Burst Packet Loss Research Articles

IPTV Multicast Over Wireless LAN Using Merged Hybrid ARQ With Staggered Adaptive FEC

<para xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> Internet Protocol TV (IPTV), an emerging internet application, provides more flexibility and interactivity for users because of its embedded return channel. Wireless 802.11 networks, used as the last hop of IPTV networks, would add another dimension of mobility, which is of great value for hot spot deployment and is increasingly required for in-home content distribution. However, a wireless channel may suffer from multi-path fading and interference, which may cause random and burst packet loss and impact users' quality of experience of an IPTV program. We report the design, implementation and evaluation of Merged Hybrid Adaptive FEC and ARQ (MHARQ) system for IPTV multicast over wireless LANs. In MHARQ, an IPTV cache server is introduced to improve the reliability for the last hop wireless distribution. MHARQ combines the advantages of receiver-driven staggered FEC and hybrid ARQ schemes to compensate the large dynamic range of WLAN channels and to achieve high reliability, scalability and wireless bandwidth efficiency for IPTV multicast. We also address backward compatible FEC encoding and decoding, audio and video synchronization issues in practical system implementation. Using the ORBIT radio grid test bed, we have investigated the performance of the proposed MHARQ system with various numbers of users per AP and different numbers of APs per IPTV cache server. It is demonstrated via real system implementation on ORBIT that MHARQ improves wireless bandwidth efficiency and scalability for reliable IPTV multicast, compared with existing reliable multicast schemes. </para>

On Understanding Transient Interdomain Routing Failures

The convergence time of the interdomain routing protocol, BGP, can last as long as 30 minutes. Yet, routing behavior during BGP route convergence is poorly understood. During route convergence, an end-to-end Internet path can experience a transient loss of reachability. We refer to this loss of reachability as <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">transient</i> <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">routing</i> <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">failure</i> . Transient routing failures can lead to packet losses, and prolonged packet loss bursts can make the performance of applications such as Voice-over-IP and interactive games unacceptable. In this paper, we study how routing failures can occur in the Internet. With the aid of a formal model that captures transient failures of the interdomain routing protocol, we derive the sufficient conditions that transient routing failures could occur. We further study transient routing failures in typical BGP systems where commonly used routing policies are applied. Network administrators can apply our analysis to improve their network performance and stability.

Markovian-based traffic modeling for mobile ad hoc networks

Mobile ad hoc networks (MANETs) show very significant difference with respect to other computer networks due to the presence of extremely large packet loss bursts. The development of protocols for mobile ad hoc networks, especially multimedia protocols, require extensive evaluation either through simulation or real-life tests. Such testing consumes a great amount of resources both in terms of time and trace file sizes. Therefore, finding efficient means of reducing the amount of data that is stored and processed is quite important to accelerate the evaluation of different audio/video streaming applications. If, moreover, we are able to model the loss pattern experienced, we can further accelerate the evaluation process. In this work we propose two models based on hidden Markov chains that are able to grasp both packet arrivals and packet loss patterns in MANETs. A simpler two-state model is proposed to model losses when proactive routing protocols are used, while a more complex three-state model is proposed for reactive routing protocols. We also introduce a new set for packet loss pattern measurements that can be of interest for the evaluation of audio/video streaming applications. Experimental results show that the proposed models can adequately reproduce extremely long packet loss patterns, typical of MANET environments, with a high degree of accuracy. Overall, we find that the proposed models are able to significantly reduce both the simulation time and the trace file sizes required.

Wireless Model Based Predictive Networked Control System

Owing to their distributed architecture, networked control systems are proven to be feasible in scenarios where a spatially distributed control system is required. Traditionally, such networked control systems operate over real-time wired networks over which sensors, controllers and actuators interact with each other. Recently, in order to achieve the utmost flexibility, scalability, ease of deployment and maintainability, wireless networks such as IEEE 802.11 LANs are being preferred over dedicated wired networks. However, basic networked control systems cannot operate over general purpose wireless networks since the stability of the system is compromised due to unbounded delays and unpredictable packet losses that are typical in the wireless medium. This work proposes a novel wireless networked control system that can achieve decent control even under unbounded delay, bursts of packet loss and ambient wireless traffic. Ambient wireless traffic is handled with modified 802.11b medium access control parameters providing the proposed system with a greater medium access priority. Packet deadlines defined at each node of the system reduce unbounded packet delay to packet loss. Performance degradation due to packet loss is kept at a minimum using the predicted plant states and control signals. The proposed system is implemented and thoroughly evaluated on a dedicated test platform under numerous scenarios. Results of the experiments show that the proposed system can work under bursts of packet loss and ambient wireless traffic levels which are intolerable for basic networked control systems while not being hindered by restraining assumptions of existing methods.

