A joint channel estimation and channel coding scheme is presented for channels with memory using polar codes. Unlike the conventional approach of first estimating all channel parameters and then performing channel decoding separately, the proposed scheme incorporates a subset of reliable estimates of channel parameters into the decoding procedure and computes decoding metrics averaged over the statistical behavior of the channel. Specifically, decoding algorithms for finite-state Markov channels of any order, for the Gauss-Markov channel and for flat-fading channels are presented. Further, by adapting list decoding to identify reliably-decoded bits within a codeword, channel estimation and decoding steps are performed iteratively to boost the reliability of channel estimation as well as error correction. In order to improve the performance even further, a new pilot arrangement scheme is developed that utilizes the structure of polar codes and sends pilot symbols embedded within the polar codewords. This construction can be viewed as a new family of shortened polar codes that can be of independent interest. Simulation results demonstrate the benefit of the proposed approach compared to existing solutions.

With the growing popularity of delay-sensitive applications (e.g., real-time conversational video, online gaming, and augmented reality) and future trends toward ultra-reliable low-latency communications such as the tactile Internet, performance analysis of wireless systems under the finite code blocklength constraint becomes extremely important. In this paper, we investigate the maximum achievable throughput of incremental redundancy-hybrid automatic repeat request (IR-HARQ) over the (correlated) Rayleigh fading channel under finite blocklength and delay-violation probability constraints as a function of the modulation scheme. The maximum number of HARQ rounds together with the transport block size specifies the underlying coding latency of the IR-HARQ scheme. A framework, namely the HARQ Markov model (HARQ-MM), is introduced to track the throughput and the probability of error of IR-HARQ over the Rayleigh fading channel as a function of the modulation scheme. The dispersion of parallel additive white Gaussian noise channels with finite input alphabets (e.g., pulse amplitude modulation) is analytically characterized. It is used to identify the state transition probabilities of the underlying HARQ-MM. An algorithm is developed to efficiently compute the steady-state distribution of the HARQ-MM. Extensive performance evaluation is conducted, which shows a good match between the throughput performance characterized by the theoretical framework and that achieved by the practical channel codes.

A joint channel estimation and channel coding scheme is presented for finite-state Markov channels using polar codes. Unlike the conventional approach of first estimating all channel parameters and then performing channel decoding separately, the proposed scheme incorporates a subset of reliable estimates of channel parameters into the decoding algorithm and computes decoding metrics averaged over the statistical behavior of the Markov channel. By adapting list decoding (without any inner code), channel estimation and decoding steps can be performed iteratively to boost the reliability of channel estimation as well as error correction. In order to improve the performance even further, a new pilot transmission scheme is developed that utilizes the structure of polar codes and sends pilot symbols along with code symbols. This construction can be viewed as a new family of shortened polar codes that can be of independent interest. Simulation results demonstrate the benefit of the proposed approach compared to existing solutions.

In a massive machine type communication (massive MTC) network, wireless connections between machine type devices (MTDs) and eNodeBs are unreliable due to the interference caused by uncoordinated access of other MTDs. Furthermore, it is with a high probability that the retransmission from the outage source MTD will fail again due to the spatiotemporally correlated interference. In this paper, we design and analyze a location-based cooperative strategy to improve the performance of massive MTC networks. In the cooperative strategy, an inactive MTD is selected as a relay if it has successfully decoded the packet and if it is located within a circular area around the eNodeB. Considering the spatial and temporal correlation of interference, the outage probability of the designed cooperative strategy is derived using stochastic geometry. Both the simulation and numerical results demonstrate that spatiotemporal correlation of interference significantly affects the performance analysis of cooperative massive MTC networks and our designed cooperative strategy can significantly reduce the outage probability compared to conventional retransmission.

For DFT-spread-OFDM or OFDM, if the delay spread varies in a wide range and the symbol duration is relatively short, adapting the cyclic prefix (CP) duration rather than using a fixed one may significantly improve the spectral efficiency while preventing inter-symbol interference (ISI). In practice, it may be beneficial to have a constant overall DFT-spread-OFDM/OFDM symbol time, which is the sum of the duration of a CP and the duration of a data portion. We propose to adapt the CP duration to the delay spread without changing the overall symbol time for DFT-spread-OFDM or OFDM, and address implementation challenges. In particular, we propose changing the clocking rate of ADC and DAC or using a Farrow filter to reduce the computational complexity of arbitrary-size DFT/IDFT resulting from the adaptation.

