The registered version of this article, first published in IEEE Internet of Things Journal, is available online at the publisher's website: Institute of Electrical and Electronics Engineers, https://doi.org/10.1109/JIOT.2026.3696106

Deep learning has achieved notable progress in vision and language domains, yet its application to tabular data remains constrained by the lack of inherent spatial structure and the profound challenge of extracting transparent explanations. This work introduces a Hybrid Neural Network (HyNN) framework focused on robust interpretability, combining symbolic reasoning on original tabular features with spatial representations derived from synthetic images. Tabular data are transformed into TINTO-generated synthetic images, enabling a CNN branch to exploit local feature interactions while an MLP branch captures global dependencies. To decode this hybrid decision process, we propose PermGrad: a unified attribution pipeline integrating permutation-based importance (MLP), Grad-CAM saliency (CNN), and a macroscopic pruning-based weighting scheme that scales feature relevance by each branch’s empirical contribution. Experiments across five heterogeneous datasets reveal that the HyNN achieves predictive performance highly competitive with SOTA deep tabular architectures (e.g., FT-Transformer). Crucially, PermGrad exposes the internal behavioural dynamics of the hybrid model: the CNN leverages the spatial layout to dominate high-dimensional contexts, while the MLP acts as a vital ”symbolic anchor” during training. Even when its explicit inference-time attribution is negligible, the auxiliary MLP stabilizes gradient flow, prevents optimization collapse, and yields consistently lower test losses. Overall, PermGrad provides a unified, scalable, and class-aware view of feature importance, demonstrating that the strategic use of synthetic images in hybrid architectures can bridge the gap between competitive predictive power and transparent, modality-aware explainability.

In this paper, we conduct an energy consumption analysis of a broadcast access mechanism over IEEE 802.11 networks. By assuming non-saturated conditions, we provide a statistical model of the energy consumed during one cycle period, defined as the duration between two successful broadcast transmissions. The model includes a characterization of the medium access transmission mechanism. Our results show that the broadcast protocol is a good candidate to be included in future amendments of the IEEE 802.11 standard.

To ensure unmanned aerial vehicle (UAV) swarm safety in areas lacking satellite navigation, accurate localization must be achieved through alternative methods such as relative positioning and clustering. A key challenge involves balancing higher localization accuracy with lower computational complexity, especially in urban environments. Additionally, maintaining up-to-date data transmission to ground stations is critical for real-time localization in dynamic scenarios.This paper proposes a cooperative relative localization scheme based on clustering for UAV swarms. The proposal employs the degree centrality metric to form clusters and connectivity information for cluster head selections, and balances intra-cluster cooperation with inter-cluster packet loss, ensuring the freshness of transmitted data.A first evaluation of the proposed scheme allows us to venture that it will achieve better performance (in terms of mean square error) compared to state-of-the-art techniques for relative localization of UAV swarms.

Virtual Learning Environments (VLE) are present in all educational modalities and levels, but their adoption in primary education remains limited. This article evaluates the acceptance of MundoCRA, a VLE that enriches traditional classrooms through a virtual world where students perform educational tasks in an engaging way. Gamified activities with rewards, points, and certificates promote motivation through quizzes, challenges, and collaborative projects. MundoCRA also includes a management module that allows teachers to create and monitor tasks easily. The evaluation, based on an extended Technology Acceptance Model (TAM), considers constructs such as perceived enjoyment, satisfaction, and gender differences. A survey of 20 teachers and 30 students from the Rural Grouped School (CRA) Sierra de Alcaraz (Spain) revealed that MundoCRA increased motivation, satisfaction, and enjoyment. Furthermore, structural equation modeling showed that perceived enjoyment and ease of use significantly influenced positive attitudes toward the platform, explaining over 70% of the variance in user attitudes.

With the growing demand for video applications, ensuring reliable broadcast services in Wireless Local Area Networks (WLANs) has become increasingly critical. Furthermore, digital video services are highly sensitive to network impairments due to timing constraints and video frame dependencies. In this paper, we consider an IEEE 802.11bc network where the Access Point (AP) provides both unicast and broadcast services, with associated stations generating unicast traffic at varying rates. We propose a Wireless Broadcast Access Protocol (WBAP) that supports multiple channel access for both unicast and broadcast traffic. We develop a Markov model to evaluate the performance of WBAP under both saturated and unsaturated conditions. The analytical results are validated using a custom-made MATLAB simulator. Our findings show that the proposed WBAP effectively manages the coexistence of unicast and broadcast traffic, significantly enhancing network performance and reliability, particularly under high-load traffic conditions.

