Blockchain technology has recently gained high popularity in data security, primarily to mitigate against data breach and manipulation. Since its inception in 2008, it has been applied in different areas mainly to maintain data integrity and consistency. Blockchain has been tailored to secure data due to its data immutability and distributive technology. Despite the high success rate in data security, the inability to identify compromised insider nodes is one of the significant problems encountered in blockchain architectures. A Blockchain network is made up of nodes that initiate, verify and validate transactions. If compromised, these nodes can manipulate submitted transactions, inject fake transactions, or retrieve unauthorized information that might eventually compromise the stored data’s integrity and consistency. This paper proposes a novel method of detecting these compromised blockchain nodes using a server-side authentication process and thwart their activities before getting updated in the blockchain ledger. In evaluating the proposed system, we perform four common insider attacks, which fall under the following three categories: (1) Those attacks targeting the Blockchain to bring it down. (2) the attacks that attempt to inject fake data into the database. (3) The attacks that attempt to hijack or retrieve unauthorized data. We described how we implement the attacks and how our architecture detects them before they impact the network. Finally, we displayed the attack detection time for each attack and compared our approach with other existing methods.

To enable virtual machines (VMs) with a large amount of memory to be flexibly migrated, split migration has been proposed. It divides a large-memory VM into small pieces and transfers them to multiple hosts. After the migration, the VM runs across those hosts and exchanges memory data between hosts using remote paging. For such a split-memory VM, however, it becomes difficult to securely run intrusion detection systems (IDS) outside the VM using a technique called IDS offloading. This paper proposes VMemTrans to support transparent IDS offloading for split-memory VMs. In VMemTrans, offloaded IDS can monitor a split-memory VM as if that memory were not distributed. To achieve this, VMemTrans enables IDS running in one host to transparently access VM's remote memory. To consider a trade-off, it provides two methods for obtaining memory data from remote hosts: self paging and proxy paging. We have implemented VMemTrans in KVM and compared the execution performance between the two methods.

Software Keyloggers are dominant class of malicious applications that surreptitiously logs all the user activity to gather confidential information. Among many other types of keyloggers, API-based keyloggers can pretend as unprivileged program running in a user-space to eavesdrop and record all the keystrokes typed by the user. In a Linux environment, defending against these types of malware means defending the kernel against being compromised and it is still an open and difficult problem. Considering how recent trend of edge computing extends cloud computing and the Internet of Things (IoT) to the edge of the network, a new types of intrusion-detection system (IDS) has been used to mitigate cybersecurity threats in edge computing. Proposed work aims to provide secure environment by constantly checking virtual machines for the presence of keyloggers using cutting edge artificial immune system (AIS) based technology. The algorithms that exist in the field of AIS exploit the immune system's characteristics of learning and memory to solve diverse problems. We further present our approach by employing an architecture where host OS and a virtual machine (VM) layer actively collaborate to guarantee kernel integrity. This collaborative approach allows us to introspect VM by tracking events (interrupts, system calls, memory writes, network activities, etc.) and to detect anomalies by employing negative selection algorithm (NSA).

The Distributed Network Protocol v3.0 (DNP3) is one of the most widely used protocols for smart grid communications. Security challenges which could cause great scale of damages to critical infrastructure like the smart grid have emerged in recent years. This paper investigates the attacks that target smart grids which utilize the DNP3 protocol, and how these attacks can be detected using the deterministic version of the Dendritic Cell Algorithm (DCA) which is an Artificial Immune System (AIS)-based algorithm. To evaluate DCAs effectiveness, we generated a testing dataset, by implementing various attacks then executing them in a DNP3 smart grid environment. We present our results, showing that these attacks can be successfully detected using an architecture based on the DCA.

This paper highlights different security threats and vulnerabilities that is being challenged in smart-grid utilizing Distributed Network Protocol (DNP3) as a real time communication protocol. Experimentally, we will demonstrate two scenarios of attacks, unsolicited message attack and data set injection. The experiments were run on a computer virtual environment and then simulated in DETER testbed platform. The use of intrusion detection system will be necessary to identify attackers targeting different part of the smart grid infrastructure. Therefore, mitigation techniques will be used to ensure a healthy check of the network and we will propose the use of host-based intrusion detection agent at each Intelligent Electronic Device (IED) for the purpose of detecting the intrusion and mitigating it. Performing attacks, attack detection, prevention and counter measures will be our primary goal to achieve in this research paper.

The IEEE 802.11 protocols are used by millions of smartphone and tablet devices to access the Internet via Wi-Fi wireless networks or communicate with one another directly in a peer-to-peer mode. Insider attacks are those originating from a trusted node that had initially passed all the authentication steps to access the network and then got compromised. A trusted node that has turned rogue can easily perform Denial-of-Service (DoS) attacks on the Media Access Control (MAC) layer by illegally capturing the channel and preventing other legitimate nodes from communicating with one another. Insider attackers can alter the implementation of the IEEE 802.11 Distributed Coordination Function (DCF) protocol residing in the Network Interface Card (NIC) to illegally increase the probability of successful packet transmissions into the channel at the expenses of nodes that follow the protocol standards. The attacker fools the NIC to upgrade its firmware and forces in a version containing the malicious code. In this paper, we present a distributed solution to detect and isolate the attacker in order to minimize the impact of the DoS attacks on the network. Our detection algorithm enhances the DCF firmware to enable honest nodes to monitor each other's traffic and compare their observations against honest communication patterns derived from a two-dimensional Markov chain. A channel hopping scheme is then used on the physical layer (PHY) to evade the attacker. To facilitate communication among the honest member stations and minimize network downtime, we introduce two isolation algorithms, one based on identity-based encryption and another based on broadcast encryption. Our simulation results show that the latter enjoys quicker recovery time and faster network convergence.

