Quality for all Abstracts   Towards a Traffic Theory Friendly Internet: the Cross-Protect Router  James Roberts, France Telecom R&D, France The key to service quality and the design of effective traffic controls is understanding the three-way "traffic-performance relation" linking demand, capacity and performance. In the talk we will summarize a number of principles derived from studies of this relation for both streaming and elastic traffic. These principles provide insight into the feasibility of proposed QoS architectures as well as their economic viability. We tend to find classical approaches deficient in several respects and have been led to suggest an alternative flow-aware networking architecture. The talk will include an outline of a new flow- aware router design called Cross-protect. Cross-protect combines implicit flow admission control and a novel per-flow scheduler to meet the respective quality requirements of streaming and elastic flows without the need for explicit service differentiation. Mobile QoS: Can we provide predictable services for unpredictably mobile users?  Jon Crowcroft, University of Cambridge, UK Surprise is the last thing the network traffic engineer wants. Whether operating at session timescales (through call admission control), load balancing (traffic engineering through load-sensitive routing) or provisioning for long term traffic matrix growth or alteration, it is the ability to extrapolate from past behaviour that allows some degree of optimisation to be achieved. The measurement study of session arrival patterns, source models and traffic trends in wireline internet now allows these tasks to be carried out well - for example, some major european phone companies now mix significant fractions of VOIP traffic with Web data in differentiated service networks and meet well described SLAs. However, wireless networks often support mobility; and mobile Internet user patterns have not been well studied yet. It is true that we have many mobile phone systems in operation, and mobility data can sometimes be obtained for study(suitably anonymized). However, much of that data is concerned with voice calls. As with fixed telephony, the user behaviour can also be controlled extensively through pricing. In wireless Internet access, where devices (e.g. cards) as well as people may be sinks or sources of data, we may see very different mobility patterns. Much research in the last few years has used naive (trivial) models of mobility (e.g.random waypoint) embedded into simulators in common use in the community. This talk is a plea to the community to accummulate and start to use realistic models, based on real measurement. While we do not have many wide area wireless mobile Internet users, we do have mobility models that can be obtained for pedestrian and vehicle traffic, and unlike the data gathered from commercial communications networks, this data is often neither propietary, nor terrifically hard to get. From it we may form parsimonious models that can then be used in simulations and analysis, to provide understanding of the performance of QoS mechanisms and therefore increase predictability of their implementation and deployment. This talk is about what we might do with some of that data. On the Impacts of Traffic Shaping on End-to-End Delay Bounds in Aggregate Scheduling Networks  Markus Fidler, Aachen University, Germany The Differentiated Services architecture allows for the provision of scalable Quality of Service by means of aggregating ows to a small number of traffic classes. Among these classes a Premium Service is defined, for which end-to-end delay guarantees are of particular interest. However, in aggregate scheduling networks the derivation of such worst case delays is significantly complicated and the derived bounds are weakened by the multiplexing of ows to aggregates. A means to minimize the impacts of interfering ows is to shape incoming traffic, so that bursts are smoothed. Doing so reduces the queuing delay within the core of the domain, whereas an additional shaping delay at the edge is introduced. In this paper we address the issue of traffic shaping analytically. We derive a form that allows to quantify the impacts of shaping and we show simulation results on the derivation of end-to-end delay bounds under different shaping options. An Adaptive RIO (A-RIO) Queue Management Algorithm  Julio Orozco, David Ros, IRISA/INRIA Rennes, France In the context of the DiffServ architecture, active queue management (AQM) algorithms are used for the differentiated forwarding of packets. However, correctly setting the parameters of an AQM algorithm may prove difficult and error-prone. Besides, many studies have shown that the performance of AQM mechanisms is very sensitive to network conditions. In this paper we present an adaptive AQM algorithm, which