Burst Packet Loss Research Articles

Intrablock Interleaving for Batched Network Coding With Blockwise Adaptive Recoding

Batched network coding (BNC) is a low-complexity solution to network transmission in multi-hop packet networks with packet loss. BNC encodes the source data into batches of packets. As a network coding scheme, the intermediate nodes perform recoding on the received packets belonging to the same batch instead of just forwarding them. A recoding scheme that may generate more recoded packets for batches of a higher rank is also called adaptive recoding. Meanwhile, in order to combat burst packet loss, the transmission of a block of batches can be interleaved. Stream interleaving studied in literature achieves the maximum separation among any two consecutive packets of a batch, but permutes packets across blocks and hence cannot bound the buffer size and the latency. To resolve the issue of stream interleaver, we design an intrablock interleaver for adaptive recoding that can preserve the advantages of using a block interleaver when the number of recoded packets is the same for all batches. We use potential energy in classical mechanics to measure the performance of an interleaver, and propose an algorithm to optimize the interleaver with this performance measure. Our problem formulation and algorithm for intrablock interleaving are also of independent interest.

Understanding commercial 5G and its implications to (Multipath) TCP

In this paper, we perform an empirical study on commercial 5G. Unlike previous works studying 5G in early deployments, we focus on fully deployed 5G networks that already serve massive numbers of users, and in particular, we perform controlled experiments to analyze TCP and multipath TCP (MPTCP) performances and behaviors over 5G. Our results show that 5G faces a dilemma in maintaining in-network buffer for TCP traffic: on one hand, TCP packets are less likely to accumulate within the network, thanks to the high data rate of 5G radio; but on the other hand, with TCP performance highly sensitive to signal quality, when signal becomes poor abruptly in mobile environment, the in-network buffer easily overflows, leading to packet loss bursts. We also confirm that employing MPTCP can aggregate capacities on 5G and 4G paths, but under the current deployment model where 5G and 4G base stations co-locate and share the same core network, MPTCP does not necessarily enhance communication reliability. In addition to active measurement experiments, we leverage a speedtest service to evaluate commercial 5G from a nationwide perspective. By exploiting over 13k crowdsourced speedtest results reported from 197 cities in China, we find that there exist considerable regional differences on 5G performances across the nation, and with the 4G co-locating deployment model, the 5G networks in some populous metropolitan areas have inferior performances. Our study provides insights for people to understand commercial 5G and to optimize upper-layer protocols and applications over 5G.

Robust High-Rate Secondary Control of Microgrids With Mitigation of Communication Impairments

The dynamic response speed of the secondary control (SC) in microgrids (MGs) is limited both by the communication network time delays and the bandwidth (BW) of the primary control (PC). By increasing the PC BW, which depends on how to design and tune the PC, the time delay issues will be multiplied due to the speed range of the communication modules and technologies. This article proposes a communication compensation block (CCB) to enhance the robustness of distributed SC against communication impairments in the MGs operated at the higher BW. The proposed method mitigates malicious time delays and communication non-ideality in distributed networked controls employed in the secondary layer of the MG by prediction, estimation and finally decision on transmitted data. A comprehensive mathematical model of the employed communication network is presented in details. Then, a robust data prediction algorithm based on the temporal and spatial correlation is applied into the SC to compensate for time delays and data packet loss. Furthermore, the effect of the number of stored packets and burst packet loss on the average success rate of the communication block, and the small signal stability analysis of the system in the presence of the CCB are investigated. Power hardware-in-the-loop (PHiL) experimental tests show the merits and applicability of the proposed method.

A Protocol Design Paradigm for Batched Sparse Codes.

Internet of Things (IoT) connects billions of everyday objects to the Internet. The mobility of devices can be facilitated by means of employing multiple wireless links. However, packet loss is a common phenomenon in wireless communications, where the traditional forwarding strategy undergoes severe performance issues in a multi-hop wireless network. One solution is to apply batched sparse (BATS) codes. A fundamental difference from the traditional strategy is that BATS codes require the intermediate network nodes to perform recoding, which generates recoded packets by network coding operations. Literature showed that advanced recoding schemes and burst packet loss can enhance and diminish the performance of BATS codes respectively. However, the existing protocols for BATS codes cannot handle both of them at the same time. In this paper, we propose a paradigm of protocol design for BATS codes. Our design can be applied in different layers of the network stack and it is compatible to the existing network infrastructures. The modular nature of the protocol can support different recoding techniques and different ways to handle burst packet loss. We also give some examples to demonstrate how to use the protocol.

