This survey provides a brief introduction to compressed sensing as well as several major algorithms to solve it and its various applications to communications systems. We firstly review linear simultaneous equations as ill-posed inverse problems, since the idea of compressed sensing could be best understood in the context of the linear equations. Then, we consider the problem of compressed sensing as an underdetermined linear system with a prior information that the true solution is sparse, and explain the sparse signal recovery based on l₁ optimization, which plays the central role in compressed sensing, with some intuitive explanations on the optimization problem. Moreover, we introduce some important properties of the sensing matrix in order to establish the guarantee of the exact recovery of sparse signals from the underdetermined system. After summarizing several major algorithms to obtain a sparse solution focusing on the l₁ optimization and the greedy approaches, we introduce applications of compressed sensing to communications systems, such as wireless channel estimation, wireless sensor network, network tomography, cognitive radio, array signal processing, multiple access scheme, and networked control.

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In recent years, technologies and markets related to data centers have been rapidly changing and growing. Data centers are playing an important role in ICT infrastructure deployment and promise to become common platforms for almost all social infrastructures. Even though research has focused on networking technologies, various technologies are needed to develop high-performance, cost-efficient, and flexible large-scale data centers. To understand those technologies better, this paper surveys recent research and development efforts and results in accordance with a data center network taxonomy that the authors defined.

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A stopping criterion is an indispensable function to reduce unnecessary power consumption and decoding delay in turbo decoding. Until now, a common design philosophy in previous works has involved using the entire block of information from the MAP decoder and its input/output information to calculate the stopping index. It is an intuitive method but suffers from heavy memory requirements and high calculation complexity. In this paper, a low-complexity stopping criterion is proposed that avoids the aforementioned disadvantages. A general abstraction model is utilized to analyze the design bottleneck of stopping criteria. Instead of using an entire block of information, a compact representation derived from the internal information of the MAP decoder at a single time instant is used as a low-complexity stopping index. Theoretical explanation is provided to justify the feasibility of the proposed criterion. Simulation results show that the proposed criterion can reduce the complexity of stopping criterion dramatically while continuing to achieve the same level of performance as previous works.

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The Start-frame-delimiter (SFD) is crucial in packet-based communications system since it indicates the end of preamble and the start of a frame. In 802.15 smart utility network system, the function of SFD is extended. Two different SFDs are used for achieving frame synchronization and at the same time, to differentiate coded and uncoded packets. This paper proposes a systematic method for the selection of SFD pair for FSK based systems. The design method is adopted by 802.15.4g Task Group with the backward compatibility to IEEE 802.15.4d systems taken into account. Four selection criteria are specified and discussed to determine the pair of sequences providing least packet error detection rate with the consideration of eliminating the adjacent channel image signal commonly occurred in the low IF receiver. The probability of false alarm and miss detection is analyzed to verify the effectiveness of the proposed selection method. The simulation results confirm that error frame detection rate of 1e-3 can be achieved with selected SFD sequences. The proposed method for selection of SFD pairs ensures a robust packet header and thus better payload protection. The SFD design approach is applicable to other packet-based wireless communication FSK systems with the support of more than one SFD sequence.

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This paper presents a novel decision feedback equalizer (DFE) with block delay detection for the joint transceiver design that uses channel state information (CSI). The block delay detection in the proposed DFE offers a degree of freedom for optimizing the precoder of the transmitter, provided the transmission power is constrained. In the proposed DFE, the feedforward matrix is devised to enable a block-based equalizer that can be cooperated with an intrablock decision feedback equalizer for suppressing the intersymbol interference (ISI) for the transmitted block with a certain block delay. In this design, the interblock interference (IBI) for the delay block is eliminated in advance by applying the recently developed oblique projection framework to the implementation of the feedforward matrix. With knowledge of full CSI, the block delay and the associated block-based precoder are jointly designed such that the average bit-error-rate (BER) is minimized, subject to the transmission power constraint. Separate algorithms are derived for directly determining the BER-minimized block delays for intrablock minimum mean-squared error (MMSE) and zero-forcing (ZF) equalization criteria. Theoretical derivations indicate that the proposed MMSE design simultaneously maximize the Gaussian mutual information of a transceiver, even under the cases of existing IBI. Simulation results validate the proposed DFE for devising an optimum transceiver with CSI, and show the superior BER performance of the optimized transceiver using proposed DFE. Relying on analytic results and simulation cases also builds a sub-optimum MMSE design of the proposed DFE using the BER-minimized block delay for ZF criterion, which exhibits almost identical BER performance as the proposed MMSE design in most of the signal-to-noise ratio (SNR) range.