In recent years, the development of Natural Language Processing (NLP) text-to-face encoders and Generative Adversarial Networks (GANs) has enabled the synthesis and generation of facial images from textual description. However, most encoders have been developed for the English language. This work presents the first study of three text-to-face encoders, namely, the RoBERTa pre-trained model and the Sent2Vec and RoBERTa models, trained with the CelebA dataset in Spanish. It then introduces customised and fine-tuned conditional Deep Convolutional Generative Adversarial Networks (cDCGANs) trained with the CelebA dataset for text-to-face generation in Spanish. To validate the results obtained, a qualitative evaluation was carried out with a visual analysis and a quantitative evaluation based on the IS, FID and LPIPS metrics. Our findings show promising results with respect to the literature, improving the numerical metrics of FID and LPIPS by 5% and 37%, respectively. Our results also show, through a quantitative–qualitative comparison of the cDCGAN training epochs, that the IS metric is not a reliable objective metric to be considered in the evaluation of similar works.

The widespread deployment of sensors interconnected by wireless links and the management and exploitation of the data collected have given rise to the Internet of Things (IoT) concept. In this article, we undertake the design and implementation of a wireless multisensor platform following the fog computing paradigm. Our main contributions are the integration of various off-the-shelf sensors smartly packaged into an air-flow module and the evaluation of the communications services offered implemented on top of two low-power radio communications technologies. Our study is complemented by evaluating the communications services over a wired link. Our results show the superiority of LoRaWAN over ZigBee in terms of power consumption despite its slightly higher computational requirements and an estimation of the gap between the resource usage of the wired link and the two wireless radio technologies.

To support the increasing demand for wireless services with efficient data rates, the IEEE 802.11be Extremely High Throughput (EHT) is being developed. This paper considers an IEEE 802.11be platform to support multiclass traffic characterized by different performance requirements. We focus on the study of the multi-link operation (MLO) being introduced by the standard. Therefore, we develop a Markov Decision Process (MDP) model to control the routing of arriving packets based on the resource requirements of the various traffic streams. We show that there exists a unique optimal stationary policy for the routing control of packets that minimizes the overall cost of the queueing system. The optimal routing control significantly decreases the Mean Sojourn Time (MST) of packets with lower load demand without interfering with the service performance of the highest resource requirements traffic.

The study of artificial learning processes in the area of computer vision context has mainly focused on achieving a fixed output target rather than on identifying the underlying processes as a means to develop solutions capable of performing as good as or better than the human brain. This work reviews the well-known segmentation efforts in computer vision. However, our primary focus is on the quantitative evaluation of the amount of contextual information provided to the neural network. In particular, the information used to mimic the tacit information that a human is capable of using, like a sense of unambiguous order and the capability of improving its estimation by complementing already learned information. Our results show that, after a set of pre and post-processing methods applied to both the training data and the neural network architecture, the predictions made were drastically closer to the expected output in comparison to the cases where no contextual additions were provided. Our results provide evidence that learning systems strongly rely on contextual information for the identification task process.

Abstract Wireless technologies are continuously evolving, including features such as the extension to mid- and long-range communications and the support of an increasing number of devices. However, longer ranges increase the probability of suffering from hidden terminal issues. In the particular case of Wireless Local Area Networks (WLANs), the use of Quality of Service (QoS) mechanisms introduced in IEEE 802.11e compromises scalability, exacerbates the hidden node problem, and creates congestion as the number of users and the variety of services in the network grow. In this context, this paper presents a configurable Colored Petri Net (CPN) model for the IEEE 802.11e protocol with the aim of analyzing the QoS support in mid- and long-range WLANs The CPN model covers the behavior of the protocol in the presence of hidden nodes to examine the performance of the RTS/CTS exchange in scenarios where the QoS differentiation may involve massive collision chains and high delays. Our CPN model sets the basis for further exploring the performance of the various mechanisms defined by the IEEE 802.11 standard. We then use this CPN model to provide a comprehensive study of the effectiveness of this protocol by using the simulation and monitoring capabilities of CPN Tools.

Abstract ZigBee is the primary standard solution for wireless sensor networks, implementing the Ad hoc On‐Demand Distance Vector (AODV) protocol in the network layer and supported by the standard IEEE 802.15.4. This study is focused on mesh topologies and the critical problems encountered when AODV is executed in conjunction with the Carrier Sense Multiple Access with Collision Avoidance protocol. These problems are mainly related to the packet overhead required to carry out route creation. To perform preliminary experiments to be able to implement AODV in a real network, a new metric is proposed herein. This metric uses fuzzy logic to help in the decision‐making process. The objective of the fuzzy routine is to determine, during the route‐discovery process, the best node to forward request/reply packets, with the aim of reducing packet overhead and energy consumption. Moreover, minor changes are also added to the discovery procedure of AODV to improve the performance of the route‐creation process. This intelligent version of AODV has provided promising experimental results, greatly reducing the number of packets required, with the consequent energy saving while selecting the best nodes to be part of the routes.

