state is initialized and a reserved resource managed with the state is released. It has been considered that resource utilization of soft-state protocols is inferior to that of hard-state protocols, since soft-state protocols may reserve resources until control states are deleted due to timeout. Therefore, some extensions to promote the performance of soft-state protocols, such as message retransmission, have been considered. In this paper, we analyze the behavior of GMPLS RSVP-TE and its variants with a Markov model and analyze the performance of RSVP-TE. From the results, we demonstrate that resource utilization of RSVP-TE can be equivalent to that of a hard-state protocol when the loss probability of signaling messages is low. We also investigate the effectiveness of message retransmission and show that using message retransmission leads to poor resource utilization in some cases. Copyright © 2012 John Wiley & Sons, Ltd.

In this paper, we analyze the behavior of GMPLS RSVP-TE and its variants with a Markov model. RSVP-TE uses a soft-state control mechanism to manage lightpaths. Since the soft-state mechanism may reserve resources until control states are deleted due to timeout, some extensions to promote the performance of soft-state protocols have been considered. We investigate the effectiveness of message retransmission through the analysis and show that the message retransmission leads to poor resource utilization.

The available unlicensed spectrum is increasingly being used by new wireless technologies, but past measurements show that the licensed spectrum is extremely underutilized. To address this issue, the IEEE 802.22 Working Group is developing a novel wireless air interface standard based on cognitive radios (CRs), i.e. IEEE 802.22 wireless regional area networks (WRANs). Moreover, over the last decade wireless multimedia applications have developed rapidly, raising significant concerns about the quality of service (QoS) of multimedia stream transmissions. In particular, the Joint Video Team (JVT) and ITU-T Video Coding Experts Group (VCEG) jointly proposed Scalable Video Coding (SVC) as the next-generation multimedia compression standard. However, the current IEEE 802.22 WRAN draft does not specify QoS mechanisms for SVC-encoded multimedia stream transmission in CR networks. To resolve this problem, we developed a cross-layer channel allocation algorithm (CLCAA) and a novel media access control (MAC) protocol to work with the algorithm. The CLCAA adapts to the characteristics of multimedia traffic and variations of wireless channels by determining the weighting of source–destination pair, which is determined by the deadlines of SVC-encoded multimedia streams, the queuing delay and channel conditions. The CLCAA then allocates transmission opportunities to source–destination pairs based on their weightings and game theory. We also conducted extensive simulations to demonstrate the efficiency of the CLCAA scheme. The simulation results show that the CLCAA scheme not only guarantees QoS for multimedia traffic but also achieves fairness across different streams. Copyright © 2012 John Wiley & Sons, Ltd.

This paper proposes a Cross-Layer Channel Allocation Algorithm (CLCAA) and a novel MAC protocol to work with the algorithm. The CLCAA allocates transmission opportunities to source-destination pairs based on their weightings and game theory.

Network coding is a novel field of information theory and coding theory. It is a breakthrough over the traditional store-and-forward routing methods by allowing coding of two or more packets together. From an information flow aspect, multiple flows could be overlapped in a routing scheme. Hence the theoretical upper bound of multicast capacity could be achieved by network coding. In this project, a complete routing and coding scheme is constructed to realize the maximum multicast transportation task. In order to implement the scheme, the paths of multiple max-flows are determined. Edges are divided into overlapped and normal type based on the merged max-flows. The transmitting data are represented using packets in a specific format. Multicast, forward and coding operations are defined to transmit data at the nodes. The nodes are classified according to the type of operation. A dynamic coding and routing algorithm is proposed to route packets gradually from source node to destinations in topological sorting order by the three operations on the path of merged max-flows. We show that the use of simple XOR operations can satisfy most of the network topologies. The running time of the algorithm presented here is less than 1 second for most of the benchmark and random datasets. Copyright © 2012 John Wiley & Sons, Ltd.

Findings: Network coding is a breakthrough over the traditional store-and-forward routing methods by allowing coding of two or more packets together. We present a complete routing and coding scheme is constructed to realize the maximum multicast transportation task which routes packets gradually from source node to destinations in topological sorting order by the three operations on the path of merged max-flows. We find that the use of simple XOR operations can satisfy most of network topologies and the running time is less than one second for most of the benchmark and random data sets.

