Small‐block interleaving for low‐delay cross‐packet forward error correction over burst‐loss channels

By adding the redundant packets into source packet block, cross‐packet forward error correction (FEC) scheme performs error correction across packets and can recover both congestion packet loss and wireless bit errors accordingly. Because cross‐packet FEC typically trades the additional latency to combat burst losses in the wireless channel, this paper presents a FEC enhancement scheme using the small‐block interleaving technique to enhance cross‐packet FEC with the decreased delay and improved good‐put. Specifically, adopting short block size is effective in reducing FEC processing delay, whereas the corresponding effect of lower burst‐error correction capacity can be compensated by deliberately controlling the interleaving degree. The main features include (i) the proposed scheme that operates in the post‐processing manner to be compatible with the existing FEC control schemes and (ii) to maximize the data good‐put in lossy networks; an analytical FEC model is built on the interleaved Gilbert‐Elliott channel to determine the optimal FEC parameters. The simulation results show that the small‐block interleaved FEC scheme significantly improves the video streaming quality in lossy channels for delay‐sensitive video. Copyright © 2013 John Wiley & Sons, Ltd.     

Estimation of accurate effective loss rate for FEC video transmission

We propose an algorithm for adjusting data transmission parameters, such as the packet size and the code rate of forward error correction (FEC), to obtain maximum video quality under dynamic channel conditions. When determining transmission parameters, it is essential to calculate an accurate effective loss rate that reflects FEC recovery failures and over-deadline packets. To this end, we analyze the delays caused by FEC coding and the potential packet size variations. In our analysis, we consider the effect of delayed transmission of video packets incurred by the parity packets as well as the encoder and decoder buffers. With the analysis reflecting the delay effect, we are able to accurately estimate the delay patterns of all video packets. Based on the analysis results, we establish an accurate model for estimating the effective loss rate. Simulations show that the proposed effective loss rate model accurately estimates the effective loss rate and significantly improves the reconstructed video quality at the receiver.     

Power allocation and transmission scheduling for a network with bidirectional relaying links

In this paper, we study packet transmission scheduling for a network with bidirectional relaying links, where the relay station can use network coding to combine packets to multiple receivers and opportunistically decide the number of packets to be combined in each transmission. Two cases are considered, depending on whether nodes are allowed to overhear transmissions of each other. A constrained Markov decision process is first formulated with an objective to minimize the average delay of packet transmissions, subject to the maximum and average transmission power limits of the relay node. The complexity for solving the constrained Markov decision process (MDP) is prohibitively high, although the computational complexity for the no‐overhearing case can be greatly reduced. Heuristic schemes are then proposed, one applies to the general case, and another applies to only the no‐overhearing case. Numerical results demonstrate that the heuristic schemes can achieve close‐to‐optimum average packet transmission delay, and furthermore, the second scheme achieves lower maximum delay while keeping the same average packet transmission delay and relay node power consumption as the first one. Copyright © 2015 John Wiley & Sons, Ltd.     

Improving link layer performance on satellite channels with shadowing via delayed two-copy selective repeat ARQ

This paper focuses on improving performance of land mobile satellite channels (LMSCs) at high band (Ka-band or EHF band), where shadowing is the primary impediment to reliable data transmission. Compared with multipath fading, shadowing exists on a longer time scale; hence, interleaving to combat shadowing introduces unacceptably large decoding delay. We use Lutz's model to investigate bit-error rate/packet-error rate (BER/PER) performance of interleaving with various forward error correction (FEC) coding as a function of different channel parameters to demonstrate its limited effectiveness for combatting burst errors whose mean duration significantly exceed a link layer (LL) packet. We propose a delayed two-copy selective repeat ARQ (DTC-SR-ARQ) scheme, whereby two copies of a packet are sent-the second with a delay relative to the first-in every transmission or retransmission. Closed-form expressions for mean transmission time, success probability, and residual loss probability are provided and simulations used to validate the analysis. Furthermore, the issue of optimum delay is addressed as well, and a simple yet effective strategy is suggested to support transmission control protocol (TCP) traffic over this data link layer. DTC-SR-ARQ is shown to achieve much shorter additional delay than interleaving and compared with normal SR-ARQ, reduces mean transmission time at expense of a small increase in residual packet loss probability. Furthermore, ns2 simulation results show that for TCP traffic, DTC-SR-ARQ acquires higher end-to-end throughput than normal SR-ARQ.     