Video communications require the use of networks capable of optimally allocating resources and adapting dynamically themselves to the changing operating conditions. When a video stream has to be delivered to various receivers, the appropriate service to accomplish the delivery process is the multicast communication service. In the particular case of the widely used IEEE 802.11 Wireless LANs (WLANs), the multicast service is simply specified as an unreliable broadcasting service, i.e., it does not include any error recovery mechanism. The absence of feedback from the receivers does not only prevent the sender from taking proper action to recover the corrupted packets, but it also makes unfeasible the implementation of a channel rate adaptation mechanism as the one used by the IEEE 802.11 unicast service. The absence of such mechanism may render the multicast service useless if the channel conditions are so that it makes unreliable the delivery process of the multicast traffic. Even though the IEEE has been working on an amendment to the standard specifying the operation of various error recovery mechanisms for the multicast service, these services have not been designed to support any rate adaptation mechanism. In this paper, we present a novel auto rate selection multicast mechanism, whose design is based on the collision prevention to increase the reliability. We have developed QoS and QoE evaluations via simulation, demonstrating that our proposal outperforms the video quality perceived by the end-user.

With the deployment of Audio-Video (AV) services and the flexibility offered by wireless networks, there is more and more an increasing interest on defining more reliable multicast mechanisms. Towards this end, the IEEE Task Group aa (TGaa) is currently working on the definition of the IEEE802.11aa amendment. The standardization efforts in this area aim to develop efficient QoS mechanisms by keeping in mind that any new mechanism should: 1) properly interoperate with the already existing unicast mechanisms; and 2) provide the robustness required by AV applications. In order to meet the former, the efforts are focusing on amending the existing multicast standard while to meet the latter there is a need of better understanding the stringent requirements of AV streaming applications. In this work, after reviewing the amendments introduced by the IEEE 802.11aa, we carried out an evaluation of the proposed multicast mechanisms when supporting video streaming applications. The main contribution of this work focuses therefore on evaluating the QoS parameters that the TGaa has taken into account to define the proposed mechanisms.

During the past few years, the development of video services over wireless networks has been driven by Quality of Service (QoS) parameters and service level agreements. Even though QoS parameters, such as packet losses, jitter and delays are good indicators of the service provided by the underlying network services to the applications, they fall short on providing a better measure of the Quality of Experience (QoE) to the end-user. In order to gain a better understanding of the benefits in terms of the QoE guarantees, it is essential to evaluate the effectiveness of the enabling protocol mechanisms. In this paper, we introduce a novel multicast service for the IEEE 802.11 WLAN standard. The proposed service combines a novel MAC scheme and layered video, guaranteeing a minimum acceptable video quality to all the members of the multicast group through the base layer. To avoid penalizing the users capable of receiving data at higher rates and the performance of an IEEE 802.11 WLAN, the enhancement layers are transmitted using a higher transmission rate. We show that by properly coupling the video and multicast control mechanisms, the QoS as well as the QoE requirements of the end-user can be guaranteed. The performance of the proposed system architecture is extensively investigated by simulations and through numerical evaluation of relevant QoS and QoE metrics. Furthermore, our results show that the real-time video streaming quality of the total system can be greatly improved while keeping downward compatibility with the IEEE 802.11 multicast mechanism.

Recent advances in communication networks have spurred interest on the definition of efficient self-configuration protocols. Such mechanisms are particularly of interest on the rapid deployment of wireless sensor networks whose nodes are characterized by their low computational and energy resources according to the green networking principles. In this paper, we introduce an adaptable fuzzy-logic assisted configuration mechanism for ZigBee networks; one of the most promising standards in the area of wireless sensor networks. Our results show that the use of fuzzy-logic principles can significantly improve the operation of tree-based routing mechanisms by reducing the set-up network time and the number of router nodes in the network, thus allowing a reduction in the energy consumed by nodes.

ZigBee in conjunction with IEEE 802.15.4 is considered the de facto standard for WSNs. The routing protocol for ZigBee mesh networks is AODV, an on-demand routing protocol which has been extensively used. However, it presents critical problems when it is used in large dense networks, mainly related to the high number of messages necessary to perform routing tasks, which causes a very high overhead. The high number of packet collisions, data loss, and its implicit energy expenditure, makes AODV unsuitable for certain applications that require efficient, fast and reliable communications. This study proposes the use of a novel fuzzy-logic-based metric to be used in the decision making process of AODV. This metric evaluates important node features during route creation and selects, hop by hop, the best option in order to obtain energy and delay efficient routes. The experiments with the improved AODV, AODV-FL show a reduction in the communication delay, number of packets and overhead, improving route efficiency and reducing packet overload.