we call Adaptive RIO (A-RIO), addressing both of these problems. Our simulation results show that A-RIO outperforms RIO in terms of stabilizing the queue occupation (and, hence, queuing delay), while maintaining a high throughput and a good protection of high-priority packets; \mbox{A-RIO} could then be used for building controlled-delay, AF-based services. These results also provide some engineering rules that may be applied to improve the behaviour of the classical, non-adaptive RIO. Deterministic End-to-End Delay Guarantees in a Heterogeneous Route Interference Environment  Florian-Daniel Otel, Jean-Yves Le Boudec, EPFL, Switzerland Some of the known results for delivering deterministic bounds on end-to-end queuing delay in networks with constant packet sizes and constant link rates rely on the concept of Route Interference. Namely, it is required to know the number of flows joining on any output link in the whole network. In this paper we extend the existing results for the more generic cases of connection-oriented networks consisting of links with different capacities, carrying different traffic classes and packets of different sizes. Delay Bounds For FIFO Aggregates: A Case Study  Luciano Lenzini, Enzo Mingozzi, Giovanni Stea, University of Pisa, Italy In a Diffserv architecture, packets with the same marking are treated as an aggregate at core routers, independently of the flow they belong to. Nevertheless, for the purpose of QoS provisioning, derivation of upper bounds on the delay of individual flows is of great importance. In this paper, we consider a case study network, composed by a tandem of rate-latency servers that is traversed by a tagged flow. At each different node, the tagged flow is multiplexed into a FIFO buffer with a different interfering flow. For the case study network, we derive an end-to-end delay bound for tagged flow traffic that, to the best of our knowledge, is better than any other applicable result available from the literature. A Dynamic Bandwidth Allocation Algorithm for IEEE 802.11e WLANs with HCF Access Method  Gennaro Boggia, Pietro Camarda, Claudio Di Zanni, Luigi Alfredo Grieco, Saverio Mascolo, Politecnico di Bari, Italy This paper proposes a dynamic bandwidth allocation algorithm for supporting QoS in IEEE 802.11e WLANs with Hybrid Coordination Function (HCF) access method. It distributes the limited WLAN capacity by taking into account the desired queueing delay that multimedia data flows would expect. The algorithm has been designed by following a control theoretic approach and its properties have been analytically investigated. The effectiveness of our approach has been also proved by computer simulations, involving both audio and video flows. Both mathematical analysis and simulation results show that the algorithm guarantees queueing delays that are bounded by the QoS specifications. Performance Analysis of An Enhanced IEEE 802.11 Distributed Coordination Function Supporting Service Differentiation  Bo Li, Roberto Battiti, University of Trento, Italy As one of the fastest growing wireless access technologies, Wireless LANs (WLANs) must evolve to support adequate degrees of service differentiation. Unfortunately, current WLAN standards like IEEE 802.11 Distributed Coordination Function (DCF) lack this ability. Work is in progress to define an enhanced version capable of supporting QoS for multimedia traffic at the MAC layer. In this paper, we aim at gaining insight into two mechanisms to differentiate among traffic categories, i.e., scaling the minimum contention window size and the length of the packet payload according to the priority of each traffic flow. We propose an analysis model to compute the throughput and packet transmission delays. In additions, we derive approximations to get simpler but more meaningful relationships among different parameters. Comparisons with simulation results show that a very good accuracy of performance evaluation can be achieved by using the proposed analysis model. Scheduling Time-sensitive Traffic on 802.11 Wireless LANs  Martin Heusse, Paul Starzetz, Franck Rousseau, Gilles Berger-Sabbatel, Andrzej Duda, LSR-IMAG Laboratory, France In contrast to the common wisdom stating that 802.11 wireless LANs are not suitable for time-sensitive traffic, we have observed that in some conditions packet traffic transmitted over 802.11b may benefit from low delays even in saturation. Our analysis and measurements show that low delays can be obtained irrespectively of the greedy behavior of other hosts and without any traffic control mechanisms: when some hosts try to gain as much as possible of the transmission capacity of the radio channel, it is still possible for other hosts to experience low delay provided their packet rates are below some threshold value. The only situation in which a time-sensitive