Parity-based loss recovery for reliable multicast transmission

We investigate how forward error correction (FEC) can be combined with automatic repeat request (ARQ) to achieve scalable reliable multicast transmission. We consider the two scenarios where FEC is introduced as a transparent layer underneath a reliable multicast layer that uses ARQ, and where FEC and ARQ are both integrated into a single layer that uses the retransmission of parity data to recover from the loss of original data packets. To evaluate the performance improvements due to FEC, we consider different loss rates and different types of loss behavior (spatially or temporally correlated loss, homogeneous or heterogeneous loss) for up to 10⁶ receivers. Our results show that introducing FEC as a transparent layer below ARQ can improve multicast transmission efficiency and scalability. However, there are substantial additional improvements when FEC and ARQ are integrated     

Performance of a priority-based dynamic capacity allocation schemefor wireless ATM systems

The performance of a priority-based dynamic capacity allocation suitable for wireless ATM systems is presented. The scheduling of ATM cell transmission in each uplink TDMA frame is based on a priority scheme with priority given to real-time traffic over nonreal-time traffic. Real-time traffic exceeding the uplink capacity is lost while nonreal-time traffic that cannot be served is stored in a first-in first-out (FIFO) queue. An analytical model is developed to evaluate the cell loss ratio (CLR) of both real-time and nonreal-time traffic. Aggregate voice, video, and data traffic is modeled by three two-state Markov-modulated Poisson processes (MMPPs). Analytical results for different system capacities and various traffic loads and scenarios are discussed. Simulation results with on-off sources and approximating MMPP sources are also presented     

Loss Tolerant Bandwidth Aggregation for Multihomed Video Streaming over Heterogeneous Wireless Networks

Bandwidth aggregation is a key research issue in integrating heterogeneous wireless networks, since it can substantially increase the throughput and reliability for enhancing streaming video quality. However, the burst loss in the unreliable wireless channels is a severely challenging problem which significantly degrades the effectiveness of bandwidth aggregation. Previous studies mainly address the critical problem by reactively increasing the forward error correction (FEC) redundancy. In this paper, we propose a loss tolerant bandwidth aggregation approach (LTBA), which proactively leverages the channel diversity in heterogeneous wireless networks to overcome the burst loss. First, we allocate the FEC packets according to the ‘loss-free’ bandwidth of each wireless network to the multihomed client. Second, we deliberately insert intervals between the FEC packets’ departures while still respecting the delay constraint. The proposed LTBA is able to reduce the consecutive packet loss under burst loss assumption. We carry out analysis to prove that the proposed LTBA outperforms the existing ‘back-to-back’ transmission schemes based on Gilbert loss model and continuous time Markov chain. We conduct the performance evaluation in Exata and emulation results show that LTBA outperforms the existing approaches in improving the video quality in terms of PSNR (Peak Signal-to-Noise Ratio).     

An error-controlled approximate analysis of a stochastic fluid flowmodel applied to an ATM multiplexer with heterogeneous on-off sources

The stochastic fluid flow approach is applied to the analysis of the cell loss performance of an ATM multiplexer. The input traffic stream offered to the multiplexer is the superposition of heterogeneous on-off sources with independent and exponentially distributed on and off times. The focus is on the numerical investigation of the steady-state behavior of models involving very large state spaces. To this end, an efficient algorithm for the evaluation of tight upper and lower bounds of the cell loss probability is developed. The algorithm allows a significant reduction of the computational burden, while yielding a guaranteed overestimate of the error implied by the proposed approximation of the cell loss probability. Numerical results are presented both to assess the tightness of the proposed bounds and to gain insight into the behavior of heterogeneous traffic mixes. The main conclusion, from the multiplexer performance evaluation point of view, is that it is not convenient to mix very different traffic streams in a completely shared FIFO buffer, without some kind of control     

On the effects of the packet size distribution on the packet loss process

Real-time multimedia applications have to use forward error correction (FEC) anderror concealment techniques to cope with losses in today’s best-effort Internet. The efficiency of these solutions is known however to depend on the correlation between losses in the media stream. In this paper we investigate how the packet size distribution affects the packet loss process, that is, the distribution of the number of lost packets in a block, the related FEC performance and the average loss run length. We present mathematical models for the loss process of the MMPP+M/D/1/K and the MMPP+M/M/1/K queues; we validate the models via simulations, and compare the results to simulation results with an MPEG-4 coded video trace. We conclude that the deterministic packet size distribution (PSD) not only results in lower stationary loss probability than the exponential one, but also gives a less correlated loss process, both at a particular average link load and at a particular stationary loss probability as seen by the media stream.Our results show that for applications that can only measure the packet loss probability, the effects of the PSD on FEC performance are higher in access networks, where a single multimedia stream might affect the multiplexing behavior. Our results show that the effects of the PSD on FEC performance are higher in access networks, where a single multimedia stream might affect the multiplexing behavior and thus can improve the queuing performance by decreasing the variance of its PSD.     