Based on our previous efforts, we introduce in this letter a high-efficient MPEG-2 to H.264 transcoder for the baseline profile in the spatial domain. Machine learning tools are used to exploit the correlation between the macroblock (MB) decision of the H.264 video format and the distribution of the motion compensated residual in MPEG-2. Moreover, a dynamic motion estimation technique is also proposed to further speed-up the decision process. Finally, we go a step further on our previous research efforts by combining the two aforementioned speed-up approaches. Our simulation results over more than 40 sequences at common intermediate format and quarter common intermediate format resolutions show that our proposal outperforms the MB mode selection of the rate-distortion optimization option of the H.264 encoder process by reducing the computational requirements by up to 90%, while maintaining the same coding efficiency. Finally, we conduct a comparative study of our approach with the most relevant fast inter-prediction methods for MPEG-2 to H.264 transcoder recently reported in the literature.

The deployment of large-scale wireless sensor networks presents major challenges whose solution requires the design and development of robust and time-efficient protocols. In this work, we focus on the design of an efficient and robust protocol aiming to rapidly configure highly populated networks. Our main effort goes towards the definition of a fast collision resolution protocol for conveying the control packets for configuring the network. Through extensive simulation work we show that the proposed mechanism is capable of quickly resolving the conflicts arising during the set-up phase.

The use of Broadband Wireless Access (BWA) technology is increasing due to the use of Internet and multimedia applications with strict requirements of end-to-end delay and jitter, through wireless devices. The IEEE 802.16 standard, which defines the physical (PHY) and the medium access control (MAC) layers, is one of the BWA standards. Its MAC layer is centralized basis, where the Base Station (BS) is responsible for assigning the needed bandwidth for each Subscriber Station (SS), which requests bandwidth competing between all of them. The standard defines a contention resolution process to resolve the potential occurrence of collisions during the requesting process. In this paper, we propose to modify the contention resolution process to improve the network performance, including end-to-end delay and throughput.

The H.264 standard achieves much higher coding efficiency than the MPEG-2 standard, due to its improved inter-and intra-prediction modes at the expense of higher computational complexity. Transcoding MPEG-2 video to H.264 is important to enable gradual migration to H.264. However, given the significant differences between the MPEG-2 and the H.264 coding algorithms, transcoding is a much more complex task and new approaches to transcoding are necessary. The main problems that need to be addressed in the design of an efficient heterogeneous MPEG-2/H.264 transcoder are: the inter-frame prediction, the transform coding and the intra-frame prediction. In this paper, we focus our attention on the inter-frame prediction, the most computationally intensive task involved in the transcoding process. This paper presents a novel macroblock (MB) mode decision algorithm for P-frame prediction based on machine learning techniques to be used as part of a very low complexity MPEG-2 to H.264 video transcoder. Since coding mode decisions take up the most resources in video transcoding, a fast MB mode estimation would lead to reduced complexity. The proposed approach is based on the hypothesis that MB coding mode decisions in H.264 video have a correlation with the distribution of the motion compensated residual in MPEG-2 video. We use machine learning tools to exploit the correlation and construct decision trees to classify the incoming MPEG-2 MBs into one of the several coding modes in H.264. The proposed approach reduces the H.264 MB mode computation process into a decision tree lookup with very low complexity. Experimental results show that the proposed approach reduces the MB mode selection complexity by as much as 95% while maintaining the coding efficiency. Finally, we conduct a comparative study with some of the most prominent fast inter-prediction methods for H.264 presented in the literature. Our results show that the proposed approach achieves the best results for video transcoding applications.

Recent developments have given birth to H.264/AVC: a video coding standard offering better bandwidth to video quality ratios than previous standards (such as H.263, MPEG-2, MPEG-4, etc.), due to its improved inter- and intraprediction modes at the expense of higher computation complexity. It is expected that H.264/AVC will take over the digital video market, replacing the use of previous standards in most digital video applications. This creates an important need for heterogeneous video transcoding technologies from older standards to H.264. In this paper, we focus our attention on the interframe prediction, the most computationally intensive task involved in the heterogeneous video transcoding process. This paper presents a novel macroblock (MB) mode decision algorithm for interframe prediction based on data mining techniques to be used as part of a very low complexity heterogeneous video transcoder. The proposed approach is based on the hypothesis that MB coding mode decisions in H.264 video have a correlation with the distribution of the motion compensated residual in the decoded video. We use data mining tools to exploit the correlation and derive decision trees to classify the incoming decoded MBs into one of the several coding modes in H.264. The proposed approach reduces the H.264 MB mode computation process into a decision tree lookup with very low complexity. For general validation purposes, we apply our algorithm to two of the most important heterogeneous video transcoders: MPEG-2 to H.264 and H.263 to H.264. Our results show that the our data-mining based transcoding algorithm is able to maintain a good video quality while considerably reducing the computational complexity by 72% on average when applied in MPEG-2 to H.264 transcoders, and by 62% on average when applied in H.263 to H.264 transcoders. Finally, we conduct a comparative study with some of the most prominent fast interprediction methods for H.264 presented in the literature. Our results show that the proposed data mining-based approach achieves the best results for video transcoding applications.