Error concealment aware streaming video system over packet-based mobile networks

A compressed video bitstream is extremely sensitive to errors when it is transmitted via networks with bursty packet loss such as 3G and mobile WiMAX. Error concealment techniques have been used to improve the recovered picture quality in cases in which parts of the coded picture are not available at the decoder for reconstruction. However, most conventional error concealment methods show low image reconstruction performance due to unavailable motion information about image blocks during burst packet loss, and require high energy consumption to find the best error concealment method in the receiver. The proposed error concealment aware error resilient streaming video system has an intelligent error concealment selector. The method recommends the best error concealment method per macroblock, by additionally transmitting its error concealment selection codes to the decoder. The streaming client conceals the corrupted MB via the chosen error concealment method, with low energy consumption, even when the frame loss is bursty. The experimental results show that the system is highly effective, in terms of the video quality and energy consumption, at the cost of a slight overhead for transmitted bit streams.

Buffer Coding for Reliable Transmissions over Wireless Networks

In-network caching is a useful technique for reducing latency and retransmission overhead of lost packets for reliable data delivery in wireless networks. However, in-network caching is challenging to implement in memory constrained devices such as RFIDs and sensors, and also in Wireless LAN (WLAN) gateways for large-scale deployments. In this paper we propose a novel technique for management of in-network caches using XOR coding for optimizing the use of limited buffer space in presence of random and burst packet losses. We identify two critical parameters, coding degree and coding distance for the coding scheme. As a case-study we implement our approach over Snoop and evaluate its performance for WLANs. We further propose a self-adaptive algorithm that tunes coding degree on the fly based on the measured coding behavior and packet loss probability. Using simulations in ns-2, we observe that in our simulation settings, when the size of the retransmission buffer in the gateway is less than 16 packets per TCP flow, the throughput can be enhanced by up to 30% for random losses and up to 20% for burst losses.

Network-adaptive Selection of Transport Error Control (NASTE) for Video Streaming over WLAN

Video streaming over wireless networks is inherently vulnerable to burst packet losses caused by dynamic wireless channel variations with time-varying fading and interference. To alleviate this limitation, especially in the transport layer, the error control schemes based on FEC (forward error correction), ARQ (automatic repeat request), interleaving, and their hybrid are essential. However, each error control mode shows different performance according to the target application requirement and the channel status. Thus, in this paper, we propose a network-adaptive selection of transport error control (NASTE), where transport-layer error control modes are dynamically selected and applied. First, an effective embedded (software-based) realization of error control modes is proposed to support the flexible mode switching for NASTE. We then present a practical yet effective mode switching algorithm by linking channel/application monitoring, table-guided mode switching decision, and subsequent fine-tuning on error-control mode parameters. Finally, we have implemented a prototype video streaming system with NASTE support and verified it over an IEEE 802.11 g WLAN (wireless LAN) environment. The experimental results indicate that the proposed mechanism can enhance the overall transport performance by recovering packet losses and thus improves the quality of video streaming over WLAN.

Predicting Properties of Congestion Events for a Queueing System With fBm Traffic

In packet networks, congestion events tend to persist, producing large delays and long bursts of consecutive packet loss resulting in perceived performance degradations. The length and rate of these events have a significant effect on network quality of service (QoS). The packet delay resulting from these congestion events also influences QoS. In this paper a technique for predicting these properties of congestion events in the presence of fractional Brownian motion (fBm) traffic is developed.

