We present new ways of producing a channel chart employing model-based approaches. We estimate the angle of arrival$\theta $and the distance$\rho $between the base station and the user equipment by employing our algorithms, inverse of the root sum squares of channel coefficients (ISQ) algorithm, linear regression (LR) algorithm, and MUSIC/MUSIC (MM) algorithm. We compare these methods with the channel charting algorithms principal component analysis (PCA), Sammon’s method (SM), and autoencoder (AE). We show that ISQ, LR, and MM surpass PCA, SM, and AE in performance. We also compare our algorithm MM with an algorithm from the literature that uses the MUSIC algorithm jointly on$\theta $and$\rho $. We call this algorithm the JM algorithm. JM performs very slightly better than MM but at a substantial increase in complexity. Finally, we introduce the rotate-and-sum (RS) algorithm which has about the same performance as the MM and JM algorithms. Unlike MUSIC, RS does not employ eigenvalue and eigenvector analysis. Thus, it is more suitable for direct register transfer logic (RTL) implementation.

We present new ways of producing a channel chart employing model-based approaches. We estimate the angle of arrival θ and the distance between the base station and the user equipment$p$by employing our algorithms, inverse of the root sum squares of channel coefficients (ISQ) algorithm, linear regression (LR) algorithm, and the MUSIC/MUSIC (MM) algorithm. We compare these methods with the channel charting algorithms principal component analysis (PCA), Sammon's method (SM), and autoencoder (AE) from [1]. We show that ISQ, LR, and MM outperform all three in performance. ISQ and LR have similar performance with ISQ having less complexity than LR. The performance of MM is better than ISQ and LR but it is more complex. Finally, we introduce the rotate-and-sum (RS) algorithm which has about the same performance as the MM algorithm but is less complex due to the avoidance of the eigenvector and eigenvalue analysis and a potential register transfer logic (RTL) implementation.

An analysis of the diversity gain for bit-interleaved coded multiple beamforming (BICMB) method in millimeter-wave (mm-Wave) massive multiple-input multiple-output (MIMO) systems is carried out for both the single-user and multi-user scenarios. We show that full diversity gain and full spatial multiplexing order can be achieved in both scenarios. Also, we show that the diversity gain in the multi-user scenario is independent of the number of users in the system. The assumption here is that the channel state information (CSI) is known at both sides of the transmitter and the receiver and the number of antennas in each remote antenna unit (RAU) tends to infinity. Based on these assumptions, the diversity gain for the single-user scenario is (Σi,jβij/)2/Σi,jβij2Lij-1where Lijis the number of propagation paths and βijis the large scale fading coefficient between the ith RAU in the transmitter and the jth RAU in the receiver. The diversity gain in the multi-user scenario for the kth user is M2/ΣjLkj-1where M represents the number of RAUs at the transmitter. Simulation results show that when the perfect CSI assumption is satisfied, the use of BICMB results in the diversity gain values predicted by the analysis.

The new era of the Internet of Things (IoT) is prompting the evolution of conventional vehicle ad-hoc networks (VANETs) into the Internet of Intelligent Vehicles (IoIV). Different from VANETs, where a vehicle is essentially considered as a node disseminating messages, the emerging IoIV paradigm is expected to regard each vehicle as a smart object equipped with a powerful multisensor platform, unprecedented communication capability, computing units, and Internet protocol (IP)-based connectivity. As such, the vehicles in IoIV are highly efficient in a broad array of vehicular and transportation applications. As a unique subset of general purpose IoT, IoIV can benefit from the existing research on VANET, which lays the foundation toward a more pervasive and ubiquitous communications and networking core that is essential for IoIV. Nevertheless, research in many aspects of IoIV, especially those that are application-driven and data-oriented ones, is still at its infancy.

In this paper, we propose a new, robust near-optimal mutual information combined zero-forcing per-group precoding (ZF-PGP) method for finite alphabets with imperfect channel state information (CSI) at the receiver. ZF-PGP achieves very high gains over ZF precoding techniques, while it simultaneously offers individually separate streams to reach individual user equipment (UE), i.e., it obliterates the need for coordinated, joint decoding by the group's UEs. The proposed robust ZF-PGP scheme comprises an inner linear beamformer (BF), which relies on second-order CSI and is based on the massive MIMO-based joint division and multiplexing for finite alphabets (JSDM-FA) to form orthogonal user groups, and an outer linear precoder that relies on instantaneous group CSI to form a per-group precoding within groups (PGP-WG) with separate data streams to each user. As we show by multiple examples, for a uniform linear array (ULA), the proposed ZF-PGP technique is very efficient, near-optimal design with reduced complexity at the UEs. Due to JSDM-FA second-order CSI inner beamforming, the groups are easily identified. We also demonstrate the robustness of the proposed precoding techniques to channel estimation errors, showing minimal performance loss.

In this paper, the problem of designing a forward link linear precoder for Massive Multiple-Input Multiple-Output (MIMO) systems in conjunction with Quadrature Amplitude Modulation (QAM) is addressed. A challenge in such system design is to consider finite alphabet inputs, especially with larger constellation sizes such as M ≥ 16. First, we employ a novel and efficient methodology that allows for a sparse representation of multiple users and groups in a fashion similar to Joint Spatial Division and Multiplexing (JSDM), thus offering an extension of JSDM to finite alphabet data symbols. We term the new approach JSDM for finite alphabets (JSDM-FA). We show that similarly to ordinary JSDM, the optimal pre-beamforming matrix for Massive MIMO in JSDM-FA becomes the DFT matrix. The proposed methodology is next applied jointly with the complexity-reducing Per-Group Processing within groups (PGP-WG) technique, on a per user group basis, in conjunction with QAM modulation and in simulations, for constellation size up to M = 64. We show by numerical results that the precoders developed offer significantly better performance than the configuration with no precoder or the plain beamformer and with M ≥ 16.16.

This work presents a single-carrier massive MIMO transmission system for the frequency selective Gaussian multi-user channel. It considers both cases of spatially uncorrelated and correlated channel and compares them in terms of the user sum-rate as well as the general performance. We consider a channel with M antennas at the base station which provides services for K single- antenna users. We develop a general expression for the achievable rate among users in the channel with a correlation among antennas at the base station. It has been shown that, when there is no correlation between base station antennas or users, the channel matched filter precoder outperforms any other precoder. In a highly- correlated massive MIMO channel however, the conventional channel matched filter precoder does not perform as expected. We show the failure of the channel matched filter in the presence of a correlation pattern among antennas at the base station with theoretical analysis and simulations. We apply three different precoders to enhance the achievable rate and their performances are determined by simulations. We show that the precoders outperform the channel matched filter in a highly-correlated channel with a large number of users. By increasing the number of users in the system, the conventional precoders show even better performance in terms of the user's sum- rates.

In this paper, we investigate the energy efficiency (EE) and spectral efficiency (SE) tradeoff in multi-cell heterogeneous networks (HetNets). Our objective is to maximize both EE and SE of the network while satisfying the rate requirement of users. We use multi-objective optimization techniques to define the objective function. We propose a three-stage algorithm in this paper. First, we select the cell-center radius for the fractional frequency reuse (FFR) method. Second, we assign the frequency resources to satisfy the rate requirements of users and maximize the objective function. Third, the power allocation subproblem is solved by using the Levenberg-Marquardt method. Our numerical results show that a Pareto-optimal solution exists for EE and SE. We present results for different rate constraints.

In this paper, the problem of sequential beam construction and adaptive channel estimation based on reduced rank (RR) Kalman filtering for frequency-selective massive multiple-input multiple-output (MIMO) systems employing single-carrier (SC) in time division duplex (TDD) mode are considered. In two-stage beamforming, a new algorithm for statistical pre-beamformer design is proposed for spatially correlated time-varying wideband MIMO channels under the assumption that the channel is a stationary Gauss-Markov random process. The proposed algorithm yields a nearly optimal pre-beamformer whose beam pattern is designed sequentially with low complexity by taking the user-grouping into account, and exploiting the properties of Kalman filtering and associated prediction error covariance matrices. The resulting design, based on the second order statistical properties of the channel, generates beamspace on which the RR Kalman estimator can be realized as accurately as possible. It is observed that the adaptive channel estimation technique together with the proposed sequential beamspace construction shows remarkable robustness to the pilot interference. This comes with significant reduction in both pilot overhead and dimension of the pre-beamformer lowering both hardware complexity and power consumption.

In this paper, the problem of designing a linear precoder for Multiple-Input Multiple-Output (MIMO) systems in conjunction with Quadrature Amplitude Modulation (QAM) is addressed. First, a novel and efficient methodology to evaluate the input-output mutual information for a general Multiple-Input Multiple-Output (MIMO) system as well as its corresponding gradients is presented, based on the Gauss-Hermite quadrature rule. Then, the method is exploited in a block coordinate gradient ascent optimization process to determine the globally optimal linear precoder with respect to the MIMO input-output mutual information for QAM systems with relatively moderate MIMO channel sizes. The proposed methodology is next applied in conjunction with the complexity-reducing per-group processing (PGP) technique to both perfect channel state information at the transmitter (CSIT) as well as statistical channel state information (SCSI) scenarios, with large transmitting and receiving antenna size, and for constellation size up to M = 64. We show by numerical results that the precoders developed offer significantly better performance than the configuration with no precoder as well as the maximum diversity precoder for QAM with constellation sizes M = 16; 32, and 64 and for MIMO channel size up to 100 ⌉x 100.

