Real-time speech enhancement on resourceconstrained devices faces a persistent challenge: many causal systems rely on dual-branch magnitude/complex refinements or attention-heavy backbones, whereas ultra-light streaming models often lack explicit cross-band interaction and long-range temporal memory, resulting in unstable artifacts. We propose Streaming Subband State-space Graph-frequency Network (S³G-Net), a causal complex-masking architecture built around a subband state-space graph-frequency bottleneck. Specifically, S³G-Net (i) allocates frequency-dependent capacity through dynamic subband gating in the encoder with no look-ahead, (ii) restores cross-band structure via banded graph-frequency residual propagation that promotes locally coherent spectral evolution, and (iii) redesigns the temporal core into a lightweight hybrid architecture that integrates a dilated causal Temporal Convolutional Network (TCN) for long-horizon modeling with a graph-conditioned diagonal state-space module. The proposed graph-conditioned diagonal state-space model (SSM) uses per frame graph-frequency descriptors to modulate band-wise memory, enabling causal cross-band interaction with small attention while stabilizing streaming behavior. S³G-Net is implemented as a single-stream encoder-bottleneck-decoder with no auxiliary branches or post-filtering, and uses a model with only ∼30k parameters requiring ∼60M MAC/s. S³G-Net outperforms or matches substantially larger baselines in terms of perceptual quality across in-domain and out-of-domain evaluations, while maintaining competitive intelligibility at significantly lower cost.

In this paper, we present a deep reinforcement learning (DRL) framework for vehicular optical camera communication (OCC) systems that ensures ultra-reliable and low-latency communication (uRLLC). We first formulate a throughput maximization problem that aims at optimizing speed of vehicles, channel code rate, and modulation order while respecting the uRLLC requirements. We model reliability by satisfying a target bit error rate and latency as transmission latency. To improve the transmission rate and provide high reliability and low latency, our scheme uses low-density parity-check codes and adaptive modulation. We then solve the optimization problem using the actor-critic-based DRL scheme with Wolpertinger framework. We employ a deep deterministic policy gradient algorithm to operate over continuous action spaces. The evaluation confirms that our proposed DRL-based optimization scheme achieves superior performance compared to radio frequency-based communication systems as well as variants of the proposed scheme. Finally, we verify through simulations that our proposed solution can maximize the communication rate while meeting the uRLLC constraints.

In this work, we study the use of non-orthogonal multiple access (NOMA) and orthogonal time frequency space (OTFS) modulation in a multiple-input multiple-output (MIMO) communication network where mobile users (MUs) with different mobility profiles are grouped into clusters. We consider a downlink scenario where a base station (BS) communicates with multiple users that have diverse mobility profiles. High-mobility (HM) users' signals are placed in the delay-Doppler (DD) domain using OTFS modulation in order to transform their time-varying channel into a sparse static channel, while low-mobility (LM) users signals are placed in the time-frequency (TF) domain. Precoding is adopted at the BS to direct focused beams towards each cluster of users. Moreover, NOMA spectrum sharing is used in each cluster to allow the coexistence of a single HM user and multiple LM users within the same resource block. LM users access disjoint subchannels to ensure their orthogonality. All users within the same cluster first detect the HM user's signal. Afterward, LM users suppress the interference from the HM user and detect their own signals. Closed-form expressions of the detection signal-to-noise ratios (SNRs) are derived. The numerical results showed that the performance of the proposed system highly depends on the number of LM users, the number of clusters and the power allocation factors between HM and LM users.

The 26th IEEE International Workshop on Multimedia Signal Processing (MMSP 2024), organized by the Multimedia Signal Processing Technical Committee (MMSP-TC) of IEEE Signal Processing Society (SPS), was held at Purdue University, West Lafayette, Indiana, U.S.A., from October 2 - 4, 2024.

In dynamic and time-varying vehicular networks, existing vehicular communication systems cannot guarantee ultra-reliable and low latency communication (uRLLC). To address this, we propose a novel deep reinforcement learning-based vehicular optical camera communication (OCC) system with an aim to maximize the throughput and ensure uRLLC. To achieve this, our scheme chooses the optimal code rate, modulation scheme and speed of vehicles for multiple vehicular links. We use OCC, which offers interference-free communication as an alternative to radio frequency systems. Moreover, we employ 5G New Radio low-density parity-check codes and an adaptive modulation scheme to support variable rates and ultra-reliability. The proposed large-scale and continuous problem is solved through an actor-critic algorithm based on Wolpertinger architecture. We extendedly evaluate the system performance and compare it with several other schemes from the literature as well as with variants of our scheme. We observe from the results that the proposed method achieves higher average throughput and lower latency than all the other schemes under comparison. Further, the proposed scheme can meet the uRLLC constraints, whereas other schemes under comparison fail to respect these constraints most of the time.

In this paper, we propose a vehicular optical camera communication system that can meet low bit error rate (BER) and ultra-low latency constraints. First, we formulate a sum spectral efficiency optimization problem that aims at finding the speed of vehicles and the modulation order that maximizes the sum spectral efficiency subject to reliability and latency constraints. This problem is mixed-integer programming with nonlinear constraints, and even for a small set of modulation orders, is NP-hard. To overcome the entailed high computational and time complexity which prevents its solution with traditional methods, we first model the optimization problem as a partially observable Markov decision process. We then solve it using an independent Q-learning framework, where each vehicle acts as an independent agent. Since the state-action space is large we then adopt deep reinforcement learning (DRL) to solve it efficiently. As the problem is constrained, we employ the Lagrange relaxation approach prior to solving it using the DRL framework. Simulation results demonstrate that the proposed DRL-based optimization scheme can effectively learn how to maximize the sum spectral efficiency while satisfying the BER and ultra-low latency constraints. The evaluation further shows that our scheme can achieve superior performance compared to radio frequency-based vehicular communication systems and other vehicular OCC variants of our scheme.

This paper studies the affine frequency division multiplexing (AFDM)-empowered sparse code multiple access (SCMA) system, referred to as AFDM-SCMA, for supporting massive connectivity in high-mobility environments. First, by placing the sparse codewords on the AFDM chirp subcarriers, the input-output (I/O) relation of AFDM-SCMA systems is presented. Next, we delve into the generalized receiver design, chirp rate selection, and error rate performance of the proposed AFDM-SCMA. The proposed AFDM-SCMA is shown to provide a general framework and subsume the existing OFDM-SCMA as a special case. Third, for efficient transceiver design, we further propose a class of sparse codebooks for simplifying the I/O relation, referred to as I/O relation-inspired codebook design in this paper. Building upon these codebooks, we propose a novel iterative detection and decoding scheme with linear minimum mean square error (LMMSE) estimator for both downlink and uplink channels based on orthogonal approximate message passing principles. Our numerical results demonstrate the superiority of the proposed AFDM-SCMA systems over OFDM-SCMA systems in terms of the error rate performance. We show that the proposed receiver can significantly enhance the error rate performance while reducing the detection complexity.