traffic source fails to obtain low delay is when its packet rate is too high with respect to its share of the channel capacity. We provide an analytical formula for determining the limiting packet rate that can be used to guide rate adaptive applications such as audio or video codecs to keep their output rates under the limiting rate and benefit in this way from low delays without any coordinated traffic control mechanisms. Bi-directional Search in QoS Routing  Fernando A. Kuipers, Piet Van Mieghem, Delft University of Technology, The Netherlands The bi-directional search method used for unicast routing is briefly reviewed. The extension of this method unicast QoS routing is discussed and an exact hybrid QoS algorithm HAMCRA that is partly based on bi-directional search is proposed. HAMCRA uses the speed of a heuristic when the constraints are loose and efficiently maintains exactness where heuristics fail. The performance of HAMCRA is simulated. The NAROS approach for IPv6 Multi-homing with Traffic Engineering  Cédric de Launois, Olivier Bonaventure, Marc Lobelle, Université catholique de Louvain, Belgium Once multihomed, an IPv6 site usually wants to engineer its interdomain traffic. We propose that IPv6 multihomed hosts inquire a so called "Name, Address and ROute System" (NAROS) to determine the source and destination addresses to use to contact a destination node. By selecting these addresses, the NAROS server roughly determines the routing. It thereby provides features like traffic engineering and fault tolerance, without transmitting any BGP advertisement and without impacting on the worldwide routing table size. The performance of the NAROS server is evaluated by using trace-driven simulations. We show that the the load on the NAROS server is reasonable and that we can obtain good load-balancing performances. Adaptive Multipath Routing Based on Local Distribution of Link Load Information  Ivan Gojmerac, Thomas Ziegler, Peter Reichl, Telecommunications Research Center Vienna, Austria Adaptive Multi-Path routing (AMP) is a new simple algorithm for dynamic traffic engineering within autonomous systems. In this paper, we describe an AMP variant which is related to the well-known Optimized Multi-Path (OMP) routing protocol. Whereas OMP requires global knowledge about the whole network in each node, the AMP algorithm is based on a backpressure concept which restricts the distribution of load information to a local scope, thus simplifying both signaling and load balancing mechanisms. The proposed algorithm is investigated using ns-2 simulations for a real medium-size network topologyand load scenarios by performing comparisons to several standard routing strategies. Statistical Point-to-Set Edge-Based Quality of Service Provisioning  Satish Raghunath, Shivkumar Kalyanaraman, Rensselaer Polytechnic Institute, USA In this paper we propose an edge-based quality of service architecture aimed at site-to-site private networks over the Internet. We extend the traditional point-to-point service model to a point-to-set service model, assuming a finite, bounded set of destination sites. Instead of provisioning point-to-point links between a source and its set of destinations, a point-to-set service allows the user to have an allocated bandwidth, which could be flexibly assigned to traffic going toward any destination within the set. The proposed point-to-set service provides low loss rates and {\em flexibility} to users while allowing providers to obtain multiplexing gains by employing a probabilistic admission control test. Simulation results are provided to demonstrate the utility of deploying such a model. Throughput Analysis of a Probabilistic Topology-Unaware TDMA MAC Policy for Ad-Hoc Networks  Konstantinos Oikonomou, INTRACOM S.A., Greece and Ioannis Stavrakakis, University of Athens, Greece The existing topology-unaware TDMA-based schemes are suitable for ad-hoc networks and capable of providing a minimum guaranteed throughput by considering a deterministic policy for the utilization of the assigned scheduling time slots. In an earlier work, a probabilistic policy that utilizes the non-assigned slots according to an access probability, common for all nodes in the network, was proposed. The achievable \textit{throughput for a specific transmission} under this policy was analyzed. In this work, the \textit{system throughput} is studied and the conditions under which the system throughput under the probabilistic policy is higher than that under the deterministic policy and close to the maximum are established. Study of the Capacity of Multihop Cellular Networks  Antonis Panagakis, Elias Balafoutis, Ioannis Stavrakakis, University of Athens, Greece Recently, the application of the peer to peer networking paradigm (typical for an ad hoc network) has been proposed for wireless