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With the fast development of mobile communication technologies, mobile multimedia services like mobile Video on Demand (VOD) are becoming prevalent. However, VOD streaming requires dedicated bandwidth to satisfy Quality of Service (QoS), and the limited wireless bandwidth will become insufficient to support the increasing number of mobile VOD users. In order to solve the problem, this paper proposes a Call Admission Control (CAC) approach which can accept new users even when the system bandwidth is insufficient. Our approach also guarantees continuous playback for subscribers by taking into account the service end time and the delay bound of the users. Simulation results demonstrate that the proposed scheme can increase the number of concurrent users and reduce the connection blocking probability significantly without playback interruption.

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This paper describes Path-Moose, a scalable tree-based shortest path bridging protocol. Both ARP-Path and Path-Moose protocols belong to a new category of bridges that we name All-path, because all paths of the network are explored simultaneously with a broadcast frame distributed over all network links to find a path or set a multicast tree. Path-Moose employs the ARP-based low latency routing mechanism of the ARP-Path protocol on a bridge basis instead of a per-single-host basis. This increases scalability by reducing forwarding table entries at core bridges by a factor of fifteen times for big data center networks and achieves a faster reconfiguration by an approximate factor of ten. Reconfiguration time is significantly shorter than ARP-Path (zero in many cases) because, due to the sharing of network paths by the hosts connected to same edge bridges, when a host needs the path it has already been recovered by another user of the path. Evaluation through simulations shows protocol correctness and confirms the theoretical evaluation results.

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To support mobility, multihoming, routing scalability, and security, there are a lot of proposals based on ID/Locator split approach not only for the current Internet but also for the future Internet. However, none of them meet the requirements for practical operation such as (1) support heterogeneous network layer protocols, (2) scalability of ID/Locator mapping system, (3) independence of mapping information management, and (4) avoidance of locator leakage beyond the administrative boundary. This paper proposes a network layer protocol called Z Network Protocol (ZNP) for the future Internet based on the clean slate approach. ZNP supports heterogeneity of network layer protocols by “Internetworking with a Common ID Space”. Its mapping systems meet the requirements (1)-(4) described above. For manipulating the mapping systems, Z Control Message Protocol (ZCMP) is designed. For resolving the link layer (L2) address from the ZNP Locator, Z Neighbor Discovery Protocol (ZNDP) is designed. We implement ZNP and ZNDP in the Linux kernel, ZCMP in the user space and measure the times needed for transmission, reception, forwarding, and locator conversion. The results show the practicability of ZNP as a network layer protocol for the future Internet.

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In many wireless sensor applications, skyline monitoring queries that continuously retrieve the skyline objects as well as the complete set of nodes that reported them play an important role. This paper presents SKYMON, a novel energy-efficient monitoring approach. The basic idea is to prune nodes that cannot yield a skyline result at the sink, as indicated by their (error bounded) prediction values, to suppress unnecessary sensor updates. Every node is associated with a prediction model, which is maintained at both the node and the sink. Sensors check sensed data against model-predicted values and transmit prediction errors to the sink. A data representation scheme is then developed to calculate an approximate view of each node's reading based on prediction errors and prediction values, which facilitates safe node pruning at the sink. We also develop a piecewise linear prediction model to maximize the benefit of making the predictions. Our proposed approach returns the exact results, while deceasing the number of queried nodes and transferred data. Extensive simulation results show that SKYMON substantially outperforms the existing TAG-based approach and MINMAX approach in terms of energy consumption.

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Recently, the number of users downloading video content on the Internet has dramatically increased, and it is highly anticipated that downloading huge size, rich content such as movie files will become a popular use of the Internet in the near future. The transmission bandwidth consumed by delivering rich content is enormous, so it is urgent for ISPs to design an efficient delivery system that minimizes the amount of network resources consumed. To deliver web content efficiently, a content delivery network (CDN) is often used. CDN providers collocate a huge number of servers within multiple ISPs without being informed of detailed network information, i.e., network topologies, from ISPs. Minimizing the amount of network resources consumed is difficult because a CDN provider selects a server for each request based on only rough estimates of response time. Therefore, an ordinary CDN is not suited for delivering rich content. P2P-based delivery systems are becoming popular as scalable delivery systems. However, by using a P2P-based system, we still cannot obtain the ideal delivery pattern that is optimal for ISPs because the server locations depend on users behaving selfishly. To provide rich content to users economically and efficiently, an ISP itself should optimally provide servers with huge storage capacities at a limited number of locations within its network. In this paper, we investigate the content deployment method, the content delivery process, and the server allocation method that are desirable for this ISP-operated CDN. Moreover, we evaluate the effectiveness of the ISP-operated CDN using the actual network topologies of commercial ISPs.