Digital transformation is increasingly reliant onservice-based operations in fog networks. The latter is a geo-dispersed form of the cloud, extending resources closer to end-users for improved privacy and reduced latency. The dispersion leverages diversity of compute-network capacities and energy prices, while promotes the coexistence of multiple providers. This drives variation in operational cost, coupled with limited information sharing across providers. Consequently, there is a critical need for an orchestration solution that preserves autonomy and optimizes operational cost across domains, while meeting service requirements. This paper proposes a novel service-based fog management and network orchestrator (sbMANO), which utilizes service metadata in enabling multi-provider resource management. The sbMANO is empowered with a novel optimization algorithm for service-based joint request mapping and response routing. The algorithm acts on partial information and preserves the edge for delay-critical services. The performance of the algorithm is evaluated analytically fordelay-awareanddelay-agnosticvariants. The results show that both achieve near-optimal performance in maximizing user satisfaction with minimum operational cost. Furthermore, the delay-aware variant outperforms the agnostic counterpart, with higher user satisfaction and lower operational cost.

Achieving ultra-reliable and low latency communication (uRLLC) in vehicular networks is challenging because of their time-varying and dynamic nature. In this paper, we propose a deep reinforcement learning (DRL) based vehicular optical camera communications (OCC) system that aims at maximizing the transmission rate. In doing so, we optimize the speed of vehicles, the channel code rate, and the modulation order while respecting the uRLLC requirements. We define reliability by satisfying a predefined bit error rate and latency as transmission latency. To improve the transmission rate and ensure reliability and low latency, we use low-density parity-check codes and adaptive modulation. We then solve the optimization problem using the actor-critic DRL framework with Wolpertinger architecture. We deal with the continuous action spaces by employing a deep deterministic policy gradient algorithm. The evaluation verifies that our proposed optimization scheme can achieve superior performance than the comparison schemes. Finally, the results further confirm that the proposed solution can maximize the communication rate while guaranteeing the uRLLC requirements.

In this paper, we study the secrecy performance of short packet secure communications over a fading wiretap channel when there are multiple receivers and eavesdroppers. In particular, we evaluate and compare the performance of colluding and non-colluding eavesdropping modes, in terms of achievable secrecy throughput. Our aim is to determine whether eavesdroppers' collusion degrades the average secrecy throughput compared to a non-colluding scenario. After deriving closed-form approximations on average secrecy throughput for both scenarios, Monte-Carlo simulations are performed to obtain the accuracy of each approximation. Our results reveal that the impact of colluding eavesdroppers on the average secrecy throughput causes more loss than a non-colluding case, especially with the increasing number. Besides, an increased number of receivers does not eliminate the negative impact of eavesdroppers.

Optical camera communications (OCC) has emerged as a key enabling technology for the seamless operation of future autonomous vehicles. In this paper, we introduce a spectral efficiency optimization approach in vehicular OCC. Specifically, we aim at optimally adapting the modulation order and the relative speed while respecting bit error rate and latency constraints. As the optimization problem is NP-hard problem, we model the optimization problem as a Markov decision process (MDP) to enable the use of solutions that can be applied online. We then relaxed the constrained problem by employing Lagrange relaxation approach before solving it by multi-agent deep reinforcement learning (DRL). We verify the performance of our proposed scheme through extensive simulations and compare it with various variants of our approach and a random method. The evaluation shows that our system achieves significantly higher sum spectral efficiency compared to schemes under comparison.

This paper studies the performance of short packet communications in the presence of multiple eavesdroppers. We start our investigation by examining the fading wiretap channel, where the communication is overheard by multiple non-colluding single antenna eavesdroppers. A closed-form expression for the average secrecy throughput is derived, when the transmitter has a single antenna. The Monte-Carlo simulations show a close match of the analytical expression with the numerical results. Moreover, the optimal blocklength value that maximizes the secrecy throughput is determined, when the communication is observed by single and multiple eavesdroppers. We then extend our analysis for the case of a multiple-antenna transmitter and consider artificial noise (AN) to confuse the eavesdroppers. A closed-form expression for the average secrecy throughput is obtained for the scenario of a two-antenna transmitter and two eavesdroppers with a single antenna. The results demonstrate the validity of the approximation when compared with Monte-Carlo simulations. The results further reveal that an increased number of antennas at the transmitter is associated with higher average secrecy throughput and applying AN helps to eliminate the harm of the eavesdroppers.

360$^{\circ }$video is an essential component of VR/AR/MR systems that provides immersive experience to the users. However, 360$^{\circ }$video is associated with high bandwidth requirements. The required bandwidth can be reduced by exploiting the fact that users are interested in viewing only a part of the video scene and that users request viewports that overlap with each other. Motivated by the findings of our recent works where the benefits of caching video tiles at edge servers instead of caching entire 360$^{\circ }$videos were shown, in this paper, we introduce the concept of virtual viewports that have the same number of tiles with the original viewports. The tiles forming these viewports are the most popular ones for each video and are determined by the users’ requests. Then, we propose a reactive caching scheme that assumes unknown videos’ and viewports’ popularity. Our scheme determines which videos to cache as well as which is the optimal virtual viewport per video. Virtual viewports permit to lower the dimensionality of the cache optimization problem. To solve the problem, we first formulate the content placement of 360$^{\circ }$videos in edge cache networks as a Markov Decision Process (MDP), and then we determine the optimal caching placement using the Deep Q-Network (DQN) algorithm. The proposed solution aims at maximizing the overall quality of the 360$^{\circ }$videos delivered to the end-users by caching the most popular 360$^{\circ }$videos at base quality along with a virtual viewport in high quality. We extensively evaluate the performance of the proposed system and compare it with that of known systems such as Least Frequently Used (LFU), Least Recently Used (LRU), First In First Out (FIFO), over both synthetic and real 360$^{\circ }$video traces. The results reveal the large benefits coming from reactive caching of virtual viewports instead of the original ones in terms of the overall quality of the rendered viewports, the cache hit ratio, and the servicing cost.

With the increasing demand for digitalization and participation in Industry 4.0, new challenges have emerged concerning the market of digital services to compensate for the lack of processing, computation, and other resources within Industrial Internet of Things (IIoTs). At the same time, the complexity of interplay among stakeholders has grown in size, granularity, and variation of trust. In this paper, we consider an IIoT resource market with heterogeneous buyers such as manufacturer owners. The buyers interact with the resource supplier dynamically with specific resource demands. This work introduces a broker between the supplier and the buyers, equipped with Distributed Ledger Technologies (DLT) providing a service for market security and trustworthiness. We first model the DLT-assisted IIoT market analytically to determine an offline solution and understand the selfish interactions among different entities (buyers, supplier, broker). Considering the non-cooperative heterogeneous buyers in the dynamic market, we then follow an independent learners framework to determine an online solution. In particular, the decision-making procedures of buyers are modeled as a Partially Observable Markov Decision Process which is solved using independent Q-learning. We evaluate both the offline and online solutions with analytical simulations, and the results show that the proposed approaches successfully maximize players’ satisfaction. The results further demonstrate that independent Q-learners achieve equilibrium in a dynamic market even without the availability of complete information and communication, and reach a better solution compared to that of centralized Q-learning.