local area networks (WLANs), instead of the traditional cellular networking paradigm. In this paper the performance of a WLAN employing the peer to peer networking paradigm is studied via simulations; the results indicate that the direct application of the peer to peer networking paradigm in a WLAN leads to a substantially decreased throughput for the traffic directed to the Access Point (AP). The study also reveals that the cumulative receiving throughput of nodes located at the periphery of relatively small circular areas around the AP is substantially higher. Thus, the capacity of the multihop cellular network may be enhanced by employing the peer to peer paradigm only outside a circular area around the AP and the cellular paradigm inside this circular area. Examples are provided of environments where the aforementioned idea of distributing the traditional AP functionality to a set of nodes at the periphery of a circular area around the AP can be effectively applied. Comparative Performance Analysis of RSVP and RMD  András Császár, Attila Takács, Budapest University of Technology and Economics, Hungary Service evolution towards QoS capable applications requires efficient resource reservation protocols. Currently, RSVP is a widely known protocol for this functionality in IntServ networks. Unfortunately, the processing power needs of RSVP make it to a less favoured candidate in high-speed environments. In these scenarios low-complexity DiffServ solutions have a clear advantage. A currently studied and industrially supported DiffServ conform resource management protocol is RMD. In this paper, we certify RMD as a simple and efficient protocol for unicast traffic by comparing the performance of RMD with RSVP, and also verify that the reduction of complexity does not entail loss in performance. Global Time for Interactive Applications over Global Packet Networks  Mario Baldi, Politecnico di Torino, Italy and Yoram Ofek, Synchrodyne Networks, Inc., USA This work presents that global time (a.k.a. time-of-day or coordinated universal time – UTC) is essential in maximizing user perceived quality of service, while eliminating both switching bottlenecks — critical in very high capacity network core — and communications link bottlenecks — at the low speed access (e.g., wireless and DSL). Global time obtained, for example, from GPS (Global Positioning System) or Galileo, is used in the design of all streaming media applications such as toll quality telephony, videotelephony and videoconferencing. The proposed solution, that can be applied to the Internet without changes to any of the existing protocols, provides a guaranteed quality service to each application without requiring nodes to keep state information on microflows. The Performance of Endpoint Admission Control Based on Packet Loss  Ignacio Más, Viktoria Fodor, Gunnar Karlsson, KTH, Sweden Endpoint admission control solutions, based on probing a transmission path, have been proposed to meet quality requirements of audio-visual applications with little support from routers. In this paper we present a mathematical analysis of a probe-based admission control solution, where flows are accepted or rejected based on the packet-loss statistics in the probe stream. The analysis relates system performance to design parameters and the experienced probe packet loss probability to the packet loss probability of accepted flows. Influence of Power Control and Link-Level Retransmissions on Wireless TCP  Niels Möller, Karl Henrik Johansson, KTH, Sweden A fundamental assumption of the TCP protocol is that packet losses indicate congestion on the network. This is a problem when using TCP over wireless links, because a noisy radio transmission may erroneously indicate congestion and thereby reduce the TCP sending rate. Two partial solutions, that improve the quality of the radio link, are power control and link-level retransmissions. By modeling these two lower layers of control loops, we derive an analytical model of the delay distribution for IP packets traversing a link. We investigate the effect on TCP, in particular the performance degradation due to spurious timeouts and spurious fast retransmits caused by delays and reorder on the link. It is shown that the models allow us to quantify the throughput degradation. The results indicate that link-level control and TCP interact, and that tuning one or the other is needed in order to improve performance. A Users´ Satisfaction Driven Scheduling Strategy for Wireless Multimedia QoS  Leonardo Badia, Michele Boaretto, Michele Zorzi, University of Ferrara, Italy In this work, we exploit game-theoretical concepts to depict the behaviour of multimedia users for the Radio Resource Management. Moreover, we also include pragmatic economic considerations, which allow studies of provider’s revenue and possible