Cellular wireless or mobile communications have seen four generations of technological developments. Starting with analog voice with the first, and then moving on to digital voice with the second generation, these generations were each marked with a clear technological advancement. For the third generation, the advancement was in incorporating data on top of a voice-based infrastructure. With the fourth generation, every service was converted into data format, or packetized transmissions, including voice and video. Today, technologists are proposing a fifth generation for around the time frame of 2020. Most consider this time to be when the infrastructure will need to be renewed. Together with this observation, it is usually argued that the new generation technology should possess a number of features. Yet, there is really no consensus on what these new features should be. Some argue that we are facing a new generation of devices that will have continuous Internet connectivity, and with that there will be more machine to-machine or machine-type communications. It is further argued that, as a result, new communication protocols should be able to support the new formats of communication that this change will require. For example, it is argued that machine-type communication will require very low latency. Others argue that the demand for services will increase by about three orders of magnitude and the new technology should be designed to support this tremendous increase, perhaps handling each order of magnitude by means of a different approach. Yet, there are others who argue that the current communications infrastructure is highly energy-inefficient and the fifth generation should be designed to solve this problem, by increasing energy efficiency by several orders of magnitude. In this talk, we will discuss the pros and cons of the approaches for defining and realizing the fifth generation cellular wireless technologies as seen today. We will discuss what fifth generation can be expected to be and, more importantly, what it cannot be. The goal in this presentation is to address both the very high expectations and the realities.

A substantial amount of research exists in the area of energy efficiency in communications and networking. In recognition of this fact, the IEEE Communications Society has decided to publish a series on Green Communications and Networking as three issues of the IEEE Journal On Selected Areas in Communications. The first issue of the Series was published in December 2015 with 39 papers. This second issue of the Series has 52 papers, covering a wide selection of topics. These topics can be characterized in five categories. The first category in this issue is cellular networks and consists of 15 papers.

A substantial amount of research exists in the area of energy efficiency in communications and networking. In recognition of this fact, the IEEE Communications Society has decided to publish a Series on Green Communications and Networking (SGCN) as three issues of the IEEE Journal on Selected Areas in Communications (JSAC). The first issue of JSAC SGCN was published in December 2015 with 39 papers. The second issue was published in May 2016 with 52 papers. This third and last issue has a total of 71 papers, covering a wide selection of topics. These topics can be characterized in six categories, with one subcategory appearing in one of the six main categories.

In this paper, the problem of designing a linear precoder for multiple-input multiple-output (MIMO) systems in conjunction with quadrature amplitude modulation (QAM) is addressed. First, a novel and efficient methodology to evaluate the input-output mutual information for a general MIMO system as well as its corresponding gradients is presented, based on the Gauss-Hermite quadrature rule. Then, the method is exploited in a block coordinate gradient ascent optimization process to determine the globally optimal linear precoder with respect to the MIMO input-output mutual information for QAM systems with relatively moderate MIMO channel sizes. The proposed methodology is next applied in conjunction with the complexity-reducing per-group processing technique to both perfect channel state information (CSI) at the transmitter as well as statistical CSI (Statistical CSI) scenarios, with large transmitting and receiving antenna sizes, and for constellation size up to M = 64. We show by the numerical results that the precoders developed offer significantly better performance than the configuration with no precoder as well as the maximum diversity precoder for QAM with constellation sizes M = 16, 32, and 64 and for MIMO channel size up to 100 × 100.

In this paper, we propose an energy-efficient resource allocation scheme for multi-channel multi- cell networks. Our objective is to maximize the network energy efficiency subject to the minimum rate constraints for each user. We take into account both the static and transmit power consumption at base stations. We employ interference pricing and dual decomposition methods. The interference pricing scheme penalizes the base stations based on their marginal costs of interference to other users. This is leveraged to mitigate inter-sector interference and determine the optimal power levels, realizing the objective. Using dual decomposition techniques, we enforce the minimum rate constraints to be satisfied. To achieve faster convergence, we employ the Levenberg-Marquardt method in the power control phase. The proposed framework employs the fractional frequency reuse (FFR) scheme. We sequentially determine the cell-center boundaries, schedule the users in the frequency domain, and solve the power control problem. Simulation results demonstrate that the network energy efficiency can be significantly improved while reducing the user outages compared to the non-cooperative power allocation scheme.

In this paper, we develop an energy-efficient base station deployment framework for heterogeneous networks. Micro base station deployment increases the total capacity of the network. However, increasing the number of micro base stations excessively may reduce the energy efficiency of the network. Therefore, in this paper, we examine the energy efficiency aspect of the micro base station deployment problem. This problem can be divided into two subproblems: choosing feasible candidate micro base station locations and selecting the optimum set of micro base stations among the candidate locations. The proposed algorithm first chooses the subset of the feasible locations as candidate locations, and then selects the micro base stations which maximize the energy efficiency of the network iteratively. It is shown that the proposed algorithm is a constant-factor approximation of the optimal solution. Our simulations demonstrate that proposed algorithm improves the energy efficiency of the network up to 12% for low-loaded scenarios and 98% for the high-loaded scenarios.

The articles in this special section focus on the technologies and applications that are facilitated by green communications and networking.

Network coding-based link failure recovery techniques provide near-hitless recovery and offer high capacity efficiency. Diversity coding is the first technique to incorporate coding in this field and is easy to implement over small networks. However, the capacity efficiency of this implementation is restricted by its systematic coding and high design complexity despite having lower complexity than the other coding-based recovery techniques. In this paper, we propose a simple column generation-based design algorithm and a novel advanced diversity coding technique to achieve near-hitless recovery over large networks. The traffic matrix, which consists of unicast connection demands, is decomposed into traffic vectors for each destination node. Further, the connection demands in each traffic vector are partitioned into coding groups. The design framework consists of two parts: a main problem and a subproblem. The main problem is solved with Linear Programming (LP) and Integer Linear Programming (ILP), whereas the subproblem can be solved with different methods. Simulation results suggest that the novel design algorithm simplifies the capacity placement problem, which enables implementing diversity coding-based recovery including the novel coding structure on large networks with arbitrary topology. It achieves near-hitless recovery with an almost optimal capacity efficiency for any single destination-based recovery.

Next generation wireless networks face the challenge of increasing energy consumption while satisfying the unprecedented increase in the data rate demand. To address this problem, we propose a utility-based energy-efficient resource allocation algorithm for the downlink transmissions in heterogeneous networks (HetNets). We consider the fractional frequency reuse (FFR) method in order to mitigate the intra- and inter-cell interference. The proposed algorithm divides the resource allocation problem into frequency and power assignment problems and sequentially solves them. The proposed power control algorithm uses the gradient ascent method to control the transmit power of macrocell base stations (MeNBs) as most of the power in the network is consumed there. We present the optimality conditions of the resource allocation problem and the convergence of the proposed algorithm. In order to mitigate the inter-cell interference further, we study the interference pricing mechanisms and obtain an upper bound to the maximum energy efficiency problem including the inter-cell interference contributions. The performance of the proposed algorithm is studied in a Long Term Evolution (LTE) system. Our simulation results demonstrate that the proposed algorithm provides substantial improvements in the energy efficiency and throughput of the network. It is also shown that interference pricing provides only marginal improvements over the proposed algorithm.

In this paper, the problem of designing a linear precoder for Multiple-Input Multiple-Output (MIMO) systems employing Low-Density Parity-Check (LDPC) codes is addressed under the constraint of minimizing the dependence between the system's receiving branches, thus reducing the relevant transmitter and receiver complexities. Our approach constitutes an interesting generalization of Bit-Interleaved Coded Modulation with Multiple Beamforming (BICMB) which has shown many benefits in MIMO systems. We start with a Pareto optimal surface modeling of the system and show the difficulty involved in the corresponding optimization problem. We then propose an alternative, practical technique, called Per-Group Precoding (PGP), which groups together multiple input symbol streams and corresponding receiving branches in the “virtual” channel domain (after singular value decomposition of the original MIMO channel), and thus results in independent transmitting/receiving streams between groups. We show with numerical results that PGP offers almost optimal performance, albeit with significant reduction both in the precoder optimization and LDPC EXIT chart based decoding complexities.

Abstract When the stations in an IEEE 802.11 infrastructure basic service set employ Transmission Control Protocol (TCP), this exacerbates per‐flow unfair access problem. We propose a novel analytical model to approximately calculate the maximum per‐flow TCP congestion window limit that prevents packet losses at the access point buffer and therefore provides fair TCP access both in the downlink and uplink. The proposed analysis is unique in considering the effects of varying number of uplink and downlink TCP flows, differing round trip times among TCP connections and the use of delayed TCP acknowledgment (ACK) mechanism. Motivated by the findings of this theoretical analysis and simulations, we design a link layer access control block to be employed only at the access point in order to resolve the unfair access problem. The proposed link layer access control block uses congestion control and ACK filtering approach by prioritizing the access of TCP data packets of downlink flows over TCP ACK packets of uplink flows. Via simulations, we show that the proposed algorithm can provide both short‐term and long‐term fair accesses while improving channel utilization and access delay. Copyright © 2013 John Wiley & Sons, Ltd.

In this paper, we address the uplink resource scheduling problem in single carrier frequency-domain multiple access systems. In particular, we focus on the efficiency and fairness trade-offs in scheduling and resource allocation for wireless cellular networks. We present an efficient implementation method that translates these scheduling problems into set partitioning problems that are well-studied in the literature. Then, we discuss a family of utility functions that enable us to investigate the performance of different frequency domain schedulers such as the sum-rate maximization, proportional fair, and max-min fair schedulers. We use the price of fairness as a metric to analytically quantify these trade-offs. Based on the intuition that fairness of resource allocation in cellular radio networks corresponds to the prioritization of cell-edge user rates, we demonstrate that the proportional fair scheduler significantly improves fairness among users, and increases the rates offered to the cell-edge and median users when compared to the sum-rate maximization scheduler. This comes at the cost of reducing the cell-center user rates and the aggregate user rate. We present the steps on how to take into account the practical implementation constraints, in particular, those related with the discrete Fourier transform implementation, in the problem formulation. Simulation results that illustrate these trade-offs are also presented. We conclude that this type of analysis can provide guidelines for the network operators to control the efficiency and fairness trade-off as the data traffic grows.