Spanners for ad hoc networks provide several benefits such as low communication cost and resource consumption. These spanners need to be fault tolerant in resource-constrained ad hoc networks. In this paper, we have proposed three spanners, called fault-tolerant local Delaunay triangulation (FTLDel), fault-tolerant relative neighborhood graph (FTRNG), and fault-tolerant Gabriel graph (FTGG). The fault-tolerant spanners provide reliability to the network by avoiding heavy packet loss and retaining useful geometric properties. The performance of fault-tolerant spanners FTLDel, FTRNG, and FTGG are evaluated using the network simulator ns2.28. Copyright © 2011 John Wiley & Sons, Ltd.

The spanners need to be fault tolerant in resource constrained ad hoc networks. This paper describes the fault tolerant version of the spanners local Delaunay triangulation, relative neighborhood graph, and Gabriel graph by introducing the concept of stability factor to increase the fault tolerance.

In local loss recovery schemes, a small number of recovery nodes distributed along the transmission paths save incoming packets temporarily in accordance with a specified cache policy and retransmit these packets if they subsequently receive a request message from a downstream receiver. To reduce the recovery latency, the cache policy should ensure that the recovery nodes are always able to satisfy the retransmission requests of the downstream receivers. However, owing to the limited cache size of the recovery nodes and the behavior of the cache policy, this cannot always be achieved, and thus some of the packets must be retransmitted by the sender. Accordingly, this paper develops a new network-coding-based cache policy, designated as network-coding-based FIFO (NCFIFO), which extends the caching time of the packets at the recovery nodes without dropping any of the incoming packets. As a result, the lost packets can be always recovered from the nearest recovery nodes and the recovery latency is significantly reduced. The loss recovery performance of the NCFIFO cache policy is compared with that of existing cache policies by performing a series of simulation experiments using both a uniform error model and a burst error model. The simulation results show that the NCFIFO cache policy not only achieves a better recovery performance than existing cache policies, but also provides a more effective solution for managing a small amount of cache size in environments characterized by a high packet arrival rate. Copyright © 2011 John Wiley & Sons, Ltd.

Owing to the limited cache size of the recovery nodes and the behavior of the cache policy, the lost packets cannot be always recovered from the nearest recovery nodes. Accordingly, this paper develops a new network-coding-based cache policy, designated as network-coding-based FIFO (NCFIFO), which extends the caching time of the packets at the recovery nodes without dropping any of the incoming packets and frees up cache space for new incoming packets by combining multiple cached packets within a single coded packet.

The ever-increasing size, complexity and heterogeneity of telecommunications networks necessitate the introduction of autonomic elements that assist providers in managing and configuring the network's resources. To tackle this increased complexity, it is expected that many specialized autonomic elements will take part in the management process. It becomes necessary for them to collaborate and communicate in order to achieve high-level, human-specified, management goals. Therefore, the need for a scalable mechanism to facilitate the interactions between autonomic elements has arisen. This article presents a communications bus, augmented with semantics through the use of ontologies and semantic reasoning, which governs the communication and collaboration between autonomic elements. It supports filtering of context based on meaning. Additionally, it facilitates matchmaking of autonomic element goals with management services using semantic definitions of their inputs, outputs, preconditions and effects. Furthermore, the delay introduced by semantic reasoning was evaluated through an implemented prototype and was shown to be limited to only a few milliseconds. Copyright © 2011 John Wiley & Sons, Ltd.

This article presents the design of a communications bus, augmented with semantics through the use of ontologies and semantic reasoning mechanisms. It supports semantic subscriptions, which filter context based on meaning rather than string patterns or keywords. Additionally, it facilitates matching of autonomic element goals with management functions based on semantic definitions of inputs, outputs, preconditions and effects. The inclusion of preconditions and effects allows these autonomic elements to better determine the influence of such functions on the managed environment.