The Internet of Things (IoT) connected by software-defined networking (SDN) promises to bring great benefits to cyber-physical systems. However, the increased attack surface offered by the growing number of connected vulnerable devices and separation of SDN control and data planes could overturn the huge benefits of such a system. This article addresses the vulnerability of the trust relationship between the control and data planes. To meet this aim, we propose an edge computing-based Blockchain as a Service (BaaS), enabled by an external BaaS provider. The proposed solution provides verification of inserted flows through an efficient, edge-distributed, blockchain solution. We study two scenarios for the blockchain reward purpose: 1) information symmetry, in which the SDN operator has direct knowledge of the real effort spent by the BaaS provider and 2) information asymmetry, in which the BaaS provider controls the exposure of information regarding spent effort. The latter yields the so-called “moral hazard,” where the BaaS may claim higher than actual effort. We develop a novel mathematical model of the edge BaaS solution and propose an innovative algorithm of a fair reward scheme based on game theory that takes into account moral hazard. We evaluate the viability of our solution through analytical simulations. The results demonstrate the ability of the proposed algorithm to maximize the joint profits of the BaaS and SDN operator, i.e., maximizing the social welfare.

$360^{o}$video is becoming an increasingly popular technology on commercial social platforms and vital part of emerging Virtual Reality/Augmented Reality (VR/AR) applications. However, the delivery of$360^{o}$video content in mobile networks is challenging because of its size. The encoding of$360^{o}$video into multiple quality layers and tiles and edge cache-assisted video delivery have been proposed as a remedy to the excess bandwidth requirements of$360^{o}$video delivery systems. Existing works using the above tools have shown promising performance for Video-on-Demand (VoD)$360^{o}$delivery, but they cannot be straightforwardly extended in a live-streaming setup. Motivated by the above, we study edge cache-assisted$360^{o}$live video streaming to increase the overall quality of the delivered$360^{o}$videos to users and reduce the service cost. We employ Long Short-Term Memory (LSTM) networks to forecast the evolution of the content requests and prefetch content to caches. To further enhance the delivered video quality, users located in the overlap of the coverage areas of multiple Small Base Stations (SBSs) are allowed to receive data from any of these SBSs. We evaluate and compare the performance of our algorithm with Least Frequently Used (LFU), Least Recently Used (LRU), and First In First Out (FIFO) algorithms. The results show the superiority of the proposed approach in terms of delivered video quality, cache-hit-ratio and backhaul link usage.

This paper presents a complete framework for content discovery and retrieval in Information-Centric Networks. For content discovery, we implement a method similar to our previously developed pull-based BFR [1], which uses Bloom filter-based signaling to inform servers about the name prefixes of available requests. For content retrieval, we propose in this paper a feedback-based cooperative protocol implementing network coding-based forwarding. The proposed network coding-based protocol provides a distributed solution to control the multisession codeblock size, i.e., the number of variables that are combined into network coded packets, by setting a capacity constraint on each node and by piggybacking the available capacity as feedback on messages sent to neighbors. The network codes are decided using linear programming. We compare the proposed network coding-based protocol with push-based BFR [2] and pull-based BFR [1]. The results show that the proposed protocol outperforms both push-based BFR and pull-based BFR in terms of content discovery overhead and average content block retrieval delay.

Despite the advances of 5G systems, the delivery of 360° video content in mobile networks remains challenging because of the size of 360° video files. Recently, edge caching has been shown to bring large performance gains to 360° Video on Demand (VoD) delivery systems, however existing systems cannot be straightforwardly applied to live 360° video streaming. To address this issue, we investigate edge cache-assisted live 360° video streaming. As videos' and tiles' popularities vary with time, our framework employs a Long Short-Term Memory (LSTM) network to determine the optimal cache placement/evictions strategies that optimize the quality of the videos rendered by the users. To further enhance the delivered video quality, users located in the overlap of the coverage areas of multiple SBSs are allowed to receive their data from any of these SBSs. We evaluate and compare the performance of our method with that of state-of-the-art systems. The results show the superiority of the proposed method against its counterparts, and make clear the benefits of accurate tiles' popularity prediction by the LSTM networks and users association with multiple SBSs in terms of the delivered quality.

A service-centric network requires a routing protocol to route service requests towards service providers. Routing operations can be divided into intra-domain and inter-domain routing. In the proposed approach, a so-called supernode is responsible for managing its own domain as well as for communicating with the supernodes of other domains to perform inter-domain routing. In order to appoint appropriate nodes as supernodes in the network topology, in this paper, we use Dominating Sets (DS) and Connected Dominating Sets (CDS). We propose fully distributed algorithms for constructing DS as well as CDS over the network topology. To prepare routing information, the nodes of each domain inform their supernodes about their available service names and resources (e.g., CPU, RAM). To this aim, the nodes use Bloom filters which reduce bandwidth and storage overhead. The performance evaluation shows that the required bandwidth overhead for DS and CDS construction algorithms increases with the topology size. The results also show that for large network topologies, CDS-based routing requires significantly less bandwidth overhead than both DS-based routing and Named Data Networking with multicast forwarding strategy. Finally, from the results we can observe that both DS-based and CDS-based routing have significantly lower service retrieval time than NDN multicast strategy.

This paper studies the problem of secure communication from a transmitter to a receiver with the use of short packets under the existence of multiple eavesdroppers, who are overhearing the transmission. We assume that the eavesdroppers are mutually independent. Further, we consider that the main channel and eavesdropper channels are Rayleigh fading, then use the performance metric of average secrecy throughput to measure how secure the communication is. We derive a closed form approximation of the average secrecy throughput in the presence of multiple eavesdroppers. Finally, the approximation is validated through simulations and we have seen that the approximation is very close to the corresponding simulation results.