Simulation study on ATM multiplexing of MPEG video sources

In this paper we study the performance of ATM multiplexing of homogeneous MPEG video sources. A source scheduling method is developed to improve the performance of ATM multiplexer for MPEG video sources. Simulation results show that the level of burstiness for the aggregated MPEG traffic is reduced and the network performance is enhanced. Based on the rationale of the source scheduling method, a simple but efficient bandwidth allocation algorithm is also derived for connection admission of MPEG video in an ATM multiplexer. Copyright © 2001 John Wiley & Sons, Ltd.     

Rate-distortion optimized hybrid error control for real-time packetized video transmission.

The problem of application-layer error control for real-time video transmission over packet lossy networks is commonly addressed via joint source-channel coding (JSCC), where source coding and forward error correction (FEC) are jointly designed to compensate for packet losses. In this paper, we consider hybrid application-layer error correction consisting of FEC and retransmissions. The study is carried out in an integrated joint source-channel coding (IJSCC) framework, where error resilient source coding, channel coding, and error concealment are jointly considered in order to achieve the best video delivery quality. We first show the advantage of the proposed IJSCC framework as compared to a sequential JSCC approach, where error resilient source coding and channel coding are not fully integrated. In the USCC framework, we also study the performance of different error control scenarios, such as pure FEC, pure retransmission, and their combination. Pure FEC and application layer retransmissions are shown to each achieve optimal results depending on the packet loss rates and the round-trip time. A hybrid of FEC and retransmissions is shown to outperform each component individually due to its greater flexibility.     

Performance modeling of FEC‐based unequal error protection for H.264/AVC video streaming over burst‐loss channels

Unequal error protection systems are a popular technique for video streaming. Forward error correction (FEC) is one of error control techniques to improve the quality of video streaming over lossy channels. Moreover, frame‐level FEC techniques have been proposed for video streaming because of different priority video frames within the transmission rate constraint on a Bernoulli channel. However, various communication and storage systems are likely corrupted by bursts of noise in the current wireless behavior. If the burst losses go beyond the protection capacity of FEC, the efficacy of FEC can be degraded. Therefore, our proposed model allows an assessment of the perceived quality of H.264/AVC video streaming over bursty channels, and is validated by simulation experiments on the NS‐2 network simulator at a given estimate of the packet loss ratio and average burst length. The results suggest a useful reference in designing the FEC scheme for video applications, and as the video coding and channel parameters are given, the proposed model can provide a more accurate evaluation tool for video streaming over bursty channels and help to evaluate the impact of FEC performance on different burst‐loss parameters. Copyright © 2014 John Wiley & Sons, Ltd.     

Improved loss calculations at an ATM multiplexer

In this paper we develop a simple and accurate analytical technique to determine the loss probability at an access node to an asynchronous transfer mode (ATM) network. This is an important problem from the point of view of admission control and network design. The arrival processes we analyze are the Markov-modulated Poisson process (MMPP) and the Markov-modulated fluid (MMF) process. These arrival processes have been shown to model various traffic types, such as voice, video, and still images, that are expected to be transmitted by ATM networks. Our hybrid analytical technique combines results from large buffer theories and quasi-stationary approaches to analyze the loss probability of a finite-buffer queue being fed by Markov-modulated sources such as the MMPP and MMF. Our technique is shown to be valid for both heterogeneous and homogeneous sources. We also show that capacity allocation based on the popular effective-bandwidth scheme can lead to considerable under-utilization of the network and that allocating bandwidth based on our model can improve the utilization significantly. We provide numerical results for different types of traffic and validate our model via simulations     

Performance evaluation of 10Base-T and 100Base-T Ethernets carryingmultimedia traffic