Audio quality in lossy networks for media‐specific forward error correction schemes

A media‐specific forward error correction (FEC) is used to prevent the loss of audio packets due to noise or interference in the transmission and buffer overflow in the router. In this FEC scheme, for nth packet, the redundant data are added in (n + ?)th packet where ? is called the offset. In this paper, we consider an M/M/1/K queueing system and derive some simple expressions for calculating the audio qualities of the aforementioned schemes. We analytically and numerically show some counter‐examples and characteristics for these FEC schemes by using our analytical model. Copyright © 2012 John Wiley & Sons, Ltd.     

Chaotic Interleaving for Robust Image Transmission with LDPC Coded OFDM

To improve the error resilience and video quality over wireless networks, we propose a novel packet-level layer-based interleaving unequal forward error correction (LIU-FEC) method. First, a scalable layer-based interleaving architecture is proposed for improving the efficiency of FEC from successive packet losses in variable channel conditions. The interleaved transmission across different scalable layers can efficiently disperse the consecutive packet losses into different scalable layers. Second, a closed form FEC assignment solution is proposed for minimizing video quality degradation using simple layer-based error propagation metric in hierarchical prediction structure. The simulation results show that the proposed algorithm offers higher PSNR values in various channel status, compared to the conventional FEC algorithm.     

Downlink media streaming with wireless fountain coding in wireline‐cum‐WiFi networks

This paper addresses the problem of streaming packetized media data in a combined wireline/802.11 network. Since the wireless channel is normally the bottleneck for media streaming in such a network, we propose that wireless fountain coding (WFC) be used over the wireless downlink in order to efficiently utilize the wireless bandwidth and exploit the broadcast nature of the channel. Forward error correction (FEC) is also used to combat errors at the application‐layer. We analytically obtain the moment generating function (MGF) for the wireless link‐layer delay incurred by WFC. With the MGF, the expected value of this wireless link‐layer delay is found and used by the access point (AP), who has no knowledge of the buffer contents of wireless receivers, to make a coding‐based decision. We then derive the end‐to‐end packet loss/late probability based on the MGF. We develop an integrated ns‐3/EvalVid simulator to evaluate our proposed system and compare it with the traditional 802.11e scheme which is without WFC capability but equipped with application‐ and link‐layer retransmission mechanisms. Through extensive simulations of video streaming, we show that streaming with WFC is able to support more concurrent video flows compared to the traditional scheme. When the deadlines imposed on video packets are relatively stringent, streaming with WFC also shows superior performance in terms of packet loss/late probability, video distortion, and video frame delay, over the traditional scheme. Copyright © 2011 John Wiley & Sons, Ltd.     

A hybrid interleaving scheme that enables packet switching onmultiple-access radio communication channels

An interleaving technique is proposed that is memory, delay, and packet efficient for demand assignment multiple-access, packet-switched transmission with frequency-hopped survivability interleaved signals over jammed or scintillated radio channels. This hybrid interleaving scheme concatenates the convolutional interleaver and the block interleaver     

Adaptive two-stage FEC scheme for scalable video transmission over wireless networks

In this paper, we present a two-stage forward error correction (FEC) scheme with an enhanced link-layer protocol especially for multimedia data transmission over wireless LANs. At the application layer, packet-level FEC (stage-one) is added across packets to correct packet losses due to congestion and route disruption. Bit-level FEC (stage-two) is then added to both application packets and stage-one FEC packets to recover bit errors from the link layer. Then at the link layer, header-CRC/FEC is used to enhance protection and to cooperate with the two-stage FEC scheme. The proposed scheme thus provides joint protection across the protocol stack. We explore both its bandwidth efficiency and video performance for the highly efficient and scalable MC-EZBC video codec using the network simulator ns-2. Our results show that the proposed scheme can effectively increase application-layer throughput, reduce both end-to-end transmission delay and application bandwidth fluctuation, and significantly improve video performance.     

Managing QoS requirements for video streaming: from intra‐node to inter‐node

Streaming video over IP networks has become increasingly popular; however, compared to traditional data traffic, video streaming places different demands on quality of service (QoS) in a network, particularly in terms of delay, delay variation, and data loss. In response to the QoS demands of video applications, network techniques have been proposed to provide QoS within a network. Unfortunately, while efficient from a network perspective, most existing solutions have not provided end‐to‐end QoS that is satisfactory to users. In this paper, packet scheduling and end‐to‐end QoS distribution schemes are proposed to address this issue. The design and implementation of the two schemes are based on the active networking paradigm. In active networks, routers can perform user‐driven computation when forwarding packets, rather than just simple storing and forwarding packets, as in traditional networks. Both schemes thus take advantage of the capability of active networks enabling routers to adapt to the content of transmitted data and the QoS requirements of video users. In other words, packet scheduling at routers considers the correlation between video characteristics, available local resources and the resulting visual quality. The proposed QoS distribution scheme performs inter‐node adaptation, dynamically adjusting local loss constraints in response to network conditions in order to satisfy the end‐to‐end loss requirements. An active network‐based simulation shows that using QoS distribution and packet scheduling together increases the probability of meeting end‐to‐end QoS requirements of networked video. Copyright © 2005 John Wiley & Sons, Ltd.     

