Wired/Wireless Internet Communications

Simulation tools are frequently used for performance evaluations of mobile ad-hoc networks. Currently the tools poorly support urban scenarios, since they do not take a spatial environment into account. In this paper, we describe a platform for the modeling of city scenarios. We extend ns-2 with corresponding mobility and wireless transmission models. By using its emulation facility, we integrate unmodified applications and real implementations of network protocols. We demonstrate the usefulness of the platform for performance evaluations by modeling a mobile application in a simulated environment of Stuttgart downtown. We show that it helps identifying application problems before deployment.

A novel approach to performing context discovery in ad-hoc networks based on the use of attenuated Bloom filters is proposed in this paper. In order to investigate the performance of this approach, a model has been developed. This document describes the model and its validation. The model has been implemented in Matlab, and some results are also shown in this document. Attenuated Bloom filters appear to be a very promising approach for context discovery in ad hoc networks.

This paper presents the Cluster-label-based mechanism for Backbones (CLaB) used in mobile ad hoc networks. The proposed mechanism provides a distributed topology control and consists of three parts: the part creating a backbone, the routing part, and the maintenance part on the backbone. The first part creates a clustered topology using a unique ID called a cluster label and establishes connections between neighboring clusters to create a backbone. The second part adapts existing routing protocols on the backbone. The third part maintains links on the backbone to minimize the influence of node movements, and needs no rerouting mechanism. The mechanism especially concentrates on maintenance by introducing constantly connected backbone elements based on cluster labels. A comparison with other backbone-based mechanisms is presented featuring different node densities and mobility levels. The results validate the effectiveness of the proposed mechanism.

The interaction between TCP flows and the IEEE 802.11 MAC protocol is analyzed in this paper. The goal is to provide comprehensive models capable of predicting the TCP performance (throughput) in WLAN Hot-Spot networks with persistent elastic flows. A model of the IEEE 802.11 MAC protocol is first introduced and validated through simulations. Using the MAC model as a basis, we then analyze several modeling strategies to catch the main interactions between TCP and MAC protocols. Results obtained from the models are compared among themselves and against simulations, and show the accuracy and the superior simplicity of the proposed models with respect to previously published work.

The deployment of high performance networks and the emergence of broadband wireless access technologies challenge the performance of standard TCP. Recent years have seen a few high performance and aggressive TCP variants being proposed. These TCP variants still suffer significant performance degradation from random/transient packet losses. In this paper, we introduce and evaluate a new method to improve high performance TCP in the presence of random packet losses. It is ‘blind’ in that our method does not attempt to differentiate between random and congestion-induced losses. Our method combines both TCP parallelisation and modification to the aggressiveness of studied AIMD algorithms. We show that our proposed method is no more aggressive than the studied TCP variants in the presence of congestion but more effectively utilises bandwidth in the presence of random packet losses.

Real-time streaming over satellite IP networks is challenging, since satellite links commonly exhibit long propagation delays and increased error rates, which impair TCP performance. In this context, we quantify the effects of satellite links on TCP efficiency and streaming video delivery. We investigate a solution-framework composed by TCP implementations which are expected to perform adequately in such environments. Furthermore, we study the supportive role of Selective Acknowledgments (SACK). Along with protocol performance, we also evaluate the impact of delayed acknowledgments. Our simulation results illustrate that most existing end-to-end solutions do not comply with the stringent QoS provisions of time-sensitive applications, resulting in inefficient bandwidth utilization and increased delays in data delivery. Finally, with the absence of a satellite-optimized TCP implementation for real-time streaming, we identify TCP Real as the most prominent solution, since it manages to alleviate most of the impairments induced by satellite links, sustaining a relatively smooth transmission rate.

With the widespread use of wireless Internet and wireless LANs, different wireless technologies, such as 3G cellular networks and WLAN, will cooperate to support more users and applications with higher data rates over wider areas. When a mobile node moves between hybrid networks, a need arises for seamless vertical handoffs between different wireless networks in order to provide high performance data transmission. When an application with a TCP connection in a mobile node performs a vertical handoff, TCP performance is degraded due to packet loss even while the previous TCP state information is maintained during handoff, as 3G and WLAN have different available bandwidths. In this paper, a new congestion control algorithm is proposed for a vertical handoff that improves TCP performance by measuring the rough end-to-end available bandwidth and calculating the slow-start threshold. By ns-2 simulation it is shown that, compared to previous algorithms, the proposed algorithm enhances TCP performance during vertical handoffs.

Classical micro-mobility solutions focus on accelerating local handovers and on minimizing signalling traffic. However, the micro-mobility components also slow down the service times in the user traffic domain. This is a critical drawback in terms of the hard delay requirements of future IP-based radio access networks. Network operators need algorithms to determine the optimal number and placement of anchor points within his network to balance both handover latency and user traffic delay.

