On turbo encodedBicm

The superior performance of the binary turbo codes has stimulated vigorous efforts in generating bandwidth efficient modulation schemes adhering to these codes. Several approaches for the integration of turbo-coding and modulation have emerged in recent years but none seem to dominate. In the bit interleaved coded modulation (Bicm) scheme is used to achieve high bandwidth and power efficiency, while separating coding and modulation. As is now well known, theBicm scheme achieves capacity remarkably close to the constellation channel capacity. The turbo-Bicm scheme enjoys high coding diversity (well suited for fading channels), high flexibility as well as design and implementation simplicity, while maintaining good power efficiency. The system comprises one standard turbo code, an interleaver, a mapper and a modulator at the transmitter, corresponding to a demodulator, a de-interleaver and a turbo decoder at the receiver. A modified system, which improves on performance by incorporating the demodulation in the iterative decoding procedure, is investigated, and some performance gain is demonstrated, especially for low rate codes. Information theoretic arguments for the somewhat minor potential improvement in performance are detailed. The preferred mapper and interleaver for this system are considered. Extending previous works, for higher level modulations, we analyze a system including a convolutional code, an interleaver, a differential encoder (De), a mapper and a modulator at the transmitter. As for theBpsk modulation, the serial concatenation of a convolutional code withDe outperforms the single convolutional code. The serial concatenation withDe approach is analyzed also for a turbo code, where it is found to fail in achieving performance improvement. Several structures for the serial concatenation withDe are examined. These results are substantiated through the ‘spectral thinning’ phenomena of the weight distribution of the convolutional and turbocodes.     

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.     

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.     

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.     

Advanced decoding algorithms for Reed-Solomon/Convolutional concatenated codes

The evaluation of the union bound for theber of Reed-Solomon/Convolutional concatenated codes indicates that their performance might largely improve through the application of soft iterative decoders. This paper presents an iterative decoding algorithm for concatenated codes consisting of an outer Reed-Solomon code, a symbol interleaver and an inner convolutional code. The performance improvement for iterative and non-iterative decoders is evaluated. Existing solutions for the different decoding stages and their interfaces are discussed and their performance is compared. A new procedure is proposed to define the feedback signal from the output of the Reed-Solomon decoder to the input of the convolutional decoder, which captures the reliability information that can be inferred from errors-and-era-suresrs decoders and includes the “state pinning” approach as a particular case. The decoding schemes are applied to the specificdvb-s concatenated code.     

Turbo soft combining hybrid ARQ techniques: Theory and application to 3G wireless networks

This paper deals with the application of Hybrid Automatic Repeat reQuest (h-arq) techniques to allow reliable data communications in wireless 3G networks. Basically, retransmission of coded data is endowed with soft combining schemes applied, respectively, to packet replicas, or to decoding algorithm outputs. In particular, the proposed coding scheme takes advantage of error correction capabilities of the turbo codes, while the combining algorithm follows the diversity approach. The performance of the proposedh-arq schemes has been derived by means of a suitable analytical approach and numerical simulations in the case of a typical umts environment. The results highlight the good behavior of the proposed scheme in term of error rate, throughput, packet delivery delay and power reduction.     

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.     

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.     

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.     

On multiple slice turbo codes

The main problem with the hardware implementation of turbo codes is the lack of parallelism in the MAP-based decoding algorithm. This paper proposes to overcome this problem by using a new family of turbo codes called Multiple Slice Turbo Codes. This family is based on two ideas: the encoding of each dimension with P independent tail-biting codes and a constrained interleaver structure that allows the parallel decoding of the P independent codewords in each dimension. The optimization of the interleaver is described. A high degree of parallelism is obtained with equivalent or better performance than thedvb-rcs turbo code. For very high throughput applications, the parallel architecture decreases both decoding latency and hardware complexity compared to the classical serial architecture, which requires memory duplication.     

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.     

Codage pour modulation d’amplitude en quadrature à grand nombre d’états avec partition à plusieurs niveaux

We present a new multilevel coding technique for the quadrature amplitude modulation with a high number of states (64-QAM and 256-QAM). The constellation is partitioned into four subsets, two bits coded with the samebch code selecting the label of each subset. We show that our system, with a double error correctingbch code, gives better results, considering redundancy, decoding simplicity and performance, than the systems known bx now (using the Reed-Solomon code (64,62), the Lee-Nakamura and Hamming codes). Preliminary differential encoding of the two coded bits and utilisation of a transparentbch code can solve the phase ambiguity problem in a simple way.     

Parallel interference cancellation efficiency for noisy signals in O-CDMA system

The aim of this paper is to evaluate the robustness of Parallel Interference Cancellation (Pic) to noise contribution for an optical Code Division Multiple Access system. The theoretical expression of thePic error probability is developed in the case of white additive Gaussian noise. From theoretical analysis, we show that, even with noise contribution, thePic receiver outperforms the Conventional Correlation Receiver (Ccr). Moreover, the results highlight that, for a given performance, thePic receiver relaxes not only the constraint on the code length, but also the Signal to Noise Ratio compared toCcr. Particularly, this is proofed in access network context, i.e. 30 users withBer lt; 10^?9.     

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.     

Multiple parallel concatenation of circular recursive systematic convolutional (Crsc) codes

This paper deals with a family of error correcting codes based on circular recursive systematic convolutional (Crsc) codes. A multiple or multidimensional parallel concatenation ofCrsc codes is proposed in order to reach minimum distances comparable to those of random codes. For information blocks of size k, minimum distances as large as k/4, for a 1/2 coding rate, may be obtained if the code dimension is raised to 4 or 5. Such codes can be decoded using an iterative (“turbo”) process relying on the extrinsic information concept.     

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.     

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.     

Predictive tree encoding of still images with vector quantization

The superior performance of tree encoding and vector quantization (vq) over scalar quantization (sq) is well known. However, large constraint length (in tree encoding) and large vector length (invq), which are required for a close-to-optimum performance, are limited by the exponentially growing complexity of the source encoder. The intention of our work has been to combine tree encoding andvq, each of moderate complexity, to preserve the advantages of both block and convolutional codes. This principle was applied to the prediction error signals of images as produced by non-adaptive and adaptive linear predictors.     

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.     

QoS multicast routing with delay constraints

Proliferation of group-based real-time applications, such as online games and video conferencing has motivated research into QoS multicast routing. These types of applications require consideration of both source-to-destination delay (i.e., packet delay from the source to all destinations) and inter-destination delay variation (i.e., the difference in packet delay from the source to different destinations) constraints. In this paper, we formulate a new combined problem for delay partitioning and multicast routing with source-to-destination delay and inter-destination delay variation constraints in a QoS framework, where a delay dependent cost function is associated with each network link. After identifying the problem asnp-complete, we introduce a Genetic Algorithm (ga) based algorithm that computes a source-based multicast tree which satisfies both constraints with near-optimal cost. We compare differentga schemes using different selection operators and find that the combination of Steady Statega and Remainder Stochastic Sampling selection operator works best for our problem. Simulation results also show that ourga heuristic consistently perfornis better than several other simple heuristics.