We will propose an adaptive distributed power management (DISPOW) algorithm that adaptively preserves network connectivity and cooperatively reduces interference for the dynamic environment and the changing network topology thereby improving the network performance. The proposed algorithm distributively builds a unique stable network topology tailored to its surrounding node density and propagation environment over random topologies in a dynamic mobile wireless model. We will show that DISPOW, with adequate power levels, provides strongly connected reliable network, reduces interference and routing overhead consequently providing a shorter packet delay and improving network throughput by as much as 37%. We will also show how it stabilizes node connectivity over the dynamic network and environment and attempts to prevent node from being completely disconnected from the network. Comparing with common node power strategy, DISPOW significantly reduces average node power level in highly attenuating environment and alleviates the dependency of common node power strategy on few nodes that are isolated in the network because of their physical location and surrounding network environment.

An innovative approach to determine the performance parameters of arbitrary multihop wireless networks is proposed. This approach can be applied to various types of multihop network topologies. In this paper we focus on a loop multihop network topology. Numerical results show that the normalized node capacity of loop networks is independent of the size of the loop. An individual node average channel busy probability is also introduced to allow for accurate performance analysis.

Video transmission is becoming an increasingly essential part of the current Internet; however before the high video quality at the client side becomes a reality, the best effort model has to adapt the video characteristic to the network conditions. In this paper, we are proposing a media middleware component that can be added to the edge routers in DiffServ network in order to have the responsibility to enhance QoS at the receivers' side. Passing such video frames to differentiated service network increases the video quality and treats the video frames based on their priorities. The media aware sub-layer has the responsibility to cache high priority video frames. We demonstrate through network simulation that the proposed technique decreases the average packet delay, and the packet loss, which yields a better video quality at the clients compared to existing techniques.

In this paper, we address the problem of collaborative video caching in ad hoc mobile networks. We consider network portraying static video server with wired interface to gateway node that is equipped with wireless interfaces, other nodes are requiring access to the video streams that is stored at video server. In order to reduce the average access latency as well as enhance the video accessibility, efficient video caching placement and replacement strategies are crucial at some of the distributed intermediate nodes across the network. Virtual backbone caching nodes will be elected by executing caching placement algorithm after running the routing protocol phase. The simulation results indicate that the proposed collaborative aggregate cache mechanism can significantly improve the video QoS in terms of packet loss and average packet delay.

Owing to the absence of any static structure, Ad-hoc networks are prone to packet losses and link failures. Selecting the shortest path may not lead to high quality routes. The consequent high loss rates can lead to the degradation of throughput and increase in the end-to-end delay on the selected path, which can lead to devastating effects for the transport of compressed video. This paper suggests a selection mechanism for multi-path routing protocols that selects routes on the basis of the expected throughput of the candidate routes. We have studied the performance of the proposed mechanism by incorporating it into a multi-path routing protocol, and examining the quality of compressed video streams in this network. Simulation results show that the proposed selection mechanism maintains the video quality under different loss rates and mobility speeds. Also we have found that our mechanism significantly outperform shortest path selection.

Enabling transmission over ad-hoc networks is more challenging than over conventional mobile networks because a connection path in an ad-hoc network is highly error-prone and the path can go down frequently. Although it is well-known that wireless channels are time varying because of user mobility and multi-path propagation effects, experiments for various types of channels show that the basic channel parameters can be stable for short time intervals. Therefore, a wireless channel can be adequately represented by a set of stationary channel models. There are number of wireless channel models proposed in literature. For simplicity, we used a two state Markov model known as a Gilbert-Elliot model and then in order to model more appropriately a real communication channel, a three state Markov model is analyzed here. We demonstrate our concept by applying it to the dynamic source routing protocol (DSR). In our proposed modified DSR, both the route discovery and route selection are based on physical layer parameter and the link monitoring function located at each node. Since routing selection based on the link quality is implemented, the minimum transfer delay from source to destination and the maximal throughput may be obtained. Simulation results show that according to the channel quality, the delay and throughput performance show remarkable performance when compared to traditional DSR.

Although network-wide flooding in reactive routing protocols like AODV facilitates the discovery of optimal routes, they engender large control overhead. Some efficient flooding methods cut down the overhead by minimizing the set of nodes participating in packet relays. When used for route discovery, however, these approaches may suppress the detection of optimal routes. A region-based routing (REGR) protocol was proposed in to effectively reduce route discovery overhead without sacrificing route optimality. REGR defines a pre-routing region and tries to find a high-quality route by broadcasting a route request message only within the region. Although the region-limited broadcasting reduces overhead, it is actually not efficient, and may experience frequent packet collision especially in dense networks. In this paper, we propose a progressive route calculation (PROQ) protocol to locate the optimal route around a preliminary route obtained from sketchy network scanning. We propose to use progressive Dijkstra's algorithm to make nodes on the preliminary route jointly derive a minimum cost spanning tree (MCST) between a source-destination pair. The topology information needed to derive the MCST is delivered along the preliminary route by using unicast packets, which makes the new protocol more efficient and reliable than broadcast approaches. We simulate PROC and compare it with various protocols. The simulation results verify the efficiency of our method.

