Wireless covert systems use low probability of detection (LPD) techniques to hide transmission and protect the transmitter. Traditionally, spread spectrum has been used to hide the covert signal in noise. Recently, research has focused on covert systems that sense the RF channel and hide their transmission in the RF band of a primary network (PN). A widely used metric to quantify covertness is the ratio of the signal-to-interference-noise ratio (SINR) of the covert signal at the intended receiver to that at an intercept receiver (IR). Analyzing this metric is challenging in the presence of PNs and, hence, has been studied in the literature for simple specific cases using simulation only. In this paper, we use stochastic geometry to provide a comprehensive analysis of this metric for three main cases: noise-limited, interference-limited with perfect sensing of the environment, and interference-limited with imperfect sensing. We compute a lower bound on the system's achievable covertness, and demonstrate the advantages of using interference, instead of noise, to increase covertness. The effect of the system parameters on the covert system are discussed. Finally, simulation results confirm the theoretical analysis and the tightness of the proposed bound.

This paper shows the impact of interference generated by a covert link on the data link layer of a packet based (infrastructure) wireless network. Infrastructure networks employ orthogonal frequency division multiplexing (OFDM), adaptive modulation and coding (AMC), and hybrid auto-repeat-request (HARQ) to allow them to achieve higher data rates and increase capacity.

There has been increasing interest in incorporating sensing systems into objects or the environment for monitoring purposes. In this work we compare approaches to performing fully-distributed anomaly detection as a means of detecting security threats for objects equipped with sensing and communication abilities. With the desirability of increased visibility into the cargo in the transport chain and the goal of improving security, we consider the approach of equipping cargo with sensing and communication capabilities as a means of ensuring the security of the cargo as a key application. We have gathered real sensor test data from a rail trial and used the collected data to test the feasibility of the anomaly detection approach. The results demonstrate the effectiveness of our approach.

With growing demand for high-speed access to mobile handheld devices, there is a significant cost benefit in deploying fixed wireless-mesh networks for backhaul access. However, enabling reliable broadband access over high-frequency radios (such as millimeter-wave networks) posses a fundamental challenge due to weather disruptions in general and rain attenuation in particular. In this paper, we present an analysis of the impact of precipitation on millimeter-wave mesh networks based on radar measurements of real storms in the Midwest US. Furthermore, we compare two novel algorithms that use physical-layer information to optimize routing at the network layer: P-WARP (Predictive Weather-Assisted Routing Protocol) and XL-OSPF (Cross-Layered Open Shortest Path First). Finally, we present simulation studies to compare the performance of the proposed protocols and evaluate the dependability of the end-user service during weather disruptions.

Existing Internet Engineering Task Force (IETF) and International Telecommunications Union (ITU) optical control plane models do not consider the service profile layer when establishing network connections, this work proposes the novel concept of a Service Profile-Aware (SPA) control plane model that is superior to existing models. SPA includes a full realization of the multi-granularity network resources, and considers services' architectures and their associated service profile feature set. Detailed comparison between the three control plane models were considered from multiple dimensions including traffic management scheme, components level interaction between (service profile, control plane, network infrastructure) layers, and network infrastructure realization from both horizontal "network domains" and vertical "resource granularity and network partitions" perspective. The service models were analyzed based on their service profile parameters from both an architectural and performance perspective. Mathematical analysis of the three control plane models was performed using multi-instance Fixed Point Approximation (FPA). The performance analysis of the new SPA traffic management schemes found a significant increase in service allowed load while maintaining lower service blocking probability and network utilization as compared to the IETF and ITU control plane models.

Modern science projects in Polar Regions have increased the data communication needs of Polar research community considerably. Existing communication facilities, which are based on geostationary and geosynchronous satellite systems, are not suitable for small science camps. We have developed a reliable, truly mobile and lightweight iridium based communication system that can provide round the clock data and Internet access to Polar Regions. Using COTS components and inverse multiplexing methodology, an integrated field unit is developed. A graphical user interface (GUI) based link management software is used to handle all aspects of the connection autonomously. This paper presents the architecture of this system along with the results of field experiments that were conducted at Greenland in summer 2003

For Internet telephony to be a viable alternative to the public switch telephone network (PSTN), efficient and high quality communications are required. This paper proposes an optimization algorithm that selects parameters like the coding scheme, packet loss bound, and maximum link utilization level in a voice over IP (VoIP) network The goal is to deliver guaranteed quality of service (for voice) while maximizing the number of users served. A VoIP architecture is also discussed that could use optimization algorithms to dynamically provision VoIP networks.

