LDPC-based space-time coded OFDM systems: Turbo-EM receiver design with channel state information guessing algorithms

In this paper we study some turbo receiver architectures employing low-density parity check (Ldpc) codes together with orthogonal frequency division multiplexing (Ofdm) for high data rate wireless transmissions. Different demodulation schemes based on expectation-maximization (Em) algorithm are studied along with the channel impulse response (Em) algorithms. We studied differentCir guessing algorithms including the EM-based algorithms such as a space-alternating generalized expectation-maximization algorithm (Sage). It is shown that the proposed turbo-Em receiver employing a soft maximum a posteriori (Map)Em demodulator and a belief propagationLdpc decoder can perform within 1 dB from the ergodic capacity of the studiedMimo ofdm channels. Besides, we find that a suboptimum structure based on a soft interference cancellationMmse filtering demodulator exhibits negligible loss in non-correlated fadingMimo channels but suffer extra performance loss in highly correlatedMimo channels.     

Distance upper bounds and true minimum distance results for turbo-codes designed with DRP interleavers

This paper presents a number of distance upper bounds for Turbo-codes designed with dithered relative prime (drp) interleavers. These bounds help determine how much dither should be applied and which input data patterns to test when designing high-distance interleavers with different lengths. Using a block length of only 640 data bits, true minimum distances of 30, 42 and 53 have been achieved for rate 1/3 Turbo-codes with 4, 8 and 16-state constituent codes, respectively. Simulated error rate results for a block length of 1504 data bits (188 bytes,mpeg packet) show excellent flare performance.     

Optimizing IP networks in a hybrid IGP/MPLS environment

In this paper, we consider a traffic engineering (te) approach toip networks in a hybridigp/mpls environment. Thoughigp (Interior Gateway Protocol) routing has proven its scalability and reliability, effective traffic engineering has been difficult to achieve in public IP networks because of the limited functional capabilities of conventionalip technologies.mpls (Multi-Protocol Label Switching) on the one hand enhances the possibility to engineer traffic onip networks by allowing explicit routes. But on the other hand it suffers from the scalability (n-square) problem. Hybridigp/mpls approaches rely onip native routing as much as possible and usempls only if necessary. In this work we propose a novel hybrid traffic engineering method based on genetic algorithms, which can be considered as an offlinete approach to handle long or medium-term traffic variations in the range days, weeks or months. In our approach the maximum number of hops anlsp (Label Switched Path) may take and the number oflsps which are applied solely to improve the routing performance, are treated as constraints due to delay considerations and the complexity of management. We apply our method to the German scientific network (b-win) for which a traffic matrix is available and also to some other networks with a simple demand model. We will show results comparing this hybridigp/mpls routing scenario with the result of pureigp routing and that of a full meshmpls with and without traffic splitting.     

Multimedia communication in wireless sensor networks

The technological advances in Micro ElectroMechanical Systems (Mems) and wireless communications have enabled the realization of wireless sensor networks (Wsn) comprised of large number of low-cost, low-power, multifunctional sensor nodes. These tiny sensor nodes communicate in short distances and collaboratively work toward fulfilling the application specific objectives ofWsn. However, realization of wide range of envisionedWsn applications necessitates effective communication protocols which can address the unique challenges posed by theWsn paradigm. Since many of these envisioned applications may also involve in collecting information in the form of multimedia such as audio, image, and video; additional challenges due to the unique requirements of multimedia delivery overWsn, e.g., diverse reliability requirements, time constraints, high bandwidth demands, must be addressed as well. Thus far, vast majority of the research efforts has been focused on addressing the problems of conventional data communication inWsn. Therefore, there exists an urgent need for research on the problems of multimedia communication inWsn. In this paper, a survey of the research challenges and the current status of the literature on the multimedia communication inWsn is presented. More specifically, the multimediaWsn applications, factors influencing multimedia delivery overWsn, currently proposed solutions in application, transport, and network layers, are pointed out along with their shortcomings and open research issues.     