charging mechanisms. These concepts are in particular applied to HSDPA scheduling procedures, whose main aim is to improve the performance of 3G networks and to allow extensions to a plethora of services. We briefly discuss a model for the users’ satisfaction that includes both perceived QoS and pricing, already proposed to determine the QoS provisioning and network dimensioning. We apply the users’ satisfaction function of this model in the scheduler. In this way we achieve improvements of the QoS as it is seen from the users’ point-of-view, i.e., by involving the satisfaction of service constraints but also paid price. This analysis will be extended also to the provider’s side, with considerations on the achievable revenue, that is an important aspect to take into account in service supplying. Effects on TCP from Radio-block Scheduling in WCDMA High Speed Downlink Shared Channels  Ulf Bodin, Arne Simonsson, Luleå University of Technology, Sweden Avoiding delay jitter is essential to achieve high throughput for TCP. In particular, delay spikes can cause spurious timeouts. Such timeouts force TCP into slow-start, which may reduce congestion window sizes drastically. Consequently, there may not always be data available for transmission on bottleneck links. For HS-DSCH, jitter can occur due to varying interference. Also, properties of the radio-block scheduling influence the jitter. We evaluate, through simulations, effects on TCP from scheduling. Our evaluation shows that round-robin (RR) schedulers can give more jitter than SIR schedulers. SIR schedulers discriminates low SIR users to improve spectrum utilization while RR schedulers distribute transmission capacity fairly. The high jitter with RR scheduling cause however both lower utilization and decreased fairness in throughput among users than with SIR scheduling. The Eifel algorithm makes TCP more robust against delay spikes and reduces thereby these problems. TFRC Contribution to Internet QoS Improvement  Nicolas Larrieu, Philippe Owezarski, LAAS-CNRS, France The Internet is on the way of becoming the universal communication network, and then needs to provide various services and QoS for all kinds of applications. We show in this paper that oscillations that are characteristic of the Internet traffic provokes huge decrease of the QoS that flows can get. After having demonstrated that such oscillations can be characterized by the Hurst (LRD) parameter, we propose an approach for improving Internet flows QoS based on smoothing sending rate of applications. TFRC is a congestion control mechanism that has been issued for this purpose. This paper then proposes an evaluation of TFRC benefits on traffic profile and flows QoS. Adaptive Bandwidth Provisioning with Explicit Respect to QoS Requirements  Hung Tuan Tran, Thomas Ziegler, Telecommunications Research Center Vienna, Austria We propose adaptive bandwidth provisioning schemes enabling quality of service (QoS) guarantees. To this end, we exploit periodic measurements and traffic predictions to capture closely traffic dynamics. We make use of the Gaussian traffic model providing available bounds for QoS to derive the associated bandwidth demands. Moreover, special attention is paid for alleviating some typical problems with adaptive provisioning like QoS degradations and signaling overhead. Numerical and simulative investigations using real traffic traces show that the proposed schemes outperform some previous ones. Wide Area Measurements of Voice Over IP Quality  Ian Marsh, Fengyi Li, Gunnar Karlsson, KTH, Sweden Time, day, location and instantaneous network conditions largely dictate the quality of Voice over IP calls. In this paper we present the results of over 18000 VoIP measurements, taken from nine sites connected in a full-mesh configuration. We measure the quality of the routes on a hourly basis by transmitting a pre-recorded call between a pair of sites. We repeat the procedure for all nine sites during the one hour interval. Based on the obtained jitter, delay and loss values as defined in RFC 1889 (RTP) we conclude that the VoIP quality is acceptable for all but one of the nine sites we tested. We also conclude that VoIP quality has improved marginally since we last conducted a similar study in 1998. Rate control and QoS-related capacity in wireless communications  Eitan Altman, INRIA, France Traditional definitions of capacity of CDMA networks are either related to the number of calls they can handle (pole capacity) or to the arrival rate that guarantees that the rejection rate (or outage) is below a given fraction (Erlang capacity). These definitions do not seem to be well adapted to 3G multi-media networks since they do not take into account the fact that transmission rates can be adapted to the system load. By slowing down the transmission rate, we can