Network coding-based link failure recovery techniques provide near-hitless recovery and offer high capacity efficiency. Diversity coding is the first technique to incorporate coding in this field and is easy to implement over small arbitrary networks. However, its capacity efficiency is restricted by its systematic coding and high design complexity even though it has lower complexity than the other coding-based recovery techniques. Alternative techniques mitigate some of these limitations, but they are difficult to implement over arbitrary networks. In this paper, we propose a novel non-systematic coding technique and a simple design algorithm to implement the diversity coding-based (or network coding-based) recovery over arbitrary networks. The design framework consists of two parts. An ILP formulation for each part is developed. The simulation results suggest that both the novel coding structure and the novel design algorithm lead to higher capacity efficiency for near-hitless recovery. The new design algorithm is able to achieve optimal results in large arbitrary networks.

For Multiple-Input Multiple-Output (MIMO) systems with frequency selective fading channels, Bit-Interleaved Coded Multiple Beamforming (BICMB) with Orthogonal Frequency Division Multiplexing (OFDM) can offer both spatial and multipath diversity, making it an important technique. It was proved that full diversity of BICMB-OFDM can be achieved with a condition of RcSL ≤ 1, where Rc, S, and L are the code rate, the number of parallel streams at each subcarrier, and the number of channel taps, respectively. In this paper, the precoding technique is applied to overcome the full diversity restriction with the minimum achievable decoding complexity.

Diversity coding is a form of network coding for link failure recovery in communication networks. Since it employs coding, there is no feedback signaling, and that feature makes it very fast. Previously, we have employed this basic technique and linear programming to come up with fast link failure recovery systems that also have small extra capacity. One approach, Diversity Coding Tree, employs mixed integer programming and results in very fast restoration. Another approach is called Coded Path Protection, and employs integer linear programming and has the advantage of small extra capacity. This latter technique is based on former work that considers a communication network as consisting of bidirectional links. However, employing coding on bidirectional links results in larger restoration times than the former technique. In this paper, we develop an improved version of our former techniques. This new technique employs a mixed integer linear programming formulation and results in restoration times as fast as Diversity Coding Tree with reduced extra capacity.

For Multiple-Input Multiple-Output (MIMO) systems with frequency selective fading channels, Bit-Interleaved Coded Multiple Beamforming (BICMB) with Orthogonal Frequency Division Multiplexing (OFDM) can be employed to offer both spatial and multipath diversity, making it an important technique. Nevertheless, analyzing its diversity is a challenging problem. In this paper, the diversity analysis of BICMB-OFDM is carried out. First, the maximum achievable diversity is derived and a full diversity condition is proved. Then, the performance degradation due to the subcarrier correlation is investigated. Finally, the subcarrier grouping technique is applied to combat the performance degradation and provide multi-user compatibility.

Coding-based restoration techniques have proactive restoration which results in time savings over other state-of-the-art restoration techniques. Diversity coding is a coding-based recovery technique which offers near-hitless restoration with a competitive spare capacity requirement with respect to other techniques. In this paper, we show that diversity coding can achieve sub-ms restoration time. In addition, we develop two optimal algorithms for pre-provisioning of the static traffic and one for the dynamic provisioning of the traffic on-demand. There is one algorithm for systematic and one for non-systematic diversity coding in pre-provisioning. An MIP formulation and an ILP formulation are developed for systematic and non-systematic cases, respectively. The MIP formulation of the systematic diversity coding requires much fewer integer variables and constraints than similar optimal coding-based formulations. In dynamic provisioning, an ILP-based algorithm covers both of the systematic and non-systematic diversity coding. In all scenarios, diversity coding results in smaller restoration time, higher transmission integrity, and much reduced signaling complexity than the existing techniques in the literature. Simulation results indicate that diversity coding has significantly higher restoration speed than Shared Path Protection (SPP) and p-cycle techniques from the literature as well as Synchronous Optical Network (SONET) rings, which are commonly deployed by service providers today. In terms of capacity efficiency, it outperforms SONET rings and 1+1 APS, whereas it may require more extra capacity than the p-cycle technique and SPP. Diversity coding offers a preferable tradeoff which offers two orders of magnitude increase in restoration speed at the expense of less than 26% extra spare capacity.

Link failures in wide area networks are common and cause significant data losses. Mesh-based protection schemes offer high capacity efficiency but they are slow and require complex signaling. Additionally, real-time reconfigurations of cross-connects threaten their transmission integrity. On the other hand, there are other schemes that are proactive. Proactivity results in higher restoration speed, lower signaling complexity, and higher transmission integrity. This paper introduces a coding-based proactive protection scheme, named Coded Path Protection (CPP). In CPP, a backup stream of the primary data is encoded with other data streams, resulting in capacity savings. In addition to a systematic approach of building valid coding structures, this paper presents an optimal and simple capacity placement and coding group formation algorithm. The algorithm converts the sharing structure of any solution of a Shared Path Protection (SPP) technique into a coding structure with minimum extra capacity. We conducted quantitative and qualitative comparisons of our technique with SPP and Shared Link Protection (SLP). Simulation results confirm that the CPP is significantly faster than both the SPP and the SLP. It is clearly more capacity efficient than the SLP while it incurs marginal extra capacity beyond that of the SPP.

Multi-Input Multi-Output (MIMO) techniques have been incorporated with Orthogonal Frequency Division Multiplexing (OFDM) for broadband wireless communication systems. Bit-Interleaved Coded Multiple Beamforming (BICMB) can achieve both spatial diversity and spatial multiplexing for flat fading MIMO channels. For frequency selective fading MIMO channels, BICMB with OFDM (BICMB-OFDM) can be employed to provide both spatial diversity and multipath diversity, making it an important technique. In our previous work, the subcarrier grouping technique was applied to combat the negative effect of subcarrier correlation. It was also proved that full diversity of BICMB-OFDM with Subcarrier Grouping (BICMB-OFDM-SG) can be achieved within the condition RcSL ≤ 1, where Rc, S, and L are the code rate, the number of parallel streams at each subcarrier, and the number of channel taps, respectively. The full diversity condition implies that if S increases, Rcmay have to decrease to maintain full diversity. As a result, increasing the number of parallel streams may not improve the total transmission rate. In this paper, the precoding technique is employed to overcome the full diversity restriction issue of RcSL ≤ 1 for BICMB-OFDM-SG. First, the diversity analysis of precoded BICMB-OFDM-SG is carried out. Then, the full-diversity precoding design is developed with the minimum achievable decoding complexity.

For broadband wireless communication systems, Orthogonal Frequency Division Multiplexing (OFDM) has been combined with Multi-Input Multi-Output (MIMO) techniques. Bit-Interleaved Coded Multiple Beamforming (BICMB) can achieve both spatial diversity and spatial multiplexing for flat fading MIMO channels. For frequency selective fading MIMO channels, BICMB with OFDM (BICMB-OFDM) can be applied to achieve both spatial diversity and multipath diversity, making it an important technique. However, analyzing the diversity of BICMB-OFDM is a challenging problem. In this paper, the diversity analysis of BICMB-OFDM is carried out. First, the maximum achievable diversity is derived and a full diversity condition RcSL ≤ 1 is proved, where Rc, S, and L are the code rate, the number of parallel steams transmitted at each subcarrier, and the number of channel taps, respectively. Then, the performance degradation due to the correlation among subcarriers is investigated. Finally, the subcarrier grouping technique is employed to combat the performance degradation and provide multi-user compatibility.

Diversity coding is a network restoration technique which offers near-hitless restoration, while other state-of-the-art techniques are significantly slower. Furthermore, the extra spare capacity requirement of diversity coding is competitive with the others. Previously, we developed heuristic algorithms to employ diversity coding structures in networks with arbitrary topology. This paper presents two algorithms to solve the network design problems using diversity coding in an optimal manner. The first technique pre-provisions static traffic whereas the second technique carries out the dynamic provisioning of the traffic on-demand. In both cases, diversity coding results in smaller restoration time, simpler synchronization, and much reduced signaling complexity than the existing techniques in the literature. A Mixed Integer Programming (MIP) formulation and an algorithm based on Integer Linear Programming (ILP) are developed for pre-provisioning and dynamic provisioning, respectively. Simulation results indicate that diversity coding has significantly higher restoration speed than Shared Path Protection (SPP) and p-cycle techniques. It requires more extra capacity than the p-cycle technique and SPP. However, the increase in the total capacity is negligible compared to the increase in the restoration speed.

Link failures in wide area networks are common and cause significant data losses. Mesh-based protection schemes offer high capacity efficiency but they are slow and require complex signaling. Additionally, real-time reconfiguration of a crossconnect threatens their transmission integrity. On the other hand, coding-based protection schemes are proactive. Therefore, they have higher restoration speed, lower signaling complexity, and higher transmission integrity. This paper introduces a coding-based protection scheme, named Coded Path Protection (CPP). In CPP, a backup copy of the primary data is encoded with other data streams, resulting in capacity savings. This paper presents an optimal and simple capacity placement and coding group formation algorithm. The algorithm converts the sharing structure of any solution of a Shared Path Protection (SPP) technique into a coding structure with minimum extra capacity. W e conducted quantitative and qualitative comparisons of our technique with the SPP and, another technique, known as p-cycle protection. Simulation results confirm that the CPP is significantly faster than the SPP and p-cycle techniques. CPP incurs marginal extra capacity on top of SPP. Its capacity efficiency is lower than the p-cycle technique for dense networks but can be higher for sparse networks. In addition, unlike p-cycle protection, CPP is inherently suitable for the wavelength continuity constraint in optical networks.