The recent surge of applications involving the use of 360° video challenges mobile networks infrastructure, as 360° video files are of significant size, and current delivery and edge caching architectures are unable to guarantee their timely delivery. In this paper, we investigate the problem of joint collaborative content-aware caching and delivery of 360° videos in a video on demand setting. The proposed scheme takes advantage of 360° video encoding in multiple tiles and layers to make fine-grained decisions regarding which tiles to cache in each Small Base Station (SBS), and where to deliver them from to the end users, as users may reside in the coverage area of multiple SBSs. This permits to cache the most popular tiles in the SBSs, while the remaining tiles may be obtained through the backhaul. In addition, we explicitly consider the time delivery constraints to ensure continuous video playback. To reduce the computational complexity of the optimization problem, we simplify it by introducing a fairness constraint. This allows us to split the original problem into subproblems corresponding to Groups of Pictures (GOP). Each of the subproblems is then solved with the method of Lagrange partial relaxation. Finally, we evaluate the performance of the proposed method for various system parameters and compare it with schemes that do not consider 360° video encoding into multiple tiles and quality layers, as well as with two variants of the proposed method: one that considers layered encoding and SBSs collaboration and another that uses tiles encoding but with no SBSs collaboration. The results showcase the benefits coming from caching and delivery decisions on per tile basis and the importance of exploiting SBSs collaboration.

In Named Data Networking (NDN), there is a need for routing protocols to populate Forwarding Information Base (FIB) tables so that the Interest messages can be forwarded. To populate FIBs, clients and routers require some routing information. One method to obtain this information is that network nodes exchange routing information by each node advertising the available content objects. Bloom Filter-based Routing approaches like BFR [1], use Bloom Filters (BFs) to advertise all provided content objects, which consumes valuable bandwidth and storage resources. This strategy is inefficient as clients request only a small number of the provided content objects and they do not need the content advertisement information for all provided content objects. In this paper, we propose a novel routing algorithm for NDN called pull-based BFR in which servers only advertise the demanded file names. We compare the performance of pull-based BFR with original BFR and with a flooding-assisted routing protocol. Our experimental evaluations show that pull-based BFR outperforms original BFR in terms of communication overhead needed for content advertisements, average roundtrip delay, memory resources needed for storing content advertisements at clients and routers, and the impact of false positive reports on routing. The comparisons also show that pull-based BFR outperforms flooding-assisted routing in terms of average round-trip delay.

In this paper, we analyze the performance of vehicular optical camera communication (OCC) towards ultra-reliable and low latency communications (uRLLC). The employed vehicular OCC model uses light-emitting diodes (LED) as transmitter and camera as receiver. In particular, we investigate the performance of the proposed system in terms of bit error rate (BER), spectral efficiency, and transmission latency at different inter-vehicular distances and angle of incidences (AoI). Further, we investigate the use of adaptive modulation to improve the spectral efficiency. From our analysis, we note that by satisfying a given target BER, higher spectral efficiency and lower latency can be achieved through adjusting the AoI towards the smaller degrees and switching into the suitable modulation order. Finally, we verify the results through simulations, which show that OCC can ensure ultra-low latency as well as satisfy the reliability requirements in automotive vehicles.

The recent surge of applications involving the use of 360° video challenges mobile networks infrastructure, as 360° video files are of significant size, and current delivery and edge caching architectures are unable to guarantee their timely delivery. In this paper, we investigate the problem of joint collaborative content-aware caching and delivery of 360° videos. The proposed scheme takes advantage of 360° video encoding in multiple tiles and layers to make fine-grained decisions regarding which tiles to cache in each small base station (SBS), and from where to deliver them to the end users, as users may reside in the coverage area of multiple SBSs. This permits to cache the most popular tiles in the SBSs, while the remaining tiles may be obtained through the backhaul. In addition, we explicitly consider the time delivery constraints to ensure continuous video playback. We evaluate the performance of the proposed method for various system parameters and compare it with schemes that do not consider 360° video encoding into multiple tiles and quality layers. The results make clear the benefits coming from caching and delivery decisions on per tile basis.

In this paper, we propose a medium access control (MAC) design method for wireless sensor networks based on decentralized coordinated reinforcement learning. Our solution maps the MAC resource allocation problem first to a factor graph, and then, based on the dependencies between sensors, transforms it into a coordination graph, on which the max-sum algorithm is employed to find the optimal transmission actions for sensors. We have theoretically analyzed the system and determined the convergence guarantees for decentralized coordinated learning in sensor networks. As part of this analysis, we derive a novel sufficient condition for the convergence of max-sum on graphs with cycles and employ it to render the learning process robust. In addition, we reduce the complexity of applying max-sum to our optimization problem by expressing coordination as a multiple knapsack problem (MKP). The complexity of the proposed solution can be, thus, bounded by the capacities of the MKP. Our simulations reveal the benefits coming from adaptivity and sensors' coordination, both inherent in the proposed learning-based MAC.

Efficient mobility management techniques are critical in providing seamless connectivity and session continuity between a mobile node and the network during its movement. However, current mobility management solutions generally require a central entity in the network core, tracking IP address movement, and anchoring traffic from source to destination through point-to-point tunnels. Intuitively, this approach suffers from scalability limitations as it creates bottlenecks in the network, due to sub-optimal routing via the anchor point. This is often termed “dog-leg” routing. Meanwhile, alternative anchorless, solutions are not feasible due to the current limitations of the IP semantics, which strongly tie addressing information to location. In contrast, this paper introduces a novel anchorless mobility solution that overcomes these limitations by exploiting a new path-based forwarding fabric together with emerging mechanisms from information-centric networking. These mechanisms decouple the end-system IP address from the path based data forwarding to eliminate the need for anchoring traffic through the network core; thereby, allowing flexible path calculation and service provisioning. Furthermore, by eliminating the limitation of routing via the anchor point, our approach reduces the network cost compared to anchored solutions through bandwidth saving while maintaining comparable handover delay. The proposed solution is applicable to both cellular and large-scale wireless LAN networks that aim to support seamless handover in a single operator domain scenario. The solution is modeled as a Markov-chain which applies a topological basis to describe mobility. The validity of the proposed Markovian model was verified through simulation of both random walk mobility on random geometric networks and trace information from a large-scale, city wide data set. Evaluation results illustrate a significant reduction in the total network traffic cost by 45 percent or more when using the proposed solution, compared to Proxy Mobile IPv6.

Bloom filter-based routing protocols for Named Data Networking (NDN) aim at facilitating content discovery in NDN. In this paper, we compare the performance of two Bloom filter-based routing protocols, namely BFR and COBRA. BFR is a push-based routing protocol that works based on Bloom filter-based content advertisements, while COBRA is a pull-based routing protocol that operates based on route traces left from previously retrieved content objects, which are stored in Stable Bloom Filters. In this paper, we show that BFR outperforms COBRA in terms of average memory needed for storing routing updates, average round-trip delay, normalized communication overhead, total Interest communication overhead, and mean hit distance.