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We address the global asymptotic stability of FAST TCP, especially considering cross traffics, time-varying network feedback delay, and queuing delay dynamics at link. Exploiting the inherent dynamic property of FAST TCP, we construct two sequences that represent the lower and upper bound variations of the congestion window in time. By showing that the sequences converge to the equilibrium point of the congestion window, we establish that FAST TCP in itself is globally asymptotically stable without any specific conditions on the tuning parameter α or the update gain γ.

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This paper presents a basic investigation of the power imbalance problem with regard to maximum ratio combining (MRC) array antennas for digital TV broadcast reception. First, the relationship between the decrease in the diversity gain and reduction in the received power was investigated using two-element and four-element dipole array antennas by means of a Monte Carlo simulation. The relationship between the decrease in the diversity gain and the number of branches imposed to reduce the received power was also investigated. Then, a simple method of predicting the reduction in the diversity gain under imbalanced power conditions is given using the simulation results. The objective is to determine a criterion associated with the gain reduction that allows us to achieve the required system performance. Finally, the proposed method is confirmed by analysis using a model representing a typical portable digital broadcasting TV set held with both hands that simulates the power imbalance condition.

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Known an a criterion that solves the trade-off between fairness and efficiency, proportional fairness is well-studied in cellular networks in the Qualcomm High Data Rate System. In multi-hop wireless networks, proportional fairness is solved by maximizing the logarithmic aggregate utility function. However, this approach can deal with instantaneous rates only where long term fairness is to be targeted. In this case, cumulative rates are more suitable. This paper proposes a framework for multi-hop wireless networks to guarantee fairness of cumulative data rates. The framework can be extended to other kinds of fairness such as max-min fairness, and to more complex networks, multi-channel multi-radio wireless networks.

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In this paper, a modified detection approach based on minimum mean-square error (MMSE) sorted QR decomposition (SQRD) for turbo coded multiple-input multiple-output orthogonal frequency division multiplexing (MIMO-OFDM) systems is presented. In conventional MMSE SQRD based detection, a component of the symbol to be detected is still present in the interference term, leading to less efficient log likelihood ratios (LLRs) passed to soft decoder. By moving this symbol component into the signal term, the modified MMSE SQRD based detection can provide more efficient LLRs for soft decoder. Simulation results show that the proposed detection can achieve significant performance gain.

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This paper presents a joint linear processing scheme for two-hop and half-duplex distributed amplify-and-forward (AF) relaying networks with one source, one destination and multiple relays, each having multiple antennas. By using the minimum mean-square error (MMSE) criterion and the Wiener filter principle, the joint relay and destination design with perfect channel state information (CSI) is first formulated as an optimization problem with respect to the relay precoding matrix under the constraint of a total relay transmit power. The constrained optimization with an objective to design the relay block-diagonal matrix is then simplified to an equivalent problem with scalar optimization variables. Next, it is revealed that the scalar-version optimization is convex when the total relay power or the second-hop SNR (signal to noise ratio) is above a certain threshold. The underlying optimization problem, which is non-convex in general, is solved by complementary geometric programming (CGP). The proposed joint relay and destination design with perfect CSI is also extended for practical systems where only the channel mean and covariance matrix are available, leading to a robust processing scheme. Finally, Monte Carlo simulations are undertaken to demonstrate the superior MSE (mean-square error) and SER (symbol error rate) performances of the proposed scheme over the existing relaying method in the case of relatively large second-hop SNR.

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In this paper, through extrinsic information transfer (EXIT) band chart analysis, an adaptive iterative decoding approach (AIDA) is proposed to reduce the iterative decoding complexity and delay for finite-length differentially encoded Low-density parity-check (DE-LDPC) coded systems with multiple-symbol differential detection (MSDD). The proposed AIDA can adaptively adjust the observation window size (OWS) of the MSDD soft-input soft-output demodulator (SISOD) and the outer iteration number of the iterative decoder (consisting of the MSDD SISOD and the LDPC decoder) instead of setting fixed values for the two parameters of the considered systems. The performance of AIDA depends on its stopping criterion (SC) which is used to terminate the iterative decoding before reaching the maximum outer iteration number. Many SCs have been proposed; however, these approaches focus on turbo coded systems, and it has been proven that they do not well suit for LDPC coded systems. To solve this problem, a new SC called differential mutual information (DMI) criterion, which can track the convergence status of the iterative decoding, is proposed; it is based on tracking the difference of the output mutual information of the LDPC decoder between two consecutive outer iterations of the considered systems. AIDA using the DMI criterion can adaptively adjust the out iteration number and OWS according to the convergence situation of the iterative decoding. Simulation results show that compared with using the existing SCs, AIDA using the DMI criterion can further reduce the decoding complexity and delay, and its performance is not affected by a change in the LDPC code and transmission channel parameters.