accept more calls at the cost of a lower the quality of real time traffic or of longer transmission delays for data (or best-effort) transfers. We thus propose new definitions for capacity that are adapted to other metrics of quality of service (QoS) and grade of service (GoS). In particular, we propose the delay aware capacity defined as the arrival rate of data calls that the system can handle such that their expected delay is bounded by a given constant. We compute both the blocking probability of the real time traffic having an adaptive GoS as well as the expected delay of the best effort traffic for an uplink multicell WCDMA system. This yields the Erlang capacity for former and the delay capacity for the latter. We further define and compute the best effort (throughput) capacity which is the rate of arrivals of best effort calls that the system can handle when the transmission rate is controlled. Scatternet formation in multi-hop Bluetooth Networks  Chiara Petrioli, Rome University 'La Sapienza', Italy The Bluetooth (BT) technology, as described in the Specifications of the Bluetooth System Version 1.1 , is one of the most promising enabling technology for pervasive computing and ad hoc networks. In this talk I will review basic results to the fundamental problems of device discovery and scatternet formation, i.e. on how Bluetooth nodes can become aware of their neighbors, can partition themselves into groups, called piconets, and finally on how such groups can be interconnected in a connected multi-hop ad hoc network called scatternet. The talk will show how a simple cluster-based approach mapped into the BlueTooth technology, named BlueStars, results in a very effective way to forms connected scatternets. Relying on the sole knowledge of a node's one-hop neighbors, BlueStars selects the piconets' masters based on how ``fit'' a node is to serve as a master. Once piconets are formed, gateway nodes are selected so that there is an inter-piconet route between all masters that are at most tree hops away (i.e., all adjacent piconets are interconnected). This condition ensures the connectivity of the BlueStars scatternet. However, BlueStars may produce scatternets whose piconets have more than 7 slaves. Solutions recently introduced, BlueMesh and LSBS, produce connected scatternets with no more than 7 slaves per piconet by exploiting geometric properties of Unit Disc Graphs, and represent good candidates when such property must be accounted for, at the price of extra complexity, higher protocol duration, higher overhead, and possibly of extra HW (LSBS) . The results of an ns2 comparative performance evaluation comparing four among the major solutions so far proposed for multi-hop scatternet formation will be presented to assess where the bottlenecks are for scatternet formation and to quantifies pros and cons of the presented approaches. Allocating and assigning OFDM sub carriers to MPEG video transmissions - Does crossing layers really help?  Holger Karl, TU Berlin, Germany A wireless OFDM system opens new flexibility in supporting several terminals in a wireless cell: Instead of simply assigning all transmission resources, i.e., the OFDM subcarriers, to one terminal at a time in a time division multiplexing operation, subcarriers can be mapped to individual terminals in a more fine-grained fashion. Different numbers of subcarriers can be allocated to each terminal, and these subcarriers can be individually assigned. While the time-varying channel characteristics evidently impact this allocation and assignment problem, it is also possible to support application-specific needs, especially for applications with real-time needs. It is, e.g., conceivable to allocate more subcarriers to a terminal which has an important video frame waiting in the access point. Consequently, an optimization problem results that has to balance the properties and dynamics of the physical layer with those of an application. As a case study, we investigate how to support MPEG4 video transmission by OFDM allocation and assignment algorithms. The application properties can be characterized very simply (e.g., by only considering the queue length of each terminal) or in a semantic-aware fashion (e.g., by taking into account the type of MPEG frames). The OFDM algorithms can be either optimal (which leads to NP-complete problems) or quick heuristics. The talk will discuss these options and present results about the achieved video quality and the number of terminals that can be supported at a given quality level. Enhancing Cellular Capacity through QoS Awareness  Anders Furuskär, Ericsson, Sweden Awareness of Quality of Service (QoS) requirements is crucial for achieving high capacity in cellular networks. This presentation discusses three QoS-based capacity enhnacing principles: radio bearer