Model for speech signal quality estimation in packet network of electricity supply industry

In this paper we present the dilemma dealing with the choice of speech segment duration and the calculation of speech signal quality in the packet telephone network of Electricity supply industry (ESI). The analyzed signal is packetized by one packet. The characteristic of this network is that disturbances of long duration in power system produce burst packet loss. The longer speech segments increase the packet delay, but decrease the number of lost packets in one burst, and vice versa. Here we present why it is better to choose speech segments of short duration. Also, we suggest the corrected method for the calculation of speech signal quality in packet network under the influence of long duration disturbances.

SOS: Standard-Compliant and Packet Loss Tolerant Security Framework for ADS-B Communications

The Automatic Dependent Surveillance - Broadcast (ADS-B) technology, already deployed by the major avionics companies (e.g., QatarAirways and AmericanAirlines), will become mandatory on board of civil and military aircraft flying in Class A, B, and C airspaces by 2020, enabling direct airplanes communications and enhanced flights monitoring. However, ADS-B has been designed without security considerations, thus being vulnerable to a variety of attacks, including message injection and messages order manipulation attacks, that can be easily performed via widely available commercial Software Defined Radios. To address these threats, we present Securing Open Skies (SOS), a standard-compliant, backward-compatible, loss-tolerant, and bandwidth efficient security framework to secure ADS-B communications. SOS leverages the real deployment of densely distributed, participatory ADS-B sensor networks such as OpenSky Network and Flight Radar, and provides message authentication and integrity security services on a time-slot basis, without resorting to any public key cryptography mechanism. Experimental performances obtained through a realistic proof-of-concept, deployed using commercial Ettus Research X310 Software Defined Radios, demonstrate the viability and effectiveness of our solution, even in presence of uniformly at random or burst packet loss events characterizing the ADS-B frequency band. For instance, SOS allows the verification of the authenticity of ADS-B messages requiring less than 50 percent of bandwidth overhead, with a percentage of verifiable slots above 80 percent, even in an highly lossy environment, characterized by a single packet loss probability of 60 percent—the process requiring less than one second: almost one tenth of similar approaches published in the literature. Finally, a thorough comparison against state of the art solutions in the literature highlights the unique security and reliability features enjoyed by SOS, as well as its practical viability.

Dependable Interference-Aware Time-Slotted Channel Hopping for Wireless Sensor Networks

IEEE 802.15.4 Time-Slotted Channel Hopping (TSCH) aims to improve communication reliability in Wireless Sensor Networks (WSNs) by reducing the impact of the medium access contention, multipath fading, and blocking of wireless links. While TSCH outperforms single-channel communications, cross-technology interference on the license-free ISM bands may affect the performance of TSCH-based WSNs. For applications such as in-vehicle networks for which interference is dynamic over time, it leads to non-guaranteed reliability of the communications over time. This article proposes an Enhanced version of the TSCH protocol together with a Distributed Channel Sensing technique (ETSCH+DCS) that dynamically detects good quality channels to be used for communication. The quality of channels is extracted using a combination of a central and a distributed channel-quality estimation technique. The central technique uses Non-Intrusive Channel-quality Estimation (NICE) technique that proactively performs energy detections in the idle part of each timeslot at the coordinator of the network. NICE enables ETSCH to follow dynamic interference, while it does not reduce throughput of the network. The distributed channel quality estimation technique is executed by all the nodes in the network, based on their communication history, to detect interference sources that are hidden from the coordinator. We did two sets of lab experiments with controlled interferers and a number of simulations using real-world interference datasets to evaluate ETSCH. Experimental and simulation results show that ETSCH improves reliability of network communications, compared to basic TSCH and the state-of-the-art solution. In some experimental scenarios NICE itself has been able to increase the average packet reception ratio by 22% and shorten the length of burst packet losses by half, compared to the plain TSCH protocol. Further experiments show that DCS can reduce the effect of hidden interference (which is not detectable by NICE) on the packet reception ratio of the affected links by 50%.