Intrusion detection systems (IDSs) are designed to monitor a networked environment and generate alerts whenever abnormal activities are detected. The number of these alerts can be very large, making their evaluation by security analysts a difficult task. Management is complicated by the need to configure the different components of alert evaluation systems. In addition, IDS alert management techniques, such as clustering and correlation, suffer from involving unrelated alerts in their processes and consequently provide results that are inaccurate and difficult to manage. Thus the tuning of an IDS alert management system in order to provide optimal results remains a major challenge, which is further complicated by the large spectrum of potential attacks the system can be subject to. This paper considers the specification and configuration issues of FuzMet, a novel IDS alert management system which employs several metrics and a fuzzy-logic based approach for scoring and prioritizing alerts. In addition, it features an alert rescoring technique that leads to a further reduction in the number of alerts. Comparative results between SNORT scores and FuzMet alert prioritization onto a real attack dataset are presented, along with a simulation-based investigation of the optimal configuration of FuzMet. The results prove the enhanced intrusion detection accuracy brought by our system. Copyright © 2011 John Wiley & Sons, Ltd.

This paper considers the specification and configuration issues of FuzMet, a novel IDS alert management system which employs several metrics and a fuzzy-logic based approach for scoring and prioritizing alerts. It also features an alert rescoring technique that leads to a further reduction in the number of alerts. Comparative results between SNORT scores and FuzMet alert prioritization onto a real attack dataset are presented, along with a simulation-based investigation of the optimal configuration of FuzMet.

Future networks will need to accommodate a significantly augmented user demand, mainly stemming from the wireless and mobile domains. In general, the emerging radio landscape will comprise multiple, collaborating radio access networks (RANs) able to operate a plethora of diverse radio access technologies (RATs), variant types of mobile terminals (MTs), with the ability to choose among various supported RANs/RATs and, in addition, both devices and networks with dynamic spectrum access capabilities that allow the sharing and/or optimization of spectrum usage among different systems. The above will stress network operators for developing mechanisms to confront the challenges and to leverage the opportunities posed by such a versatile radio environment. In particular, the situation calls for adaptive and flexible management paradigms that are able to dynamically manage network elements and terminals, thus ensuring the great availability and efficient usage of spectrum and other radio resources. Framed within the above, this paper considers a cognitive network management architecture, which is destined for optimized management of future wireless networks operating in versatile radio environments, and presents a performance evaluation methodology, which was set up for measuring the signalling loads that the operation of the architecture will bring to the managed network. The methodology is analytically described, and useful results with respect to the signalling load produced for management signalling purposes in an indicative scenario are presented and analysed. Copyright © 2011 John Wiley & Sons, Ltd.

The paper considers a cognitive network management architecture, which aims at ensuring efficient usage of spectrum and radio resources in future, complex wireless networks. It also presents a methodology for measuring the signalling loads and associated delays that the operation of the architecture will bring into the network. Results from the application to an indicative ‘new spectrum assignment’ scenario provide conclusions to be taken into account when deploying the architecture, so that the overall network operation will not be aggravated from such deployment.

One important problem when deploying interdomain path selection is advertising metrics that hold for a long period of time. In this paper we propose a method to aid interdomain path selection mechanisms in that sense. We present a means of computing a bound on the end-to-end delay of traversing a domain considering that the traffic varies within a given uncertainty set. This provides a robust and a verifiable quality of service value for traversing the autonomous system (AS), without revealing confidential information. Consequently, the bound can be safely conceived as a metric to be announced by each AS in the process of interdomain path selection. We show how the maximum delay value is obtained for an interdomain bandwidth demand and we propose an exact method and a numerical approximation method for computing it, neither of which rely on a complex monitoring infrastructure. Simulations with real data that illustrate the problem and validate our results are also presented. Copyright © 2011 John Wiley & Sons, Ltd.

This work studies the worst case of the end-to-end delay of traversing an Autonomous System when Traffic Matrix is uncertain. An exact method for computing the bound and a numerical approximation are presented. Simulation studies are performed with real networks, showing that delay can suddenly increase. The method provides an end-to-end metric that holds for a long period of time and can thus be used by interdomain path selection mechanisms with quality of service constraints.