In this paper, we propose a novel decentralized framework for optimizing the transmission strategy of Irregular Repetition Slotted ALOHA (IRSA) protocol in sensor networks. The proposed method is inspired by reinforcement learning algorithms. To deal with sensor nodes limited lifetime and communication range, we allow sensor nodes to decide how many packet replicas to transmit and when to transmit them considering only their own buffer state. We show that this information is sufficient and can help avoiding packets' collisions and improving the throughput significantly. We solve the problem using the decentralized partially observable Markov Decision Process (POMDP) framework, where we allow each node to decide independently of the others how many packet replicas to transmit and when. The performance of the proposed method is compared with the native IRSA protocol. The results make clear that our method leads to large throughput gains without imposing nodes coordination, in particular, when network traffic is heavy.

We propose a novel network coding (NC) enabled named data networking (NDN) architecture for scalable video delivery. Our architecture utilizes NC in order to address the problem that arises in the original NDN architecture, where optimal use of the bandwidth and caching resources necessitates the coordination of the Interest forwarding decisions. To optimize the performance of the proposed NC-based NDN architecture and render it appropriate for transmission of scalable video, we devise a novel rate allocation algorithm that decides on the optimal rates of Interests sent by clients and intermediate nodes. The flow of Data packets achieved by this algorithm maximizes the average quality of the video delivered to the client population. To support the handling of Interest and Data packets when intermediate nodes perform NC, we introduce the use of Bloom filters, which store efficiently additional information about the Interest and Data packets, and modify accordingly the standard NDN architecture. We also devise an optimized Interest forwarding strategy that implements the target rate allocation. The proposed architecture is evaluated for transmission of scalable video over PlanetLab topologies. The evaluation shows that the proposed scheme exploits optimally the available network resources.

Seamless connectivity plays a key role in realizing QoS-based delivery in mobile networks. However, current handover mechanisms hinder the ability to meet this target, due to the high ratio of handover failures, packet loss and service interruption. These challenges are further magnified in Heterogeneous Cellular Networks (HCN) such as Advanced Long Term Evolution (LTE-Advanced) and LTE in unlicensed spectrum (LTE-LAA), due to the variation in handover requirements. Although mechanisms, such as Fast Handover for Proxy Mobile IPv6 (PFMIPv6), attempt to tackle these issues; they come at a high cost with sub-optimal outcomes. This primarily stems from various limitations of existing IP core networks. In this paper we propose a novel handover solution for mobile networks, exploiting the advantages of a revolutionary IP over Information-Centric Networking (IP-over-ICN) architecture in supporting flexible service provisioning through anycast and multicast, combined with the advantages of random linear coding techniques in eliminating the need for retransmissions. Our solution allows coded traffic to be disseminated in a multicast fashion during handover phase from source directly to the destination(s), without the need for an intermediate anchor as in exiting solutions; thereby, overcoming packet loss and handover failures, while reducing overall delivery cost. We evaluate our approach with an analytical and simulation model showing significant cost reduction compared to PFMIPv6.

The fast and huge increase of Internet traffic motivates the development of new communication methods that can deal with the growing volume of data traffic. To this aim, named data networking (NDN) has been proposed as a future Internet architecture that enables ubiquitous in-network caching and naturally supports multipath data delivery. Particular attention has been given to using dynamic adaptive streaming over HTTP to enable video streaming in NDN as in both schemes data transmission is triggered and controlled by the clients. However, state-of-the-art works do not consider the multipath capabilities of NDN and the potential improvements that multipath communication brings, such as increased throughput and reliability, which are fundamental for video streaming systems. In this paper, we present a novel architecture for dynamic adaptive streaming over network coding enabled NDN. In comparison to previous works proposing dynamic adaptive streaming over NDN, our architecture exploits network coding to efficiently use the multiple paths connecting the clients to the sources. Moreover, our architecture enables efficient multisource video streaming and improves resiliency to Data packet losses. The experimental evaluation shows that our architecture leads to reduced data traffic load on the sources, increased cache-hit rate at the in-network caches and faster adaptation of the requested video quality by the clients. The performance gains are verified through simulations in a Netflix-like scenario.

Information-centric networking (ICN) facilitates content identification in networks and offers parametric representation of content semantics. This paper proposes an ICN/WDM network architecture that uses these features to offer superior network utilization, in terms of performance and power consumption. The architecture introduces an ICN publish/subscribe communication approach to the wavelength layer, whereby content is aggregated according to its popularity rank into wavelength-size groups that can be published and subscribed to by multiple nodes. Consequently, routing and wavelength assignment (RWA) algorithms benefit from anycast to identify multiple sources of aggregate content and allow optimization of the source selection of light paths. A power-aware algorithm, maximum degree of connectivity, has been developed with the objective of exploiting this flexibility to address the tradeoff between power consumption and network performance. The algorithm is also applicable to IP architectures, albeit with less flexibility. Evaluation results indicate the superiority of the proposed ICN architecture, even when utilizing conventional routing methods, compared with its IP counterpart. The results further highlight the performance improvement achieved by the proposed algorithm, compared with the conventional RWA methods, such as shortest-path first fit.

Multicast data delivery can significantly reduce traffic in operators' networks, but has been limited in deployment due to concerns such as the scalability of state management. This paper shows how multicast can be implemented in contemporary software defined networking (SDN) switches, with less state than existing unicast switching strategies, by utilising a Bloom Filter (BF) based switching technique. Furthermore, the proposed mechanism uses only proactive rule insertion, and thus, is not limited by congestion or delay incurred by reactive controller-aided rule insertion. We compare our solution against common switching mechanisms such as layer-2 switching and MPLS in realistic network topologies by modelling the TCAM state sizes in SDN switches. The results demonstrate that our approach has significantly smaller state size compared to existing mechanisms and thus is a multicast switching solution for next generation networks.

Content-Centric Networking (CCN) naturally supports multi-path communication, as it allows the simultaneous use of multiple interfaces (e.g. LTE and WiFi). When multiple sources and multiple clients are considered, the optimal set of distribution trees should be determined in order to optimally use all the available interfaces. This is not a trivial task, as it is a computationally intense procedure that should be done centrally. The need for central coordination can be removed by employing network coding, which also offers improved resiliency to errors and large throughput gains. In this paper, we propose NetCodCCN, a protocol for integrating network coding in CCN. In comparison to previous works proposing to enable network coding in CCN, NetCodCCN permits Interest aggregation and Interest pipelining, which reduce the data retrieval times. The experimental evaluation shows that the proposed protocol leads to significant improvements in terms of content retrieval delay compared to the original CCN. Our results demonstrate that the use of network coding adds robustness to losses and permits to exploit more efficiently the available network resources. The performance gains are verified for content retrieval in various network scenarios.