When the Channel State Information (CSI) is known by both the transmitter and the receiver, beamforming techniques employing Singular Value Decomposition (SVD) are commonly used in Multiple-Input Multiple-Output (MIMO) systems. Without channel coding, there is a trade-off between full diversity and full multiplexing. When channel coding is added, both of them can be achieved as long as the code rate Rcand the number of employed subchannels S satisfy the condition RcS ≤1. By adding a properly designed constellation precoder, both full diversity and full multiplexing can be achieved for both uncoded and coded systems with the trade-off of a higher decoding complexity, e.g., Fully Precoded Multiple Beamforming (FPMB) and Bit-Interleaved Coded Multiple Beamforming with Full Precoding (BICMB-FP) without the condition RcS ≤1. Recently discovered Perfect Space-Time Block Code (PSTBC) is a full-rate full-diversity space-time code, which achieves efficient shaping and high coding gain for MIMO systems. In this paper, a new technique, Bit-Interleaved Coded Multiple Beamforming with Perfect Coding (BICMB-PC), is introduced. BICMB-PC transmits PSTBCs through convolutional coded SVD systems. Similarly to BICMB-FP, BICMB-PC achieves both full diversity and full multiplexing, and its performance is almost the same as BICMB-FP. The advantage of BICMB-PC is that it can provide a much lower decoding complexity than BICMB-FP, since the real and imaginary parts of the received signal can be separated for BICMB-PC of dimensions 2 and 4, and only the part corresponding to the coded bit is required to acquire one bit metric for the Viterbi decoder.

Perfect Space-Time Block Codes (PSTBCs) achieve full diversity, full rate, nonvanishing constant minimum determinant, uniform average transmitted energy per antenna, and good shaping. However, the high decoding complexity is a critical issue for practice. When the Channel State Information (CSI) is available at both the transmitter and the receiver, Singular Value Decomposition (SVD) is commonly applied for a Multiple-Input Multiple-Output (MIMO) system to enhance the throughput or the performance. In this paper, two novel techniques, Perfect Coded Multiple Beamforming (PCMB) and Bit-Interleaved Coded Multiple Beamforming with Perfect Coding (BICMB-PC), are proposed, employing both PSTBCs and SVD with and without channel coding, respectively. With CSI at the transmitter (CSIT), the decoding complexity of PCMB is substantially reduced compared to a MIMO system employing PSTBC, providing a new prospect of CSIT. Especially, because of the special property of the generation matrices, PCMB provides much lower decoding complexity than the state-of-the-art SVD-based uncoded technique in dimensions 2 and 4. Similarly, the decoding complexity of BICMB-PC is much lower than the state-of-the-art SVD-based coded technique in these two dimensions, and the complexity gain is greater than the uncoded case. Moreover, these aforementioned complexity reductions are achieved with only negligible or modest loss in performance.

Link failures in wide area networks are common. To recover from such failures, a number of methods such as SONET rings, protection cycles, and source rerouting have been investigated. Two important considerations in such approaches are the recovery time and the needed spare capacity to complete the recovery. Usually, these techniques attempt to achieve a recovery time less than 50 ms. In this paper we introduce an approach that provides link failure recovery in a hitless manner, or without any appreciable delay. This is achieved by means of a method called diversity coding. We present an algorithm for the design of an overlay network to achieve recovery from single link failures in arbitrary networks via diversity coding. This algorithm is designed to minimize spare capacity for recovery. We compare the recovery time and spare capacity performance of this algorithm against conventional techniques in terms of recovery time, spare capacity, and a joint metric called Quality of Recovery (QoR). QoR incorporates both the spare capacity percentages and worst case recovery times. Based on these results, we conclude that the proposed technique provides much shorter recovery times while achieving similar extra capacity, or better QoR performance overall.

In Multiple-Input Multiple-Output (MIMO) systems, Sphere Decoding (SD) can achieve performance equivalent to full search Maximum Likelihood (ML) decoding with reduced complexity. Several researchers reported techniques that reduce the complexity of SD further. In this paper, a new technique is introduced which decreases the computational complexity of SD substantially, without sacrificing performance. The reduction is accomplished by deconstructing the decoding metric to decrease the number of computations and exploiting the structure of a lattice representation. Simulation results show that this approach achieves substantial gains for the average number of real multiplications and real additions needed to decode one transmitted vector symbol. As an example, for a 4 × 4 MIMO system, the gains in the number of multiplications are 85% with 4-QAM and 90% with 64-QAM, at low SNR.

Multiple beamforming is realized by singular value decomposition of the channel matrix which is assumed to be known to both the transmitter and the receiver. Bit-Interleaved Coded Multiple Beamforming (BICMB) can achieve full diversity as long as the code rate Rcand the number of employed subchannels S satisfy the condition RcS ≤ 1. Bit-Interleaved Coded Multiple Beamforming with Constellation Precoding (BICMB-CP), on the other hand, can achieve full diversity without the condition RcS ≤ 1. However, the decoding complexity of BICMB-CP is much higher than BICMB. In this paper, a reduced complexity decoding technique, which is based on Sphere Decoding (SD), is proposed to reduce the complexity of Maximum Likelihood (ML) decoding for BICMB-CP. The decreased complexity decoding achieves several orders of magnitude reduction, in terms of the average number of real multiplications needed to acquire one precoded bit metric, not only with respect to conventional ML decoding, but also, with respect to conventional SD.

The computational complexity of optimum decoding for an orthogonal space-time block code GNsatisfying GNHGN= c(Σk=1K|sk|2)INwhere c is a positive integer is quantified. Four equivalent techniques of optimum decoding which have the same computational complexity are specified. Modifications to the basic formulation in special cases are calculated and illustrated by means of examples. This paper corrects and extends, and unifies them with the results from the literature. In addition, a number of results from the literature are extended to the case c >; 1.

Beamforming techniques that employ Singular Value Decomposition (SVD) are commonly used in Multi-Input Multi-Output (MIMO) wireless communication systems. In the absence of channel coding, when a single symbol is transmitted, these systems achieve the full diversity order provided by the channel; whereas when multiple symbols are simultaneously transmitted, this property is lost. When channel coding is employed, full diversity order can be achieved. For example, when Bit-Interleaved Coded Modulation (BICM) is combined with this technique, full diversity order of NM in an M x N MIMO channel transmitting S parallel streams is possible, provided a condition on S and the BICM convolutional code rate is satisfied. In this paper, we present constellation precoded multiple beamforming which can achieve the full diversity order both with BICM-coded and uncoded SVD systems. We provide an analytical proof of this property. To reduce the computational complexity of Maximum Likelihood (ML) decoding in this system, we employ Sphere Decoding (SD). We report an SD technique that reduces the computational complexity beyond commonly used approaches to SD. This technique achieves several orders of magnitude reduction in computational complexity not only with respect to conventional ML decoding but also, with respect to conventional SD.

ABSTRACT In multiple‐input multiple‐output (MIMO) systems, sphere decoding (SD) can achieve a performance equivalent to a full‐search maximum likelihood decoding, with reduced complexity. Several researchers reported techniques that reduce the complexity of SD further. In this paper, a new technique that decreases the computational complexity of SD substantially, without sacrificing performance, is introduced. The reduction is accomplished by deconstructing the decoding metric to decrease the number of computations and by exploiting the structure of a lattice representation. Furthermore, an application of SD employing a proposed smart implementation with very low computational complexity is introduced. This application calculates the soft bit metrics of a bit‐interleaved convolutional‐coded MIMO system in an efficient manner. On the basis of the reduced complexity SD, the proposed smart implementation employs the initial radius acquired by zero‐forcing decision feedback equalization, which ensures no empty spheres. Other than that, a technique of a particular data structure is also incorporated to efficiently reduce the number of executions carried out by SD. Simulation results show that these approaches achieve substantial gains in terms of the computational complexity for both uncoded and coded MIMO systems. Copyright © 2011 John Wiley & Sons, Ltd.

The Golden Code is a full-rate full-diversity space-time code, which achieves maximum coding gain for Multiple-Input Multiple-Output (MIMO) systems with two transmit and two receive antennas. Since four information symbols taken from an M-QAM constellation are selected to construct one Golden Code codeword, a maximum likelihood decoder using sphere decoding has the worst-case complexity of O(M4), when the Channel State Information (CSI) is available at the receiver. Previously, this worst-case complexity was reduced to O(M2.5) without performance degradation. When the CSI is known by the transmitter as well as the receiver, beamforming techniques that employ singular value decomposition are commonly used in MIMO systems. In the absence of channel coding, when a single symbol is transmitted, these systems achieve the full diversity order provided by the channel. Whereas this property is lost when multiple symbols are simultaneously transmitted. However, uncoded multiple beamforming can achieve the full diversity order by adding a properly designed constellation precoder. For 2 × 2 Fully Precoded Multiple Beamforming (FPMB), the general worst-case decoding complexity is O(M). In this paper, Golden Coded Multiple Beamforming (GCMB) is proposed, which transmits the Golden Code through 2 × 2 multiple beamforming. GCMB achieves the full diversity order and its performance is similar to general MIMO systems using the Golden Code and FPMB, whereas the worst-case decoding complexity of O(√(M)) is much lower. The extension of GCMB to larger dimensions is also discussed.