基于H.264 SVC无线视频传输的自适应差错保护方法

提出了一种针对H.264可分级编码(H.264 SVC)的自适应前向纠错编码保护方案.通过比较不同的纠错方案,提出了划分丢包率区间的概念,并根据不同区间的丢包率自适应地选择最佳的纠错方案.仿真结果表明,与单一保护方法相比,所提自适应方法能够取得更好的保护效果,更适于在无线信道中进行视频传输.     

Cross‐layer design for energy efficient communication in wireless sensor networks

There is a plethora of recent research on high performance wireless communications using a cross‐layer approach in that adaptive modulation and coding (AMC) schemes at wireless physical layer are used for combating time varying channel fading and enhance link throughput. However, in a wireless sensor network, transmitting packets over deep fading channel can incur excessive energy consumption due to the usage of stronger forwarding error code (FEC) or more robust modulation mode. To avoid such energy inefficient transmission, a straightforward approach is to temporarily buffer packets when the channel is in deep fading, until the channel quality recovers. Unfortunately, packet buffering may lead to communication latency and buffer overflow, which, in turn, can result in severe degradation in communication performance. Specifically, to improve the buffering approach, we need to address two challenging issues: (1) how long should we buffer the packets? and (2) how to choose the optimum channel transmission threshold above which to transmit the buffered packets? In this paper, by using discrete‐time queuing model, we analyze the effects of Rayleigh fading over AMC‐based communications in a wireless sensor network. We then analytically derive the packet delivery rate and average delay. Guided by these numerical results, we can determine the most energy‐efficient operation modes under different transmission environments. Extensive simulation results have validated the analytical results, and indicates that under these modes, we can achieve as much as 40% reduction in energy dissipation. Copyright © 2008 John Wiley & Sons, Ltd.     

Robust video multicast with joint network coding and video interleaving

In this work, we propose a cross-layer solution to robust video multicast in erasure networks based on random linear network coding (RLNC) in the network layer and video interleaving (VI) in the application layer, and call it the joint RLNC-VI scheme. In the RLNC implementation, we partition one video coding unit (VCU) into several priority levels using scalable properties of H.264/SVC video. Packets from the same priority level of several VCUs form one RLNC generation, and unequal protection is applied to different generations. RLNC provides redundancy for video packets in the network layer and has proved to be useful in a multicast environment. Then, we propose a new packet-level interleaving scheme, called the RLNC-facilitated interleaving scheme, where each received packet corresponds to a new constraint on source packets. As a result, it can facilitate the RLNC decoding at the destination node. Furthermore, we study the problem of optimal interleaving design, which selects the optimal interleaving degree and the optimal redundancy of each generation. The tradeoff between delay and received video quality due to the choice of different VCUs is also examined. It is shown by simulation results that the proposed RLNC-VI scheme outperforms the pure RLNC method for robust video multicast in erasure networks. This can be explained by two reasons. First, the VI scheme distributes the impact of the loss (or erasure) of one VCU into partial data loss over multiple neighboring VCUs. Second, the original video content can be easily recovered with spatial/temporal error concealment (EC) in the joint RLNC-VI scheme.     

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.     

On Broadcast Stability of Queue-Based Dynamic Network Coding Over Erasure Channels

The transmission of packets is considered from one source to multiple receivers over single-hop erasure channels. The objective is to evaluate the stability properties of different transmission schemes with and without network coding. First, the throughput limitation of retransmission schemes is discussed and the stability benefits are shown for randomly coded transmissions, which, however, need not optimize the stable throughput for finite coding field size and finite packet block size. Next, a dynamic scheme is introduced for distributing packets among virtual queues depending on the channel feedback and performing linear network coding based on the instantaneous queue contents. The difference of the maximum stable throughput from the min-cut rate is bounded as function of the order of erasure probabilities depending on the complexity allowed for network coding and queue management. This queue-based network coding scheme can asymptotically optimize the stable throughput to the max-flow min-cut bound, as the erasure probabilities go to zero. This is realized for a finite coding field size without accumulating packet blocks at the source to start network coding. The comparison of random and queue-based dynamic network coding with plain retransmissions opens up new questions regarding the tradeoffs of stable throughput, packet delay, overhead, and complexity.      