Mobile ad hoc networks (MANETs) are of much interest to both the research community and the military because of the potential to establish a communication network in any situation that involves emergencies. Examples are search-and-rescue operations, military deployment in hostile environments, and several types of police operations. One critical open issue is how to route messages considering the characteristics of these networks. The nodes act as routers in an environment without a fixed infrastructure, the nodes are mobile, the wireless medium has its own limitations compared to wired networks, and existing routing protocols cannot be employed, at least without modifications. Over the last few years, a number of routing protocols have been proposed and enhanced to address the issue of routing in MANETs. It is not clear how those different protocols perform under different environments. One protocol may be the best in one network configuration but the worst in another. This article provides an analysis and performance evaluation of those protocols that may be suitable for military communications. The evaluation is conducted in two phases. In the first phase, we compare the protocols based on qualitative metrics to locate those that may fit our evaluation criteria. In the second phase, we evaluate the selected protocols from the first phase based on quantitative metrics in a mobility scenario that reflects tactical military movements. The results disclose that there is no routing protocol in the current stage without modifications that can provide efficient routing to any size of network, regardless of the number of nodes and the network load and mobility. Copyright © 2011 John Wiley & Sons, Ltd.

This article differs from other works mainly in three areas. First, the evaluation process is based on both qualitative and quantitative metrics. Second, the simulated mobility and traffic scenarios reflect those in military tactical operations, and third we run extensive simulation sets in which we differentiate all the possible attributes of the network. After studying several routing protocols and running extensive simulations we conclude that efficient routing in MANETs is still an open area with many unsolved issues.

Today the policy-based management (PBM) approach is recognized as an efficient solution to simplify the complex task of managing and controlling networks. To this end, the Internet Engineering Task Force (IETF) has introduced a reference framework to build PBM systems. However, this framework only addresses the provisioning of relatively long validity period services based on predefined service-level agreements. Furthermore, very little work addresses the scalability properties of the instantiation of this framework in a real network. This work aims to extend the IETF PBM framework in order to support dynamic provisioning of short-term services (end-system signalling) as well as an instantiation scheme that is scalable (distributed provisioning of edge routers). This instantiation scheme is based on the distribution of the provisioning process while keeping centralized only the parts that involve critical resources, namely bandwidth brokerage. The performance properties of the proposed scheme are then demonstrated throughout both extensive experimentation and an analytical study. The extension of this performance analysis to the case where multiple bandwidth brokers are used is also discussed. The results of this work are intended to be used as a guideline to help network operators to design a scalable PBM system in order to offer to their customers services with quality of service assurance on an on-demand basis. Copyright © 2011 John Wiley & Sons, Ltd.

This work aims to extend the IETF PBM framework in order to support dynamic provisioning of short-term services. We proposed an instantiation scheme is based on the distribution of the provisioning process while keeping centralized the bandwidth brokerage. The performance analysis are then demonstrated throughout both extensive experimentation and an analytical study. The results are intended to be used as a guideline to design a scalable PBM system in order to offer to their customers services with quality of service assurance on an on-demand basis.

The development and integration of an alarm interface between network elements and a network management system is a costly process, largely because of the informal way in which alarm interfaces are expressed and communicated. Low-quality alarm documentation and confusion around fundamental concepts like alarm states and alarm types are typical consequences of current practices. If alarm interfaces were expressed in a more formal manner, costs could be reduced and more advanced analysis and automation would be enabled. We present a novel approach to alarm interfaces by providing a formal alarm model together with a domain-specific language that allows us to specify both the alarm models and the constraints placed on the alarm models in a consistent manner. This means that we can verify the consistency of an alarm interface and automatically generate artifacts such as alarm correlation rules or alarm documentation based only on the model. Copyright © 2011 John Wiley & Sons, Ltd.

Current approaches for alarm integration suffer from informal alarm interface documentation and expensive knowledge acquisition processes. In contrast to this, we present a novel approach to alarm interfaces by providing a formal alarm model together with a domain-specific language that allows us to specify both the alarm models and the constraints placed on the alarm models in a consistent manner. This means that we can verify the consistency of an alarm interface and automatically generate artifacts such as alarm correlation rules or alarm documentation based only on the model.