Multipath routing is typically utilized in wireless mesh networks for fault tolerance, throughput enhancement, and load balancing purposes. To achieve good route resilience and optimal end-to-end transmission performances, most multipath routing algorithms tend to establish link-disjoint or node-disjoint paths with minimized path costs. Although multiple paths created by these algorithms do not share common links or nodes, they are usually located close to each other. If all these paths are activated simultaneously to boost end-to-end throughput, serious inter-path interferences will happen. In this paper, we first introduce an AODV-based decoupled multipath routing protocol named AODV-DM. AODV-DM builds an insulating region around each established path to prevent new paths from being formed in the neighborhood of the established path. By this means, several "isolated" paths can be created and utilized for simultaneous data transmissions. We also present a novel destination-polling scheme to improve medium access efficiency in destination area, and at the same time alleviate the out-of-sequence problem caused by concurrent packet transmissions through multiple paths. By optimizing the functions crossing network and MAC layers, we are able to complete the design of an efficient multipath structure targeting at end-to-end throughput enhancement. Simulation results demonstrate that the decoupled multipath structure can achieve much higher throughput than single-path and coupled-path cases

Designed and developed for military communications in their early stage, multi-hop wireless ad hoc network is attracting interest from commercial applications and becoming a research hot spot. Although the technology has been studied for about a decade, there still exist some problems to be solved. Route discovery latency for reactive routing protocols is one of them. In this paper, we propose a dynamic RTT-based algorithm which can dramatically reduce the route discovery latency. This algorithm can be generally applied to any source selection based reactive routing protocols. Our simulation study has also shown some interesting results about packet transmission time and route discovery latency in a contention-based channel like IEEE 802.11 which have not been discussed before.

The standard SCTP, RFC 2960, assumes that all packet losses are congestion losses. Thus, it does not perform well in wireless ad hoc networks where congestion losses and wireless losses coexist. This paper designs an alternative of SCTP, which is wireless-aware (W-A). The W-A SCTP is the first that simultaneously considers i) the nature of wireless losses (random or bursty losses), ii) the energy efficiency of battery powered ad hoc nodes, and iii) the goodput performance.

This paper introduces a resource management application of a probabilistic-based ant routing algorithm for mobile ad hoc networks (ARAMA) that is inspired from the ant's life . Mobile ad hoc networks (MANETs) are highly dynamic, self-configured and self-built networks. The goal of this paper is to present ARAMA ability to manage MANET's resources by achieving fair network resources distribution, while considering the dynamic characteristics of MANETs and the need for low control overheads. This paper provides a description for the algorithm. In this algorithm, the nodes' (node's energy, processing power, ...) and links' (bandwidth, ...) parameters are measured and collected in the nodes' indices. A path index is used to measure the path total resources and serves to minimize the forward control packet (ant) size. The concepts of negative backward ant destination trail are introduced to enhance the performance of the algorithm. The simulation results show the potential of ARAMA to achieve fair energy usage across the network nodes as an example of the network resource management. More, the results show the general ability of the algorithm to solve MANET's routing problem.

In wireless sensor networks, the multihop routes between sensors and sinks (data collectors) suffer breakage or damage because of the movement of targets and sinks, or the power exhaustion of some sensors. An efficient route maintenance and update protocol should give attention to both route rediscovery overhead and route optimization. In most flooding based routing protocols, route discovery/rediscovery packets are broadcast omni-directionally, which causes significant communication overhead as well as wireless channel contention. Query localization (QL) protocol (Castenada, R. et al., ACM/Kluwer Wireless Networks Journal, vol.8, no.2, p.137-51, 2002) limits the broadcast of route rediscovery packets within the local region of the old routes. However, as new routes formed under QL have to share most parts with old routes, route optimization is penalized. We propose to update the route by using the information of the old route, but at the same time, select a new route not necessarily overlapping the old one. This virtue is especially important to route-wide update cases, such as continuous movement of targets and sinks, and renovation of worn-out routes. Simulations show that, our new protocol always builds a fresher and better route with a very low control overhead.

Under many circumstances, broadcasting is the only way of service discovery, device paging, and even data transfer in wireless ad hoc networks. However, broadcasting may lead to "broadcast storm" problems if it is done by simple flooding in a single common channel. The problems include redundant rebroadcast (overlaying), contention and collision. When the network density is high or the network is getting congested, the problems become severe and may directly lead to degraded performance of the network. We propose a novel broadcast scheme that is simple, highly efficient and scalable. Most importantly, it is a fully distributed scheme which introduces no control overheads (in terms of control traffic), and can adapt to network movement at no extra cost. We evaluate our scheme by OPNET simulation and compare it with existing methods.

With the popularity of multi-homed hosts, which can be reached through multiple paths, a new challenge arises on how to efficiently and simultaneously use multiple paths between multi-homed communication hosts to explore potential benefits of multi-homing feature, such as to improve connection reliability, to increase throughput etc. Standard SCTP cannot support the multi-homing feature efficiently due to its per association congestion control structure. Per association congestion control structure makes it unfeasible to extend standard SCTP for load sharing, load balancing and other multi-homing applications. In this paper, we identified and analyzed limitations of the congestion control mechanism of standard SCTP in the context of using multiple paths simultaneously. We proposed an independent per path congestion control principle for standard SCTP to overcome these limitations. Algorithms are provided to realize independent per path congestion control in standard SCTP without modifying SCTP DATA chunk format and without introducing overhead to the network. SCTP with independent per path congestion control not only overcomes existing flaws of standard SCTP in supporting the multi-homing feature but also demonstrates its applicability to various multi-homing applications.