Effective quality-of-service (QoS) metrics must relate to end-user experience. For multimedia services these metrics should focus on phenomena that are observable by the end user. Once a congestion event occurs in the network it tends to persist, resulting in long bursts of consecutive packet loss. Such an event is observable to the network customer. There is a need to increase our understanding of the temporal characteristics of congestion. It has become increasingly apparent that the temporal characteristics of congestion events have the dominant effect on user-perceived QoS. A rigorous definition of the time between congestion events is given here, as well as an associated prediction methodology. The inter-congestion event time or the rate of congestion events per unit time provides a network quality metric that is easily understandable to network users and is conveniently predicted and measured. The contribution of this paper is the definition of a metric to characterize congestion events and development of an analytic methodology to predict the expected number of congestion events per unit time. The proposed methodology is evaluated for a variety of traffic models.

This paper describes the impact of traffic handling mechanisms on network capacity for support of quality of service (QoS) in multi-service networks. The choice of which traffic handling strategy to employ requires a methodology that can be used to capture the trade-off between the different schemes and this is the focus of this paper. One key result of this work is that on the basis of capacity requirements, there is no significant difference between class-based traffic handling and per-flow traffic handling.

Overloads that occur during times of network stress result in blocked access to all users, independent of importance. These overloads can occur because of degraded resource availability or abnormally high demand. Public broadband networks must dynamically recognize some multimedia connections as having greater importance than others and allocate resources accordingly. A new approach to connection admission control is proposed that uses an upper limit policy to optimize the admission of connections based on the weighted sum of blocking across traffic classes. This results in a simple algorithm suitable for multimedia and packet networks. This work is also the first to demonstrate that the use of an upper limit policy is superior to traditional approaches of adding extra capacity or partitioning capacity, both in terms of the amount of resources required and sensitivity to load variations. An upper limit policy can also be deployed much faster when a large overload occurs from a disaster event.

For broadband networks to be widely useful to society, they must dynamically recognize some connections, like those that deal with emergencies or natural disasters, as having greater importance than others. This paper proposes an architecture of geographically distributed ticket servers that issue importance tickets for connection admission. User agents contact ticket servers using an agent communication language, then a ticket server intelligent agent determines how valuable of a ticket to issue within the current context of an emergency or natural disaster. Use of ticket servers and agent communication enables quick adaptation to dynamic context changes and provides high user satisfaction.

The rapidly deployable radio network (RDRN) is an architecture and experimental system to develop and evaluate hardware and software components suitable for implementing mobile, rapidly deployable, and adaptive wireless communications systems. The driving application for the RDRN is the need to quickly establish a communications infrastructure following a natural disaster, during a law enforcement activity, or rapid deployment of military force. The RDRN project incorporates digitally controlled antenna beams, programmable radios, adaptive protocols at the link layer, and mobile node management. This paper describes the architecture for the RDRN and a prototype system built to evaluate key system components.

To assist in the design and understanding of future global networks, this paper describes the practical and simulation experiences gained from a TCP/IP on ATM network over a high speed satellite link and presents performance comparison studies of such networks with the same host/traffic configurations over local area (LAN) and wide area (WAN) networks. It was found that the satellite systems deliver performance similar to the terrestrial networks regardless of their path latencies in cases where the communication channels exhibit a low bit error rate (BER). NASA's Advanced Communications Technology Satellite (ACTS), with its special characteristics and high data rate satellite channels, and the ACTS ATM Internetwork (AAI) were used in these experiments to deliver broadband traffic. Network performance tests were carried out using application-level software (ttcp, Netspec) on OC-3 and OC-12 ATM satellite links.

Experiments in asynchronous transfer mode (ATM) wide area network (WAN) environments are discussed in this work. The lessons learned through practical experiences with ATM systems and how these can be applied to improving performance are presented. Factors ranging from the transport protocol to the equipment used and the network conditions are considered.

A new method for in-service cell loss ratio (CLR) QoS estimation for asynchronous transfer mode (ATM) networks has been developed. For a typical CLR, a large number of cells must be observed before statistically meaningful results can be achieved. These results may be obsolete resulting in ineffective network management reactions. For a variety of arrival processes, many analyses have shown there exists a relationship between the logarithm of the cell loss probability and buffer size. For models that do not possess long-range dependence, this relationship is often linear. On the other hand, for the fractional Brownian motion model that captures long-range dependent traffic behavior, this relationship has a polynomial form. The proposed method uses these relationships and observations of cell loss for several small pseudo-buffers to characterize the behavior of the actual system. Specifically, a real-time technique to dynamically detect the failure of meeting a cell loss quality of service (QoS) objective has been developed. The method requires a short observation period and is suitable for in-service monitoring of CLR QoS. Simulation studies show the effectiveness of this method for both modeled traffic and measured network trace data.