In this paper, we propose the use of data mining algorithms to create a macroblock partition mode decision algorithm for inter-frame prediction, to be used as part of a high-efficient H.263 to H.264 transcoder. We use machine learning tools to exploit the correlation and derive decision trees to classify the incoming H.263 MC residual into one of the several coding modes in H.264. The proposed approach reduces the H.264 MB mode computation process into a decision tree lookup with very low complexity. Experimental results show that the proposed approach reduces the inter-prediction complexity by as much as 60% while maintaining the coding efficiency.

The two key problems in video transcoding are complexity reduction and quality management. The complexity of H.264 encoding makes complexity reduction even more important. A good transcoder design reduces complexity with negligible loss in quality. While motion estimation complexity can be easily reduced by reducing the search range or eliminating the coding modes, these approaches will cause significant quality drop. This paper presents reduced complexity motion estimation for MPEG-2 to H.264 transcoding. The proposed approach uses MPEG-2 motion vectors to dynamically reduce the search range and the motion vector refinement window. This approach is based on the hypothesis that longer MPEG-2 motion vectors indicate higher motion and hence a larger search window as well as a larger refinement window is necessary. Experimental results show that the proposed approach reduces the motion estimation complexity by as much as 85% with negligible loss in PSNR. We show that the results hold across video resolutions and bitrates.

Multicast is an efficient paradigm for transmitting data from a sender to a group of receivers. According to the IEEE 802.11 standard, the multicast service is defined as an unreliable service, that is, it does not include the use of ACK frames. Furthermore, different to the unicast service, the multicast service makes use of a single rate out of the various rates included in the basic service set defined by the IEEE 802.11 standard. Even though various proposals have recently appeared in the literature addressing these issues, none of them has come out with a structured set of control mechanisms taking into account the varying conditions characterising the wireless channels as well as the requirements of various applications. A novel cross-layer auto rate selection multicast mechanism for multi-rate wireless LANs, namely auto rate selection for multicast, capable of adapting the data transmission to the varying conditions of the channel and taking into account the characteristics of various applications, is introduced. The simulation results show that our proposal outperforms the IEEE 802.11 standard and the mechanisms recently proposed in the literature.

Multicast video streaming over multirate wireless LANs imposes strong demands on video codecs and the underlying network. It is not sufficient that only the video codec or only the underlying protocols adapt to changes in the wireless link quality. Research efforts should be applied in both and in a synchronized way. Cross layer design is a new paradigm that addresses this challenge by optimizing communication network architectures across traditional layer boundaries. This paper presents cross-layer architecture for adaptive video multicast streaming over multirate wireless LANs where layer-specific information is passed in both directions, top-down and bottom-up. The authors jointly consider three layers of the protocol stack: the application, data link and physical layers. The authors analyze the performance of the proposed architecture and extensively evaluate it via simulations. The results show that the real-time video quality of the overall system can be greatly improved by cross-layer signaling.

Though H.264 is highly efficient compared to MPEG-2, the wide and deep penetration of MPEG-2 creates a need for co-existence of these technologies and hence creates a need for transcoding. In this paper, we introduce and evaluate a novel macroblock partition mode decision algorithm for inter-frame prediction to be used as part of a high-efficient MPEG-2 to H.264 transcoder. The proposed tools are used to compute an optimal MB coding mode decision with significantly reduced computational complexity. Specifically, we achieve the computational savings by using the following MB information coming from MPEG-2: the MB coding modes in MPEG-2 and the mean and variance of the 16 4×4 sub blocks of the MPEG-2 residual MBs. We use data mining algorithms to develop a decision tree for obtaining the H.264 coding decision modes. Our results show that the proposed algorithm is able to maintain a good picture quality while considerably reducing the number of operations to be performed.

In this paper, we introduce and evaluate a low complexity macroblock partition mode decision algorithm for inter-frame prediction in MPEG-2 to H.264 transcoder. The proposed tools are used to compute an optimal MB coding mode decision with significantly reduced computational complexity. Specifically, we achieve the computational savings by using the following MB information coming from MPEG-2: the MB coding modes, the coded block pattern (CBPC) in MPEG-2, and the mean and variance of the 16 4times4 sub blocks of the MPEG-2 residual MBs. We use data mining algorithms to develop a decision tree for H.264 coding mode decisions. The decision trees are built using RD optimized mode decisions and result in highly efficient mode decisions. The proposed transcoder is 35% faster than the RD optimized H.264 reference transcoder without a significant PSNR degradation. The proposed transcoder performs over 3 dB better than the SAE cost based H.264 transcoding