This paper presents a very fast and exact mathematical method for selecting the optimal number and placement of nodes supporting micro-mobility functionality for telecommunication networks. The method is in particular applicable for large networks, which is demonstrated by means of an optimization of a real wireless access network with more than 150 nodes. In order to ensure the applicability in real network scenarios we additionally provide a solution to estimate the stability of the optimal configuration. The presented method allows the autonomous and optimal configuration of the network’s micro-mobility functionality as described in concepts of zero-touch networks. Due to the optimization network operators can reduce the administration cost and maximize the quality of service as well.

In this paper, we propose an enhanced handover mechanism with new additional primitives and parameters to the media independent handover (MIH) services as defined in IEEE 802.21. The proposed scheme can reduce handover latency for MIPv6 (mobile IPv6) by removing the router discovery time. Moreover, when the proposed mechanism is applied to a FMIPv6 (fast handovers for MIPv6), we can increase the probability that FMIPv6 can be per-formed in a predictive mode by reducing the handover initiation time. As a result, we can get the reduced handover latency in FMIPv6. In addition, with the proposed scheme, we can design the cost-effective network by reducing cover-age overlap between adjacent cells. This is due to the decreased handover initiation time in FMIPv6. From simulation results, it is proven that the proposed scheme can provide a higher goodput for UDP and TCP because it has less service disruption during handovers.

The definition of the IEEE 802.1Q and IEEE 802.1p standards provided Class of Service (CoS) capabilities to Ethernet networks and, consequently, allowed new QoS services to be deployed. This is used by the Subnet Bandwidth Management (SBM) protocol, an RSVP-based protocol that provides IntServ-like services at Ethernet level. This paper proposes an alternative way to quality of service provision and resource allocation in Ethernet networks, based on the emerging IETF NSIS framework. The proposed approach was validated as a proof of concept by simulation, showing the ability of NSIS to provide QoS differentiation in Ethernet scenarios.

The provisioning of IP services in the mobile telecommunication and the IP world differs in many aspects, especially with prepaid charging de termining an important useability distinction. Thus, the state of the art of pre paid charging systems have been analyzed in detail, elaborating on their key ca pabilities and major limitations. Additionally, economic aspects in terms of op erator’s risks and major costs, addressing the technical overhead to implement prepaid charging, have been evaluated. Therefore, to enable prepaid charging for QoS-enabled and all-IP environments, key concerns for a future IP service provisioning have been determined, which remain infeasible with existing prepaid solutions, and the evaluation lead to a new scheme proposed.

In this paper we discuss IP layer packet filtering and an application level gateway approach used to secure handheld devices when providing and using web services. We propose a firewall management plane as a means for cross layer interaction. In our approach the application level gateway updates the IP layer firewall rules based on its knowledge about whether or not a certain source is sending malicious packets. We show that such a cross layer interaction can significantly decrease the CPU load in case of attacks, i.e., if many malicious packets arrive at the handheld device. Our measurement results show that the additional overhead for IP layer filtering is less than 10 per cent, if the number of applied rule sets is less than 200. In addition our cross layer approach can reduce the CPU load caused by the application layer gateway by about 10 up to 30 per cent.

The convergence of mobile domain and data networks has been under focus in the standardization forums. However, dynamic interworking of wired infrastructure, wireless access systems and small scale Personal Area Networks has been challenging due to their heterogeneous nature. One of the most important problems to be solved is name resolution between different terminals and networks. This paper presents a new mechanism for name resolution, which relies on existing naming mechanisms. In particular the focus is on the scalability of the solution from signaling load and latency point of view.

Results for Enhanced UMTS (E-UMTS) cost/revenue optimisation are obtained, as a function of the coverage distance, R. E-UMTS traffic generation and activity models are described and characterised in an urban scenario based on population and service penetration values. By using a System Level Simulator results were obtained for blocking and handover failure probabilities. Models for the supported fraction of active users and for the supported throughput, as a function of active users, were obtained. When one amplifier is used, the maximum throughput per BS is around 600kb/s. However, it achieves values up to 2000kb/s when three amplifiers per BS are considered. Generally, the profit in percentage is a decreasing function with R. The use of three amplifiers per BS is strongly advised in order to get cheaper communications, with prices that vary from 0.016 to 0.07 €/min, for R=250 and 1075m, respectively.