The computational complexity of optimum decoding for an orthogonal space-time block code is quantified. Four equivalent techniques of optimum decoding which have the same computational complexity are specified. Modifications to the basic formulation in special cases are calculated and illustrated by means of examples.

Diversity analysis of a number of multiple-input multiple-output (MIMO) applications requires the calculation of the expectation of a function whose variables are the ordered multiple eigenvalues of the Wishart matrices. To solve this, we need the marginal pdf of an arbitrary subset of the ordered eigenvalues. The marginal pdf shown in the literature is useful in numerical analysis, but not beneficial to diversity analysis. In this paper, we derive an upper bound to the marginal pdf of the eigenvalues. The derivation is based on the multiple integration of the well-known joint pdf, which is very complicated due to the exponential factors of the joint pdf. We suggest an alternative function that provides simpler calculation of the multiple integration. As a result, the marginal pdf is shown to consist of a multivariate polynomial with a given degree. By applying the marginal pdf to the calculation of the expectation, the diversity order for a number of MIMO systems can be calculated. Simulation results that support the analysis are presented.

In this paper, we present the diversity order analysis of bit-interleaved coded multiple beamforming (BICMB) combined with the constellation precoding scheme. Multiple beamforming is realized by singular value decomposition of the channel matrix which is assumed to be perfectly known to the transmitter as well as the receiver. Previously, BICMB is known to have a diversity order bound related with the product of the code rate and the number of parallel subchannels, losing the full diversity order in some cases. In this paper, we show that BICMB combined with the constellation precoder and maximum likelihood detection achieves the full diversity order. We also provide simulation results that match the analysis.

In this paper, diversity analysis of bit-interleaved coded multiple beamforming (BICMB) is extended to the case of general spatial interleavers, removing a condition on their previously known design criteria. We provide a method to get diversity order, simplifying the calculation of pairwise error probability (PEP). By using the Singleton bound, we also show the maximum achievable diversity for given code rate and the number of subchannels.

We present and analyze the performance of constellation precoded beamforming. This multi-input multi-output transmission technique is based on the singular value decomposition of a channel matrix. In this work, the beamformer is precoded to improve its diversity performance. It was shown previously that while single beamforming achieves full diversity without channel coding, multiple beamforming results in diversity loss. In this paper, we show that a properly designed constellation precoder makes uncoded multiple beamforming achieve full diversity order. We also show that partially precoded multiple beamforming gets better diversity order than multiple beamforming without constellation precoder if the subchannels to be precoded are properly chosen. We propose several criteria to design the constellation precoder. Simulation results match the analysis, and show that precoded multiple beamforming actually outperforms single beamforming without precoding at the same system data rate while achieving full diversity order.

Cooperative relaying has been proposed as a way to create a virtual antenna array, hence providing spatial diversity in slow fading wireless environment. From the practical implementation perspective, one needs to design a relaying system that can achieve the maximum diversity order with a low system complexity. One promising candidate is to incorporate equal gain combining (EGC) technique into relaying systems and it is the main focus of this paper. In particular, the techniques of EGC and smart relaying are combined in the decode-and-forward (DF) relay processing method. It is shown that for a system with one relay and M phase-shift-keying (M-PSK) modulation, a maximum diversity order of 2m is achieved in Nakagami-m fading environment. With K relays, numerical results suggest that the diversity order is m(K + 1).

In this letter, we propose a low complexity maximum likelihood (ML) decoding algorithm for orthogonal spacetime block codes (OSTBCs) based on the real-valued lattice representation and QR decomposition.We show that for a system with rate r = K/T, where K is the number of transmitted symbols per T time slots, the proposed algorithm decomposes the original complex-valued system into a parallel system represented by 2K real-valued components, thus allowing for a simple and independent detection of the real and imaginary parts of each complex transmitted symbol. We further show that for square L-QAM constellations, the proposed algorithm reduces the decoding computational complexity from O(L) for conventional ML to O(radicL) without sacrificing the performance.

In this letter, we propose a low complexity Maximum Likelihood (ML) decoding algorithm for quasi-orthogonal space-time block codes (QOSTBCs) based on the real-valued lattice representation and QR decomposition. We show that for a system with rate r = ns/T, where nsis the number of transmitted symbols per T time slots; the proposed algorithm decomposes the original complex-valued system into a parallel system with ns2 times 2 real-valued components, thus allowing for a simple joint decoding of two real symbols. For a square QAM constellation with L points (L-QAM), this algorithm achieves full diversity by properly incorporating two-dimensional rotation using the optimal rotation angle and the same rotating matrix for any number of transmit antennas (N ges 4). We show that the complexity gain becomes greater when N or L becomes larger. The complexity of the proposed algorithm is shown to be linear with the number of transmitted symbols ns.

In this letter, we propose a reordering of the channel representation for Sphere Decoding (SD) where the real and imaginary parts of each jointly detected symbol are decoded independently. Making use of the proposed structure along with a scalar quantization technique, we reduce the decoding complexity substantially. We show that this approach achieves 85% reduction in the overall complexity compared to the conventional SD for a 2 times 2 system, and 92% reduction for the 4 times 4 and 6 times 6 cases at low SNR values, and almost 50% at high SNR, thus relaxing the requirements for hardware implementation.

We propose an analytical model for the performance analysis of the enhanced distributed channel access (EDCA) function of the IEEE 802.11e standard. The proposed discrete-time Markov chain (DTMC) model incorporates the main quality-of-service (QoS) features of 802.11e, namely, contention window (CW), arbitration interframe space (AIFS), and transmit opportunity (TXOP) differentiation. Due to its specific design, the proposed DTMC model jointly considers the state of the medium access control (MAC) layer buffer and the MAC differentiation mechanisms which facilitates a novel performance analysis framework at an arbitrary traffic load. Analytical and simulation results are compared to demonstrate the accuracy of the proposed approach for varying traffic load, EDCA parameters, and MAC layer buffer space.

We propose a multimedia capacity analysis framework for the Enhanced Distributed Channel Access (EDCA) function of the IEEE 802.11e standard. Our analysis shows that the multimedia capacity of the EDCA function for each Access Category (AC) can accurately be estimated by appropriately weighing the service time predictions of a saturation model over different number of active stations. We propose a simple and generic cycle time model to derive the service time in saturation which we employ in the calculation of an accurate station- and AC-specific queue utilization ratio. Based on the estimated queue utilization ratio, we design a simple model-based admission control scheme. We show that the proposed call admission control algorithm maintains satisfactory user-perceived quality for coexisting voice and video connections in an infrastructure Basic Service Set (BSS) and does not present over-or under-admission problems of previously proposed models in the literature.

This paper addresses the performance of bit-interleaved coded multiple beamforming (BICMB) [1], [2] with imperfect knowledge of beamforming vectors. Most studies for limited-rate channel state information at the transmitter (CSIT) assume that the precoding matrix has an invariance property under an arbitrary unitary transform. In BICMB, this property does not hold. On the other hand, the optimum precoder and detector for BICMB are invariant under a diagonal unitary transform. In order to design a limited-rate CSIT system for BICMB, we propose a new distortion measure optimum under this invariance. Based on this new distortion measure, we introduce a new set of centroids and employ the generalized Lloyd algorithm for codebook design. We provide simulation results demonstrating the performance improvement achieved with the proposed distortion measure and the codebook design for various receivers with linear detectors.We show that although these receivers have the same performance for perfect CSIT, their performance varies under imperfect CSIT.

The complexity of sphere decoding (SD) has been widely studied due to the importance of this algorithm in obtaining the optimal maximum likelihood (ML) performance with lower complexity. In this paper, we propose a proper tree search traversal technique that reduces the overall SD computational complexity without sacrificing the performance. We exploit the similarity among the complex symbols in a square QAM lattice representation and rewrite the squared norm ML metric in a simpler form allowing significant reduction of the number of operations required to decode the transmitted symbols. We also show that this approach achieves > 45% complexity gain for systems employing 4-QAM, and that this gain becomes bigger as the constellation size is larger.

In this paper, we propose a new decoding algorithm for quasiorthogonal space-time block codes (QOSTBCs) which achieves near maximum likelihood (ML) performance while substantially reducing the decoding complexity. We show that for a system with rate r = ns/T, where nsis the number of transmitted symbols per T time slots; the proposed algorithm decomposes the original complex-valued system into a parallel system with ns2times2 real-valued components, thus allowing for a simple decoding of one real symbol at a time. For a square QAM constellation with L points (L-QAM), this algorithm achieves full diversity by properly incorporating two-dimensional rotation using the optimal rotation angle and the same rotating matrix for any number of transmit antennas (Nges4). We show that the complexity of this algorithm is linear with the number of transmitted symbols ns, and is independent of the constellation size L.

In this paper, we propose a low complexity Maximum Likelihood (ML) decoding algorithm for orthogonal space-time block codes (OSTBCs) based on the real-valued lattice representation and QR decomposition. We show that for a system with rate r = K/T, where K is the number of transmitted symbols per T time slots; the proposed algorithm decomposes the original complex-valued system into a parallel system represented by 2 K real-valued components, thus allowing for a simple and independent detection of the real and imaginary parts of each complex transmitted symbol. We further show that for a system employing the well-known Alamouti OSTBC and 16-QAM modulation scheme, the new approach achieves >82% reduction in the overall complexity compared to conventional ML detection. Moreover, we show that for square L-QAM constellations, the proposed algorithm reduces the decoding computational complexity from O(L) for conventional ML to O(radicL) without sacrificing the performance.