Parametric Planning Model for Video Quality Evaluation of IPTV Services Combining Channel and Video Characteristics

Parametric planning models are designed for estimating the video quality, which can be applied to effective planning, implementation, and management of network video applications and communication networks. However, different from the bitstream-based evaluation models, the planning models are not allowed to exploit the video streams, with only limited information available for use, i.e., a few general parameters predetermined by the service providers and network operators. In this paper, a parametric planning model combining channel and video characteristics is proposed to estimate the video distortion caused by packet loss for Internet protocol television (IPTV) services. More specifically, the probability distribution of the channel states is determined by detailed analysis of the channel characteristics. Then, considering the influence of burst packet loss and the temporal dependence between frames, several sequence-level and frame-level parameters for video quality evaluation are derived from the perspective of the probability distribution of the channel states. Utilizing these parameters, the proposed model approximates the video quality considering the effects of direct packet loss and error propagation. Experimental results show that the proposed model has a superior performance for video quality estimation than the three commonly used parametric planning models.

Adaptive Flow Assignment and Packet Scheduling for Delay-Constrained Traffic Over Heterogeneous Wireless Networks

Flow assignment and packet scheduling are two processes of deploying limited wireless resources to support high-data-rate transmissions. Optimizing these processes is critical in improving the traffic performance and network utilization. In this paper, we investigate the flow assignment and packet scheduling for multihomed communication of delay-constrained traffic over heterogeneous wireless networks. With regard to the inherent channel diversity and unreliability, large end-to-end delay and burst packet losses pose crucial challenges to guarantee goodput performance of real-time traffic. To address these critical issues, we present an Adaptive Flow Assignment and Packet Scheduling (AFAPS) framework. First, we introduce a “horizontal water filling” algorithm, which effectively integrates the channel resources in heterogeneous wireless networks to maximize the aggregate goodput. Second, we propose an alternative path interleaving scheme, which spreads out the packets' departures over multiple communication paths within the delay constraint to mitigate burst losses. The performance of the proposed AFAPS framework is evaluated through semi-physical emulations in EXata involving real Internet traffic traces. Experimental results show that AFAPS outperforms existing multipath data distribution models, in terms of goodput and end-to-end delay.

Reducing impact of network induced perturbations in remote control systems

The practical realization of remote control systems enforces handling network-induced effects: information transfer and processing delay, delay variability, packet loss and reordering, etc. The paper presents a comparative study of algorithmic methods that assist the control law in achieving higher regulation quality by reducing the detrimental impact of network-related uncertainties. Three popular families of methods, encompassing dynamic delay compensators, multiple control loops, and adjustable sampling rate, are investigated in a common experimental framework that involves a benchmark plant – structurally unstable inverted pendulum-on-a-cart system – and commonly available modules and communication technologies. The method performance in relation to computational footprint and network load is discussed. For the delay with well-established trend of variation the dynamic compensator proves the most efficient option, while more sophisticated methods, involving the exchange of multiple pieces of information, are required under burst packet loss and in stochastic settings with aggravated randomness.

Trading Delay for Distortion in One-Way Video Communication Over the Internet

We study the problem of one-way video communication in a single-source, multiple-destination scenario over the lossy Internet. Forward error correction (FEC) coding is commonly adopted for data protection in implementing loss-resilient video transmission systems. However, the burst packet losses over the Internet frequently degrade FEC performance and induce video quality deteriorations. To address the challenging problem, we propose a novel transmission scheme dubbed trading delay for distortion (TELFORD) that includes three components: 1) adaptive multidestination status estimation; 2) delay-constrained transmission rate assignment; and 3) differentiated FEC packet spreading. We analytically formulate and solve the problem of FEC packet allocation and scheduling to minimize the end-to-end video distortion. The proposed TELFORD is able to cope with multiple-destination scheduling separately. We conduct performance evaluation through semiphysical emulations in Exata using real-time H.264 video streaming. Experimental results show that TELFORD outperforms existing error control transmission schemes in improving the video peak signal-to-noise ratio and mitigating packet transmission impairments.

Packet dispersion strategy evaluation from the perspective of packet loss pattern and VoIP quality

The appearance of burst packet losses and its devastating effect on Voice over IP (VoIP) service have imposed a requirement for the implementation of loss recovery mechanisms to address VoIP quality during periods when high packet loss is exhibited. Existing loss recovery mechanisms are dependent on end point capabilities, whereas Quality of service (QoS) routing protocols suffer from complexity and scalability issues. In this paper, we examine packet dispersion?s ability to address burst losses and provide a computational model, which is verified using real network testing. A study has been carried out to investigate the effect of different packet dispersion strategies on burst losses, which clearly shows dispersion?s qualitative superiority over single path routing. Furthermore, an analytical approach is proposed resulting in quality estimation obtained by individual strategies. Practical evaluation has shown that each strategy copes differently with various burst scenarios in order to maximize VoIP quality.