The performance of 10Base-T and 100Base-T Ethernet segments carrying audio/video traffic is presented. End-to-end delay requirements suitable for a wide range of multimedia applications are considered (ranging from 20-500 ms). Given the specific nature of the network considered and the maximum latency requirement, some data is lost. Data loss at the receiver causes quality degradations in the displayed video in the form of discontinuities, referred to as glitches. We define various quantities characterizing the glitches, namely, the total amount of information lost in glitches, their duration, and the rate at which glitches occur. We study these quantities for various network and traffic scenarios, using a computer simulation model driven by real video traffic generated by encoding video sequences. We also determine the maximum number of video streams that can be supported for given maximum delay requirement and glitch rate. We consider various types of video contents (video conferencing, motion pictures, commercials), two encoding schemes (H.261 and MPEG-1), and two encoder control schemes [constant bit rate (CBR) and constant-quality variable bit rate (CQ-VBR)] and compare their results. Furthermore, we consider scenarios with mixtures of video and data traffic (with various degrees of burstiness), and determine the effect of one traffic type over the other     

Reliable transmission of high-quality video over ATM networks

The development of broadband networks has led to the possibility of a wide variety of new and improved service offerings. Packetized video is likely to be one of the most significant high-bandwidth users of such networks. The transmission of variable bit-rate (VBR) video offers the potential promise of constant video quality but is generally accompanied by packet loss which significantly diminishes this potential. We study a class of error recovery schemes employing forward error-control (FEC) coding to recover from such losses. In particular, we show that a hybrid error recovery strategy involving the use of active FEC in tandem with simple passive error concealment schemes offers very robust performance even under high packet losses. We discuss two different methods of applying FEC to alleviate the problem of packet loss. The conventional method of applying FEC generally allocates additional bandwidth for channel coding while maintaining a specified average video coding rate. Such an approach suffers performance degradations at high loads since the bandwidth expansion associated with the use of FEC creates additional congestion that negates the potential benefit in using FEC. In contrast, we study a more efficient FEC application technique in our hybrid approach, which allocates bandwidth for channel coding by throttling the source coder rate (i.e., performing higher compression) while maintaining a fixed overall transmission rate. More specifically, we consider the performance of the hybrid approach where the bandwidth to accommodate the FEC overhead is made available by throttling the source coder rate sufficiently so that the overall rate after application of FEC is identical to that of the original unprotected system. We obtain the operational rate-distortion characteristics of such a scheme employing selected FEC codes. In doing so, we demonstrate the robust performance achieved by appropriate use of FEC under moderate-to-high packet losses in comparison to the unprotected system.     

A Recursive Approximation Technique for a Combined Source Queueing Model

The model studied in this paper captures the combined effects of finite and infinite source traffic-often used to model interactive and batch traffic, respectively-when they contend for a single server resource. The finite source traffic is modeled by heterogeneous finite sources, the infinite source traffic by a stationary Poisson process, and the single server is assumed to have exponentially distributed service times with distinct service rates for the different customer types. All customers share a common queue and are serviced in FIFO order. A special case of this model where theNfinite sources are identical combines two fundamental and widely used models (the repairman andM/M/1models) in a natural manner. Regardless of the homogeneous or heterogeneous nature of the finite sources, the combined source model is not product form due to the realistic assumption that service rates are distinct for different customer types (batch and interactive traffic typically have different CPU processing requirements). In this paper, we show how to recursively calculate all mean quantities of interest in an approximate but quite accurate manner for the general heterogeneous model. The accuracy of the recursive technique is established in part by contrasting the approximate solution to simulation results for a wide parameter range, and in part by studying the asymptotic behavior of the approximation.     

Adaptive Rate Control and QoS Provisioning in Direct Broadcast Satellite Networks

Adaptive rate control, if properly employed, is an effective mechanism to sustain acceptable levels of Quality of Service (QoS) in wireless networks where channel and traffic conditions vary over time. In this paper we present an adaptive rate (source and channel) control mechanism, developed as part of an Adaptive Resource Allocation and Management (ARAM) algorithm, for use in Direct Broadcast Satellite (DBS) networks. The algorithm performs admission control and dynamically adjusts traffic source rate and Forward Error Correction (FEC) rate in a co-ordinated fashion to satisfy QoS requirements. To analyze its performance, we have simulated the adaptive algorithm with varying traffic flows and channel conditions. The traffic flow is based on a variable bit rate (VBR) source model that represents Motion Picture Expert Group (MPEG) traffic fluctuations while the DBS channel model is based on a two-state Additive White Gaussian Noise (AWGN) channel. For measures of performance, the simulator quantifies throughput, frame loss due to congestion during transmission as well as QoS variations due to channel (FEC) and source (MPEG compression and data transmission) rate changes. To show the advantage of the adaptive FEC mechanism, we also present the performance results when fixed FEC rates are employed. The results indicate significant throughput and/or quality gains are possible when the FEC/source pairs are adjusted properly in co-ordination with source rate changes.     