We propose a novel and simple multimedia capacity analysis framework for the IEEE 802.11e Enhanced Distributed Channel Access (EDCA) function. We design a simple model- based admission control scheme which is based on the accurate queue utilization ratio we calculate by appropriately weighing the service time predictions of our previously developed saturation cycle time model for different number of active stations. We show that the proposed call admission control algorithm maintains satisfactory user-perceived quality for coexisting voice and video connections in an infrastructure Basic Service Set (BSS) and does not present over- or under-admission problems of previously proposed models in the literature.

In this paper, we propose a low complexity Maximum Likelihood (ML) decoding algorithm for quasi-orthogonal space-time block codes (QOSTBCs) based on the real-valued lattice representation and QR decomposition. We show that for a system with rate r = ns/T, where ns is the number of transmitted symbols per T time slots; the proposed algorithm decomposes the original complex-valued system into a parallel system with ns 2 x 2 real-valued components, thus allowing for a simple joint decoding of two real symbols. For a square QAM constellation with L points (L-QAM), this algorithm achieves full diversity by properly incorporating two-dimensional rotation using the optimal rotation angle and the same rotating matrix for any number of transmit antennas (N ges 4). We further show that for N = 8 and 16-QAM modulation scheme, the new approach achieves > 98% reduction in the overall complexity compared to conventional ML detection, and > 93% reduction compared to the most competitive reported algorithms in the literature. This complexity gain becomes greater when N or L becomes larger. We also show that the complexity of the proposed algorithm is linear with L and ns.

We present the unfairness problem between the uplink and the downlink flows in the IEEE 802.11e infrastructure basic service set (BSS) when the default settings of the enhanced distributed channel access (EDCA) parameters are used. First, we propose an analytical model to calculate the EDCA parameter set that ideally achieves a predetermined utilization ratio between uplink and downlink flows. Next, we design a simple and practical model-assisted measurement-based dynamic parameter adaptation algorithm. In the design of the proposed algorithm, we also consider the interactions of the bi-directional communication structure of transmission control protocol (TCP) with the medium access control (MAC) layer algorithm. Via simulations, we show that our solution provides short- and long-term fair access for uplink and downlink TCP and user datagram protocol (UDP) flows with different wired link propagation delays in a wired-wireless heterogeneous scenario. In the meantime, the quality-of-service (QoS) requirements of coexisting real-time flows can also be maintained.

We illustrate the transport layer unfairness problem in the IEEE 802.11 Wireless Local Area Networks (WLANs). We design a link layer access control block for the Access Point (AP) in order to provide fair Transmission Control Protocol (TCP) access in an 802.11 infrastructure Basic Service Set (BSS). The novel and simple idea of the proposed control block is employing a congestion control and filtering algorithm on TCP Acknowledgment (ACK) packets of uplink flows, thereby prioritizing the access of TCP data packets of downlink flows at the AP. We quantify the parameters of the proposed ACK congestion control and filtering algorithm based on the measured average downlink TCP data access rate. Via simulations, we show that the introduction of the proposed link layer access control block to the protocol stack enhances short- and long-term fairness in a wide range of scenarios.

We illustrate the transport layer unfairness problem in the IEEE 802.11 wireless local area networks (WLANs). We propose a novel and simple analytical model to calculate the per-flow congestion window limit that provides fair TCP access in a wired/wireless scenario where the wireless hop is an 802.11 link. The proposed analysis is unique in that it considers the effects of varying number of uplink and downlink TCP flows, varying round trip times (RTTs) of TCP connections, and the use of delayed TCP acknowledgment (ACK) mechanism. The comparison with simulation results validates the accuracy of the analytical estimations on the congestion window limit. When the TCP connections use the congestion window limits calculated from the proposed model, not only fair access can be provisioned, but also the channel can be utilized efficiently in a wide range of scenarios.

Sphere decoding (SD) is a low complexity maximum likelihood (ML) detection algorithm, which has been adapted for different linear channels in digital communications. The complexity of SD has been shown to be exponential in some cases, and polynomial in others and under certain assumptions. The sphere radius and the number of nodes visited throughout the tree traversal search are the decisive factors for the complexity of the algorithm. The radius problem has been addressed and treated widely in the literature. In this paper, we propose a new structure for SD, which drastically reduces the overall complexity. The complexity is measured in terms of the floating point operations per second (FLOPS) and the number of nodes visited throughout the algorithm's tree search. This reduction in complexity is due to the ability of decoding the real and imaginary parts of every jointly detected symbol independently of each other, making use of the new lattice representation. We further show by simulations that the new approach achieves 80% reduction in the overall complexity compared to the conventional SD for a 2times2 system, and almost 50% reduction for the 4times4 and 6times6 cases, thus relaxing the requirements for hardware implementation.

The Enhanced Distributed Channel Access (EDCA) function of IEEE 802.11e standard defines multiple Access Categories (AC) with AC-specific Contention Window (CW) sizes, Arbitration Interframe Space (AIFS) values, and Transmit Opportunity (TXOP) limits to support MAC-level Quality-of- Service (QoS). In this paper, we propose an analytical model for the EDCA function which incorporates an accurate CW, AIFS, and TXOP differentiation at any traffic load. The proposed model is also shown to capture the effect of MAC layer buffer size on the performance. Analytical and simulation results are compared to demonstrate the accuracy of the proposed approach for varying traffic loads, EDCA parameters, and MAC layer buffer space.

The recently ratified IEEE 802.11e standard defines the enhanced distributed channel access (EDCA) function for quality-of-service (QoS) provisioning in the wireless local area networks (WLANs). The EDCA uses carrier sense multiple access with collision avoidance (CSMA/CA) and slotted binary exponential backoff (BEB) mechanism. We present a simple mathematical analysis framework for the EDCA function. Our analysis considers the fact that the distributed random access systems exhibit cyclic behavior where each station successfully transmits a packet in a cycle. Our analysis shows that an AC-specific cycle time exists for the EDCA function. Validating the theoretical results via simulations, we show that the proposed analysis accurately captures EDCA saturation performance in terms of average throughput, medium access delay, and packet loss ratio. The cycle time analysis is a simple and insightful substitute for previously proposed more complex EDCA models.

The IEEE 802.11e standard has recently been ratified. This standard provides quality-of-service (QoS) guarantees in the medium access control (MAC) layer of 802.11 wireless local area networks (WLANs). In other words, the 802.11e standard updates the MAC layer of the former 802.11 standard for QoS. In particular, the enhanced distributed channel access (EDCA) function of 802.11e is an enhancement of the distributed coordination function (DCF) of 802.11. In this paper, we present an analytical model for EDCA in saturation mode. Our model incorporates an accurate contention window (CW) and arbitration interframe space (AIFS) differentiation. In addition, it provides virtual collision treatment for an access category (AC). Validating the theoretical results via simulations, we show that i) the proposed model accurately captures EDCA saturation throughput, and ii) this model is substantially more accurate than the models previously proposed in the literature.

Most of the deployed IEEE 802.11 wireless local area networks (WLANs) use infrastructure basic service set (BSS) in which an access point (AP) serves as a gateway between wired and wireless domains. We present the transport layer uplink unfairness problem that occurs due to uneven bandwidth sharing between upstream transmission control protocol (TCP) data and downstream TCP acknowledgment (ACK) packets. We propose ACK congestion control and filtering in the WLAN downlink to improve fairness between uplink flows. Through extensive simulations, we show that the proposed heuristic algorithms not only enhance both short- and long-term fairness but also increase channel utilization in the 802.11 WLAN. Another attractive feature for the proposed algorithms is the independence of the performance from variable TCP round trip times (RTTs).

In this paper, we investigate the performance of bit-interleaved coded multiple beamforming (BICMB). We provide interleaver design criteria such that BICMB achieves full spatial multiplexing of min( N, M) and full spatial diversity of NM with N transmit and M receive antennas over quasi-static Rayleigh flat fading channels. If the channel is frequency selective, then BICMB is combined with orthogonal frequency division multiplexing (OFDM) (BICMB-OFDM) in order to combat ISI caused by the frequency-selective channels. BICMB-OFDM achieves full spatial multiplexing of min(N, M), while maintaining full spatial and frequency diversity of NML for an NtimesM system over L-tap frequency-selective channels when an appropriate convolutional code is used. Both systems analyzed in this paper assume perfect channel state information both at the transmitter and the receiver. Simulation results show that, when the perfect channel state information assumption is satisfied, BICMB and BICMB-OFDM provide substantial performance or complexity gains when compared to other spatial multiplexing and diversity systems.

As his term as Editor-in-Chief (EIC) of the IEEE Transactions on Communications comes to an end with the December 2007 issue the out-going EIC would like to take this opportunity to give an update on the status of the Transactions over the last five years.

The IEEE 802.11e standard defines the Hybrid Coordination Function (HCF) which specifies the QoS algorithms in the MAC layer of the next generation 802.11 wireless LAN (WLAN). The HCF is composed of two access functions: A distributed contention-based channel access function (EDCA) providing prioritized QoS and a centralized polling-based channel access function (HCCA) providing parameterized QoS. In this paper, we address the efficient medium access control (MAC) of the mostly variable bit rate (VBR) and bursty multimedia traffic in HCCA and propose a novel application-aware adaptive 802.11e QoS scheduler. The proposed scheduler coordinates the resource allocation of the associated flows using adaptive service intervals, transmit opportunities, and polling order. The service schedule is based not only on the traffic classification done according to the packet size, interval distribution, and the direction but also on the instantaneous buffered traffic conditions. The proposed scheduler is shown to have significant performance gains in terms of multimedia traffic channel utilization, packet loss ratio, delay, and jitter when compared with the other schemes.