Information-centric networking (ICN) has been proposed to cope with the drawbacks of the Internet Protocol, namely scalability and security. The majority of research efforts in ICN have focused on routing and caching in wired networks, while little attention has been paid to optimizing the communication and caching efficiency in wireless networks. In this work, we study the application of Raptor codes to Named Data Networking (NDN), which is a popular ICN architecture, in order to minimize the number of transmitted messages and accelerate content retrieval times. We propose RC-NDN, which is a NDN compatible Raptor codes architecture. In contrast to other coding-based NDN solutions that employ network codes, RC-NDN considers security architectures inherent to NDN. Moreover, different from existing network coding based solutions for NDN, RC-NDN does not require significant computational resources, which renders it appropriate for low cost networks. We evaluate RC-NDN in mobile scenarios with high mobility. Evaluations show that RC-NDN outperforms the original NDN significantly. RC-NDN is particularly efficient in dense environments, where retrieval times can be reduced by 83% and the number of Data transmissions by 84.5% compared to NDN.

In this paper, we deal with the problem of jointly determining the optimal coding strategy and the scheduling decisions when receivers obtain layered data from multiple servers. The layered data is encoded by means of prioritized random linear coding (PRLC) in order to be resilient to channel loss while respecting the unequal levels of importance in the data, and data blocks are transmitted simultaneously in order to reduce decoding delays and improve the delivery performance. We formulate the optimal coding and scheduling decisions problem in our novel framework with the help of Markov decision processes (MDP), which are effective tools for modeling adapting streaming systems. Reinforcement learning approaches are then proposed to derive reduced computational complexity solutions to the adaptive coding and scheduling problems. The novel reinforcement learning approaches and the MDP solution are examined in an illustrative example for scalable video transmission . Our methods offer large performance gains over competing methods that deliver the data blocks sequentially. The experimental evaluation also shows that our novel algorithms offer continuous playback and guarantee small quality variations which is not the case for baseline solutions. Finally, our work highlights the advantages of reinforcement learning algorithms to forecast the temporal evolution of data demands and to decide the optimal coding and scheduling decisions .

Interactive multiview video (IMV) applications offer to users the freedom of selecting their preferred viewpoint. Usually, in these systems texture and depth maps of captured views are available at the user side, as they permit the rendering of intermediate virtual views. However, the virtual views' quality depends on the distance to the available views used as references and on their quality, which is generally constrained by the heterogeneous capabilities of the users. In this context, this work proposes an IMV scalable system, where views are optimally organized in layers, each one offering an incremental improvement in the interactive navigation quality. We propose a distortion model for the rendered virtual views and an algorithm that selects the optimal views' subset per layer. Simulation results show the efficiency of the proposed distortion model, and that the careful choice of reference cameras permits to have a graceful quality degradation for clients with limited capabilities.

Intra-session network coding has been shown to offer significant gains in terms of achievable throughput and delay in settings where one source multicasts data to several clients. In this paper, we consider a more general scenario where multiple sources transmit data to sets of clients over a wireline overlay network. We propose a novel framework for efficient rate allocation in networks where intermediate network nodes have the opportunity to combine packets from different sources using randomized network coding. We formulate the problem as the minimization of the average decoding delay in the client population and solve it with a gradient-based stochastic algorithm. Our optimized inter-session network coding solution is evaluated in different network topologies and is compared with basic intra-session network coding solutions. Our results show the benefits of proper coding decisions and effective rate allocation for lowering the decoding delay when the network is used by concurrent multicast sessions.

In this work, we propose a distributed rate allocation algorithm that minimizes the average decoding delay for multimedia clients in inter-session network coding systems. We consider a scenario where the users are organized in a mesh network and each user requests the content of one of the available sources. We propose a novel distributed algorithm where network users determine the coding operations and the packet rates to be requested from the parent nodes, such that the decoding delay is minimized for all clients. A rate allocation problem is solved by every user, which seeks the rates that minimize the average decoding delay for its children and for itself. Since this optimization problem is a priori non-convex, we introduce the concept of equivalent packet flows, which permits to estimate the expected number of packets that every user needs to collect for decoding. We then decompose our original rate allocation problem into a set of convex subproblems, which are eventually combined to obtain an effective approximate solution to the delay minimization problem. The results demonstrate that the proposed scheme eliminates the bottlenecks and reduces the decoding delay experienced by users with limited bandwidth resources. We validate the performance of our distributed rate allocation algorithm in different video streaming scenarios using the NS-3 network simulator. We show that our system is able to take benefit of inter-session network coding for simultaneous delivery of video sessions in networks with path diversity.

In free viewpoint applications, the images are captured by an array of cameras that acquire a scene of interest from different perspectives. Any intermediate viewpoint not included in the camera array can be virtually synthesized by the decoder, at a quality that depends on the distance between the virtual view and the camera views available at decoder. Hence, it is beneficial for any user to receive camera views that are close to each other for synthesis. This is however not always feasible in bandwidth-limited overlay networks, where every node may ask for different camera views. In this work, we propose an optimized delivery strategy for free viewpoint streaming over overlay networks. We introduce the concept of layered quality-of-experience (QoE), which describes the level of interactivity offered to clients. Based on these levels of QoE, camera views are organized into layered subsets. These subsets are then delivered to clients through a prioritized network coding streaming scheme, which accommodates for the network and clients heterogeneity and effectively exploit the resources of the overlay network. Simulation results show that, in a scenario with limited bandwidth or channel reliability, the proposed method outperforms baseline network coding approaches, where the different levels of QoE are not taken into account in the delivery strategy optimization.

Growth codes are a subclass of Rateless codes that have found interesting applications in data dissemination problems. Compared to other Rateless and conventional channel codes, Growth codes show improved intermediate performance which is particularly useful in applications where partial data presents some utility. In this paper, we investigate the asymptotic performance of Growth codes using the Wormald method, which was proposed for studying the Peeling Decoder of LDPC and LDGM codes. Compared to previous works, the Wormald differential equations are set on nodes' perspective which enables a numerical solution to the computation of the expected asymptotic decoding performance of Growth codes. Our framework is appropriate for any class of Rateless codes that does not include a precoding step. We further study the performance of Growth codes with moderate and large size codeblocks through simulations and we use the generalized logistic function to model the decoding probability. We then exploit the decoding probability model in an illustrative application of Growth codes to error resilient video transmission. The video transmission problem is cast as a joint source and channel rate allocation problem that is shown to be convex with respect to the channel rate. This illustrative application permits to highlight the main advantage of Growth codes, namely improved performance in the intermediate loss region.

In a previous paper, Chang et al. presented a method for iterative decoding of FEC-based multiple description codes in image transmission. In this correspondence, we clarify that an outer interleaver used in the above research was previously proposed for the iterative decoding and optimization of product codes in image transmission.

In this paper, we present a pull-based dissemination protocol for efficient distribution of scalable video content in overlay peer-to-peer networks with mesh structures. The proposed protocol employs prioritized network coding, where the network coded packets belong to classes that represent packets of different priorities. For a receiver, the pull procedure begins with the reception of buffer vector messages from the senders, which bring information about the numbers and classes of available packets. The receiver node decides on the rate allocation of the different classes to be requested from each of the senders. The rate allocation is cast as a video quality maximization problem and solved using a hill-climbing algorithm. The simulation results show that the proposed mechanism, which is able to fully adapt to network dynamics, accounts for the unequal packet importances and utilizes the network resources efficiently.