In wireless networks such as those based on IEEE 802.11, packet losses due to fading and interference are often misinterpreted as indications of congestion by the congestion control protocol at higher layers, causing an unnecessary decrease in the data sending rate. For delay-constrained applications such as video telephony, packet losses may result in excessive artifacts or freeze in the decoded video. We propose a simple and yet effective mechanism to detect and reduce channel-caused packet losses by adjusting the retry limit parameter of the IEEE 802.11 protocol while taking into account the delay requirement of the traffic. Since the retry limit is left configurable in the IEEE 802.11 standard and does not require cross-layer coordination, our scheme can be easily implemented and incrementally deployed. Experimental results of applying the proposed scheme to a WebRTC-based real-time video communication prototype show significant performance gain compared to the case where the retry limit is configured statically.

This paper presents an end-to-end algorithm for detecting a shared congested router in the internet when communication is established using multiple paths between wireless devices endowed with multiple interfaces (e.g., LTE, and WiFi). The proposed technique is based on one way end-to- end delay observation of packets received through multiple paths at the end hosts to infer whether or not the flows share the same congested router. With very different statistics on the wireless access delays encountered across different interfaces in practical scenarios, this paper describes techniques for estimating queuing delay variations for each path from one way end-to-end delay observations. The proposed scheme does not make use of exclusive probing packets, but instead uses the data packets that are transmitted and received through multiple paths to infer the path's one way delay characteristics. Simulation results assessing the performance of our proposed detection algorithm are provided for scenarios where wireless devices exchange traffic through its LTE and WiFi interface. With the recently proposed protocols such as multi path transmission control protocol (MPTCP), and multi path real time protocol (MPRTP) beginning to be embraced with increasing interests, this study becomes important as it enables the end hosts to perform efficient congestion control.

The human visual system (HVS) cannot perceive spatial frequency components in an image that are above a certain limit, the value of which is affected by factors such as the viewing distance. This has been exploited to improve the video coding efficiency by first filtering out redundant frequency components and then doing conventional video encoding. To facilitate the deployment of this scheme, we propose to implement it in a network entity such as a Multipoint Control Unit (MCU). Specifically, by analyzing the video sent from a client, the MCU infers that client's viewing conditions, which are then used to adapt the encoding of the video destined to that client. The scheme is implemented in a real-world video teleconferencing system. Experimental results show that our approach can result in significant savings in bandwidth without affecting subjective video quality.

Recently HTTP based adaptive video streaming has gained significant attention. A video client predicts the future available bandwidth and then selects the appropriate bit rate for the video content in order to maximize the video quality while avoiding rebuffering. The often severe fluctuation in the available bandwidth in mobile wireless networks poses a challenge to the bit rate selection algorithm. In this paper, we propose to utilize the regularity in a user's movement to improve the bit rate selection algorithm. We use Kalman filters to predict a user's speed and available bandwidth, which are then fed into a bit rate adaptation algorithm. The proposed method is implemented in JavaScript, and tested in real LTE networks. Experimental results show that our method significantly improves the quality of DASH-based video streaming.

This paper presents an Active Queue Management (AQM) algorithm for improving the Quality of Experience (QoE) of video telephony over packet switched networks. The algorithm exploits the characteristics of the video coding structure, and builds on the Controlled Delay (Codel) active queue management algorithm recently proposed by Nichols and Jacobson to address the prevalent ‘bufferbloat’ problem in the current Internet. The proposed algorithm, Loss Concentration based controlled Delay (LC-Codel), maintains low queuing delay which is essential for video telephony, while using loss concentration to improve video QoE. Simulation results show significant gains in QoE with negligible impact on cross traffic.

It has been shown that various statistics of the peak signal-to-noise ratio (PSNR) time series of a video sequence can be used to construct fairly accurate Quality of Experience (QoE) models. To predict QoE, it is sufficient to predict the PSNR time series. The possibility of predicting QoE further enables QoE-based network resource allocation. We propose two approaches to packet-based prediction of PSNR time series to overcome the limitations of frame-based approaches. The proposed first approach adopts a parametric model for the impact on the video quality due to losing a packet, while the second proposed approach is parameter-free. Simulation results show that both approaches significantly outperform the simple mean or median algorithms and are close to their respective performance bounds.

In wireless networks such as those based on IEEE 802.11, packet losses due to fading and interference are often misinterpreted as indications of congestion, causing unnecessary decrease in the data sending rate due to congestion control at higher layer protocols. For delay-constrained applications such as video teleconferencing, packet losses may result in excessive artifacts or freeze in the decoded video. We propose a simple and yet effective mechanism to detect and reduce channel-caused packet losses by adjusting the retry limit parameter of the IEEE 802.11 protocol. Since retry limit is left configurable in the IEEE 802.11 standard, and does not require cross-layer coordination, our scheme can be easily implemented and incrementally deployed. Experimental results of applying the proposed scheme to a WebRTC-based realtime video communication prototype show significant performance gain compared to the case where retry limit is configured statically.