Bit interleaved coded multiple beamforming (BICMB) was previously designed to achieve full spatial multiplexing of min(N, M) and full spatial diversity of NM for N transmit and M receive antennas over flat fading channels. Furthermore, BICMB when combined with orthogonal frequency division multiplexing (OFDM) achieves full spatial multiplexing and full diversity order of NML over L-tap frequency-selective channels. BICMB requires full channel state information (CSI) both at the transmitter and receiver, however it uses uniform power and rate over the parallel channels established by multiple beamforming. In this paper, our main goal is to investigate the performance of the previously analyzed BICMB system via adaptive modulation and coding (AMC) to further utilize CSI and to improve system throughput performance. Hence, we name the new systems as adaptive BICMB (ABICMB) over flat fading, and ABICMB-OFDM over frequency-selective channels. Simulation results show that adaptive system achieves 4-13 dB performance gain compared to non-adaptive case depending on the antenna configuration and environment. Systems analyzed require perfect CSI both at the transmitter and receiver, which may be difficult to obtain in a practical scenario. However, high performance gains achieved makes it worthwhile to study the performance of the proposed systems, leaving room for significant gain with limited feedback

It is well-known that using multiple antennas provides a substantial capacity and diversity increase for wireless communication systems. A multi-input multi-output (MIMO) technique that utilizes the channel knowledge both at the transmitter and the receiver is known as beamforming. Beamforming separates a MIMO channel into parallel subchannels. It was previously shown that uncoded beamforming achieves a diversity order of (N - S + 1)(M - S + 1) if S symbols are transmitted simultaneously for N transmit and M receive antennas. Hence, there is a significant drop in the diversity order (and performance) of the system with increased spatial multiplexing. In this paper, we introduce bit interleaved coded multiple beamforming and name the system BICMB. We provide interleaver design criteria such that the resulting system achieves full spatial multiplexing of min(N, M) and full spatial diversity of NM. Simulation results show that BICMB, due to its ability of maintaining the maximum diversity order even at full spatial multiplexing, provides substantial performance gain when compared to the best spatial multiplexing systems

Orthogonal frequency-division multiplexing (OFDM) is known as an efficient technique to combat frequency-selective channels. In this paper, we show that the combination of bit-interleaved coded modulation (BICM) and OFDM achieves the full frequency diversity offered by a frequency-selective channel with any kind of power delay profile (PDP), conditioned on the minimum Hamming distance dfree of the convolutional code. This system has a simple Viterbi decoder with a modified metric. We then show that by combining such a system with space-time block coding (STBC), one can achieve the full space and frequency diversity of a frequency-selective channel with N transmit and M receive antennas. BICM-STBC-OFDM achieves the maximum diversity order of NML over L-tap frequency-selective channels regardless of the PDP of the channel. This latter system also has a simple Viterbi decoder with a properly modified metric. We verify our analytical results via simulations, including channels employed in the IEEE 802.11 standards

In this letter, we study two techniques, known as single and multiple beamforming, to exploit the perfect channel state information (CSI) available both at the transmitter and the receiver of a multiantenna wireless system. Assuming N and M are the number of antennas at the transmitter and the receiver, respectively, we show that single beamforming (transmission of a single symbol from all transmit antennas at the same time, employing the best subchannel) can achieve the maximum spatial diversity order in the channel (NM). We extend our analytical results to multiple beamforming (transmission of S symbols simultaneously, S>1) and calculate that the diversity order achievable for this system is (N-S+1)(M-S+1).

In this paper we show and quantify both analytically and via simulations that the use of channel knowledge at the transmitter, the technique known as beamforming, achieves the maximum diversity in space when the best eigenmode is used (single beamforming). Furthermore, we investigate beamforming in conjunction with next generation wireless local area networks (WLAN). It is known that the widely used technique in WLAN, bit interleaved coded modulation (BICM) with orthogonal frequency division multiplexing (OFDM), can achieve the maximum frequency diversity order that is inherited in the channel. We show that the combination of BICM, single beamforming, and OFDM also leads to the maximum diversity order in space and frequency domains. In other words, for systems with N transmit and M receive antennas, BICM beamforming-OFDM (BBO) can achieve a diversity order of NML over L-tap frequency selective channels by using an appropriate convolutional code. In addition to having a substantial diversity order, simulation results show that beamforming and BBO introduce significant coding gains when compared to other systems based on space time block codes (STBC) with the same diversity order.

It has been shown by Zehavi that the performance of coded modulation can be improved over a Rayleigh fading channel by bit-wise interleaving at the encoder output, and by using an appropriate soft-decision metric for a Viterbi decoder at the receiver. Caire et al presented the details of the theory behind bit-interleaved coded modulation (BICM). In this paper we show that for Gray encoded M-ary quadrature amplitude modulation (QAM) systems, the bit metrics of BICM can be further simplified. In QAM systems, the maximum likelihood (ML) detector for BICM uses the minimum distance between the received symbol and M/2 constellation points on the complex plane as soft-decision metrics. We show that soft-decision bit metrics for the ML decoder can be further simplified to the minimum distance between the received symbol and /spl radic/M/2 constellation points on the real line /spl Ropf/1. This reduces the number of calculations needed for each bit metric substantially, and therefore reduces the complexity of the decoder without compromising the performance. Simulation results for single carrier modulation (SCM), and multi-carrier modulation (MCM) systems over additive white Gaussian noise (AWGN) and Rayleigh fading channels agree with our findings. In addition, we tie this result to the decoding methods for bit interleaved convolutional code standards used in industry.

In this paper, we propose an automatic repeat request (ARQ) technique for enhancing the reliability and throughput of broadband wireless Internet access applications, and provide examples of its performance. ARQ is a method for making a link and flow more robust to errors. Other than enhancing link quality, ARQ provides early error recovery, so that the transport layer does not lower the data throughput. The cost of these enhancements is additional overhead. In this work, we have extensively analyzed our proposed ARQ scheme along with realistic wireless channel and protocol stack models. We have quantified the performance gains of our ARQ scheme as reduced end-to-end delay, increased throughput, SNR gain and fade margin, also providing the overhead costs. We have proposed and analyzed enhanced features namely, segmentation and reassembly (SAR) and bitmap compression. We showed that these features reduce the overhead, the extent of which depends on the nature of the errors.

In this paper we describe a coding, modulation, and spatial processing technique for fixed broadband wireless Internet access applications and provide examples of its performance. This technique is built on orthogonal frequency division multiplexing (OFDM) and is known as vector OFDM (VOFDM). We compare VOFDM with conventional single carrier modulation (SCM), and show that it provides substantial performance improvements over SCM.

This paper concerns connection provisioning for optical networks employing wavelength division multiplexing. A heuristic algorithm is developed and numerically studied for routing and wavelength assignment of a set of static connection requests. The algorithm runs much faster than the optimum solution of this problem. An adaptation of the algorithm is proposed to design restorable networks which can handle a specified set of failures. The proposed algorithm is based on taking all failures into consideration simultaneously, and performs better than developing independent designs for each failure.

This paper concerns connection provisioning for optical networks employing wavelength division multiplexing. A heuristic algorithm is developed and numerically studied for routing and wavelength assignment of a set of static connection requests. The algorithm runs much faster than the optimum solution of this problem. An adaptation of the algorithm is proposed to design restorable networks which can handle a specified set of failures. The proposed algorithm is based on taking all failures into consideration simultaneously, and performs better than developing independent designs for each failure.

Wavelength division multiplexed point-to-point transport is becoming commonplace in wide area networks. With the expectation that the next step is end-to-end networking of wavelengths (in the optical domain without conversion to electronics), there is a need for new design techniques, a new understanding of the performance issues, and a new performance evaluation methodology in such networks. This paper describes approaches to that end, summarizes research results, and points to open problems.

Modems designed for the public switched telephone network (PSTN) have conventionally been based on modeling assumptions which view the PSTN connection as an essentially analog medium. However, as the PSTN evolves toward all digital transport and switching, and particularly as major traffic sources, such as Internet service providers, increasingly have direct digital connections to the PSTN, it is appropriate to revisit the model assumptions. Recently, several modem and chipset manufacturers have announced "56 K" modems based on an emerging system paradigm in which one user (a residential Internet subscriber) has an analog connection to the PSTN, and the other (an Internet service provider) has a digital one. ITU-T is expected to finalize details of a corresponding recommendation in 1998. With this configuration, modem designs based on signaling with the /spl mu/-law alphabet become feasible, and the conventional Shannon limit disappears as the quantization noise is avoided. Thus, the conventional Shannon limit of about 36 kb/s can be beaten, and it is possible to approach the digital transmission rate of 64 kb/s. Modems employing this general approach have become known as /spl mu/-law or pulse-code modulation (PCM) modems. In this paper we present a signaling technique and the sampling theory based on this technique, display the structure and operating principles of a PCM modem equalizer, and show how this equalizer problem can be cast in the language of multiinput-multioutput (MIMO) system theory.