We present a novel receiver-driven p2p system for delivery of multiple concurrent time constrained data streams in overlay networks. We propose an effective combination of rateless coding with intra- and inter-session network coding to efficiently exploit the path diversity in the streaming overlay. Network nodes can decide to forward rateless coded packets or to code them in intra or inter-session mode before transmission. The transmission strategy is determined based on the availability of data sources and the demands of the children nodes. Each network node solves independently a simple flow maximization problem in order to determine the optimal coding policy. The overall system is evaluated for various networks and the results outline the advantages of the proposed approach over intra-session network coding based schemes in terms of clients' satisfaction, innovative flow rate and decoding delay.

We consider a multi-source delivery system, where Raptor coding at sources and linear network coding in overlay nodes work in concert for efficient data delivery in networks with diversity. Such a combination permits to increase throughput and loss resiliency in multicast scenarios with possibly multiple sources. The network coding operations however change the degree distribution in the set of packets that reach the receivers, so that the low complexity decoding benefits of Raptor codes are unfortunately diminished. We propose in this paper to change the degree distribution at encoder, in such a way that the degree distribution after network coding operations recovers a form that leads to low complexity decoding. We first analyze how the degree distribution of the encoded symbols is altered by network coding operations and losses in a regular network. Then we formulate a geometric optimization problem in order to compute the best degree distribution for encoding at sources, such that the decoding complexity is low and close to Raptor decoders' performance. Simulations show that it is possible to maintain the low complexity decoding performance of Raptor codes even after linear network coding operations, as long as the coding at sources is adapted to the network characteristics.

Network coding permits to deploy distributed packet delivery algorithms that locally adapt to the network availability in media streaming applications. However, it may also increase delay and computational complexity if it is not implemented efficiently. We address here the effective placement of a limited number of nodes that implement randomized network coding in overlay networks, so that the goodput is kept high while the delay for decoding stays small in streaming applications. We first estimate the decoding delay at each client, which depends on the innovative rate in the network. This estimation permits to identify the nodes that have to perform coding in order to reduce the decoding delay. We then propose two iterative algorithms for selecting the nodes that should perform network coding. The first algorithm relies on the knowledge of the full network statistics. The second algorithm uses only local network statistics at each node. Simulation results show that large performance gains can be achieved with the selection of only a few network coding nodes. Moreover, the second algorithm performs very closely to the central estimation strategy, which demonstrates that the network coding nodes can be selected efficiently with help of a distributed innovative flow rate estimation solution. Our solution provides large gains in terms of throughput, delay, and video quality in realistic overlay networks when compared to methods that employ traditional streaming strategies as well as random network coding nodes selection algorithms.

We address the problem of prioritized video streaming over lossy overlay networks. We propose to exploit network path diversity via a novel randomized network coding (RNC) approach that provides unequal error protection (UEP) to the packets conveying the video content. We design a distributed receiver-driven streaming solution, where a client requests packets from the different priority classes from its neighbors in the overlay. Based on the received requests, a node in turn forwards combinations of the selected packets to the requesting peers. Choosing a network coding strategy at every node can be cast as an optimization problem that determines the rate allocation between the different packet classes such that the average distortion at the requesting peer is minimized. As the optimization problem has log-concavity properties, it can be solved with low complexity by an iterative algorithm. Our simulation results demonstrate that the proposed scheme respects the relative priorities of the different packet classes and achieves a graceful quality adaptation to network resource constraints. Therefore, our scheme substantially outperforms reference schemes such as baseline network coding techniques as well as solutions that employ rateless codes with built-in UEP properties. The performance evaluation provides additional evidence of the substantial robustness of the proposed scheme in a variety of transmission scenarios.

Network coding has been recently proposed as an efficient method to improve throughput, minimize delays and remove the need for reconciliation between network nodes in distributed streaming systems. It permits to take advantage of the path and node diversity in the network when the network coding nodes are placed efficiently. In this paper, we investigate networks consisting of nodes that autonomously determine whether they should perform network coding or not as well as their set of parent nodes. Each node makes its decisions that maximize its quality of service. The decisions include the selection of operation mode (i.e., network coding mode, simple data forwarding mode) and the selection of extra connections. The resulting interactions among the nodes are modeled as a congestion game, thereby ensuring an equilibrium, i.e., stable multimedia stream flow. The experimental results show that the proposed scheme is appropriate for distributed multimedia transmission since it provides a stable quality without imposing centralized control.

Network coding has been proposed recently as an efficient method to increase network throughput by allowing network nodes to combine packets instead of simply forwarding them. However, packet combinations in the network may increase delay, complexity and even generate overly redundant information when they are not designed properly. Typically, the best performance is not achieved when all the nodes perform network coding. In this paper, we address the problem of efficiently placing network coding nodes in overlay networks, so that the rate of innovating packets is kept high, and the delay for packet delivery is kept small. We first estimate the expected number of duplicated packets in each network node. These estimations permit to select the nodes that should implement network coding, so that the innovating rate increases. Two algorithms are then proposed for the cases where a central node is aware of the full network statistics and where each node knows the local statistics from its neighbor, respectively. The simulation results show that in the centralized scenario the maximum profit from network coding comes by adding only a few network coding nodes. A similar result is obtained with the algorithm based on local statistics, which moreover performs very close to the centralized solution. These results show that the proper selection of the network coding nodes is crucial for minimizing the transmission delay in streaming overlays.

We present a system for collaborative video streaming in wired overlay networks. We propose a scheme that builds on both rateless codes and network coding in order to improve the system throughput and the video quality at clients. Our hybrid coding algorithm permits to efficiently exploit the available source and path diversity without the need for expensive routing nor scheduling algorithms. We consider specifically an architecture where multiple streaming servers simultaneously deliver video information to a set of clients. The servers apply Raptor coding on the video packets for error resiliency, and the overlay nodes selectively combine the Raptor coded video packets in order to increase the packet diversity in the system. We analyze the performance of selective network coding and describe its application to practical video streaming systems. We further compute an effective source and channel rate allocation in our collaborative streaming system. We estimate the expected symbol diversity at clients with respect to the coding choices. Then we cast a minmax quality optimization problem that is solved by a low-cost bisection based method. The experimental evaluation demonstrates that our system typically outperforms Raptor video streaming systems that do not use network coding as well as systems that perform decoding and encoding in the network nodes. Finally, our solution has a low complexity and only requires small buffers in the network coding nodes, which are certainly two important advantages toward deployment in practical streaming systems.