Most algorithms proposed for controlling traffic prior to entering ATM networks are based on static mechanisms. Such static control mechanisms do not account for the dynamics of the user traffic or the network state. Some dynamic control algorithms have been proposed, but most of these algorithms are extremely complex and may make it difficult to provide real time control. In this paper, we present an adaptive rate control algorithm that has been implemented in hardware. The algorithm controls the traffic submitted by a source based on the indirectly observed average rate and burst size for the source. The algorithm is highly efficient and thereby provides real time control at high speed. Our implementation, in concert with flow control in the local area network, provides the basis for ATM-based high performance distributed systems.

This paper describes performance measurements taken in the MAGIC gigabit testbed relating to the performance of TCP in wide area ATM networks. The behavior of TCP with and without cell level pacing is studied. In particular, we focus on results that indicate that the TCP rate control mechanism alone is inadequate for congestion avoidance and control in wide-area gigabit networks. We also present results showing that TCP augmented by cell-level pacing addresses these problems and allows the full bandwidth capacity to be utilized. These results demonstrate the viability of high performance distributed systems based on wide area ATM networks given the proper ATM traffic management infrastructure.

This paper describes a gigabit LAN/WAN gateway being developed for the MAGIC gigabit testbed. The gateway interfaces a gigabit LAN developed by Digital's Systems Research Center and the MAGIC SONET/ATM wide area network. The gateway provides 622 Mb/s throughput between the LAN and WAN environments, and supports either a single STS-12c or four STS-3c tributaries. Traffic measurement capability and support for multiple bandwidth management schemes are provided by this architecture.>

The assessment of the cell loss performance of networks using asynchronous transfer mode (ATM) via Monte Carlo simulation incurs an enormous computational burden due to the need to estimate an event that has a very small probability of occurrence. Although importance sampling (IS) techniques have been proven useful in simulations of rate events related to bit error rate in digital communications and false alarm rate in radar systems, its application to ATM queuing problems with correlated input traffic has yet to be demonstrated. It is established that significant computational savings can be obtained using IS for correlated traffic by using regenerative properties of the underlying system and biasing the conditional arrival process. The results show that IS can reduce the computational burden by more than three orders of magnitude. Extensions of the methodology to more complex arrival processes are discussed. The foundation for applying IS to ATM systems given can be used to study congestion control as well as networks of ATM queues in the future.>

The authors describe the modeling and simulation of a frequency-division multiple-access (FDMA) satellite bandwidth-on-demand (BOD) service. The class of resource allocation processes to which the satellite BOD application belongs is identified and contrasted with more common resource allocation processes. Several unique features of BOD service complicate the performance specification, modeling, and analysis. The authors divide the modeling problem into three fundamental components: modeling the resource algorithm and modeling the processing of individual resource requests. These modeling components are illustrated with the satellite BOD application. In addition, simulation of the resource allocation requires three additional components: an easy-to-use user input interface, performance measures and statistics gathering, and presentation of simulation results (user output interface). The architecture of the simulation software (resource allocation analysis program, or RAAP) is presented in terms of these six components. To illustrate RAAP's capabilities, example simulation results are given.>

Overload control in asynchronous transfer mode (ATM) networks is dealt with via priority cell discarding mechanisms governed by a set of nested queue fill thresholds. An efficient search technique is developed which is applicable to any number of discarding priorities for determining the set of thresholds that optimize the offered load when constraints on cell loss (discarding) probabilities are specified. Results are presented as a function of queue capacity for given traffic distribution and arrival statistics. The results indicate that nested threshold discarding systems can support significantly higher traffic loads than systems without priorities. The sensitivity of the performance advantage to traffic mix and cell loss constraints is also illustrated. The efficient search procedure is shown to provide substantial computational gain when there are more than two discarding priorities.>

The performance of two high speed network protocols, IEEE 802.6 distributed queue dual bus MAN and fiber distributed data interface are compared based on their ability to support integrated traffic. The comparison includes the asynchronous priority characteristics of the two protocols as well as the performance characteristics of their isochronous service classes. In the integrated network considered, voice stations are assigned to the isochronous class of both protocols. Video sources, which are models of video CODEC systems for ATM environments, are given the highest priority of the asynchronous class for both protocols. The asynchronous data traffic is distributed equally among three asynchronous classes. The two protocols are then compared on the basis of their throughput, delay, network utilization, and fairness.>