Wavelength-division multiplexing (WDM) technology is emerging as the transmission and switching mechanism for future optical mesh networks. In these networks it is desired that a wavelength can be routed without electrical conversions. Two technologies are possible for this purpose: wavelength selective cross-connects (WSXC) and wavelength interchanging cross-connects (WIXC), which involve wavelength conversion. It is believed that wavelength converters may improve the blocking performance, but there is a mix of results in the literature on the amount of this performance enhancement. We use two metrics to quantify the wavelength conversion gain: the reduction in blocking probability and the increase in maximum utilization, compared to a network without converters. We study the effects of wavelength routing and selection algorithms on these measures for mesh networks. We use the overflow model to analyze the blocking probability for wavelength-selective (WS) mesh networks using the first-fit wavelength assignment algorithm. We propose a dynamic routing and wavelength selection algorithm, the least-loaded routing (LLR) algorithm, which jointly selects the least-loaded route-wavelength pair. In networks both with and without wavelength converters the LLR algorithm achieves much better blocking performance compared to the fixed shortest path routing algorithm. The LLR produces larger wavelength conversion gains; however, these large gains are not realized in sufficiently wide utilization regions and are diminished with the increased number of fibers.

The performance of error control based on forward error correction (FEC) for MPEG-2 video transmission in an indoor wireless ATM LAN is studied. A multipath fading model is used to investigate the effect of errors on video transport. Combined source and channel coding techniques that employ single layer and scalable MPEG-2 coding to combat channel errors are compared. Simulation results indicate that FEC-based error control in combination with 2-layer video coding techniques can lead to acceptable quality for indoor wireless ATM video.

We quantify the benefits of wavelength interchanging in terms of the required capacity and the blocking probability. Most of the previous work on this subject assumed single-fiber links; we study networks with multiple fibers, extend Barry's (see Proceedings of IEEE INFOCOM '95, vol.2, p.402, 1995) probability analysis to the multifiber case, and compare those results with fiber gain calculations in exemplary networks. We quantify the gain in the blocking probability when wavelength interchanging cross-connects are used compared to the case when wavelength-selective cross-connects are used, and show that this gain decreases rapidly with an increasing number of fibers on a link. The formulation captures the effects of the topology and the traffic on this gain. Calculations and simulations are performed which verify the analytical results.

Wireless ATM LANs have the potential to support multi-Mb/s bandwidths to mobile users with guaranteed quality of service. However, the lossy nature of the wireless medium will pose problems for loss-sensitive applications. Techniques to minimize the effect of these losses will therefore be required. In this paper, we examine the use of combined source and channel coding for MPEG video transport in a wireless ATM environment. Although forward error correction (FEC) provides protection against channel bit errors, the bandwidth overhead can become a significant drawback in a fixed bandwidth scenario. Additional protection against losses can be realized by using two-layer video coding. In this work, we compare the performance of a 1-layer main profile and 2-layer data partitioning and SNR scalable MPEG-2 encoders in a system with random channel errors and forward error correction. The results indicate that if the channel bit error rate is known an optimum FEC level can be chosen for the 1-layer case; However, at this fixed FEC level, if a critical bit error rate is exceeded then video quality degrades dramatically. The 2-layer cases appear to lead to more graceful degradation in quality at this critical bit error rates. In particular, SNR scalability may lead to better video quality over a larger range of bit error rates than a 1-layer approach.

pg 100 diversity coding, as introduced in Ayanoglu et al. (1990), is a method of protection against failures in a communication network or a storage system, which is based on introducing a digital error-correcting code across independent links. This technique makes efficient use of the extra network capacity needed for coding and has the additional advantages of being nearly instantaneous, not requiring a feedback channel, rerouting, or resynchronization. In high-speed (multi Gbps) networks, digital coding will be difficult to implement, and the purpose of the present paper is to demonstrate how diversity coding may be implemented in the analog domain using the discrete fourier transform (DFT). In particular, the authors show that the DFT is a continuous-amplitude maximum-distance separable code over the field of complex numbers when the transform kernel is a prime root of unity. This code can be used to generate self-healing or fault-tolerant communication networks for continuous- or discrete-amplitude signals, as long as continuous-amplitude parity channels are available. The authors describe electrical and optoelectronic implementations, and a signal estimation approach to combat channel noise and thereby improve the performance of the analog diversity coding system. The most important advantage of this technique is in greatly simplifying the encoders and decoders of diversity coding systems for high-speed networks, such as fiber-optic wavelength division multiplexed networks. Application of analog diversity coding to systems with analog sources, such as telemetry systems is also possible.>

The authors analyze a threshold priority access statistical multiplexing technique with a Markov modulated fluid source model for variable-bit-rate (VBR) video traffic, and an exact constant-bit-rate (CBR) model. Exact expressions for loss probabilities and delay are obtained. Based on these quality-of-service parameters, buffer size requirements of the statistical multiplexer are obtained, and the performance of a fixed threshold algorithm is compared with that of a rate-based threshold algorithm. The results provide the tradeoffs between buffer size and processing complexity for control. The exact analytical results can be used to calculate buffer size requirements.>

A channel coding approach called diversity coding is introduced for self-healing and fault-tolerance in digital communication networks for nearly instantaneous recovery from link failures. To achieve this goal, the problem of link failures is treated as an erasure channel problem. Implementation details of this technique in existing and future communication networks are discussed.>

Transcoding algorithms that eliminate distortion accumulation due to tandem transcodings between memoryless, finite-state, and predictive vector quantization and pulse code modulation (PCM) are presented. The algorithms can be implemented using table lookups for memoryless and finite-state vector quantization, whereas predictive vector quantization requires online calculations. Computer simulations indicate a 6 dB improvement in the case of 16 kb/s predictive vector quantizers, 48 kb/s PCM, and four tandems for speech.>

Techniques are presented that can cancel ghosts in received analog TV (for IDTV, EDTV, and HDTV) signals. The authors utilize the fact that there are short periods of time without the analog signal (the horizontal flyback interval between the lines) to periodically cleanse a finite impulse response (FIR) or an infinite impulse response (IIR) equalizer. This line-by-line processing (cleansing) overcomes the limitations of standard equalizers to allow adequate suppression of ghosts, even with nulls in the spectrum.>

Techniques that can cancel ghosts in received analog TV (for improved-definition TV, extended-definition TV, and high-definition TV) signals are presented. The fact that there are short periods of time without the analog signal (the horizontal flyback interval between the lines) is utilized to periodically cleanse a finite impulse response (FIR) or an infinite impulse response (IIR) equalizer. This line-by-line processing (cleansing) overcomes the limitation of standard equalizers to allow for 40-50 dB of suppression of ghosts, even with nulls in the spectrum, as long as the ghost delay is less than the period of time without the analog signal. Furthermore, by using time inversion in combination with line-by-line processing, the stability problem of the conventional IIR equalizer can be eliminated. It is shown that it may be possible to implement this IIR equalizer on a single digital integrated circuit. Alternatively, an FIR equalizer can be used; although it requires multiple chips (i.e. more taps), it can acquire and adapt to the ghosted channel more rapidly than an IIR equalizer.>

An error-control-based approach, called diversity coding, that provides nearly instantaneous self-healing digital communication networks is presented. This is achieved by constructing an error-correcting code across logically independent channels and by treating link failures within the framework of an erasure channel model. Diversity coding is more efficient than previous approaches to self-healing communication networks since it is nearly instantaneous, is transparent to the end user, minimizes the required extra capacity, and does not need rerouting, resynchronization, or a feedback channel. It is applicable to both circuit-switched and packet-switched networks and to a wide variety of network topologies. Diversity coding can be extended to provide protection from short-duration environmental disruptions, such as multipath fading in radio networks and polarization dispersion in fiber-optic networks, or, in conjunction with previous error detection schemes, to provide forward error correction for random and burst errors.>

Problems in optimal multidimensional quantization of sources corrupted by noise are addressed. Expressions for the optimum quantizer values and the optimum quantization rule for the weighted squared error distortion measure are found and calculated for the Gaussian signal in additive independent Gaussian noise problem. Some properties of the optimum quantizer, and its relations with the optimal estimator for the general problem, are derived.>

Problems in optimal multidimensional quantization of noise-corrupted sources are addressed. The optimal quantization rule, properties of the optimal quantizer, and its relations with the optimal estimator for the weighted squared error distortion measure are derived. An equivalent quantization technique is developed and the optimum codewords are calculated for the Gaussian signal in the additive independent Gaussian noise problem. Simulation results are presented.>

The generalized Lloyd algorithm is applied to the design of joint source and channel trellis waveform coders to encode discrete-time continuous-amplitude stationary and ergodic sources operating over discrete memoryless noisy channels. Experimental results are provided for independent and autoregressive Gaussian sources, binary symmetric channels, and absolute error and squared error distortion measures. Performance of the joint codes is compared with the tandem combination of a trellis source code and a trellis channel code on the independent Gaussian source using the squared error distortion measure operating over an additive white Gaussian noise channel. It is observed that the jointly optimized codes achieve performance close to or better than that of separately optimized tandem codes of the same constraint length. Performance improvement via a predictive joint source and channel trellis code is demonstrated for the autoregressive Gaussian source using the squared error distortion measure.

Trellis source codes consist of a finite-state machine decoder and a trellis search algorithm, such as the Viterbi algorithm, as the encoder. The encoder experiments with a local copy of the decoder and determines the best channel path map in the sense that it will yield the smallest average distortion between the source sequence and the reproduction sequence given the codebook. In this paper we present a coding system and a design algorithm for predictive trellis coding. Results obtained via simulation are compared for trellis and predictive trellis codes designed for first-order autoregressive sources with Gaussian and Laplacian innovations and for sampled speech. On a random source which models speech, simulation results of the predictive and nonpredictive trellis codes designed by the generalized Lloyd algorithm and those obtained by other researchers are compared. Issues related to computational complexity, the effects of initial codebook selection, training sequence segmentation, search length, channel errors, and algorithm convergence are addressed.