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.     

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.     

An evaluation oftcp with explicit congestion notification

We study the effect of Explicit Congestion Notification (ecn) ontcp for relatively large but finite file transfers inip networks, and compare it to other congestion avoidance mechanisms, namely Drop Tail (dt) and Random Early Detection (red). We use simulation to measuretcp performance for transfers initiated by a varying number of end hosts. In contrast to previous work, we focus on situations in which all nodes in the network operate uniformly under the same mechanism (dt orred orecn). Our results show that under such uniform conditionsecn does not necessarily lead to significant improvement intcp goodput, although in no case does it lead to an actual degradation in performance. Our results also show that, withecn, tcp flows benefit from lower overhead for unsuccessful transmissions. Furthermore, lockouts are largely avoided. In other words, in an all-ecn network resources are shared more fairly. Finally, we show that global synchronization is no longer an issue, and argue that currenttcp versions have essentially solved the problem, regardless of the queue management scheme employed.     

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.     

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.     

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.     

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.     

IP-based transport of voice traffic in the UMTS radio access network: analytical study and empirical validation

In this paper, we investigate theIp protocol as a transport option for the user traffic in the UMTS Terrestrial Radio Access Network (Utran), where stringent delay bounds are to be met for both real-time and non real-time traffic. We focus on real-time voice traffic and present an analytical model for the multiplexing and transport of voice channels in theUtran usingIp. The novelty of our model is that it analytically includes and quantifies the performance of the timer used in multiplexing arriving Frame Protocol (Fp) frames into largerIp packets. We then validate our work through empirical results on a test-bed emulating theUtran transport functionalities. We show the trade-offs between performance, in terms of delay and link utilization, and quantify optimal values for the timer as well as the number ofFp frames perIp packet for a given output link capacity.     

WLAN standards and evolutions

Wireless Local Area Networks technologies have known an important technological and commercial development. Multiplicity of standards and variety of domains of use make necessary to compose with different technologies that can be seen either as concurrent or complementary. In this article, after positioning the different types of wireless networks (IEEE 802.11, HomeRF,HIPERLAN/2, Bluetooth) for mass market and professional applications destination, some generalities are briefly reminded such as centralised and ad-hoc architectures, regulatory constraints in the 2.45 and 5GHZ frequency bands used forWLAN, typical ranges, mobility and security features and limitations. Then the differentIEEE (802.11, 802.11a et 802.11b) andETSI (HIPERLAN/2) standards are described in details as well as their foreseen evolutions. It appears that 802.11 family of standards would take benefit of the currently existing products to evolve smoothly while integrating new features (broadband 802.11a physical layer, necessary radio features to meet European regulatory requirements, future introduction of Quality of Service schemes…). In the meantime,HIPERLAN/2 which has been specified as a complete system already supports most of those important features and is able to be adapted to various kinds of higher network layers. Lastly, it is shown that interworking schemes between 3G cellular systems andWLAN currently under investigations in 3GPP andETSI BRAN should permit in the future to easily operate wide area and multi-access technology based mobile networks.     

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.     

Robust time-frequency distributions based on the robust short time fourier transform

Design of time-frequency distributions (Tfds) that are robust to the impulse noise influence is considered. The robustTfds based on the robust short-time Fourier transform (Stft) are proposed. An efficient procedure to evaluate the robustStft is given. RobustTfds based on the robustStft have better energy concentration around the signal instantaneous frequency (If) than the robustStft itself. Also, theseTfds are more resistant to higher impulse noise than the robustTfds obtained using the local autocorrelation function (Laf) based minimization problem.     

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.     

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.     

Mastering quality of service in GPRS/UMTS an overview

Quality of Service (QoS) has become a very important issue in networking, covering many performance aspects and numerous measures. The deployment of next generation wireless system includes 2.5G General Packet Radio Service (Gprs), which is the packet-switched extension of the Global System for Mobile communications (Gsm), and Third-Generation (3G) Universal Mobile Telecommunications System (Umts) to meet the needs of larger capacity and higher bit rates. AnUmts packet core network is an IP-based network. The Internet Engineering Task Force (Ietf) Forum developed several IP QoS related mechanisms available for IP transport networks. Service Quality Management (Sqm), one component of Telecommunication Management Network (Tun), will enable providers to manage QoS against objectives set out in customer Service Level Agreements (Slas) and will enable customers to compare the service offerings of different service providers.     