Rate-distortion based real-time wireless video streaming

This paper presents wireless video streaming techniques that exploit the characteristics of video content, transmission history, and physical layer channels to enable real-time efficient video streaming over wireless networks to a wireless client. The key contribution of the proposed video streaming techniques is the use of rate-distortion based, but simplified, low complexity packet scheduling as well as forward error correction (FEC) rate selection. To this end, we develop an optimization framework that jointly schedules the packets and selects the FEC rates. The rate-distortion optimized packet scheduling and FEC rate selection provides the optimum quality video on the receiver side albeit at a high computational cost. By some intelligent approximations, rate distortion optimized packet scheduling and FEC rate selection technique is transformed into two sub-optimal but low complexity video streaming techniques that can provide high video quality. We perform extensive simulations to understand the performance of our proposed techniques under different scenarios. Results show that, the proposed techniques improve video quality on the average by 4 dB. We conclude that significant benefits to end-user experience can be obtained by using such video streaming methods.     

Real-time traffic measurement on MAGNET II

Real-time traffic measurements on MAGNET II, an integrated network testbed based on asynchronous time sharing, are reported. The quality of service is evaluated by monitoring the buffer-occupancy distribution, the packet time-delay distribution, the packet loss, and the gap distribution of the consecutively lost packets. The experiments show that both time-delay and buffer-occupancy distributions of multiplexed video sources display a marked bimodal behavior, which does not seem to depend on the buffer size. The reliance of the network designer on traffic sources that do not exhibit substantial correlations can lead to implementations with serious congestion problems. For asynchronous-time-sharing-based networks with different traffic classes, the impact of a traffic class on the performance of the other classes tends to be diminished when compared to single-class-based asynchronous transfer mode (ATM) networks     

Efficiency of error-control schemes for real-time wireless applications on the Gilbert channel

The design of a bandwidth-efficient physical layer for wireless access has always been a challenging task, due to the harsh environment, characterized by impairing phenomena such as radio interference, fading, and shadowing. With circuit switching, a bit-error rate suitable for real-time applications such as voice and video is guaranteed by adopting robust forward error correction (FEC) codes and proper power-budget margins to face fading problems. With this approach, automatic repeat request (ARQ) is used only for applications that require a much lower error rate and can tolerate high delays. The introduction of the packet technique allows the use of ARQ even for real-time traffic. We compare the efficiency of three error-recovering techniques in the presence of traffic with delay constraints, when the memory property of the wireless segment is represented by the Gilbert-Elliot channel. The techniques compared are FEC with interleaving, real-time ARQ, and erasure coding (EC). The comparisons are performed by using both analytical and simulation tools. Two new analytical models are introduced to evaluate the performance of FEC and EC. Simulation is used to validate the analytical results and to derive the performance of real-time ARQ. The numerical results show that when the channel memory increases well beyond the packet-transmission time, the performance of FEC impairs due to the limited interleaving depth, while ARQ and EC remain effective.     

VBR Video Source Characterization and a Practical Hierarchical Model

There are many ways to build up traffic models for VBR video sources. A frequently applied methodology is to use mathematical analysis based on realistic assumptions to set up a source model that generates traffic according to a stochastic process. In this case, the critical issue is the validation of the synthetic trace by comparing statistics to results obtained from measurements on the real source. In this paper, we choose a different and more practical approach to model the behavior of the real traffic source. Our model building philosophy is that we analyze and understand what happens with the video information on its way from the ingress to the multimedia terminal to the egress of the network card. Throughout this journey the information is processed by several mechanisms and we build an empirical model step by step based on our measurement-based observations. Besides understanding the traffic generation procedure, statistical analysis of VBR traffic traces captured from a number of video sequences was also carried out in several scenarios. Using the knowledge of encoding, encapsulation and scheduling processes and results of the trace analysis, a hierarchical source model is set up for modeling the multimedia terminal. Thereby our model imitates the generation of video frames and the inner working of each level of protocol hierarchy and tries to reproduce the complex behavior of the real source. We use the leaky bucket analysis for verification of the model in order to capture directly the behavior of the traffic in a queue.