This paper presents a novel macroblock mode decision algorithm for inter-frame prediction based on machine learning techniques to be used as part of a very low complexity MPEG-2 to H.264 video transcoder. Since coding mode decisions take up the most resources in video transcoding, a fast macro block (MB) mode estimation would lead to reduced complexity. The proposed approach is based on the hypothesis that MB coding mode decisions in H.264 video have a correlation with the distribution of the motion compensated residual in MPEG-2 video. We use machine learning tools to exploit the correlation and derive decision trees to classify the incoming MPEG-2 MBs into one of the 11 coding modes in H.264. The proposed approach reduces the H.264 MB mode computation process into a decision tree lookup with very low complexity. The proposed transcoder is compared with a reference transcoder comprised of a MPEG-2 decoder and an H.264 encoder. Our results show that the proposed transcoder reduces the H.264 encoding time by over 95% with negligible loss in quality and bitrate.

The H.264 standard, jointly developed by the ITU-T and the MPEG committees, is highly efficient offering perceptually equivalent quality video at about 1/3 to 1/2 of the bitrates offered by the MPEG-2 format. These significant bandwidth savings open the market to new products and services, including HDTV services at lower bitrates. It is expected that the H.264/AVC takes over the digital video market, replacing the use of MPEG-2 in most digital video applications. The complete migration to the new video-coding algorithm takes several years given the wide scale use of MPEG-2 in the market place today. This creates an important need for transcoding technologies for converting the large volume of existent video material from the MPEG-2 into the H.264 format and vice versa. However, given the significant differences between the MPEG-2 and the H.264 encoding algorithms, the transcoding process of such systems is much more complex to other heterogeneous video transcoding processes. In this paper, we introduce and evaluate a novel fast intra-frame prediction algorithm to be used as part of a high-efficient MPEG-2 to H.264 transcoder. Our results show that the proposed algorithm is able to maintain a good picture quality while considerably reducing the number of operations to be performed.

MPEG-2 is the most widely digital video-encoding standard in use nowadays. It is being widely used in the development and deployment of digital TV services, DVD and video-on-demand services. However, recent developments have given birth to the H.264/AVC, offering better bandwidth to video quality ratios than MPEG2. It is expected that the H.264/AVC will take over the digital video market, replacing the use of MPEG-2 in most digital video applications. The complete migration to the new video-coding algorithm will take several years given the wide scale use of MPEG-2 in the market place today. This creates an important need for transcoding technologies for converting the large volume of existent video material from the MPEG-2 into the H.264 format and vice versa. However, given the significant differences between the MPEG-2 and the H.264 encoding algorithms, the transcoding process of such systems is much more complex to other heterogeneous video transcoding processes. In this paper, we introduce and evaluate a novel intra-frame prediction algorithm to be used as part of a high-efficient MPEG-2 to H.264 transcoder. Our evaluation results show that the proposed algorithm considerable reduces the complexity involved in the intra-frame prediction: a key operation in the transcoding process

HIPERLAN/2 is a recent standard from ETSI for highspeed wireless LANs. This work explores this standard when providing QoS support to different traffic types. We describe a new bandwidth allocation protocol, which distinguishes among five types of connections, each one of them having its own QoS requirements. We carry out a performance evaluation via simulations that reflects the behavior of the novel MAC protocol. The bandwidth allocation protocol is evaluated in terms of its flexibility and efficiency in supporting delay sensitive traffic. We look particularly to MPEG-4 video communications, an application which is expected to be widely deployed in broadband wireless LANs.

Personal communication systems (PCS) aim to provide ubiquitous service to its subscribers. In order to fulfil the ubiquity requirement for service, PCS rely on two main features: use of wireless communications technology and mobility management. The former sets the basis to untie subscribers from a wire allowing them to freely move around. The latter allows subscribers to gain access to the network services at all times regardless of their location. A handover is the process used to allow a call in progress to continue as the associated subscriber moves around. A large number of handover mechanisms have been reported in the literature. In this article, we carry out a comprehensive performance study of the handover mechanisms as proposed in the literature. A detailed simulation model has been developed allowing us to set up parameters such as the speed and motion path of the mobile terminal. The performance metrics of interest are: the length of the disruption time caused by the handover mechanism and the number of lost packets caused by the handover mechanism. Under the scenario under consideration, the handovers using multicasting mechanism and full reestablishment mechanism exhibit the best results.

Recent advances in wireless technology are enabling the design and deployment of multiservice wireless networks. In order to be able to meet the QoS requirements of the various applications, it is essential to deploy QoS provisioning mechanisms. In this paper, we present various QoS provisioning mechanisms. Our simulation results show the effectiveness of the mechanisms when supporting various types of services.

Recent advances on wireless technology are enabling the design and deployment of multiservice wireless networks. In order to be able to meet the QoS requirements of the various applications, it is essential to deploy QoS provisioning mechanisms. To be most effective, these mechanisms require to monitor the status of the various active sources enabling them to better distribute the available resources. We propose a QoS provisioning mechanism for time constrained applications. Our simulation results show the effectiveness of the proposed mechanism when supporting time constrained services, such as MPEG-4 video communications.