This paper presents an analytical approach for characterizing the aggregated traffic carried in the UMTS Terrestrial Radio Access Network (UTRAN). The characteristic of the incoming traffic stream of UTRAN is studied based on the measured trace traffic from the simulations. The main idea of the aggregated traffic modelling is to employ Batch Markov Arrival Process (BMAP) model as an analytically tractable model, which considers different lengths of packets and batch arrivals. In this paper, the setup and customization of the BMAP model for characterizing the aggregated traffic in UTRAN is presented. The accuracy of the BMAP model is demonstrated by comparing with simulations and Poisson traffic model. At the end the potential application of the presented approach and its advantages is briefly discussed.

Orthogonal Frequency Division Multiplexing (OFDM) is used in both 802.11 and 802.16 [1] wireless network standards and is being evaluated for Fourth Generation (4G) networks. Its tolerance to frequency selective fading has made it an attractive choice for these broadband wireless networks. In both downlink and uplink directions, resources can be allocated in three dimensions, frequency, time and power. In both cases the resource allocation is performed primarily by the BS. In this paper we focus on the allocation of downlink resources when such a network is used for Quality of Service (QoS) applications. We formulate and solve the corresponding optimization problem and use the solution to design simple, practical algorithms. In doing so we take into account practical details that have not been addressed in previous work.

We propose a novel resource allocation scheme, which can reduce MAP overhead and maximize the throughput in the MIMO-OFDM systems. Message based broadband access systems are needed to minimize the MAP overhead since the excessive MAP overhead causes degradation of system throughput. Increasing the size of resource allocation unit can reduce MAP overhead. However, multiuser diversity gain becomes smaller as the size of resource allocation unit increases. Therefore, we investigate joint optimization between multiuser diversity gain and MAP overhead size. The proposed scheme can reduce the MAP overhead size as well as achieve high throughput.

Voice over WLAN (VoWLAN) is widely acknowledged as one of the key, emerging applications for wireless LANs. As with any multi-service network, there is the need to provision the WLANs with the QoS mechanisms capable of guaranteeing the requirements of various services. The upcoming IEEE 802.11e (EDCA) standard is on the way to define the required QoS mechanisms into the protocol architecture of IEEE 802.11 WLANs. However, recent studies have shown that EDCA performs poorly when the medium is highly loaded. This situation has led many researchers to design techniques aiming to improve the performance of EDCA. Unfortunately, the main deficiency of many of the proposed mechanisms to date comes from their inability to properly interoperate with legacy IEEE 802.11 (DCF) systems. Moreover, the implementation of many of the proposed QoS mechanisms requires important and incompatible changes to the IEEE 802.11e specifications. In this paper we introduce a novel IEEE 802.11e-based QoS protocol capable of providing QoS support to voice communications, and able to properly interoperate with legacy DCF based stations. Our design has been based on an in-depth analysis of the several operation modes of both standards. This should ensure full compatibility of operation: an important feature since the transition from the IEEE 802.11 to the IEEE 802.11e will take some time making more likely the existence of hybrid scenarios where both standards will have to coexist. Our results show that the proposed mechanism outperforms the IEEE 802.11e standard by offering better QoS guarantees to the voice service over a wider range of load conditions and traffic mix scenario.

With the growth of broadband networks, Video-on-Demand (VoD) has become realistic. Most of the significant broadcasting schemes have been proposed to reduce the server bandwidth requirement for Constant Bit-Rate (CBR) encoded videos. The few existing proposals, which support of the bandwidth-efficient Variable Bit-Rate (VBR) encoded videos, smooth and minimize the server bandwidth requirement at the cost of data losses. The lossless and bandwidth efficient (LLBE) scheme states that minimization of the server bandwidth as a shortest path problem on a directed acyclic graph, and uses a dynamic programming method to solve the problem. However, it cannot be used for support live video broadcasting. In this paper, we propose a live harmonic broadcasting (LHB) scheme to support VBR-encoded live video broadcasting. The ideas behind the scheme are (1) support VBR-encoded videos by asynchronous downloading and playing, and hybrid division by length and size, and (2) support live videos by postponing recasting process. It is simple and effective. Using data from real videos, we conduct a simulation to evaluate our scheme. The experiment results indicate that the LHB scheme requires slightly higher server bandwidth than LLBE scheme.

Efficient H.264 video transmission over wireless/mobile networks, which is used as a dominant video coding standard in Digital Multimedia Broadcasting (DMB) and in other advanced video conferencing, becomes dominant. However, wireless video transmission suffers from deficient wireless channel conditions such as high bit error rate, error bursts due to channel fading and bandwidth limitations. In this paper, a novel design and analysis of unified error-control with cross-layer interaction over IEEE 802.11e Wireless LAN, in order to improve error protection performance, is proposed. This framework combines cross-layer error protection techniques, i.e., error correction code in the link/MAC layer, erasure code in the application layer and automatic retransmission across the link/MAC layer and the application layer. The performance of the proposed scheme in comparison to the generic scheme is demonstrated by simulation.