Bounds on the error probability ofml decoding for block and turbo-block codes

The performance of either structured or random turbo-block codes and binary, systematic block codes operating over the additive white Gaussian noise (Awgn) channel, is assessed by upper bounds on the error probalities of maximum likelihood (Ml) decoding. These bounds on the block and bit error probability which depend respectively on the distance spectrum and the input-output weight enumeration function (Iowef) of these codes, are compared, for a variety of cases, to simulated performance of iterative decoding and also to some reported simulated lower bounds on the performance ofMl decoders. The comparisons facilitate to assess the efficiency of iterative decoding (as compared to the optimalMl decoding rule) on one hand and the tightness of the examined upper bounds on the other. We focus here on uniformly interleaved and parallel concatenated turbo-Hamming codes, and to that end theIowefs of Hamming and turbo-Hamming codes are calculated by an efficient algorithm. The usefulness of the bounds is demonstrated for uniformly interleaved turbo-Hamming codes at rates exceeding the cut-off rate, where the results are compared to the simulated performance of iteratively decoded turbo-Hamming codes with structured and statistical interleavers. We consider also the ensemble performance of ‘repeat and accumulate’ (Ka) codes, a family of serially concatenated turbo-block codes, introduced by Divsalar, Jin and McEliece. Although, the outer and inner codes possess a very simple structure: a repetitive and a differential encoder respectively, our upper bounds indicate impressive performance at rates considerably beyond the cut-off rate. This is also evidenced in literature by computer simulations of the performance of iteratively decodedRa codes with a particular structured interleaver.     

Web-based interorganizational information systems for logistics outsourcing

This article examines how web-based interorganizational information systems(ios)can efficiently support coordination mechanisms between outsourcer and third party logistics (3PL). First, we review the literature on coordination mechanisms andios. Second, we report on the methodology used for gathering information on outsourcers and 3PLs. Third, armed with limited but quality data, we identified two fundamental dimensions: 3PL involvement andios impacts on logistics outsourcing decisions. By combining the two dimensions, we propose a conceptual framework that highlights four main categories ofios that we characterize as neutralios, supply chainios, strategicios and dynamicios. After characterizing and analyzing each category, we discuss how these web basedios support outsourcers and 3PL along the supply chain. Finally, the article discusses the framework’s relevancy and its limits.     

Bounds on mutual information for simple codes using information combining

For coded transmission over a memoryless channel, two kinds of mutual information are considered: the mutual information between a code symbol and its noisy observation and the overall mutual information between encoder input and decoder output. The overall mutual information is interpreted as a combination of the mutual informations associated with the individual code symbols. Thus, exploiting code constraints in the decoding procedure is interpreted as combining mutual informations. For single parity check codes and repetition codes, we present bounds on the overall mutual information, which are based only on the mutual informations associated with the individual code symbols. Using these mutual information bounds, we compute bounds on extrinsic information transfer (exit) functions and bounds on information processing characteristics (ipc) for these codes.     

Improving parallel EKF-based nonlinear channel equalization using unscented transformation

The paper presents a new review of parallel Kalman filtering for nonlinear channel equalization. A Network of Extended Kalman Filters (nekf) has already been suggested for this purpose. This equalizer gives recursively a minimum mean squared error (mmse) estimation of a sequence of transmitted symbols according to a state formulation of a digital communication scheme. It is essentially based on two mechanisms: the approximation of the non Gaussiana posteriori probability density function (pdf) of the symbol sequence by a Weighted Gaussian Sum (wgs); and the local linearization of the nonlinear channel function for each branch of the network. Since the linearization, bearing on scattered symbol states, is one of the major limitations of thenekf, a new Kalman filtering approach, the Unscented Kalman Filter (ukf) suggested by Julier and Uhlman is considered in this paper for an interesting adaptation to the equalization context. Theukf algorithm is based on the equations of a Kalman filter, as the optimal linear minimum variance estimator, and on determining conditional expectations based on a kind of deterministic Monte-Carlo simulations. The new equalizer referred to as the Network ofukf (nukf), thus combines density approximation by awgs and the Unscented Transformation (ut) principle to circumvent the linearization brought within eachekf and is shown to perform better than thenekf based equalizer for severe nonlinear channels. Also, an adaptive version of thenukf is developed using the k-means clustering algorithm for noise-free channel output identification, since thenukf-based algorithm does not require the knowledge of the channel nonlinearity model.     