Les outils du commerce électronique

Electronic commerce (ec) is a combination of practices, strategies, processes, applications and technologies, enabling business transactions. This paper focuses on software and technologies forec. In the first part, the different categories of applications forec are reviewed. A fully integrated solution does not seem to be the major trend. However, the “best of breed” approach implies an integration problem. The second part of the paper includes modeling languages andec applications core technologies. Theuml appears to be used at different levels in the development of anec solution. Web technologies are continually evolving:XML could be the language of the future for information exchanges between heterogeneous applications. The third part deals with component and application integration. In spite of converging standards for languages as well as for models, system interoperability is a major problem, when informations are to be shared and processes to be jointed. Two major trends can be identified: Web services andEDI approaches.     

nOBS: an ns2 based simulation tool for performance evaluation of TCP traffic in OBS networks

Performance evaluation of tcp traffic in obs networks has been under intensive study, since tcp constitutes the majority of Internet traffic. As a reliable and publicly available simulator, ns2 has been widely used for studying tcp/ip networks; however ns2 lacks many of the components for simulating optical burst switching networks. In this paper, an ns2 based obs simulation tool (nobs), which is built for studying burst assembly, scheduling and contention resolution algorithms in obs networks is presented. The node and link objects in obs are extended in nobs for developing optical nodes and optical links. The ingress, core and egress node functionalities are combined into a common optical node architecture, which comprises agents responsible for burstification, routing and scheduling. The effects of burstification parameters, e.g., burstification timeout, burst size and number of burstification buffers per egress node, on tcp performance are investigated using nobs for different tcp versions and different network topologies.     

Codage conjoint source-canal des paramètres lsf du codeur FS1016: application de la quantification vectorielle par treillis optimisée pour un canal bruité

In this paper, we present an optimized trellis coded vector quantization (tcvq) coding system designed for the effective and robust coding of lsf spectral parameters at low bit rate. The aim of this system, called at the beginning « lsf-otcvq Encoder », is to achieve a low bit rate transparent quantization of lsf parameters of the us Federal Standard (fs1016) speech coder. Once the effectiveness of our lsf-otcvq encoders (with weighted distance) was proven in the case of ideal transmissions over noiseless channel, we were interested after in the improvement of their robustnesses for real transmissions over noisy channel. To implicitly protect the transmission indices of our lsf-otcvq coders incorporated in the Fsl016, we used a joint source-channel coding carried out by the channel optimized vector quantization (covq) method. In the case of transmissions over noisy channel, we will show that our new encoding system, called “covq-lsf-otcvq Encoder”, would be able to contribute significantly in the improvement of the fs1016 performances by ensuring a good coding robustness of its lsf spectral parameters.     

Design and performance of a product code turbo encoding-decoding prototype

This paper presents the latest results on a block turbo decoder design. We propose a block turbo decoder circuit for the error protection of small data blocks such asAtm cells on anAwgn (additive white Gaussian noise) channel with a code rate close to 0.5. A prototype was developed atEnst Bretagne. It allowsBer (bit error rate) measurements down to 10^?9 and uses programmable gate arrays (Fpga Xilinx circuits). The elementary extendedBch code and the data block size can be modified to fit specifications of different applications.     

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.     

Architecture d’un réseau actif à base de services Web

This paper presents a novel active architecture for building and deploying network services:aswa, Web Services based Active network Architecture. At the architectural level,aswa defines an active node whose functionalities are divided into the Node Operating System, the Execution Environment, and the Active Applications. At the implementation level,aswa is a Web Services based platform where new components could be added and deployed, in order to dynamically modify network nodes behavior. Applications can be developed with any language and communicate across heterogeneous environments, and across Internet and Intranet structures. At the deployment levelaswa uses an active node approach, and offers a controlled deployment mode. In terms of security, Authentication of deployed code and protection of the nodes is achieved by the use ofhttps and the header extensions of thesoap envelope. Finally to validate this architecture,aswa defines a Firewall as an Active Application to secure the code deployment.