In this paper, we address various aspects related to extending and engineering stream control transmission protocol (SCTP) to utilize its multi-homing support for simultaneous transmission of data chunks on different path, while maintaining the congestion control on each path to ensure fair integration with other traffic in the network. The extended SCTP, referred to as load sharing-SCTP (LS-SCTP), is able to aggregate the bandwidth of all the active transmission paths between the communicating endpoints. It monitors the paths, and accordingly chooses the paths that are suitable for load sharing. In addition, LS-SCTP retransmission mechanism accelerates the delivery of missing data to the receiver to prevent stalling the transport connection while waiting for missing data chunks. Simulation results show that LS-SCTP is extremely beneficial for networks with limited bandwidth and failure prone links.

We propose a mechanism that combines retransmission-based error control with path diversity, to provide different levels of protection to interactive video in ad-hoc networks. The mechanism factors in the importance of the retransmitted packets to the reconstructed video quality as well as the end-to-end latency constraints to minimize the overhead and maximize the reconstructed video quality at the receiver. Simulation results show that the proposed retransmission mechanism maintains the video quality under different loss rates and with less overhead compared to error control methods that depend on controlling the intra-update rate. In addition, the mechanism is shown to be more robust to wireless losses than schemes that combine layered coding with path diversity.

The current stream control transmission protocol (SCTP) does not work well over lossy links. To make SCTP congestion control algorithms robust in lossy networks, the paper first introduces a fine-tuned explicit congestion notification (ECN) mechanism for SCTP in such environment, and then discusses the ECN-D SCTP that can differentiate noncongestion losses from congestion losses. We identify the optimal value of the congestion window for an SCTP source in response to ECN messages in order to maximize the throughput and maintain relatively small end-to-end delay. A simple and practical method to achieve the optimal value is developed by carefully choosing the threshold of queues that support ECN. Because the total goodput performance of SCTP associations is not sensitive to window reduction policies when the network load is heavy, and because fine-tuning SCTP or transmission control protocols congestion window in response to congestion indications using complicated methods may not be worth the increase in complexity of the protocol, the simplified method becomes attractive in achieving the optimal congestion window.

Routing algorithms in mobile ad-hoc networks (MANETs) should match the special characteristics of MANETs. It should be A self-configured, self-built and distributed routing algorithm. In addition, it should be able to optimize more than one quality of service parameter to achieve good network performance. The remaining energy in each node across the network should be fairly distributed. We are propose the ant routing algorithm for Mobile Ad-hoc networks (ARAMA), which is a biological based routing algorithm. This algorithm is inspired from the ant colonies' life. Forward packets are used to collect information about the network and backward packets are used to update the routing information in the nodes. The paper provides a description of the algorithm. Simulation results show the ability of the algorithm to fairly distribute the remaining energy across the network nodes and solve the MANETs routing problem under different dynamic conditions.

Bluetooth scatternets, integrating polling, and frequency hopping spread-sprectrum in their medium access control protocol, provide a contention-free environment for Bluetooth devices to access the medium and communicate over multihop links. Currently, most available scatternet formation protocols tend to interconnect all Bluetooth devices at the initial network startup stage and maintain all Bluetooth links thereafter. Instead of this "big scatternet" approach, we propose a scatternet-route structure to combine the scatternet formation with on-demand routing, thus eliminating unnecessary link and route maintenances. To the best of our knowledge, this is the first effort to address on-demand scatternet formation with every detail. We introduce an extended ID (EID) connectionless broadcast scheme, which, compared with original Bluetooth broadcast mechanism, achieves very much shortened route discovery delay. We also propose to synchronize the piconets along each scatternet route to remove piconet switch overhead and obtain even better channel utilization. Furthermore, we present a route-based scatternet scheduling scheme to enable fair and efficient packet transmissions over scatternet routes. Network performance analysis and simulations show that scatternet routes can provide multihop wireless channels with high network utilization and extremely stable throughput, being especially useful in the transmission of large batches of packets and real time data in wireless environment.

In this paper, we report and reveal an unfairness problem among TCP connections in an IEEE 802.11-based wireless multi-hop network. This problem is not the same as those unfairness problems reported before. In each of the cases we identified, the one TCP connection is completely shut down, even if it starts much earlier than the competing TCP traffic. By illustrating the TCP layer and MAC layer traces, we show that this kind of unfairness problem is rooted in the IEEE 802.11 MAC layer. The hidden node problem and the exposed node problem along with the exponential back-off scheme in the MAC layer are the major causes for that problem.

Abstract Wireless packet ad hoc networks are characterized by multi‐hop wireless connectivity and limited bandwidth competed among neighboring nodes. In this paper, we investigate and evaluate the performance of several prevalent TCP algorithms in this kind of network over the wireless LAN standard IEEE 802.11 MAC layer. After extensively comparing the existing TCP versions (including Tahoe, Reno, New Reno, Sack and Vegas) in simulations, we show that, in most cases, the Vegas version works best. We reveal the reason why other TCP versions perform worse than Vegas and show a method to avoid this by tuning a TCP parameter— maximum window size. Furthermore, we investigate the performance of these TCP algorithms when they run with the delayed acknowledgment (DA) option defined in IETF RFC 1122, which allows the TCP receiver to transmit an ACK for every two incoming packets. We show that the TCP connection can gain 15 to 32 per cent good‐put improvement by using the DA option. For all the TCP versions investigated in this work, the simulation results show that with the maximum window size set at approximately 4, TCP connections perform best and then all these TCP variants differ little in performance. Copyright © 2001 John Wiley & Sons, Ltd.