FEC在3G多媒体实时传输系统中的应用

3G无线传输是多媒体实时传输的新方式,比传统方式更加灵活、便捷。为了克服在3G无线网络传输中的数据丢包问题,提出了利用FEC和交织来提高数据传输正确率的方法,主要研究了交织和RS(204,188)纠错码的在3G传输系统中的设计和实现,最后对整个系统在不同丢包率信道下的性能进行了对比测试。     

适合TCP的网络编码重传机制

通过理论分析,看出基于反馈的重传方法比定量重传的方法有更低的解码延迟。提出了一种新型的基于反馈的网络编码(FNC)重传机制,利用seen机制中的隐含信息来获取接收方解码所需的重传分组个数,并改变了编码规则使部分分组可以提前解码。该机制不仅可以处理有固定误码率的随机分组丢失,还可以有效地应对大量突发性分组丢失。仿真结果显示,该机制在高误码率下也能保持较高的吞吐量,且极大地减少了解码延迟,传输过程基本不受分组丢失的影响,有效地对拥塞控制协议隐藏了链路错误。算法简单有效,更适于在实际系统中应用。     

A channel reuse strategy with adaptive channel allocation for all-optical WDM networks

The fundamental problems of WDM networks are: (1) high rate of control packet loss and (2) high propagation delay for each (re)transmission. In this paper, we minimize the station randomness to access the control architecture introducing a collisions-free access scheme. We propose a synchronous protocol according which at the end of the propagation delay each station applies a distributed algorithm for packet transmission following the data channel collisions and the receiver collisions avoidance algorithms. We introduce two data transmission stages. The time difference between them is one packet transmission time. At the end of the first stage all data channels are free and can be reused by the remaining data packets during the second stage. The proposed protocol ensures a totally collisions-free performance. The main advantage is that the data channels reuse strategy applied during the second stage provides enhanced transmission probability to the rejected packets during the first stage. This allows the data packets to try retransmission in the same cycle without requiring control packets re-coordination that increases propagation delay. Thus, we achieve large number of data packets transmission, even more than the data channels number, providing throughput improvement and delay reduction, comparing with other studies.     

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).     

Transform domain inter-block interleaving schemes for robust image and video transmission in ATM networks

Data interleaving schemes have proven to be an important mechanism in reducing the impact of correlated network errors on image/video transmission. Current interleaving schemes fall into two main categories: (a) schemes that interleave pixel intensity values and (b) schemes that interleave JPEG/MPEG transform blocks. The schemes in the first category suffer in terms of lower compression ratio since highly correlated information in the spatial domain is de-correlated prior to compression. The schemes in the second category interleave DCT transformed blocks. In this case, in the absence of ARQ, if a packet is lost, an entire block may be lost thus yielding poor image quality and making the error concealment task difficult. Interleaving transform coefficients is tricky and error concealment in the presence of lost coefficients is challenging. In this paper, we develop three different interleaving schemes, namely Triangular, Quadrant, and Coefficient, that interleave frequency domain transform coefficients. The transform coefficients within each block are divided into small groups and groups are interleaved with the groups from other blocks in the image, hence they are referred to as inter-block interleaving schemes. The proposed schemes differ in terms of group size. In the Triangular interleaving scheme AC coefficients in each block are divided into two triangles and interleaving is performed among triangles from different blocks. In the Quadrant interleaving scheme, coefficients in each block are divided into four quadrants and quadrants are interleaved. In the Coefficient interleaving scheme, each coefficient in a block is a group and it is interleaved with the coefficients in other blocks. The compression ratio 3 of the proposed interleaving schemes is impressive ranging from 90 to 98% of the JPEG standard compression while providing much higher robustness in the presence of correlated losses. We also propose two new variable end-of-block (VEOB) techniques, one based on the number of AC coefficients per block (VAC-EOB) and the other based on the number of bits per block (VB–EOB). Our proposed interleaving techniques combined with VEOB schemes yield significantly better compression ratios compared to JPEG (2–11%) and MPEG-2 (3–6.7%) standards while at the same time improve the resilience of the coded data in the presence of transmission errors.