Increase of WLAN network deployments lead to the need of developing tools to predict coverage in terms of available service. In this paper we propose to establish service coverage based only on approximate floor plans by using the so-called PixelFlow algorithm. This algorithm is based on discrete version of the Huygens principle and appears to be rather robust to approximation in the floor plan. Measurements of service performances have been undertaken and used to calibrate the prediction results. Since the conventional calibration based on prediction errors is not the goal of service coverage prediction, a new metric has been developed to quantify differences between predicted and measured boundaries of the service coverage. Despite the complex impact of inaccuracies related to floor plan, wall material, radio propagation and WLAN protocols, it appears that the service coverage prediction proposed here is suitable to ease the radio network design of indoor Wi-Fi system.

We consider the state of the wireless channel in terms of the covariance stationary signal-to-noise ratio (SNR) process and parameterize it using the probability distribution function of SNR and lag-1 autocorrelation coefficient of associated autocorrelation function (ACF). In order to discriminate the state of the wireless channel we apply methods of statistical process control. Particularly, we use exponential weighted moving average (EWMA) change-point statistical test to detect shifts in the mean of the SNR process. The proposed approach is verified using SNR measurements of IEEE 802.11b wireless channel.

In this paper we propose the use of statistical QoS guarantees for transmission over the wireless channel. Here, instead of QoS assurances we propose to guarantee the percentage of time the QoS requirements are satisfied. We present an associated scheduling algorithm for the opportunistic multiple access system. We compare the proposed scheduler with popular schedulers from the literature. We observe that the statistical QoS guarantee is an attractive alternative to the assured QoS for the wireless platform since such strict QoS assurances decrease the wireless system performance significantly.

Location determination of mobile users within a building has attracted much attention lately due to its many applications in mobile networking including network intrusion detection problems. However, it is challenging due to the complexities of the indoor radio propagation characteristics exacerbated by the mobility of the user. A common practice is to mechanically generate a table showing the radio signal strength at different known locations in the building. A mobile user’s location at an arbitrary point in the building is determined by measuring the signal strength at the location in question and determining the location by referring to the above table using a LMSE (least mean square error) criterion. Obviously, this is a very tedious and time consuming task. This paper proposes a novel and automated location determination method called ARIADNE. Using a two dimensional construction floor plan and only a single actual signal strength measurement, ARIADNE generates an estimated signal strength map comparable to those generated manually by actual measurements. Given the signal measurements for a mobile, a proposed clustering algorithm searches that signal strength map to determine the current mobile’s location. The results from ARIADNE are comparable and may even be superior to those from existing localization schemes.

Power saving mechanisms in wireless ad hoc network nodes mainly switch off the transmission and reception hardware for a maximal amount of time and turn it on again within a given interval. Many approaches aim to synchronize the state changes of the nodes in the network through distributed beacon generation and introduce mechanisms where nodes synchronously wake up at designated points of time to exchange announcements about pending traffic. Synchronization however is difficult to achieve, in particular in ad hoc networks.

This paper describes the simulation, evaluation and refinement of a recently proposed power saving approach based on asynchronous wake-up patterns and wake-up announcements integrated with AODV. We show that significant improvements of the connectivity under low wake ratios can be achieved by carefully designed forwarding strategies of AODV route request messages.

This paper proposes a new power-aware routing protocol, TDPR(Traffic load and lifetime Deviation based Power-aware Routing protocol), that does not only consider the residual battery capacity and transmission power, but also the traffic load of nodes and deviation among node lifetimes. It helps to extend the entire network’s lifetime and to achieve load balancing. Simulation using ns-2 shows better the performance of the proposed routing protocol in terms of load balancing of the entire network, the consumed energy capacity of nodes, and paths reliability. TDPR has dead nodes 72% less than AODV, and 58% less than PSR. TDPR consumes residual energy capacity 29% less than AODV, 15% less than PSR. Error messages are sent maximum 38% less than PSR, and maximum 41% less than AODV.

The performance of an energy-efficient object tracking system depends on its accuracy in predicting the next destination of a mobile event. Unfortunately, a sophisticated prediction method cannot be operated in sensor nodes which have low computational power and storage. Moreover, precise prediction alone cannot be guaranteed to eliminate error in the future destination of the object in real circumstances. In this paper, we present a location error handling technique to prevent and handle this error efficiently. Real situations such as an unexpected change in the mobile event’s direction, failure of event-detection and failure of transmitting an error message are considered when designing the error handling technique. This simple yet effective solution complements the weakness of the energy-efficient object tracking paradigm. From experiments on both real hardware and simulation, our method outperformed the existing work.