A large number of studies have been conducted to evaluate the benefits of using layered video coding schemes as a means to improve the robustness of video communications systems. In this article, we study a frame-aware layering scheme for the transport of DCT-based video over packet switched networks. Under this scheme, the relevance of the different elements of the video sequence is taken into account in order to compose the encoded video sequence. We conduct a detailed study over a large set of video streams encoded using the MPEG-2 video standard. Our results show that it is possible to reduce the video traffic while maintaining an acceptable video quality.

More efficient transmission of coded speech over packet switching systems can be achieved by channel coding. The redundancy introduced by channel coding to recover lost or impaired packets results in an increased traffic and an increased queuing loss. Hence there is a critical tradeoff between channel coding rate and queuing loss. We show that the channel loss rate, the traffic level and congestion rate affect the optimum channel coding rate.

Two major issues in providing true end-to-end wireless/mobile video capabilities are: interoperability among network platforms and robustness of video compression algorithms in error-prone environments. In this paper, we mainly focus on the second issue and show how error resilience techniques can be used to improve the video quality. We argue that the error resilient tools provided within the MPEG-4 standard are not sufficient to provide acceptable quality in wireless/mobile networks, but that this quality can be significantly improved by the inclusion of hierarchical MPEG-4 video coding techniques. We present a novel hierarchical MPEG-4 video scheme particularly designed for video communications over QoS-capable wireless/mobile network.

Transmission of G.729A coded speech over packet switching networks including forward error correction is considered. Three buffer management policies are applied: simple, random and privileged. The privileged policy plays an important role to achieve quasi-uniform packet loss among multiple users. Its effect on spreading the loss among various sources and in avoiding consecutive or closely timed losses for any source is described. In this paper, the subjective quality improvement by correctly adapting the loss pattern to the sensitivity of the G.729A algorithm is described.

MPEG-2 video communications over ATM networks is one of the most active research areas in the field of computer communications. In the transmission process of a variable bit rate video signal over an ATM network, cells are inevitably exposed to delays, errors, and losses due to the statistical multiplexing used in these networks. These phenomena affect the quality of the video signal, and without adequate measures to control the propagation of the impairments, the quality of the service may fall below acceptable levels. In this paper, we study the impact of cell losses on the quality of an MPEG-2 video sequence encoded in a variable bit rate mode. We introduce a set of control mechanisms at different levels of the protocol architecture to be used in MPEG-2-based video communications systems using ATM networks as their underlying transmission mechanism. Our results (using different video sequences) show the effectiveness to improve the video quality by using a structured set of control mechanisms to overcome for the loss of cells carrying VBR MPEG-2 video streams. We argue that in order to be able to create video systems able to cope with cell losses encountered in computer communications systems, a structured set of error-resilient protocol mechanisms is needed.

With the ever evolving mobile cellular technology, more standards are being defined and upgraded with the goal to provide better quality of service as well as higher capacity. This paper focuses on the performance of the mobile/wireless network interface for IS-136, a North-American standard for digital cellular systems. There is no doubt that the voice service can be affordable but how the interface reacts to the addition of a short message service (SMS) is still under investigation. A study of the reverse access control protocol of the IS-136 digital control channel (DCCH) is provided and performance results are presented taking into consideration different set-ups for the access control parameters and support for SMS. Finally, guidelines are provided on how to set-up the system.

Personal communications services (PCS) key objective is to provide ubiquitous communication services. The success of PCS will depend on the network resources to ensure the quality-of-service required by service subscribers. Performance evaluation studies of different system configurations and implementation alternatives will play a major role in the planning and dimensioning of successful PCS networks. We undertake the analysis of some major implementation issues of a PCS architecture based on the IN system architecture. Different system configurations and traffic scenarios are considered in the study. The overall system performance is evaluated in terms of utilization rate of the key network elements and the end-to-end response time of the signaling network for various IN services.

We evaluate the performance of two schemes to control the load imposed to the service control point (SCP), namely, call gapping and credit-based control and introduce a two level load-balancing control mechanism to provide better quality of services guarantees to IN services. Our study differs from previous studies as we consider in detail the internal organization of the SCP. Furthermore, we consider the operation of the system when servicing IN services of various levels of complexity. The complexity being measured by the number of visits to the database required to complete the service. We have been particularly interested in evaluating the call response time and system utilization since these are the two performance metrics of concern to the users and service provider, respectively.

We introduce a set of control mechanisms at different levels of the protocol architecture to be used in MPEG-2-based video communications systems using ATM networks as their underlying transmission mechanism. Our results show the effectiveness of using a structured set of control mechanisms to overcome for the loss of cells carrying VBR video streams.

