IPTV Video Perception Quality Based on Packet Loss Distribution

Mobile Internet Protocol Television (IPTV) provides users with multimedia content through wireless access technologies whenever they want and wherever they are. However, vertical handover in heterogeneous networks results in long interruption time and large packet loss rate, which poses a challenge for seamless mobile IPTV service. The existing solutions that make use of the heterogeneous access networks to transmit multimedia content have two drawbacks: high bandwidth consumption of the wireless networks and lots of packets loss when a user resides in the overlapped area of the boundary. To solve these issues, we propose a new soft handover mechanism, based on the IPTV server and the consumer electronic devices. The proposed Parallel Soft Handover (PSH) divides the Real-time Transport Protocol (RTP) packets according to the packet loss rates, and exploits different networks efficiently by using parallel transmission. Moreover, the selective retransmission for RTP packets is introduced to reduce the packet loss. Simulation results demonstrate that the PSH can reduce the packet loss rate, as well as obtain a more efficient use of the wireless bandwidth.

An FEC scheme combined with weighted scheduling to reduce multicast packet loss in IPTV over PON

Residual packet loss rate analysis of 2-D parity forward error correction

Packet loss and delay are the major network impairments for transporting real-time voice over IP networks. In the proposed system, multiple descriptions of the speech are used to take advantage of the packet path diversity. A new objective method is presented for predicting the perceived quality of multi-stream voice transmission. Also proposed is a multi-stream playout buffer algorithm, together with an adaptive parameter adjustment scheme, that maximizes the perceived speech quality via delay-loss trading. Experimental results showed that, compared to FEC-protected single-path transmission, the proposed multi-stream transmission scheme achieves significant reductions in delay and packet loss rates as well as improved speech quality.

In this paper, the consensus problem is studied for a class of multi-agent systems with sampled data and packet losses, where random and deterministic packet losses are considered, respectively. For random packet losses, a Bernoulli-distributed white sequence is used to describe packet dropouts among agents in a stochastic way. For deterministic packet losses, a switched system with stable and unstable subsystems is employed to model packet dropouts in a deterministic way. The purpose of this paper is to derive consensus criteria, such that linear multi-agent systems with sampled-data and packet losses can reach consensus. By means of the Lyapunov function approach and the decomposition method, the design problem of a distributed controller is solved in terms of convex optimization. The interplay among the allowable bound of the sampling interval, the probability of random packet losses, and the rate of deterministic packet losses are explicitly derived to characterize consensus conditions. The obtained criteria are closely related to the maximum eigenvalue of the Laplacian matrix versus the second minimum eigenvalue of the Laplacian matrix, which reveals the intrinsic effect of communication topologies on consensus performance. Finally, simulations are given to show the effectiveness of the proposed results.In this paper, the consensus problem is studied for a class of multi-agent systems with sampled data and packet losses, where random and deterministic packet losses are considered, respectively. For random packet losses, a Bernoulli-distributed white sequence is used to describe packet dropouts among agents in a stochastic way. For deterministic packet losses, a switched system with stable and unstable subsystems is employed to model packet dropouts in a deterministic way. The purpose of this paper is to derive consensus criteria, such that linear multi-agent systems with sampled-data and packet losses can reach consensus. By means of the Lyapunov function approach and the decomposition method, the design problem of a distributed controller is solved in terms of convex optimization. The interplay among the allowable bound of the sampling interval, the probability of random packet losses, and the rate of deterministic packet losses are explicitly derived to characterize consensus conditions. The obtained criteria are closely related to the maximum eigenvalue of the Laplacian matrix versus the second minimum eigenvalue of the Laplacian matrix, which reveals the intrinsic effect of communication topologies on consensus performance. Finally, simulations are given to show the effectiveness of the proposed results.