In the underlay cognitive radio approach, secondary users (SUs) are allowed to transmit their messages as long as the negative impact on the performance of the primary user (PU) is below a certain threshold. We consider a MIMO underlay cognitive radio network with a single PU and multiple SUs. We propose a practical interference alignment and cancellation scheme that not only avoids interference at the PU, but also optimizes SUs' performance in terms of degrees of freedom. Given any SU, the interference from the other SUs is aligned, and the interference at the SU from the PU is cancelled in the spatial domain. We give the feasibility condition for achieving interference alignment and cancellation. Due to the NP-hardness of the problem, we resort to approximation algorithms. We also consider practical issues for implementation. The effectiveness of the proposed scheme is shown by simulation results.

The ultimate goal of network resource allocation for video teleconferencing is to optimize the Quality of Experience (QoE) of the video. We consider the IPPP video coding structure with macroblock intra refresh, which is widely used for video teleconferencing. Generally, the loss of a current frame causes error propagation to subsequent frames due to the video coding structure. Therefore, to optimize the QoE, a communication network needs to be able to accurately predict the consequence of each of its resource allocation decisions. We propose a QoE prediction scheme by considering QoE models that use the per-frame PSNR time series as the input, thus reducing the QoE prediction problem to a per-frame PSNR prediction problem. The QoE prediction scheme is jointly implemented by the video sender (orMCU) and the communication network. Simulation results show that the proposed per-frame PSNR prediction method is fairly accurate, with an average error well below 1dB.

This paper considers a downlink LTE system where a basestation (eNodeB) is serving video traffic (generated possibly due to video teleconferencing applications) to many users in a cell. The video traffic is separated into multiple sub-streams (logical channels) based on the coding structure or priority of video packets. The objective is to maximize video quality by servicing appropriate video sub-streams and users while taking into account the resource constrained wireless channel. It is shown that one can obtain significant gains in video quality by determining the quality of service (QoS) parameters on a per sub-stream basis, rather than in `user-based' approach where all the sub-streams are lumped into one single stream. Motivated by video teleconferencing applications, we provide simulation results for the case where video traffic is separated into logical channels based on a Hierarchical-P encoding structure. Furthermore, we demonstrate the gains in video quality that can be obtained by allocating resources across logical channels. With the exploding growth in mobile video traffic expected to happen in 4G systems, this study suggests potential benefits of a logical channel based approach as compared to conventional user-based schemes.

The existing dynamic spectrum access network architecture fails to offer incentives to primary users (PUs). Therefore, PUs are unwilling to cooperate with secondary users (SUs) and intend to set up stringent requirements on SUs' spectrum access. This diminishes the benefit of dynamic spectrum access and hinders its commercialization success. In this paper, we propose a novel architecture for dynamic spectrum access networks, termed incentivized cooperative dynamic spectrum access network (IC-DSAN), to motivate PUs to cooperate with SUs, such that both PUs and SUs achieve significantly higher performance. In an IC-DSAN, SU nodes serve as relays between PU nodes and while relaying PU packets, utilize network coding to encode SU packets onto PU packets, i.e., SU packets get a `free ride' via network coding. To further improve the performance, we propose a new network coding scheme termed coding over coded packets. At last, an optimization model is developed to analyze the throughput gain of IC-DSAN. The performance evaluation indicates that the throughputs of both PUs and SUs significantly increase.

Energy detection is widely used by cognitive radios for spectrum sensing. During a silent period, secondary users (SUs) are kept silent so that the energy detector does not confuse SU signals for primary user (PU) signals. Due to imperfect coordination, an SU may transmit during a silent period and cause possible false alarms. We propose to leverage matched filters that already exist in many SUs to alleviate the impact of such SU interference by combining the matched filtering result and the energy detection result. The analysis shows that for practical purposes, our algorithm virtually eliminates all of the negative impact of SU interference with only negligible penalty in delay and energy consumption.

The overlay approach to dynamic spectrum access recently proposed in information theory allows both primary users (PUs) and secondary users (SUs) to simultaneously access the same spectrum with comparable power levels while ensuring no degradation to the performance of PUs. However, this approach is based on a number of idealized assumptions that are difficult to satisfy in practice, and existing efforts to address this issue fall outside physical layer processing. In this paper, we propose a number of physical layer mechanisms to make the overlay approach practical. In particular, we leverage the broadcast nature of the wireless medium and the latest breakthrough in full-duplex radios to resolve the synchronization problem and to get around the non-causal assumption while naturally offering delay diversity.