In gigabit passive optical networks (GPONs), the ports of the optical line terminal (OLT) support passive optical networks (PONs). An optical split ratio supported on an OLT chassis determines the number of optical network units (ONUs) which can share PON link capacity. Generally, network planners tend to do dimensioning for PON link capacity (OLT port) based on the number of subscribers and their type (i.e. residential or business). Although this dimensioning approach is simple, it does not guarantee a selection of optical split ratio which can optimally allocate bandwidth to end-subscribers. In this paper, we develop an integrated mathematical framework for optimally dimensioning resources in an GPON access network, namely OLT capacity. This framework comprises three resource-dimensioning approaches which are based on user requirements: GPON link utilization and capacity optimization. Our mathematical framework has been integrated into software for GPON resource dimensioning, which we have developed to evaluate the support and performance of services in GPON access networks. Copyright © 2011 John Wiley & Sons, Ltd.

To realize optimal local and global resource dimensioning in GPON access networks, network planners have to consider multiple parameters; such as, busy hour rate and subscriber type (i.e, residential or business), allocated capacity for each end-subscriber and penetration rate of demanding services. The PON link capacity should be allocated for subscribers such that generated traffic load can be smaller than this capacity by at least a minimum capacity threshold to guarantee a minimum throughput for supported traffic classes.

This work presents novel design schemes for the code gateway between the electronic product code (EPC) and ubiquitous ID (uID) for radio-frequency identification (RFID) technology solutions. Interfaces of the gateway that are implemented include the system interface, EPC discovery service (DS), uID DS, and the certification card mechanism. Five major operations are also developed, including inquiry of EPC DS, uID DS, code diversion mechanism, PIN-IC interface, and code conversion mechanism. Simulation results indicate that in addition to providing codes diversion and conversion, the proposed code gateway achieves low-order conformity between these two systems. Moreover, the implemented EPC DS and uID DS interface provide immediate commodity news and exchange of historical information. In addition to delivering inevitable processing efficiency, the integrated system between these two systems significantly improves ubiquitous computing environments. Copyright © 2011 John Wiley & Sons, Ltd.

This work presents novel design schemes for the code gateway between electronic product code (EPC) and ubiquitous ID (uID) for RFID technology solutions. Interfaces of the gateway that are implemented include the system interface, EPC Discovery Service (DS), uID DS, and the certification card mechanism. Five major operations are also developed, including inquiry of EPC DS, uID DS, code diversion mechanism, PIN-IC interface, and code conversion mechanism. Simulation results indicate that in addition to providing codes diversion and conversion, the proposed code gateway achieves low-order conformity between these two systems.

Information technology (IT) service availability is at the core of customer satisfaction and business success for today's organisations. Many medium- to large-size organisations outsource part of their IT services to external providers, with service-level agreements describing the agreed availability of outsourced service components. Availability management of partially outsourced IT services is a non-trivial task since classic approaches for calculating availability are not applicable, and IT managers can only rely on their expertise to fulfil it. This often leads to the adoption of non-optimal solutions. In this paper we present A²thOS, a framework to calculate the availability of partially outsourced IT services in the presence of SLAs and to achieve a cost-optimal choice of availability levels for outsourced IT components while guaranteeing a target availability level for the service. Copyright © 2011 John Wiley & Sons, Ltd.

Availability management of partially outsourced IT services is a non trivial task since classic approaches for calculating availability are not applicable. This often leads to the adoption of non optimal solutions. In this paper we present A2THOS, a framework to calculate the availability of partially outsourced IT services in the presence of SLAs and to achieve a cost-optimal choice of availability levels for outsourced IT components while guaranteeing a target availability level for the partially outsourced IT service.

In the telecommunications service engineering world, the feature interaction (FI) problem is defined as the situation in which running services at the same time undesirably alters their behaviour. It is also known as the service interaction problem. FIs may occur between various types of services, ranging from signalling protocol features to value-added end-user services. The problem has been widely investigated for those types of services. However, sufficient attention has not been paid to operation, administration, management and provisioning (OAM&P) features like creating a new customer account or registering a new gateway as part of the network. The present paper addresses the detection of OAM&P FIs in Internet telephony architectures. The solution consists in an adapted version of a known FI detection method. A simple feature modelling language and an FI detection procedure are proposed. A validation with a case study is also reported. Copyright © 2011 John Wiley & Sons, Ltd.