This paper presents a receiver-driven video delivery algorithm that exploits a novel Randomized Network Coding (RNC) scheme for unequal error protection (UEP). The main idea of our approach is to account for the unequal importance of media packets in the network coding algorithm for efficient stream delivery in lossy overlay networks. Based on the requests from their neighbours, the network nodes properly combine packets and forward them to their children nodes. The network coding operations at every node are formulated as a log-concave optimization problem, which is solved with a greedy algorithm in only a few iterations. Our experimental results demonstrate that the proposed scheme permits to respect the priorities between the different packet classes. It further outperforms baseline network coding techniques for video streaming in overlay networks.

We investigate the problem of collaborative video streaming with Raptor network coding over overlay networks. We exploit path and source diversity, as well as basic processing capabilities of network nodes to increase the overall throughput and improve the video quality at the clients. We consider an architecture where several streaming servers simultaneously deliver video information to a set of clients. The servers apply Raptor coding on the video packets for error resiliency, and the forwarding peer nodes further combine the Raptor coded video packets in order to increase the packet diversity in the network. We find the optimal source and channel rate allocation in such a collaborative streaming system. The resulting scheme efficiently exploits the available network resources for improved video quality. The experimental evaluation demonstrates that it typically outperforms Raptor video streaming systems that do not use network coding.

This paper addresses the problem of frame interpolation for Wyner-Ziv video coding. A novel frame interpolation method based on block-adaptive matching algorithm for motion estimation is presented. This scheme enables block size adaptation to local activity within frames using block merging and splitting techniques. The efficiency of the proposed method is evaluated in transform domain Wyner-Ziv video coding. The experimental results demonstrate the superiority of the proposed method over existing frame interpolation techniques.

We propose a novel scheme for error resilient image transmission. The proposed scheme employs a product coder consisting of LDPC codes and RS codes in order to deal effectively with bit errors. The efficiency of the proposed scheme is based on the exploitation of determinate symbols in Tanner graph decoding of LDPC codes and a novel product code optimization technique based on error estimation. Experimental evaluation demonstrates the superiority of the proposed system in comparison to recent state-of-the art techniques for image transmission.

We present a novel scheme for the transmission of H.264/AVC video streams over lossy packet networks. The proposed scheme exploits the error resilient features of H.264/AVC codec and employs Reed-Solomon codes to protect effectively the streams. The optimal classification of macroblocks into slice groups and the optimal channel rate allocation are achieved by iterating two interdependent steps. Simulations clearly demonstrate the superiority of the proposed method over other recent algorithms for transmission of H.264/AVC streams

The transmission of JPEG2000 images over wireless channels is examined using reorganization of the compressed images into error-resilient, product-coded streams. The product-code consists of Turbo-codes and Reed-Solomon codes which are optimized using an iterative process. The generation of the stream to be transmitted is performed directly using compressed JPEG2000 streams. The resulting scheme is tested for the transmission of compressed JPEG2000 images over wireless channels and is shown to outperform other algorithms which were recently proposed for the wireless transmission of images.

We propose a novel scheme for error-resilient image transmission. The proposed scheme employs a product coder consisting of low-density parity check (LDPC) codes and Reed-Solomon codes in order to deal effectively with bit errors. The efficiency of the proposed scheme is based on the exploitation of determinate symbols in Tanner graph decoding of LDPC codes and a novel product code optimization technique based on error estimation. Experimental evaluation demonstrates the superiority of the proposed system in comparison to recent state-of-the-art techniques for image transmission.

A novel image transmission scheme is proposed for the communication of set partitioning in hierarchical trees image streams over wireless channels. The proposed scheme employs turbo codes and Reed-Solomon codes in order to deal effectively with burst errors. An algorithm for the optimal unequal error protection of the compressed bitstream is also proposed and applied in conjunction with an inherently more efficient technique for product code decoding. The resulting scheme is tested for the transmission of images over wireless channels. Experimental evaluation clearly demonstrates the superiority of the proposed transmission system in comparison to well-known robust coding schemes.

Ultrasound imaging allows the evaluation of the degree of emergency of a patient. However, in some instances, a well-trained sonographer is unavailable to perform such echography. To cope with this issue, the Mobile Tele-Echography Using an Ultralight Robot (OTELO) project aims to develop a fully integrated end-to-end mobile tele-echography system using an ultralight remote-controlled robot for population groups that are not served locally by medical experts. This paper focuses on the user interface of the OTELO system, consisting of the following parts: an ultrasound video transmission system providing real-time images of the scanned area, an audio/video conference to communicate with the paramedical assistant and with the patient, and a virtual-reality environment, providing visual and haptic feedback to the expert, while capturing the expert's hand movements. These movements are reproduced by the robot at the patient site while holding the ultrasound probe against the patient skin. In addition, the user interface includes an image processing facility for enhancing the received images and the possibility to include them into a database.

A novel scheme is proposed for the transmission of JPEG2000 image streams over wireless channels. The proposed scheme exploits the block-based coding structure of the JPEG2000 streams and employs optimized product codes consisting of Turbo codes and Reed-Solomon codes in order to deal effectively with burst errors. The optimization is based on information extracted directly from the compressed JPEG2000 streams. Experimental evaluation demonstrates that the proposed scheme outperform other recent algorithms for the wireless transmission of images.

A novel image transmission scheme is proposed for the communication of SPIHT image streams over wireless channels. The proposed scheme employs turbo codes and erasure-correction codes in order to deal effectively with burst errors. An algorithm for the optimal unequal error protection of the compressed bitstream is also proposed. The resulting scheme is tested for the transmission of images over wireless channels. Experimental evaluation clearly demonstrates the superiority of the proposed scheme in comparison to well-known robust coding schemes.

A novel embedded wavelet coding scheme is proposed for the transmission of images over unreliable channels. The proposed scheme is based on the partitioning of information into a number of layers which can be decoded independently provided that some important and highly protected information is initially errorlessly transmitted to the decoder. Forward error correction is used in conjunction with the error-resilient source coder for the protection of the compressed stream. Unlike many other robust coding schemes presented to date, the proposed scheme is able to decode portions of the bitstream even after the occurrence of uncorrectable errors. This coding strategy is very suitable for application with block coding schemes such as defined by the JPEG2000 standard. The proposed scheme is compared with other robust image coders and is shown to be very suitable for transmission of images over memoryless channels.

An error-resilient coding scheme is proposed for the transmission of images over unreliable channels. Forward error correction is used in conjunction with the error-resilient source coder for the protection of the compressed stream. Unlike almost all other robust coding schemes presented to date, the proposed scheme is able to decode portions of the bitstream even after the occurrence of uncorrectable errors. The resulting coder is shown to be very efficient for image transmission over noisy channels.

The decoding of convolutional codes using the list Viterbi algorithm is proposed for data hiding applications. The performance of this technique is evaluated for wavelet-domain information hiding and is shown in many cases to be advantageous in comparison to the widely used turbo codes for efficient extraction of information embedded in digital images.