The overlay approach to dynamic spectrum access recently proposed in information theory allows both primary users (PUs) and secondary users (SUs) to simultaneously access the same spectrum with comparable power levels. However, the information theory approach is based on idealized assumptions that are difficult to be satisfied in practice. We propose a practical scheme with the overlay flavor for multi-hop wireless networks subject to two challenging design constraints: (1) PUs can not be modified, and (2) the performance of PUs can not degrade. Using the notion of credit in our scheme, SUs first help with the traffic delivery for PUs, and in return SUs later are allowed to access the spectrum in a manner disruptive to PUs. Specifically, SUs earn credit for helping PUs and consume credit for disrupting PUs. Moreover, we propose a distributed credit diffusion algorithm for SUs to share credit, and we modify a widely used reactive routing protocol to satisfy the design constraints. This overlay scheme is then integrated with the widely used interweave scheme to form the Credit-based Overlay and Interweave (CONI) protocol. Simulation results show that CONI can significantly benefit both PUs and SUs.

In this paper, we present a rendezvous scheme for dynamic spectrum access (DSA) networks that can achieve approximately optimal throughput while not relying on a common control channel. Compared with existing rendezvous schemes for DSA networks that use a common control channel, the proposed scheme avoids congestion and jamming of the control channel, and also reduces control complexity and overhead in DSA. We prove that the proposed rendezvous scheme achieves approximately optimal throughput, and analyze the convergence time of this scheme to obtain the approximately optimal throughput.

In dynamic spectrum access (DSA), secondary users (SUs) dynamically search and access spectrum bands unused by primary users to communicate. We propose a DSA scheme where it does not require a control channel for coordination and SU nodes do not need to exchange control messages to rendezvous. Every SU node selects its operational channel independently. When a node wants to rendezvous with another node, the former estimates the operational channel of the latter. The scheme ensures that such rendezvous has a high probability of success, while operational channels are diversified to reduce co-channel interference and hence increase throughput. We develop a mathematical model to analyze the performance of the scheme. Both analytical and simulation results show that the scheme achieves very good performance.

The dynamic spectrum access (DSA) capability of cognitive radio networks (CRN) promises to resolve both the spectrum scarcity and the low spectrum utilization problems caused by today's static spectrum access (SSA) policy. With DSA, CRN nodes search the dynamically accessible spectrum bands for communication. In this paper, we study a random DSA scheme, where each node randomly selects its operating band based on the locally detected accessible spectrum bands. This scheme does not need the coordination or exchange of control messages to select a communication band between a sender and a receiver, and is desirable in certain scenarios. We analyze the performance of this random DSA scheme. Numerical results show that the random DSA scheme can achieve 60% of the theoretical maximum performance.

Dynamic spectrum access (DSA) promises to resolve spectrum scarcity and low spectrum utilization caused by today's static spectrum access (SSA) policy. In DSA, secondary users dynamically search and access spectrum bands that are temporarily unused by primary users. In this paper, we propose a control-free DSA algorithm for cognitive radio networks (CRN). Our algorithm enables each CRN node to select its operation band without coordination and exchange of control messages with neighbors. The contribution of our algorithm is that such an independently selected band can reach neighbors with high probability, while streamlining control complexity and overhead in DSA. We develop an analytical model to evaluate performance. Numerical results show that our control-free DSA algorithm can achieve very good performance.

The paper focuses on the queue size estimation problem in random early detection (RED) gateways. Queue size estimation plays a critical role in gateways' packet dropping/marking decisions. Conventional RED gateways use exponentially weighted moving averages (EWMA) to estimate the queue size. These IIR filters require very small weights in order to avoid nonlinear instabilities and accommodate transient congestion. Small weights, however, lead to failure of gateways to track rapid queue size depletion closely and thus causes link under utilization. We use adaptive weighted median filters for queue size estimation and study the corresponding queue dynamics. Simulation results show that the proposed algorithm provides better stability in queue dynamics, greater network power, less global synchronization, and a fairer treatment to bursty traffic than the RED algorithm.

This letter presents a new retransmission timeout (RTO) algorithm based on recursive weighted median (RWM) filters for the transmission control protocol (TCP). The RTO algorithm utilized in current TCP implementations is Jacobson's algorithm, which is based on recursive linear filtering. While linear filters are adequate for estimation in Gaussian signal environments, the round trip time (RTT) signals filtered to determine the RTOs are often impulsive. Thus, Jacobson's algorithm is not effective in many cases. The proposed algorithm employs RWM filters that yield improved performance when operating on RTT signals with heavy tailed statistics. Simulation results show that the proposed method yields significantly tighter RTT bounds than Jacobson's method over Internet traffic with heavy tailed statistics.