In this paper we showed that many OAM&P software defects can be avoided at design time if they are considered under the FI problem approach.

We have developed an incentive-rewarding mechanism that stimulates activities in social networking services (SNSs), including content uploading and link establishment. We particularly focus on changing the reward assignment ratio based on the different risks users perceive when uploading content with different privacy settings: public-open and friend-limited. Learning-based simulation allowed us to observe that SNS activity, which we measured as the amount of browsed content within a certain period, can be controlled by a rewarding assignment ratio. We then analyzed how the amount of uploaded content and the increase of established links affect SNS activity. Results suggested that the optimal reward assignment ratio to maximize SNS activity changes depending on the total amount of available reward resources. Copyright © 2011 John Wiley & Sons, Ltd.

We have developed an incentive-rewarding mechanism that stimulates activities in social networking services (SNSs), including content uploading and link establishment. We particularly focus on changing the reward assignment ratio based on the different risks users perceive when uploading content with different privacy settings: public-open and friend-limited. Simulation results suggested that the optimal reward assignment ratio to maximize SNS activity changes depending on the total amount of available reward resources.

This paper proposes a stateless open-digest spam fingerprinting at the packet level (layer 3) based on an open-digest fingerprinting algorithm Nilsimsa. Spam emails show several characteristics when viewed at gateway level, which are suitable for spam fingerprinting: (a) content invariance and (b) recipient address dispersion. In this paper, Nilsimsa is adapted to support both fingerprinting and fast email class estimation, on a per-packet basis. Email packets are incrementally fingerprinted on a per-packet basis, without the need for reassembly. Spam detection status is tagged to the last packet of each email. This in turn allows fast email class estimation (spam detection) at receiving email servers to support more effective spam handling on both inbound and outbound (relayed) emails. The work presented in this paper focuses on evaluating the accuracy of spam fingerprinting at the packet level with consideration on the constraints of processing byte streams over the network, including packet reordering, fragmentation, overlapped bytes, different packet sizes, and possibilities of random addition attacks. Results show that the proposed packet-level fingerprinting can detect spam with 100% random addition when the similarity threshold is set to between 36 and 59. This method gives 0% false positive and 100% true negative, which equals the performance attained for spam fingerprinting at full email abstraction (layer 7). This shows that classifying emails at the packet level can differentiate non-spam from spam with high confidence for a viable spam control implementation on middleboxes. Copyright © 2011 John Wiley & Sons, Ltd.

This paper proposes a stateless open-digest spam fingerprinting for implementation on middleboxes for fast spam detection at receiving email servers to support more effective spam handling of both inbound and outbound (relayed) emails. The work focuses on evaluating the accuracy of spam fingerprinting at the packet level with consideration on the constraints of processing byte streams over the network.

Detecting failed nodes is important in a sensor network, because a failed node can cause coverage and connectivity holes. Since sensors are resource-limited devices, lack of a clocking mechanism imposes difficulties in implementing synchronous failure detection techniques. To counter this, asynchronous methods are needed for detecting failed sensor nodes. For example, in traditional distributed systems sent data packets and received acknowledgements are matched to identify any malfunctions. A similar approach can be adopted in sensor networks. An asynchronous failed sensor node detection (AFSD) method is proposed, where a numeric counter variable is used to track the received and sent data packets between active nodes. This variable acts as bounded for an active sensor node and unbounded for an out-of-communication or failed node. By observing the value of the counter, a decision can be made on a deployed sensor node. Whenever the counter value of a neighbour crosses a predefined threshold, that neighbour is suspected. However, measuring a counter value can always lead to a false positive. To avoid false suspicion, a consensus protocol is needed. Such a technique is described, which is proved as accurate and complete. The complexity analysis shows that the control, energy, and time overheads of the proposed method are linear and in the order of number of neighbours per node. Copyright © 2011 John Wiley & Sons, Ltd.

This paper addresses the failed sensor node detection issues and proposes a novel method to detect failed nodes asynchronously. The new approach has taken the trivial considerations of a sensor network such as the resource scarcity, appropriately and proves to be accurate and complete.