In this paper, we report and reveal a so-called "incompatibility" problem among TCP connections in an IEEE 802.11-based wireless multi-hop network. This problem is not the same as those unfairness problems reported before. In each of the cases we identified, one TCP connection is completely shut down at a time, and, after a while, it will kill the other TCP connection-just like a turn-over. There is no unfairness in the long-term average throughput. By illustrating the TCP layer and MAC layer traces, we show that this kind of incompatibility problem is rooted in the IEEE 802.11 MAC layer. The hidden node problem and the exposed node problem, along with the exponential back-off scheme in the MAC layer, are the major causes for that problem.

The regional foreign agent Mobile-IP approach was introduced to reduce the frequency by which registration with the remotely located home agent is needed. Providing an efficient system that support IP multicast, in an environment where the multicast group members frequently change their locations, is a challenge for systems providing mobility support. In addition, sensitive applications require a reliable multicast service such that each intended receiver is guaranteed to receive a copy of each transmitted packet. We propose an architecture that supports IP multicast in an environment where mobility is supported using regional foreign agents. The proposed scheme takes advantage of the inherent characteristics of the local registration to enhance the efficiency of the multicast support. Performance aspects such as multicast datagram delay, delivery cost and robustness are analyzed. Simulation results are used to describe the performance under various conditions. In addition, the proposed approach is extended to support reliable multicast. The proposed scheme tolerates the failure of regional foreign agents, and compensates for those datagrams undelivered due to such failures and for those due to the mobility of the receivers.

We report the so-called "TCP instability problem" in 802.11-based wireless mobile multi-hop networks. Our results show that the throughput performance of multi-hop TCP traffic in such a network may suffer from severe oscillation. By illustrating the multi-layer traces, we show that this problem is rooted in the MAC layer. Furthermore, a resolution is proposed to eliminate this problem.

In multi-hop wireless networks, the TCP data packet stream has to compete for limited bandwidth with its ACK stream. In this paper, we investigate and evaluate the performance of several prevalent TCP algorithms using the delayed acknowledgment (DA) option defined in IETF RFC 1122, which allows the TCP receiver to transmit an ACK for every two incoming packets. After extensively comparing the existing TCP versions (including Tahoe, Reno, New Reno, Sack and Vegas) by simulation, we show that with the DA option, the TCP connection can gain 15-32% output improvement. We strongly recommend using the DA option in TCP applications in wireless multi-hop networks.

Generally speaking, the parameter-based call admission control (CAC) approaches are conservative and lead to low network utilization since they use a worst-case model to ensure complete commitment conformance. Even if additional information of the new connection is available, they cannot take advantage of it. In this paper, we introduce some enhancements to the existing parameter based CAC algorithms for those applications in which one or two simple additional traffic characteristics are available, i.e. mean rate of the new connection and its rate variance. As well as showing how to do this, we give the reasons why we do this. We show that with these simple additional parameters, a much higher network utilization can be achieved. The numerical results demonstrate that the number of calls admitted to the network with our schemes is very close to (even equals) the best value.

In this paper, a modular clutter rejection technique using region-based principal component analysis (PCA) is proposed. Our modular clutter rejection system uses dynamic ROI (region of interest) extraction to overcome the problem of poorly centered targets. In dynamic ROI extraction, a representative ROI is moved in several directions with respect to the center of the potential target image to extract a number of ROIs. Each module in the proposed system applies region-based PCA to generate the feature vectors, which are subsequently used to decide about the identity of the potential target. We also present experimental results using real-life data evaluating and comparing the performance of the clutter rejection systems with static and dynamic ROI extraction.

We study the link-sharing between the assured real-time traffic and the throughput intensive best-effort services. Our simulation shows that the throughput intensive best-effort services can degrade the QoS performance of real-time services if both are forwarded to a single FIFO queue. Therefore, it is advantageous to assign separate queues for individual classes of traffic. A novel link-sharing scheme (delay-sensing) is proposed to improve the QoS measures such as delay and jitter for the real-time traffic while preserving the fairness among different traffic classes. The new method guarantees an explicit delay bound for the real-time traffic when bandwidth borrowing is allowed. Another advantage of the method is that it can offer different priority classes with user specified fairness.

We investigate and evaluate the performance of these two TCP variants in a wireless mobile ad hoc network. The results in this paper are based on simulations using the NS2 network simulator from Lawrence Berkeley National Laboratory (LBNL), with extensions from the MONARCH project at Carnegie Mellon. We present our experimental results in two parts. The first part of the experiments does not include node movement. The effect of mobility and the link breakage are considered in the second part experiments. We consider one type of topology in the first part experiments: a string topology with 8 nodes as shown. The distance between any two-neighbor nodes is equal to 200 m, which lets a node can only connect to its neighbor node. In other words, only those nodes between which a line exists can directly communicate. In this performance study, we set up a single TCP connection between a chosen pair of sender and receiver nodes and measured the successively received packets over the lifetime of the connection. The TCP good-put result was measured for each connection, averaged over ten runs. We compare these two TCP variants and discuss the effect of TCP maximum window size window.