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This paper proposes an enhanced feature detection method for the OFDM signals of digital TV (DTV) standards, namely Digital Video Broadcasting-Terrestrial (DVB-T) and Integrated Services Digital Broadcasting-Terrestrial (ISDB-T). The proposed method exploits property of time-domain sliding correlation results of DTV signals with the pilots that are inserted into OFDM symbols. Some correlation outputs are much larger than the remaining outputs and are called correlation peaks here, and, the distance between their positions in the correlation output sequence keep constant regardless of the received DTV timings. The proposed method then derives sensing test statistic with improved SNR by aggregating the correlation peaks based on their positions. Performance of the proposed method is evaluated by both computer simulation and hardware implementation. Simulation results for DVB-T detection verify that compared to the optimal conventional sensing method, the proposed method achieves superior sensing performance. It reduces sampling time by about 25% for the same sensing performance while increasing computational complexity by around 0.0001%. Hardware performance further verifies that the proposed method is able to accurately detect ISDB-T at the low SNR of -14.5dB by employing 8 OFDM symbol durations of samples.

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For opportunistic spectrum access (OSA), spectrum management is a key function to effectively utilize white space without causing harmful interference to incumbent receivers. Geo-location database approaches using radio propagation estimation have been regarded as practical spectrum management methods. However, propagation models inevitably fail to accurately estimate the path loss in actual radio environments, resulting in estimation error of carrier to interference ratio (CIR) of the incumbent receivers. This could prevent white space from being efficiently utilized, because the allowable transmit power of the opportunistic system has to be limited to keep the CIR at the required level. To improve the accuracy of CIR estimation, we propose the new concept of Interference Monitoring which works in combination with spectrum management. In this method, a monitoring node located near the incumbent receivers actually measures both the interference signals and the incumbent signals. Using the measurement results, the CIR estimates are corrected based on the minimum mean square error (MMSE) criterion. The proposed Interference Monitoring can be extended to establish cooperation among multiple monitoring nodes and thus spatial diversity. Analytical evaluations assuming a simple cellular system model show that Interference Monitoring can more accurately estimate CIR, and thus it can significantly increase the allowable transmit power. For an urban macro cell, Interference Monitoring with a single node achieved about a 6.5dB increase in the transmit power; Cooperative Interference Monitoring with 4 nodes achieved about a 13.5dB increase. Thus, Interference Monitoring-based spectrum management can maximize opportunities for white space utilization without imposing additional interference to the incumbent system.

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We examine the relationship between 116VHF sensor events recorded by the VHF sensor on the Maido-1 satellite and lightning strokes detected by the World Wide Lightning Location Network (WWLLN) to show that most of the VHF sensor events were caused by lightning discharges. For each VHF sensor event, the WWLLN events within 1400km from the subsatellite point and within 1sec, 30sec, and 300sec of the VHF sensor trigger time are analyzed. We find that the coincidence rates in the North and South American continents, and in Southeast/East Asia and the Australian continent are greater than 0.90. Those in the African and European continents, and in the Pacific and Atlantic Oceans are less than 0.61. These high enough coincidence rates indicate that the VHF sensor events were emitted from lightning, although the coincidence rates in the other regions are quite low because of the low detection efficiency of the WWLLN in the regions. We also focus on 6 coincident events measured by both the VHF sensor and the WWLLN. The incidence angles of the EM waves for the VHF sensor are estimated from the group delay characteristics of the recorded EM waveforms. Compared with the WWLLN lightning locations, the two incidence angles are temporally and spatially coincident. These results indicate that a large fraction of the VHF sensor events are emitted by lightning discharges.