A scheduler for relative delay service differentiation

We propose a new delay-based scheduler called asRD-VC (Relative Delay VirtualClock). Since it performs a delay-based service differentiation among flow aggregates, the quality at microflow level is the same as that at aggregate level. This is not easily achievable when the service differentiation is bandwidth-based or loss-based. Unlike theEDF (Earliest Deadline First) scheduler [1], our proposed scheduler self-regulates and adapts the delays according to load changes. This characteristic permits us to implement it in an AF-likePHB providing the relative quantification service in a DiffServ network. Finally, we compare our proposedrd-vc scheduler with two important existing propositions:WTP (Waiting Time Priority) [2, 3] andex-vc (Extended VirtualClock) [4]. Both these propositions are delay-based and have self-regulation property. All three schedulers (RD-VC, WTP andEX-VC) maintain the required service differentiation among aggregates and have comparable long term average performance like mean throughput per aggregate and packet loss ratio etc. However,RD-VC and WTP take an edge overEX-VC at short-term performance like jitter. Bothrd-vc andWTP have good long term and short-term performance. Our proposedrd-vc, compared to existingWTP, has two additional characteristics, i.e. unlike WTP which is limited to architectures with one queue per Qos class, it has no limitation on implementation scope (with or without separate queues per class) and it has lower complexity. This rendersRD-VC an interesting proposition.     

Securing communications overAtm networks: the remoteAtm private networks interconnection example

When remoteAtm sites communicate through anAtm public network, a number of security problems arise, such as hacking, eavesdropping and traffic tampering. This paper proposes three contributions to these security problems. Firstly, risks due toAtm technology usage are detailed. Secondly, a survey of existing techniques aiming at securingAtm communications is presented with emphasis on theAtm Forum’s security specifications. Thirdly, a new solution called Safe (which stands for Solution for anAtm Frequent communications Environment) developed in the Démostène project is described. Safe realizes both firewall’s filtering functions and communications protection over theAtm network. The main idea of Safe is to use signaling (Uni 3.1) as a means to exchange security information over the network. This idea has been implemented and introduced to theAtm Forum.     

Rogue-base station detection in WiMax/802.16 wireless access networks

We address the problem of detecting a rogue base station (Bs) in WiMax/802.16 wireless access networks. A rogueBs is a malicious station that impersonates a legitimate access point (Ap). The rogueBs attack represents a major denial-of-service threat against wireless networks. Our approach is based on the observation that inconsistencies in the signal strength reports received by the mobile stations (Mss) can be seen if a rogueBs is present in a network. These reports can be assessed by the legitimate base stations, for instance, when a mobile station undertakes a handover towards anotherBs. Novel algorithms for detecting violations of received signal strength reports consistency are described in this paper. These algorithms can be used by an intrusion detection system localized on the legitimateBss or on a global network management system operating theBss.     

Characterization and evaluation of TCP and UDP-based transport on real networks

StandardTcp (RenoTcp) does not perform well on fast long distance networks, due to its AMD congestion control algorithm. In this paper we consider the effectiveness of various alternatives, in particular with respect to their applicability to a production environment. We then characterize and evaluate the achievable throughput, stability and intra-protocol fairness of differentTcp stacks (Scalable,Hstcp,Htcp, FastTcp, Reno,Bictcp, hstcp-lp andLtcp) and aUdp based application level transport protocol (Udtv2) on both production and testbed networks. The characterization is made with respect to both the transient traffic (entry and exit of different streams) and the steady state traffic on production Academic and Research networks, using paths withRtts differing by a factor of 10. We also report on measurements made with 10 Gbit/secNics with and withoutTcp Offload Engines, on 10 Gbit/s dedicated paths set up forSc2004.     