The video quality degradation in wireless network is much changeable for complex packet loss scenes of wireless channels. However, few Video Quality Metrics (VQM) analyze the quality degradation under different wireless packet loss conditions. This paper proposes a novel Universal Video Quality Metric (UVQM), which is suitable for both distributed loss situation and burst loss situation. The major contributions of this paper can be concluded as follows. Firstly, this metric is an end-to-end measurement which considers the impacts of encoding parameter (quantization parameter, q-scale) and transmission parameters (Packet Loss Rate, PLR and Maximum Transmission Unit, MTU) to make a complete VQM. Then, parameters used in UVQM are extracted from bitstreams which reduces the use of computation resource for decoding at the receiver. Finally, UVQM coalesces the impact of content-aware features by introducing a developed Just Noticeable Distortion (JND) which makes the prediction much accuracy. The verification results show that the predicted video quality measured by Mean Opinion Score (MOS) has a high Pearson correlation (higher than 0.74 in distributed loss situation and 0.65 in burst loss situation).

Mobile Ad-hoc Networks (MANETs) usually suffer from high packet loss and high link failure rates. These challenges usually increase congestion on some links while other links are almost free. In this paper, we propose a novel mechanism to enhance QoS in multipath routing protocols in MANETs based on the InfiniBand (IB) QoS architecture. The basic idea of our approach is to enhance the load balancing and thus reduce congestion on overloaded links. This mechanism has enabled us to (1) give critical applications higher priority when routing their packets across the network, (2) effectively manage frequent connections and disconnections and thus reduce link failures and packet loss rates, and (3) reduce the overall power consumption as a consequence of the previous gains. We have tested the scheme on the (IBsim) simulator and achieved significant improvements in QoS parameters compared to two well- known routing protocols: AODV and AOMDV.

In the last two decades, Multi-Cast schemes in mobile ad-hoc networks have been widely studied and lots of research results have been put into use and changed our everyday life. However, packet loss for Multi-Cast schemes in mobile ad-hoc networks has not been well studied yet. The influence of node density and relay schemes on packet loss is not clear. Thus, we study the packet loss problem for Multi-Cast schemes in MANETs in this paper. First, we present a general Multi-Cast probabilistic model to get a better understanding of the packet loss problem. Second, based on the general model and Chernoff Bounds, we prove that the up'er bound of the general packet loss for Multi-Cast is Ω(me <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">- 2</sup> ), where n is the node number and m denotes number of hops. Thus we can conclude that large node density decreases the packet loss rate by exponential factor, and bigger hops increase packet loss rate linearly. In addition, we present a reliable Multi-Cast protocol(RMP). Through an ACK Aware Tree to complete packet acknowledgement and repair host selection, RMP is able to minimize the impact of packet loss on the Multi-Cast throughput in the simulation.

In this paper we discuss a methodology for performance evaluation of the solutions for vertical handovers between different wireless access technologies in IP network. The performance evaluation is based on simple analytical models and covers both the ideal case (no packet loss) and the real case where there is a given packet loss rate. The methodology is applied to a comparison among three solutions, namely MIPv4, classical SIP mobility management using re-INVITE messages and a SIP based solution called MMUSE (Mobility Management Using SIP Extensions) proposed by the authors.

Internet Protocol Television (IPTV) is a service to present digital TV over Internet. The IPTV network architecture consists of two main components: core network and access network. Many types of access networks can be used for IPTV including Ethernet PON (EPON). In this paper, a new mechanism is proposed for bandwidth allocation to various IPTV services over the EPON that decreases IPTV packet loss. This mechanism is implemented in optical line terminal (OLT) and weights are allotted to queues in the OLT to reduce packet loss. Then, weighted round robin (WRR) is used to scheduling IPTV packets.

In the broad area of performance evaluation of multimedia networking mechanisms, quality of experience (QoE) assessments are often used as a performance metric. Even though intensive efforts have been made to understand the impact of packet loss on QoE for different application scenarios, there is no consensus about how specific packet loss patterns impact the QoE for RTP video streaming. In this paper, we conduct an extensive study on the impact of uniformly distributed random and burst loss on the QoE-metrics Peak-Signal-to-Noise-Ratio (PSNR), Structural Similarity (SSIM), and Video Multi-Method Assessment Fusion (VMAF). Overall, our study indicates that for QoE-based performance evaluations, it is in most cases sufficient to only consider easier to understand and easier to model uniformly distributed random loss.