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We show an improved throughput scaling law for an ultra-wide band (UWB) ad hoc network by using a modified hierarchical cooperation (HC) strategy; the n wireless nodes are assumed to be randomly sited. In a dense network of unit area, our result indicates that the derived throughput scaling depends on the path-loss exponent α for certain operating regimes due to the power-limited characteristics. It also turns out that the use of HC is helpful in improving the throughput scaling of our UWB network in some conditions. More specifically, assuming that the bandwidth scales faster than n^α+1(log n)^α/2, it is shown that the HC protocol outperforms nearest multi-hop routing for 2<α<3 while using nearest multi-hop routing leads to higher throughput for α≥3.

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Many broadcast encryption schemes have been proposed for conventional networks. However, those schemes are not suitable for wireless sensor networks, which have very limited resources such as communication, computation, and storage. In this paper, we propose an efficient and practical identity-based broadcast encryption scheme for sensor networks by exploiting the characteristics of sensor networks: in the deployment stage, the set of neighboring sensor nodes are determined and most communications are conducted among the neighbors due to radio power limitations of the nodes. The proposed scheme features the following achievements: (1) all of the public keys and private keys are of constant size; (2) it satisfies all the security requirements for sensor networks. The proposed scheme is optimal in the sense that it requires no pairing operation when adopting pre-computation.

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In our previous work [2], we proposed a new concept of utility functions for rate control in communication networks. Unlike conventional utility-based rate control in which the utility function of each user is defined as a function of its transmitting data rate, in [2], we defined the utility function of each user as a function of not only its transmitting data rate but also it receiving data rate. The former is called a session-level utility function and the latter is called a user-level utility function. The user-level utility function reflects the satisfaction with the service of a user with two-way communication, which consists of transmitting and receiving sessions, better than the session-level utility function, since user's satisfaction depends on not only the satisfaction with its transmitting session but also that for its receiving session. In [2], an algorithm that required each user to know the exact utility function of its correspondent was developed. However, in some cases, this information might not be available due to some reasons such as security and privacy issues, and in such cases, the algorithm developed in [2] cannot be used. Hence, in this paper, we develop a new distributed algorithm that does not require each user to know the utility function of its correspondent. Numerical results show that our new algorithm, which does not require the utility information of the correspondent, converges to the same solution to that with the algorithm that requires the utility information of the correspondent.

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We evaluate the rich-club property of the Internet topology at the autonomous system (AS) level by comparing the Internet AS graphs of traceroute and BGP, and the synthetic graphs of PFP model. The results indicate that, for rich-club coefficient, PFP model can exactly match traceroute AS graphs in the early years around 2002, but it has significantly deviated from the grown AS graphs since about 2010.

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This letter proposes a new mechanism that supports adaptive sending control using Real-Time Streaming Protocol (RTSP) and Transmission Control Protocol (TCP) for IPTV service over heterogeneous networks. The proposed mechanism is implemented on a mobile IPTV device and its performance is verified for providing seamless television watching in heterogeneous networks, even when in motion.

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This letter proposes a low-complexity soft-detection algorithm for modified dual-carrier modulation (MDCM) in WiMedia ultra-wideband (UWB) systems. In order to reduce the complexity of soft-output maximum-likelihood detection (soft-MLD), which gives the optimal performance for MDCM symbols, the proposed algorithm utilizes the following three methods: real/imaginary separation, multiplierless distance calculation, and candidate set reduction. Through these methods, the proposed algorithm reduces the complexity of soft-MLD by 97%, while preventing the deterioration of its optimality. The performance of the proposed algorithm is demonstrated by simulations of 640-1024Mbps transmission modes of the latest Release 1.5 standard of the WiMedia UWB.

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In this study, 2-location distance-based registration (2DBR) is proposed to improve the performance of traditional distance-based registration. In distance-based registration, when a mobile station (MS) enters a new cell, the MS calculates the distance from the last registered cell and registers its location if the calculated distance reaches a prescribed distance threshold D. In 2DBR, an MS stores not only the last registered location area (LA) but also the second-to-last LA, and then no registration is performed when the MS crosses the two stored LAs. The 2DBR may increase paging cost but it may decrease registration cost. Simulation results show that our proposed 2DBR outperforms current distance-based registration in most cases.

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Among the cognitive radio technologies, cooperative spectrum sensing has been corroborated to be an effective approach to counter channel fading. Recent research about it is mainly with the assumption that secondary users (SUs) are synchronous with primary users (PUs). In this letter, we discuss the asynchronous situation for the first time, which means SUs have no idea about the communication time table of PUs' network. Based on the ON/OFF channel model, we derive the detection and false alarm probabilities, and the optimal sensing parameters under such asynchronous scenario. Simulation results are shown in the end.

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