ATM supports a wide range of multimedia traffic. While ATM provides increased flexibility in supporting various types of traffic the traffic control problems become more difficult to solve when trying to achieve efficient use of network resources. One of such problem is the bandwidth allocation. This paper presents a new approach to the bandwidth allocation problem for video traffic using radial basis functions neural networks (RBFNN). Estimation of an accurate amount of bandwidth to support this traffic has been a challenging task using conventional algorithmic approaches. In this paper, we show that a radial basis function neural network (RBFNN) is capable of learning the non-linear multi-dimentional mapping between different multimedia traffic patterns quality of service (QoS) requirements and the required bandwidth to support each call. Our method employs "on-line" measurements of the traffic count process over a monitoring period which is determined such that the error in estimating the bandwidth is minimized to less than 3% of the actual value. In order to simplify the design of the RBFNN, the input traffic is preprocessed through a low pass filter in order to smooth all high frequency fluctuations. A large set of training data, representing different traffic patterns with different QoS requirements, was used to ensure that RBFNN can generalize and produce accurate results when confronted with new data. The reported results prove that the neurocomputing approach is effective in achieving more accurate results than other traditional methods, based upon mathematical or simulation.

The Base Mobile-IP, the Mobile-IP with Route Optimization extension and Local Registration Mobile-IP are different approaches to supporting the mobility of IP hosts. Each one of those schemes has its own advantages and drawbacks. The route optimization solves the triangular route problem and the corresponding larger delay associated with the Base Mobile-IP with the expense of larger overhead for location management purposes. The Local Registration Mobile-IP scheme was introduced to enhance the performance by processing the MN's registration requests at a local agent. The local registration approach may effect other aspects of the Mobile-IP systems as the fault tolerance. The Base Mobile-IP and Mobile-IP with the Route Optimization schemes can be implemented on top of Hierarchical Local Registration Mobile-IP, thus it is important to tackle the issues affecting the performance in those environments. We present a mechanism to enhance the performance of the Route Optimization Mobile-IP systems and a platform to support foreign agents fault tolerance in systems supporting local registration.

With the growing interests in distributed applications, efficient and scalable multicast routing is of great concern. Source-based tree multicast routing protocols, however, become very inefficient when used with large number of sources over large scale internetworks. Shared-tree based multicast routing protocols like PIM and CBT being developed by the IDMR working group of the IETF, trade efficiency of multicast delivery trees with scalability, but there are some issues open to further study. The work presented in this paper, the core-manager based multicast routing (CMMR) architecture, adopts shared-tree based techniques to achieve good scalability. It worked out an innovative approach to issues like core selection and multicast delivery tree management. The basic idea behind CMMR is that the core-manager (CM) keeps tracking cores of each multicast delivery tree in order to maintain a brief image of the tree. Consequently, new members will almost always be directed to a nearby core so that tree cost is well confined. However, not every new member has to actually consult the CM thanks to a self-growing scheme, maintaining control overhead low. Simulations show that the CMMR outperforms other existing shared-tree based multicast routing protocols like PIM, and can do as well as expensive near-optimal Steiner tree heuristics when average costs of multicast delivery trees are compared. Further improving on scalability is sought by using a hierarchical architecture, which requires the multicast addresses be administratively scoped in accordance with the hierarchy. Management of candidate cores are restricted within individual hierarchical regions. In addition, the CMMR scheme is recursive, meaning that all routers run the same algorithm regardless of their hierarchical levels.

Through an intensive experimental study, the characteristics of cell delay variation (CDV) and CTD in an ATM network transporting constant bit rate (CBR) traffic under diverse operating conditions are evaluated. We examine the effect of changing network parameters, such as the number of nodes and background load conditions, on the performance parameters of both CDV and CTD. Our results revealed the distributions of CTD and CDV could not be fit to any of the standard distributions (normal, gamma, etc.). The results also present, the autocorrelation behavior of CTD as a function of both load and number of travelled hops. A verification study of the square root law (SRL) of the ATM Forum, on estimating the CTD standard deviation with increasing number of network nodes is also presented. Our results indicate that the SRL reveal very close estimates only under the assumption of a uniform load conditions across all nodes in the network.

This paper proposes a connection managing scheme, called the connection server (CS), for supporting intra-domain connectionless multicast over ATM. The intention of using CS is to relieve multicasting hosts from directly being involving in ATM call setup and release, so that the dynamic membership transition of a host group has minimal impact on all steady members in the group. It also eliminates the needs for dynamic address resolution, and enables virtual circuit (VC) reuse at membership management level. The paper presents the proof-of-concept implementation of the CS system, including the server/client signaling protocol and the ATM user datagram protocol (AUDP) which is designed as a typical CS application to provide multicast users with efficient transport services.

The goal of this paper is to propose a new perceptually-based objective technique that uses radial basis functions neural networks, instead of regression algorithms, to estimate the nonlinear mapping function that best represents the relationship among input (perceptual parameters) and output (speech quality) variables in a database. In the proposed technique, the perceptual parameters are obtained by: (1) emulating several known features of perceptual processing of speech sounds by the human ear (including critical-band masking, equal loudness, and the intensity-loudness power law operations) to map the speech power spectrum into the auditory power spectrum (bark domain), (2) deriving the perceptual LPC coefficients from the auditory spectrum that is used to calculate, for each frame, the cepstrum distance between the input and the output coded speech signals; (3) using the radial basis functions neural network to map the perceptual cepstrum distance per frame into the corresponding estimated speech quality. After extensive experimentation and validation of the proposed techniques, the results indicate that the proposed technique is shown to be effective for estimating the coded speech quality.