Congestion Detection in the European Internet

In this paper we present a study about the utilization of one-way delay measurements to detect and characterize network congestion in the european Internet. The experiments have been made using the ETOMIC platform that allows one-way delay measurement with high precision timestamps. We have found a peculiar router behaviour in which the bottleneck is not the available bandwidth but it is the packet processing power of the router (backplane and CPU constraints). This router has been characterized with several network parameters. Some of them are the dependency of this limitation with the input data rate in packets per second, the size of burst packet losses measured in packets or time and the absence of specific scheduling algorithms in the router that could affect to larger flows.

Performance Analysis of Mobile Hotspots with Heterogeneous Wireless Links

Mobile hotspot enabling Internet access services in moving vehicles is an important service for ubiquitous computing. In this paper, we propose an analytical framework for studying the packet loss behavior and throughput in a mobile hotspot with heterogeneous wireless links. We first develop a two-state Markov model for the integrated wireless wide area network (WWAN) and wireless local area network (WLAN). We then derive the expressions that describe the experienced packet loss probability, packet loss burst length, and throughput. Finally, we present simulation results to verify the accuracy of our analysis. It is concluded that adaptive and cross-layer approaches should be deployed to improve the performance of mobile hotspots.

Towards improving the robustness of distributed speech recognition in packet loss

This work addresses the problem of achieving robust distributed speech recognition (DSR) performance in the presence of packet loss. The nature of packet loss is analysed by examining packet loss data gathered from a GSM mobile data channel. This analysis is then used to examine the effect of realistic packet loss conditions on DSR systems, and shows that the accuracy of DSR is more sensitive to burst-like packet loss rather than the actual number of lost packets. This leads to the design of a three-stage packet loss compensation scheme. First, interleaving is applied to the transmitted feature vectors to disperse bursts of packet loss. Second, lost feature vectors are reconstructed prior to recognition using a variety of reconstruction techniques. Third, a weighted-Viterbi decoding method is applied to the recogniser itself, which modifies the contribution of the reconstructed feature vectors according to the accuracy of their reconstruction. Experimental results on both a connected digits task and a large-vocabulary task show that simple methods, such as repetition, are not as effective as interpolation methods. Best performance is given by a novel maximum a posteriori (MAP) estimation, which utilizes temporal statistics of the feature vector stream. This reconstruction method is then combined with weighted-Viterbi decoding, using a novel method to calculate the confidences of reconstructed static and temporal components separately. Using interleaving, results improve significantly, and it is shown that a limited level of interleaving can be applied without increasing the delay to the end-user. Using a combination of these techniques for the connected digits task, word accuracy is increased from 49.5% to 95.3% even with a packet loss rate of 50% and average burst length of 20 feature vectors.

A Hybrid FEC Method Using Packet-Level Convolution and Reed-Solomon Codes

Efficient real-time contents distribution services on the Internet are only possible by suppressing the influence of packet losses. One solution for UDP transmission is the use of Forward Error Correction (FEC) based on Reed-Solomon codes. However, a more efficient method is required since this causes the increase of network traffic and includes the weakness to burst packet losses. In this paper, we propose a data recovery method that generates redundant data with the combination of Reed-Solomon codes and convolution of neighboring blocks. We realize the small amount of redundancy and the high reliability in data transmission compared with using only Reed-Solomon codes in the environment that burst packet losses are occurred frequently. We implement proposal method into the network bridge and confirm its efficiency from the viewpoint of data reconstruction from burst packet losses.

A novel content-aware interleaving for wireless video transmission

In wireless network environments, the effect of transmission errors and losses on the video quality vary, depending on the intensity of the burst, and which parts of the video stream are lost. Among the existing transmission error control techniques, FEC and ARQ represent good solutions for combating transmission errors, but require redundant data. Interleaving can improve subjective video quality without wasting additional bandwidth, even though the method has no error correcting capability, because it allows spreading successive errors. In this paper, a content-aware packet-level interleaving method is proposed, which uses a quantitative index, in order to indicate the degree of content-importance of the video content, to ensure the effect of burst packet loss is distributed intelligently. The proposed scheme improves overall video quality, compared with content-blind interleaving methods.