Centralised Enhanced forward Error Correction Mechanism on the Contention Window to Improve the Transmission Quality in Vehicular AD Hoc Networks

Background: Vanet the recent emerging technology in the recent industry has grabbed the attention with advanced techniques and extricate framework of sophisticated equipments. Initially absence of an infrastructure in this network has communication constraints for the nodes involved therein. Adding to this the dynamic topology makes the network more vulnerable for potential targets of security and reliability of data diffusion within these nodes. The communication standard i.e. 802.11p have some optional channel rejection in the Orthogonal Frequency Division Multiplexing in the Physical layer which motivated to work on the reliability methods of data dissemination. Methods: To start with the hidden terminal issues of these networks have to be addressed which can be resolved by frame capture. This frame capture quality in turn is determined by the Contention window of the Physical layer. Hence an adaptive mechanism that involves forward Error correction (AFEC) which helps to overcome packet losses in an efficient way in midst of channel fluctuation, burst packet loss. The results are simulated in NS2 (2.34) with static packet size of 512 bytes. Findings: The experimental results demonstrated that the FEC mechanism is more adaptive in adjusting and controlling the data redundancy across the network for static and dynamic channel fluctuations. The redundancy rate was further reduced with an adaptive technique on the Contention Window size, which had considerable effects of every transmission on the delay, through put and jitter. The centralized FEC algorithm is set on an adaptive platform where each vehicle updates its back off window size at the end of each Observation Time depending on the priority of the message received. The feasible part of this algorithm is that it can be implemented on the MAC layer, which has a Reception Module. Consequently, the proposed mechanism under various network conditions performs well in terms of QoS for emergency messages than the conventional methods for a less volume of data. Application/Improvements: The algorithm can be analyzed from a traffic scenario of two-way highway and rural areas. The progress of the work can also be carried out for multimedia messages.

A protection scheme for multimedia packet streams in bursty packet loss networks based on small block low-density parity-check codes

This paper proposes an enhanced forward error correction (FEC) scheme based on small block low-density parity-check (LDPC) codes to protect real-time packetized multimedia streams in bursty channels. The use of LDPC codes is typically addressed for channels where losses are uniformly distributed (memoryless channels) and for large information blocks. This work suggests the use of this type of FEC codes at the application layer, in bursty channels (e.g., Internet protocol (IP)-based networks) and for real-time scenarios that require low transmission latency. To fulfil these constraints, the appropriate configuration parameters of an LDPC scheme have been determined using small blocks of information and adapting the FEC code to be capable of recovering packet losses in bursty environments. This purpose is achieved in two steps. The first step is performed by an algorithm that estimates the recovery capability of a given LDPC code in a burst packet loss network. The second step is the optimization of the code: an algorithm optimizes the parity matrix structure in terms of recovery capability against the specific behavior of the channel with memory. Experimental results have been obtained in a simulated transmission channel to show that the optimized LDPC matrices generate a more robust protection scheme against bursty packet losses for small information blocks.

BACnet/WS ビル設備遠隔監視の通信性能に対するバーストパケットロスの影響

BACnet/WS ビル設備遠隔監視の通信性能に対するバーストパケットロスの影響

Robust FMO algorithm and adaptive redundant slice allocation for wireless video transmission

Visual quality of compressed video may suffer great degradation when transmitted over lossy wireless networks. Flexible macro-block ordering (FMO) is a new error resilient tool adopted by H.264/AVC. It has a good performance of error resilience by changing the coding order of macro-blocks in the frame. redundant slice (RS) is another tool which adds redundant copy of slices into the stream to take precautions against packet loss. However, we shouldn't only care about peak signal to noise ratio (PSNR) of the video; the robustness of video streams to burst packet loss of wireless channel is also worth considering. In applications, such as real-time video transmission services, degradation of video quality may be tolerable, but collapse of decoder due to burst packet loss will greatly lower user's quality of experience. This paper proposes a robust FMO (RFMO) algorithm which takes gradient feature of frames into consideration to enhance robustness of video streams, and the adaptive RS allocation (ARSA) helps to increase the PSNR with only a little increase in bit rate. Experiment results show that the RFMO algorithm can significantly reduce the collapse times of decoder with invisible decrease in visual quality, and the ARSA can still guarantee a high PSNR in the case of high packet loss rate.