Network-controlled mobility within radio access networks based onWLAN technologies

This article presents a network-controlled approach of user terminal mobility within anIP based WirelessLAN Access Network. In a first part, this article makes a review of the mobility support, on the subject of emergingWLAN technologies asHIPERLAN/2 andIEEE 802.11, on the one hand, and, regardingIP networks as currently studied withinIETF, on the other hand. Both types ofIP mobility protocols are presented, either global mobility protocols such as MobileIP, or local mobility management protocols (micro mobility). In the next part, the overall principles of our mobility management approach are explained; this approach is based on the implementation of a new network entity dedicated to the control of user terminal mobility. The last part details a practical implementation of this approach. The implementation is carried out on the basis of Hierarchical MobileIPv6 (HMIPv6). The experimental results confirm the importance to carefully plan and control the user terminal mobility within largeIP based Access Networks, as this brings benefit to the user as well as to the operator.     

Performance evaluation of DCCP: A focus on smoothness and TCP-friendliness

Recent years have seen dramatic increases of the use of multimedia applications on the Internet, which typically either lack congestion control or use proprietary congestion control mechanisms. This can easily cause congestion collapse or compatibility problems. Datagram Congestion Control Protocol (Dccp) fills the gap betweenUdp andTcp, featuring congestion control rather than reliability for packet-switched rich content delivery with high degree of flexibility. We present aDccp model designed and implemented withOpnet Modeler, and the experiments and evaluation focused on largely the smoothness of the data rates, and the fairness between concurrentDccp flows andTcp flows. We foundDccp-ccid3 demonstrates stable data rates under different scenarios, and the fairness betweenDccp andTcp is only achieved under certain conditions. We also validated that the throughput ofDccp-Ccid3 is proportional to the average packet size, and relatively fixed packet size is critical for the optimal operation ofDccp. Problems in the slow start phase and insufficient receiver buffer size were identified and we hereby proposed solutions on this.     

Décodage Syndromique à Complexité Réduite Pour Codes Convolutifs

The decoding of convolutional codes in the maximum likelihood sense is carried out in a traditional way with the Viterbi algorithm (Va). We proposed a soft and hard input decoder where theVa, associated with an relevant metric, is applied to identify the error vector rather than the information message. In this paper, we show that, with this type of decoding, the exhaustive computation of a majority ofAcs (Add Compare Select) is unnecessary. Moreover, we show that optimal performance is achieved in the case of a hard input decoder, and that performance closed to the optimum is achieved in the case of a soft input decoder, while offering of a reduction of the complexity which is all the more important than the Ec/No ratio is high (e. g. for ratio Ec/No greater than 3 dB, more than 80 % of theAcs can be avoided). We also propose an algorithm allowing rejecting a frame without having to carry out any iteration of theVa.     

A MC-CDMA system analysis in a software radio context

This paper presents a Multi-Carrier Code Division Multiple Access (Mc-Cdma) system analysis in a software radio context. Based on a combination of multi-carrier modulation and code division multiple access,Mc-Cdma benefits from the main advantages from both schemes: high spectral efficiency, high flexibility, multiple access capabilities, etc. It is firstly shown why, nowadays,Mc-Cdma is undoubtedly a high potential candidate for the air interface of the 4G cellular networks. TheMc-Cdma concept and the block-diagrams of the transmitter and the receiver are presented first. Afterwards, the technical issues concerning the processing devices for the implementation ofMc-Cdma systems in a software radio context are analysed. The advantages and disadvantages of Digital Signal Processors (Dsps) and Field Programmable Gate Arrays (Fgpas) components are discussed. The implementation ofMc-Cdma systems and the integration of signal processing algorithms as Fast Hadamard Transform (Fht) and Inverse Fast Fourier Transform (Ifft) are considered and analysed for the first time. Finally, implementation results with a mixed prototyping board are presented. Then, it is shown that a new combination of the flow graphs ofFht andIfft leads to interesting computation savings and that hardware structures asFgpas are more adapted thanDsps to those intensive computation functions. Finally, for the completeMc-Cdma modem implementation, the necessity of a Co-Design methodology is highlighted in order to obtain the best matching between algorithms and architecture.     