realization, per-service-capacity balancing, and multi-access service allocation. Simple capacity gain estimates are presented for GSM and WCDMA systems. Providing Transport-Layer Fairness in 802.11 WLANs  Carla-Fabiana Chiasserini, Politecnico di Torino, Italy Wireless Local Area Networks (WLANs) based on the IEEE 802.11 technology are becoming increasingly popular and widely deployed. However, the growing need for Quality of Service (QoS) guarantees is difficult to implement in distributed systems like WLANs, where the random access protocol and the unpredictability of the wireless channel make it difficult their interaction with well-established architectures like DiffServ. Even without trying to provide deterministic QoS guarantees, simpler requirements are hard to get by. For example, the basic requirement of providing fair access to all users is conflicting with the nature of higher-layer protocols: TCP is fair only under certain conditions, hardly met by 802.11b WLANs. Another basic requirement is the protection for short-lived TCP flows, that are sensitive to losses during the early stages of the TCP window growth. The main contribution of this work is the proposal of an LLC-layer algorithm that can be implemented on both AP and WSs. The algorithm aims at guaranteeing fair access to the medium to every user, by awarding longer transmission opportunities to WSs that experienced short channel failures. We outline the proposed solution and present a simulation study that shows the effectiveness of the new algorithm in comparison to the standard 802.11b implementation. Unattended Ground Sensor Ad Hoc Networks -- A Real-World Test Bed  Leif Axelsson,Ericsson Microwave, Sweden Cheap Unattended Ground Sensors (UGS's) using ad hoc network communication will give great benefits in surveillance and control of areas for military and civilian purposes. To demonstrate the concept and the benefits of such a system we have successfully developed an operative ground sensor ad hoc network test bed and development platform. The assumed scenario is surveillance of roads to a protected area. In order to avoid the need of a planned distribution of UGS nodes, the sensors need to determine their own position and establish communication with their neighbours. Power efficiency is a demand to increase the lifetime, which also requires an efficient communication system. Another desire is low probability of detection. An elegant solution to this is an ad hoc network. When a user is within communication range of any other node in the network the user will receive alarms from the sensors. Upstream core-network connection can in remote areas be obtained via satellite communication. The different utilized sensors in the test bed are surveillance cameras and vehicle sensors that classify passing vehicles. In order to locate the sensors, they are equipped with GPS receivers. Among the sensors pure communication nodes are distributed to increase the coverage region of the network. The sensors are represented in the network as services, enabled by the Network Based Defence (NBD) kernel and the used ad hoc routing protocol implementation is a reactive protocol, Dynamic Source Routing (DSR). Application of congestion pricing to mobile networks  Vasilios Siris, Univ. Crete and FORTH, Greece Flexible and efficient resource control in mobile wireless networks is an increasingly important issue, due to the limited capacity of mobile networks, and their increasing use for delay and loss sensitive applications. A unique property of mobile networks, and CDMA networks in particular, is that resource usage is determined by the transmission rate and the transmission power, which can be different for different mobile users. In Wideband CDMA (WCDMA), resource control procedures include rate control, outer-loop power control, which is responsible for adjusting the target bit-energy-to-noise-density ratio (signal quality), and fast closed-loop power control, which is responsible for adjusting the transmission power to achieve the target signal quality. Economic modelling, and congestion pricing in particular, has been identified as a flexible framework for efficient resource control in wired networks, and has recently has been applied to wireless networks. We consider the application of this framework to mobile WCDMA networks, considering the joint optimization of the transmission rate and the target signal quality, and we identify how the two control procedures for adjusting these quantities can obtain a simple and attractive form that takes into account, through shadow prices, the level of demand and supply in order to achieve efficient resource utilization. The approach also takes into account different user requirements and characteristics, in the form of utility functions, in terms of both average throughput and data loss rate.