MPEG-2 video communications over ATM networks is one of the most active research areas in the field of computer communications. In this paper, we introduce a set of control mechanisms at different levels of the protocol architecture to be used in MPEG-2-based video communications systems using ATM networks as their underlying transmission mechanism. We argue that in order to be able to create video systems able to cope with some of the errors encountered in computer communications systems, a structured set of error-resilient protocol mechanisms is needed. Our results show the effectiveness of using a structured set of control mechanisms to overcome for the loss of cells carrying VBR MPEG-2 video streams.

The available bit rate (ABR) service has been proposed by the ATM Forum in 1994 and also adopted by the ITU shortly afterwards. This paper presents the performance of a rate-based ABR in the presence of bursty data applications. Based on measurements reported in the literature, we have developed a simulation tool to evaluate the performance of the rate-based ABR mechanism. The simulation model includes a complete protocol stack allowing us to study the interaction of the traffic control mechanisms implemented by protocols at different layers, i.e., TCP and ABR. Under realistic traffic models, it is shown that the rate-based ABR mechanism provides a high link utilization, minimum buffering requirements and cell transfer delays.

This paper presents a binary feedback flow control scheme which uses the gradient of the system performance function to generate feedback from the network to the data sources. The setting of the single bit congestion indication communicated back to the users is based upon the sign of the gradient of the system performance function. The proposed scheme uses a simple gradient optimization method and a neural model of the system dynamics to determine the gradient of the performance function. Simulation results are presented to compare the performance of the proposed scheme to the conventional queue thresholding approach.

Many studies have been conducted involving the transport of constant and variable bit rate MPEG-2 video in asynchronous transfer mode (ATM) networks; however, many of these studies have considered fairly homogeneous scenarios in which the only other traffic present in the ATM network, if any, are other MPEG-2 video sources. In this study the requirements of supporting MPEG-2 video communications in an ATM network under unloaded and heavily loaded network conditions, in which the generated traffic sources are bursty in nature, are considered. To complete the study an experimental video on demand (VoD) testbed, developed as part of a collaborative research effort between the Communications Research Center of Canada (CRC), Nortel and the University of Ottawa, was employed. To determine the characteristics of the MPEG-2 video traffic generated by the VoD application in the ATM network in the presence of other traffic, cell interarrival time measurements and cell rate snapshots at the ATM cell level were considered. Results obtained show that the end-to-end flow control implemented in the application layer in the video-on-demand (VoD) system and the traffic controls implemented intermediate network elements (e.g., routers, switches) have significant impacts on the characteristics of the MPEG-2 traffic carried in the ATM network. Moreover, the impact of the intermediate network elements on the characteristics of the MPEG-2 traffic increases with the amount of non-MPEG-2 video.

The authors present a study whose main objective has been to explore the use of high-bandwidth multimedia communications technology for medical applications physically distributed among hospitals, that is to say an interhospital multimedia communications system. The design and development issues of a prototype are described, as well as the results of an in-field trial. From the results of this trial, a performance study has been developed for evaluating the expected communications requirements of a large-scale configuration of the system.>

The problem of communicating the radiologist's findings to the attending physician is introduced, and it is shown why it is essentially a multimedia communication problem. A multimedia communications system that was designed with a focus on database and medical reports organization and architecture is described. A report is presented of some of the key results of an in-house trial linking the Department of Radiological Sciences and the Emergency Department at the Ottawa Civic Hospital, Canada, a 950-bed tertiary-care teaching hospital.>

A performance study is presented of the proposed IEEE 802.6 standard (DQDB MAN) used as a backbone loop network in packet-switching mode to interconnect token rings that operate as specified in the ANSI/IEEE 802.5 standard. Interconnection between DQDB and token rings is provided by gateways that use assembly and segmentation processors and limited buffer sizes. The authors consider a network architecture in which Type 2 of the IEEE 802.2 logical-link-control (LLC) standard is used as end-to-end protocol between stations on the token rings. The use of the former and current DQDB slot formats are compared. Several window sizes, different policy schemes, and various traffic intensities and bit rates in token rings are also considered. Using the current DQDB slot size, results show that higher access priority should be allocated to the gateways in the token rings and also show the importance of reducing the time caused by the segmentation of packets in the gateways.>

The authors present a multimedia medical communications system based on packet local area networks. This system is intended to facilitate the communications between radiologists and physicians within a hospital. Radiographs are taken, digitized, and then stored in the database. Radiologists retrieve radiographic images from the database for diagnosis, and make diagnostic reports to be stored in the database. Physicians can then consult these radiographic images and diagnostic reports. A real-time conference can be set up between a radiologist and a physician who are in different locations, for discussing a particular case or radiographic images can be retrieved from the database for diagnosis and conferencing. Simulations indicate the delay for displaying an image of 1 Mb from the local storage (hard disk) at a workstation is approximately 6.5 s.>