Bandwidth sharing under the assured forwarding Per-Hop behavior

The DiffServ’s Assured Forwarding (af) Per-Hop Behavior (phb) Group defines a differentiated forwarding of packets in four independent classes, each class having three levels of drop precedence. Specific end-to-end services based on thisphb are still being defined. A particular type of service that could assure a given rate to a traffic aggregate has been outlined elsewhere. In such a service, a fair distribution of bandwidth is one of the main concerns. This paper presents experimental work carried out to evaluate howaf distributes bandwidth among flows under different load conditions and traffic patterns. We focused on the effect that marking mechanisms have on bandwidth sharing among flows within a singleaf class. The traffic types we used includeudp flows, individual and aggregatedtcp flows, mix oftcp andudp, tcp sessions with heterogeneous round-trip times, as well as color-blind and color-aware re-marking at the aggregation point fortcp flows. Tests were performed on real and simulated networks. We have found certain conditions under whichaf distributes bandwidth fairly among nonadaptiveudp flows andtcp aggregates. Finally, we evaluate a basic rule for setting the parameters of the two-rate Three-Color Marker conditioning algorithm (trtcm) in order to achieve a better bandwidth distribution fortcp flows.     

Iterative successive interference cancellation for multi-user DS/CDMA detectors in multipath channels

This paper deals with uplink Direct-Sequence Code Division Multiple Access (DS-CDMA) transmissions over mobile radio channels. A new interference cancellation scheme for multiuser detection, calledSIC/RAKE, is presented. It is based on a modified multistage Successive Interference Cancellation (sic) structure that enables efficient detection in multipath propagation environments, thanks to a single userRAKE receiver incorporated in each unit of thesic structure. Furthermore, a modified version of thesic structure, calledSIC/MMSE, that ensures convergence to theMMSE detector rather than to the decorrelating detector has been suggested. The convergence of theSIC/RAKE andSIC/MMSE methods is proved. Simulation results for the Universal Mobile Telecommunication System (UMTS) have been carried out for flat fading Rayleigh multipath channels, showing that the proposed detector is resistant to the near-far effect and that low performance loss is obtained compared to the single-user bound.     

Modeling of submicrometer gate GaAs field effect transistors

Before describing the mainFet modelings today available, the main technological evolutions ofMesfet andTegfet are summarized. It is brought some information on the various physical effects that occur in the devices and that must be taken into account in the models. It is shown that the different kinds of modelings (Monte Carlo, two dimensional, one dimensional) constitute a continuous chain, where the different elements appear strictly complementary. Finally, the present situation concerning modeling ofMesfet andTegfet will be presented.     

Optimisation de la quantification vectorielle codée par treillis: application au codage des paramètres LSF

Speech coders operating at low bit rates necessitate efficient encoding of the linear predictive coding (Lpc) coefficients. Line spectral Frequencies (Lsf) parameters are currently one of the most efficient choices of transmission parameters for theLpc coefficients. In this paper, an optimized trellis coded vector quantization (Tcvq) scheme for encoding theLsf parameters is presented. When the selection of a proper distortion measure is the most important issue in the design and operation of the encoder, an appropriate weighted distance measure has been used during theTcvq construction process. We further applied the optimizedTcvq system for encoding theLsf parameters of the us Federal Standard (Fs1016) 4.8 kbps speech coder. At lower bit rates, objective and subjective evaluation results show that the incorporatedLsf tcvq encoder performs better than the 34 bits/frameLsf scalar quantizer used originally in the fs1016 coder. The subjective tests reveal also that the 27 bit/frame scheme produces equivalent perceptual quality to that when theLsf parameters are unquantized.