An unequal packet loss resilience scheme for video over the Internet

We present an unequal packet loss resilience scheme for robust transmission of video over the Internet. By jointly exploiting the unequal importance existing in different levels of syntax hierarchy in video coding schemes, GOP-level and Resynchronization-packet-level Integrated Protection (GRIP) is designed for joint unequal loss protection (ULP) in these two levels using forward error correction (FEC) across packets. Two algorithms are developed to achieve efficient FEC assignment for the proposed GRIP framework: a model-based FEC assignment algorithm and a heuristic FEC assignment algorithm. The model-based FEC assignment algorithm is to achieve optimal allocation of FEC codes based on a simple but effective performance metric, namely distortion-weighted expected length of error propagation, which is adopted to quantify the temporal propagation effect of packet loss on video quality degradation. The heuristic FEC assignment algorithm aims at providing a much simpler yet effective FEC assignment with little computational complexity. The proposed GRIP together with any of the two developed FEC assignment algorithms demonstrates strong robustness against burst packet losses with adaptation to different channel status.

An End-to-End Multipath Smooth Handoff Scheme for Stream Media

Supporting transmission of stream media over wireless mobile networks is often difficult because packets may be lost due to the rerouting of packets during handoff, and also because bursts of packet loss may occur during handoff due to the disparity in the amount of available bandwidth among different cells. In this paper, we propose an end-to-end multipath handoff scheme that provides smooth handoff for stream media in wireless networks with different amounts of available bandwidth from cell to cell. In the proposed scheme, multiple paths are established during handoff to reach a mobile destination node. The stream media sources are equipped with an adaptive multilayer encoder, and important layers in the encoded video stream are duplicated and transmitted over multiple paths during handoff. The effectiveness of the proposed multipath handoff scheme is verified and compared with existing schemes through extensive simulations. The simulation results show that the proposed scheme provides higher throughput and better quality for stream media.

Robust error concealment for visual communications in burst-packet-loss networks

Compressed video sequences are very vulnerable to channel disturbances when they are transmitted through the burst-packet-loss networks such as a wireless channel. An effective solution for packet loss is to perform concealment at the receiver. Most of the spatial and/or temporal concealment methods estimate the true motion vector (MV) associated with a missing block by using the MV's of the adjacent blocks. However, these methods cannot conceal effectively the missing blocks, since the neighboring blocks can be easily lost by the burst packet losses destroying a large area of coded video. In this paper, we present a robust error concealment method of estimating the true MV of the missing block by using the bidirectional motion tracking technique within neighboring frames of a damaged frame. This motion tracking technique uses the linear trajectory feature of the image block in the forward and backward prediction directions. Experimental results sow that the proposed algorithm has superior subjective and objective video quality than conventional error concealment methods in the burst-packet-loss environments.

Quantifying the temporal characteristics of network congestion events for multimedia services

Effective quality-of-service (QoS) metrics must relate to end-user experience. For multimedia services these metrics should focus on phenomena that are observable by the end user. Once a congestion event occurs in the network it tends to persist, resulting in long bursts of consecutive packet loss. Such an event is observable to the network customer. There is a need to increase our understanding of the temporal characteristics of congestion. It has become increasingly apparent that the temporal characteristics of congestion events have the dominant effect on user-perceived QoS. A rigorous definition of the time between congestion events is given here, as well as an associated prediction methodology. The inter-congestion event time or the rate of congestion events per unit time provides a network quality metric that is easily understandable to network users and is conveniently predicted and measured. The contribution of this paper is the definition of a metric to characterize congestion events and development of an analytic methodology to predict the expected number of congestion events per unit time. The proposed methodology is evaluated for a variety of traffic models.

Unequal loss protection for robust transmission of motion compensated video over the internet

This paper presents an unequal loss protection (ULP) scheme for robust transmission of motion compensated video over the Internet. By exploiting the temporal dependency between frames, forward error correction (FEC) codes across packets are assigned to different frames in a group of pictures in the sense of minimizing the effect of error propagation, thus improving video quality significantly. To achieve optimal allocation of FEC codes, we formulate the effect of packet loss on video quality degradation as an expected length of error propagation (ELEP) model, which makes sense intuitively, as fewer frames corrupted implies better quality of reconstructed video. Experimental results show the validity of the proposed ELEP model and that the associated ULP scheme is robust to burst packet loss in the Internet. More importantly, graceful degradation of video quality is achieved by the proposed scheme as the packet loss rate of an Internet connection increases.