Multiple Description Coding and Recovery of Free Viewpoint Video for Wireless Multi-Path Streaming

By transmitting texture and depth videos captured from two nearby camera viewpoints, a client can synthesize via depth-image-based rendering (DIBR) any freely chosen intermediate virtual view of the 3D scene, enhancing the user's perception of depth. During wireless network transmission, burst packet losses can corrupt the transmitted texture and depth videos and degrade the synthesized view quality at the client. In this paper, we propose a multiple description coding system for multi-path transmission of free-viewpoint video, with joint interview and temporal description recovery capability. In particular, we encode separately the even frames of the left view and the odd frames of the right view, and transmit them as one description on one path. The second description comprises the remaining frames in the two views and is transmitted over a second path. If the receiver receives only one description due to burst loss in the other path, the missing frames in the other description are partially reconstructed using our frame recovery procedure. First, we construct two recovery candidates for each lost pixel in a frame. The first candidate is generated via temporal super-resolution from its predecessor and successor frames in the same view. The second candidate is generated via DIBR from the received frame of the same time instance in the other view. Next, we select the best pixel candidates one patch at a time, where an image patch corresponds to a neighborhood of pixels with similar depth values in the 3D scene. Near-optimal source and channel coding rates for each description are selected using a branch-and-bound method, for given transmission bandwidth on each path. Experimental results show that our system can outperform a traditional single-description/single-path transmission scheme by up to 5.5 dB in Peak Signal-to-Noise Ratio (PSNR) of the synthesized intermediate view at the client.

An Improved Packet Loss Concealment Method for Mobile Audio Coding

In this paper, we propose an improved packet loss concealment (PLC) algorithm for ACELP-based speech coders in order to improve the quality of decoded speech at the receiver. Conventional receiver-based PLC algorithms in the AMR-WB+ speech codec are usually based on speech correlations to reconstruct the decoded speech of the lost frames by using parameter information obtained from the previous or adjacent correctly received frames. However, this approach has difficulties in reconstructing accurate voice signals when consecutive packet loss and bursts of packet loss occur. Thus, we propose a new PLC approach based on data-driven principle to recover the Immittance Spectral Frequen- cy (ISF) coefficients of lost packet, referring to pitch period parameters of adjacent correctly received voice packets. The proposed PLC algorithm is designed for ACELP-based audio codec. We make the performance comparison between the data-driven method and the PLC algorithm currently employed in AMR-WB+ speech codec through the PESQ objective evaluation test and MOS subjective evaluation test. The results prove that our algorithm can achieve better speech recov- ery quality than original PLC algorithm employed in AMR-WB+ speech codec.

Impact of Random and Burst Packet Losses on H.264 Scalable Video Coding

This paper presents an encoder-based prediction model to study the impact of packet loss on H.264 scalable video coding (SVC). A Markov Chain (MC) with 2 N states is developed to describe the error propagation process inside a group of pictures (GOP). Using the packet loss rate reported by RTCP (RTP Control Protocol) and analyzing the inter-frame prediction rules, we evaluate packet-loss-induced quality degradation in SVC as the probability that a given scalable layer can be correctly received. Based on the proposed model, the performance of the SVC hierarchical B-frame structure, the zero delay encoding/decoding structure, and the advance video coding (AVC) IPPP structure (compatible base layer in SVC) are evaluated and compared under both random and burst packet loss events.

광대역 무선 네트워크에서 효율적인 멀티미디어 전송을 위한 혼잡 제어 기법

TCP does not ensure the bandwidth and delay bound required for multimedia streaming services in broadband wireless network environments. In this paper, we propose a new congestion control scheme for efficient multimedia transmission, called COLO TCP (Concave Increase Slow Start Logarithmic Increase Congestion Avoidance TCP). The COLO TCP prevents the burst packet loss by applying the concave increase algorithm in slow start phase. In the congestion avoidance phase, COLO TCP uses the logarithmic increase algorithm that quickly recovers congestion window after packet loss. To highly utilize network bandwidth and reduce packet loss ratio, COLO TCP uses additive increase algorithm and adaptive decrease algorithm . Through simulation results, we prove that our COLO TCP is more robust for random loss. It is also possible for efficient multimedia transmission.