Management of long-distance, high-speed optical backbones spanning multiple administrative domains requires new solutions for challenging tasks. In particular, it is not trivial to negotiate, monitor and continuously enforce the required quality of service (QoS) for applications that span multiple domains. This paper proposes GigaManP2P: a novel peer-to-peer (P2P) management architecture for high-speed QoS-aware backbones. GigaManP2P peers provide management services in a ubiquitous fashion through modules that interface with both the communication infrastructure and network users. In particular, we describe management services for on inter-domain QoS monitoring and resilient routing. After detecting a QoS constraint violation trend, a proactive rerouting strategy is triggered based on redundant virtual circuits, allowing both full and partial rerouting. The P2P overlay implementation is the basis for allowing transparent communication across autonomous systems. Experimental results showing the overhead of the P2P infrastructure in comparison to raw Simple Network Management Protocol, and the performance of the rerouting strategy, are presented. Copyright © 2011 John Wiley & Sons, Ltd.

GigaManP2P is a management overlay for long-distance, high-speed optical backbones that span multiple administrative domains. GigaManP2P peers provide management services in a ubiquitous fashion, in particular services to negotiate, monitor and continuously enforce the required quality of service (QoS) for applications that span multiple domains. After detecting a QoS constraint violation trend, a proactive rerouting strategy is triggered based on redundant virtual circuits, allowing both full and partial rerouting. The system has been deployed on the Brazilian RNP backbone.

Video-on-Demand (VoD) deployment over existing IP networks has recently gained significant popularity. Typically, the deployment of VoD is done in an arbitrary manner, without utilizing a proper engineering approach. In this paper, we present an engineering approach to deploy VoD services over IP-based hospitality networks, such as those networks seen in hotels and hospitals. In particular, our approach aims to determine the total number of VoD sessions that can be sustained by an existing hospitality network, while satisfying the QoS requirements of all network services, and at the same time leaving adequate capacity for future growth. We gauge the capacity of the hospitality network to sustain VoD services using both analysis and simulation. The capacity is gauged considering VoD quality of service requirements of throughput and delay constraints. Our analysis utilizes the principles of queuing theory, and our simulation is performed using OPNET simulation. Copyright © 2011 John Wiley & Sons, Ltd.

The deployment of VoD over existing IP networks is done these days in an arbitrary manner without applying proper engineering principles. This paper presents an engineering approach to deploy VoD services over IP-based hospitality networks, as those networks seen in hotels and hospitals. The proposed approach determines the total number of VoD sessions that can be sustained by an existing hospitality network, while satisfying the QoS requirements of all network services, and at the same time leaving adequate capacity for future growth.

Existing pervasive applications are based on time series data that possess the form of time-ordered series of events. Such applications also embody the need to handle large volumes of unexpected events, often modified on-the-fly, containing conflicting information, and dealing with rapidly changing contexts while producing results with low latency. Correlating events across contextual dimensions holds the key to expanding the capabilities and improving the performance of these applications. In this paper we analyze complex-event semantic correlation that examines epistemic uncertainty in computer networks by using Dempster–Shafer theory to support a high-volume, event-based, in-network and non-deterministic pervasive network management. We consider imprecision and uncertainty when an event is detected and associate a belief parameter with the semantics and the detection of composite events. The approach taps into in-network processing capabilities of pervasive computer networks and can withstand missing or conflicting information gathered from multiple participating entities. In the end, we establish that a lightweight, distributed, large-volume, event-based technique which exploits epistemic uncertainty to correlate events along contextual dimensions provides a successful technique for enabling management of large-scale and pervasive contemporary and future computer networks. Copyright © 2011 John Wiley & Sons, Ltd.

This paper analyzes complex-event semantic correlation that examines epistemic uncertainty in computer networks by using Dempster-Shafer theory to support a high-volume, event-based, in-network and non-deterministic pervasive network management. The work establishes that a lightweight, distributed, large-volume, event-based technique which exploits epistemic uncertainty to correlate events along contextual dimensions provides a successful technique for enabling management of large-scale and pervasive contemporary and future computer networks.