A problem associated with overload control by packet discarding in integrated packet networks such as a broadband integrated services digital network (B-ISDN) is considered. The question of optimality in priority packet discarding is examined in order to minimize the service degradation associated with the discarding. An analysis model which is consistent with asynchronous transfer mode (ATM) operation of such a network is presented. It is shown how dynamic programming can be used with this model to find an optimal discarding policy, and initial results of such an analysis for queuing systems with a single buffer position are presented. It is shown by example that the optimal policy can perform significantly better over a wide range of parameter values than the default policy of discarding arriving packets only when the queue is full.>

A technique based on importance sampling is presented for extrapolating the end-to-end delay performance of a high-level data link control (HDLC) network. The lost packets can arise from bit errors or discarding due to network congestion. The technique works well for a substantial range of loads, as long as the desired performance is at an error rate less than the observed rate (the environment most frequently encountered). It is applicable across a range of error rates. The technique was tested on a simple HDLC link but would generalize to other link-level protocols. The extrapolation technique can be based on simulation or measurement. Simulation results are presented to demonstrate its effectiveness.>

A replacement technique for lost speech packets is presented. This technique is based on the classification of the packets into four distinct classes: background noise, voiced speech, fricatives, and other. Different encoding schemes and lost packet replacement techniques are used for each class. Results of subjective tests indicate that giving preferential delivery treatment to packets based on class can be used to improve subjective quality. The replacement strategy renders the reconstructed signal indistinguishable from the original utterance for packet loss rates up to 47% for background noise packets, 8% for fricative packets, and 4% for other packets. For voiced packets, the replacement is distinguishable from the original signal even for packet loss rates lower than 5%, but significant improvement may be possible by reducing the memory associated with the voiced speech coding process.>

This study analyzes traffic on a 10-Mb/s Ethernet local area network that connects PCs, minicomputers (running UNIX), engineering workstations, and other equipment in an electronics manufacturing environment. The network is artificially loaded by having up to 32 PCs exchanging large files with seventeen minicomputers. Traffic levels of over 50% of network bandwidth are generated. Several parameters including network utilization, collision rate, deferment rate, and channel acquisition time are measured. The results indicate that Ethernet is capable of handling large amounts of traffic with minimal delays.>

The authors discuss the control of short-term congestion, which is referred to as overload, in integrated packet networks (IPNs) containing a mix of data, speech, and possibly other types of signals. A system model is proposed that assigns a delivery priority to each packet (speech or otherwise) at the transmitter and discards speech packets according to delivery priority at any point in the network in response to overload. This model attempts to minimize per-packet processing at networks nodes. The research described is guided by two principles for IPN design: minimal per-packet processing and flexibility due to signal structure. The quality of the received speech is maintained by classifying speech segments according to their structure and coding them in a way that ensures ease of lost-packet regeneration at the receiver. The results of an experiment are reported that confirmed the general validity of this model from the standpoint of transmitter and receiver processing and subjective quality.>

The statistical nature of Monte Carlo techniques presents significant computational problems in the simulation of computer communications networks. A variety of techniques has been developed over the years to improve the computational efficiency of simulation models of general queuing networks, some of which have been applied to communication networks. The authors review the status of efficiency-enhancing techniques as they relate to the simulation of computer communications networks, present some novel approaches, and suggest promising areas for future research.>

The design of ICSSM, a nonreal time computer-aided simulation and analysis tool for communications systems, is presented, ICSSM is capable of supporting modeling, simulation, and analysis of any system representable in terms of a network of multiport functional blocks. Its applicability is limited only by the modeler's ingenuity to decompose the system to functional blocks and to represent these functional blocks algorithmically. ICSSM has been constructed modularly, consisting of five subsystems to facilitate the tasks of formulating the model, exercising the model, evaluating and showing the simulation results, and storing and maintaining a library of modeling elements, analysis, and utility subroutines. It is written exclusively in ANSI Standard Fortran IV language, and is now operational in a Honeywell DPS 7/80 M computer under the MULTICS Operating System. Description of a recent simulation using ICSSM and some generic moduels of general interest developed as a result of the modeling work are also presented.

Standard image processing techniques which are used to enhance noncoherent optically produced images are not applicable to radar images due to the coherent nature of the radar imaging process. A model for the radar imaging process is derived in this paper and a method for smoothing noisy radar images is also presented. The imaging model shows that the radar image is corrupted by multiplicative noise. The model leads to the functional form of an optimum (minimum MSE) filter for smoothing radar images. By using locally estimated parameter values the filter is made adaptive so that it provides minimum MSE estimates inside homogeneous areas of an image while preserving the edge structure. It is shown that the filter can be easily implemented in the spatial domain and is computationally efficient. The performance of the adaptive filter is compared (qualitatively and quantitatively) with several standard filters using real and simulated radar images.