Numerous studies have shown that haptic interaction plays a key role in enriching the sense of immersion and copresence of distributed users in collaborative virtual environments (CVEs). However, to ensure high-fidelity haptic interaction in CVEs, a high packet rate is required, resulting in considerable increase in overall traffic over the network. While perceptual deadband approach for haptic signals can successfully reduce high packet rates, this method is vulnerable to packet loss because the losing of single perceivable update packet results in a succession of wrong predictions. In this paper, we improve the perceptual deadband model by incorporating a packet loss resilient scheme using acknowledgement (ACK) packets. In case that an ACK packet is not returned to the sender within a trigger time, the sender will send an additional haptic data packet to the receiver to offset the effect caused by packet loss. By carefully selecting trigger time, buffer length and acknowledge time, the proposed scheme can be applied in a network environment with variable packet loss rates. The ACK-packets-based scheme is experimented by using different packet loss rates and lengthes of burst loss. Experimental results show that the proposed scheme can maintain stable and robust haptic interaction in terms of both objective and subjective measurements in a lossy network environment, and meanwhile perform well in haptic data reduction.

Mobile Ad hoc Networks (MANETs) in the form of wireless local area networks is becoming a common platform for communication among internet users. As compared with wired networks, the rate of random packet loss due to wireless channel errors or node mobility is very high in MANET. Misinterpretation of random packet loss as loss due to congestion by TCP causes significant degradation in the performance (i.e. aggregate throughput) of the network. Numerous solutions are proposed in literature, but they either are cross layer solution or focused on specific issues of TCP. In this paper, we present a new energy efficient, end-to-end, layered solution named as "Proposed-TCP", as transport layer protocol. The "Proposed-TCP" can identify packet loss due to congestion, wireless channel error and route failure, by using multiple end-to-end metrics and handles incipient congestion in the network also. The "Proposed-TCP" is implemented and simulated in Network Simulator (NS2). In order to achieve accuracy in results, the Proposed-TCP is simulated multiple times and compared with TCP Reno, TCP Santa Cruz and ADTCP. For the present analysis, the simulation is carried out by introducing 0%, 5% and 10% random noise. The simulation results show that Proposed-TCP outperforms TCP Reno, TCP Santa Cruz, and ADTCP by producing better performance in terms of throughput and fairness.

In recent years, ground‐based aerospace tracking, telemetry and control (TT&amp;C) systems have been shifting towards an IP‐based cloud TT&amp;C network architecture to improve signal transmission efficiency and system flexibility. However, this new architecture presents new challenges for the performance of tracking loops in the cloud‐based TT&amp;C receiver. The authors present a predictive compensation tracking loop that addresses the problem of packet loss. The loop utilises radial basis function (RBF) online network identification for robust carrier and code frequency offset tracking. To account for packet loss, the phenomenon is modelled as a Bernoulli random process, and the predictive compensation loop uses the trained RBF network to maintain the system's tracking state. When normal data are received again, the loop can track the received information well and align the frequency offset locking state. The authors compare the performance of the proposed predictive compensation tracking loop with traditional loop tracking performance using data hold strategies under different packet loss rates. The results demonstrate the effectiveness of the proposed approach in maintaining stable tracking and solving in the discontinuous reception of the loop under packet loss environments. The proposed predictive compensation tracking loop offers a practical solution for addressing packet loss, which is a common issue in real‐world applications.

The Internet of things, including Internet technology, including wired and wireless networks. Internet of Things and the Internet is the relationship between the parent and the child. In this paper, we aim to study the Investigation on the network packet loss’s long-range dependence and QOE and gain a good result and conclusion. In order to better establish no-reference video quality assessment model considering the network packet loss and further gain a better QoE evaluation, so we build NS2 + MyEvalvid simulation platform to study the scale characteristic of the network packet loss, scale characteristic of packet loss through the influence of packet loss rate to influence QoE. The experimental results show that, packet loss processes have long-range dependence, the number of superimposed source N, shape parameter, Hurst parameter, the output link speed have impacts on long-range dependence. We came to the conclusion that when superimposed source N is more, the shape parameter is smaller, Hurst parameter is bigger, the output link speed is smaller, packet loss’s long range dependence is larger, packet loss rate is high.