We propose a new neurocomputing call admission control (CAC) algorithm for asynchronous transfer mode (ATM) networks. The proposed algorithm employs neural networks (NNs) to calculate the bandwidth required to support multimedia traffic with multiple quality-of-service (QoS) requirements. The NN controller calculates the bandwidth required percall using on-line measurements of the traffic via its count process, instead of relying on simple parameters such as the peak, average bit rate and burst length. Furthermore, to enhance the statistical multiplexing gain, the controller calculates the gain obtained from multiplexing multiple streams of traffic supported on separate virtual paths (i.e., class multiplexing). In order to simplify the design and obtain a small reaction time, the controller is realized using a hierarchical structure of a bank of small size, parallel NN units. Each unit is a feed-forward back-propagation NN that has been trained to, learn the complex nonlinear function relating different traffic patterns and QoS, with the corresponding received capacity. The reported results prove that the neurocomputing approach is effective in achieving more accurate results than other conventional methods that are based upon mathematical or simulation analysis. This is primarily due to the unique learning and adaptive capabilities of NNs that enable them to extract and memorize rules from previous experience. Evidently such unique capabilities poise NNs to solve many of the problems encountered in the development of a coherent ATM traffic management strategy.

Objective measure techniques that provide reliable performance evaluation of speech coders is indispensable to designers, to end users when selecting telecommunication services and to providers in the designing stage of the telecommunications facilities. We introduce our objective measure technique that: (1) emulates several known features of perceptual processing of speech sounds by the human ear to map the speech power spectrum into an auditory power spectrum that is used to calculate the bark spectral distance per band (BSDB) between the input and the output coded speech signals; and (2) uses the abductive networks to estimate the speech quality from the BSDB. The results indicate that our proposed technique is reliable and robust in evaluating the coded speech quality and it is highly correlated to human responses across a wide range of quality levels and for a wide range of speech processing.

This paper discusses major issues involved in the design and implementation of a multipoint multimedia conference system, such as system architecture, conference management, session control, and intramedium and intermedia synchronization. In particular, emphasis is given to conference management and adaptive synchronization algorithms. The management of multiparticipants is based upon a distributed architecture for greater flexibility. The proposed synchronization algorithm is adaptive to network changes, eliminates the need for a global clock, and is immune to the clock frequency drift, while its realization is very simple and the involved overhead is minimal. The essence of the algorithm is partitioning the vicinity of the arrival epochs of multimedia objects into three regions and counting arrivals at each region. The function of the synchronizer is to shift the playback clock (PBC) according to the individual counter contents. The ideas proposed are implemented within a teleconference system on the Ethernet/FDDI using the TCP/UDP. Experimental results show that the proposed synchronization algorithm performs well in our network testbed environment.

An algorithm related to synchronization problems arising in a multimedia multiparty conference is presented in this paper. The proposed algorithm determines the set of packets generated periodically from different participants that are arriving at a node, either for mixing at the master of a conference, or for simply playing back at a regular participant of a conference. No global synchronization of the clocks is assumed. In this paper, the statistical approach rather than the deterministic is used to determine the proper set of packets that have to be mixed at a given time. The statistics are derived from the time stamp each packet carries along with it. The essence of the proposed algorithm is to estimate the expected packet arrival time (or reference time) for each participant. With the reference time at hand, the maximum jitter and the optimum waiting time for a mixer to wait packets from all participants can be determined. By employing feedback, the proposed algorithm is shown to lead to an optimum waiting/delay time. The error of the proposed algorithm is enumerated by the Chernoff bound, demonstrated by simulation, and shown to be acceptable in practical application.

In this paper, an algorithm to determine the set of packets generated continuously and periodically from different participants that are arriving at a node either for mixing at the master of a conference, or for simply playing back at a regular participant of a conference, is proposed. The essence of the algorithm is to estimate the average packet arrival time (or reference time) for each participant. With the reference time at hand, the maximum jitter and the optimum waiting time for a mixer to wait packets from all participants can be determined. The error of the proposed algorithm is enumerated by the Chernoff bound and is shown to be acceptable in practical application.

In this paper, an algorithm to determine the set of packets generated continuously and periodically from different participants that are arriving at a node either for mixing at the master of a conference, or for simply playing back at a regular participant of a conference, is proposed. The essence of the algorithm is to estimate the average packet arrival time (or reference time) for each participant. With the reference time at hand, the maximum jitter and the optimum waiting time for a mixer to wait packets from all participants can be determined. An enhancement to improve synchronization which deals with the estimation of the time offsets between the individual periods of the sources and the period of the receiver is also presented. The error of the proposed algorithm is enumerated by the Chernoff bound and is shown to be acceptable in practical application.>

Experimental results are presented for both passive and active overhead infrared systems to detect and monitor vehicular road traffic. The active system developed uses lasers and detectors operating in the near infrared spectral range. The passive system uses detectors operating in the thermal infrared region. Both systems make use of computerized signal processing and correlation techniques to determine presence of vehicles, traffic volume, vehicle speed measurement and length classification. Preliminary road tests confirmed the potential of both systems to provide effective, rugged, low cost approaches to vehicular traffic monitoring suitable for use on multilane highways. The comparative advantages of both systems are discussed and their potential for integration in intelligent vehicle highway systems are examined.>

Discusses some of the options for compressed video communications like SMDS, ISDN, VSATs and POTS. CCTV cameras mounted along the freeway are the probes to provide 100 percent coverage of the highway in real time. The major purpose of video surveillance is monitoring traffic flow and the detection and verification of an incident in real time. This incident detection and verification in real time, will be very helpful for the law enforcing agencies, so that they can pinpoint the incident location, rescue the people with minimum time delay and restore the traffic flow to its normal. The pan, tilt and zoom function of a CCTV camera can be automatic as well as manual. The application is typical and does not require high picture quality at higher costs. The authors consider freeze frame system for video transmission at low data rates on wire/wireless due to its low cost, ease of installation and maintenance. The system transmits data or picture @ 5 fps rate dependent upon the activity in the picture with SNR greater than 30 dB at very low cost.>

ATM has been recommended by the CCITT as the transport vehicle for the future B-ISDN networks. In ATM-based networks, a set of user declared parameters that describes the traffic characteristics, is required for the connection acceptance control (CAC) and traffic enforcement (policing) mechanisms. At the call set-up phase, the CAC algorithm uses those parameters to make a call acceptance decision. During the call progress, the policing mechanism uses the same parameters to control the user's traffic within its declared values in order to protect the network's resources and avoid possible congestion problems. A novel policing mechanism using neural networks (NNs) is presented. This is based upon an accurate estimation of the probability density function (pdf) of the traffic via its count process and implemented using NNs. The pdf-based policing is made possible only by NNs because pdf policing requires complex calculations, in real-time, at very high speeds. The architecture of the policing mechanism is composed of two interconnected NNs. The first one is trained to learn the pdf of "ideal nonviolating" traffic, whereas the second is trained to capture the "actual" characteristics of the "actual" offered traffic during the progress of the call. The output of both NNs is compared. Consequently, an error signal is generated whenever the pdf of the offered traffic violates its "ideal" one. The error signal is then used to shape the traffic back to its original values.>

Accurate characterization of the multimedia traffic is essential, in asynchronous transfer mode (ATM) broadband networks, in order to develop a robust set of traffic descriptors. Such a set is required, by the usage parameter control (UPC) algorithm, for traffic enforcement (policing). The authors present a novel approach to characterize and model the multimedia traffic using neural networks (NNs). A backpropagation NN is used to characterize and predict the statistical variations of the packet arrival process resulting from the superposition of N packetized video sources and M packetized voice sources. The accuracy of the results is verified by matching the index of dispersion for counts (IDC), the variance, and the autocorrelation of the arrival process to those of the NN output. The results show that the NNs can be successfully utilized to characterize the complex nonrenewal process with extreme accuracy.>

Variable bit rate (VBR) coding techniques have received great research interest as very promising tools for transmitting bursty multimedia traffic with low bandwidth requirements over a communication link. Statistically multiplexing the multimedia bursty traffic is a very efficient method of maximizing the utilization of the link capacity. The application of computer simulation techniques in analyzing a rate-based access control scheme for multimedia traffic such as voice traffic is discussed. The control scheme regulates the packetized bursty traffic at the user network interface of the link. Using a suitable congestion measure, namely, the multiplexer buffer length, the scheme dynamically controls the arrival rate by switching the coder to a different compression ratio (i.e., changing the coding rate). VBR coding methods can be adaptively adjusted to transmit at a lower rate with very little degradation in the voice quality. Reported results prove that the scheme greatly improves the link performance, in terms of reducing the probability of call blocking and enhancing the statistical multiplexing gain.>

A circuit-switched network with two types of traffic is modeled as a finite two-dimensional Markov chain. The problem of trunk congestion under different routing schemes is analyzed. It is shown that the trunk reservation technique can be used to alleviate trunk congestion in this heterogeneous traffic environment. It is also shown that two control schemes, namely, the restricted access and preemption, when properly applied, may minimize the blocking of one type of traffic while keeping that of the other at a certain level. Some numerical examples are presented.>

A continuous-time Markov-chain model for an asynchronous communication spread-spectrum code-division-multiple-access (CDMA) packet radio network is developed. The network is composed of mutually independent users. The receiver-based code is considered; a terminal with a packet to send looks up the destination's code and transmits on that code. Each user senses the channel load and refrains from transmission if the channel load exceeds the channel threshold. The model makes it possible to study the threshold effect of channel load on the performance of the CDMA packet radio network. Improvements in performance of spread-spectrum packet radio networks due to channel-load sensing are shown. Steady-state results for throughput are obtained.>

The authors present a continuous-time Markov chain model for a single-hop network using the unslotted Aloha scheme. Two interconnected networks using the unslotted Aloha scheme have also been modeled, assuming that the gateway has a finite buffer. The mathematical model derived from the unslotted Aloha system is a special case of the model developed by R.R. Roy and T.N. Saadawi (1984). Throughput, delay, blocking probability, and throughput-delay tradeoffs are presented for the gateway, each of the networks, and the entire system of interconnected networks.>

The service quality of a circuit-switched network is usually measured in terms of probability of rejecting a call and the delay encountered by the customers in accessing the common resources. In this paper, a simplen-dimensional birth-death steady-state traffic model is used to evaluate the blocking probability experienced byndifferent classes of customers in a two-way interactive distributed community antenna television (CATV) communication system. In the computation of the end-to-end blocking probability, we have assumed that the links between the call's originating node and the destination node are statistically independent. It is assumed, without loss of generality, that the number of channels (servers) is the same for all the links. In addition to the closed-form solution obtained for different cases, a simple recursive formula that simplifies the computational complexity is presented.

Most of tile present approaches to two-way interactive CATV focus on the centralized approach, in which all traffic must travel to the CATV head end. In this paper, we present a distributed approach, in which intelligent store-and-forward packet switches are to be installed at various locations on the cable. The analysis emphasizes three main criteria: the maximum traffic flow on the cable, the throughput of each switch, and the average message delay. Two possible schemes are compared. In one, the switches appear on the main trunk only. In the second, switches are used at branch locations as well.

The commentors point out a problem with equation (6) of the above-named work (ibid., vol. COM-27, pp. 857-868, June 1979) that affects the subsequent values determined in the derivations presented and may make the paper's results inconclusive.