Adaptive hybrid error correction model for video streaming over wireless networks

The Hybrid ARQ (HARQ) mechanism is the well-known error packet recovery solution composed of the Automation Repeat reQuest (ARQ) mechanism and the Forward Error Correction (FEC) mechanism. However, the HARQ mechanism neither retransmits the packet to the receiver in time when the packet cannot be recovered by the FEC scheme nor dynamically adjusts the number of FEC redundant packets according to network conditions. In this paper, the Adaptive Hybrid Error Correction Model (AHECM) is proposed to improve the HARQ mechanism. The AHECM can limit the packet retransmission delay to the most tolerable end-to-end delay. Besides, the AHECM can find the appropriate FEC parameter to avoid network congestion and reduce the number of FEC redundant packets by predicting the effective packet loss rate. Meanwhile, when the end-to-end delay requirement can be met, the AHECM will only retransmit the necessary number of redundant FEC packets to receiver in comparison with legacy HARQ mechanisms. Furthermore, the AHECM can use an Unequal Error Protection to protect important multimedia frames against channel errors of wireless networks. Besides, the AHECM uses the Markov model to estimate the burst bit error condition over wireless networks. The AHECM is evaluated by several metrics such as the effective packet loss rate, the error recovery efficiency, the decodable frame rate, and the peak signal to noise ratio to verify the efficiency in delivering video streaming over wireless networks.     

Hybrid Erasure-Error Protocols for Wireless Video

Many recently proposed cross-layer protocols for wireless video, have advocated the relay of corrupted packet to higher layers. Such protocols lead to both errors and erasures at the compressed video application layer. We generically refer to such schemes as hybrid erasure-error protocols (HEEPs). In this paper, we analyze the utility of HEEPs for efficient transmission of video over wireless channels. In order to maintain the generic nature of the deductions in this paper, we base our analysis on two (rather abstract) communication schemes for wireless video: hybrid error-erasure cross-layer design (CLD) and hybrid error-erasure cross-layer design with side-information (CLDS). We make a comparative analysis of the channel capacities of these schemes over single and multi-hop wireless channels to identify the conditions under which the HEEPs can provide improved performance over conventional (CON) protocols. In addition, we employ Reed Solomon (RS) and low-density parity check (LDPC)-code-based forward-error correction (FEC) schemes to illustrate that the improvement in capacity can easily enable an FEC scheme employed in conjunction with a HEEP to provide improved throughput. Finally we compare the performance of CON, CLD, and CLDS in terms of video quality using the H.264 video standard. The simulation results show a significant advantage for the HEEPs     

Delay Constraint Error Control Protocol for Real-Time Video Communication

Real-time video communication over wireless channels is subject to information loss since wireless links are error-prone and susceptible to noise. Popular wireless link-layer protocols, such as retransmission (ARQ) based 802.11 and hybrid ARQ methods provide some level of reliability while largely ignoring the latency issue which is critical for real-time applications. Therefore, they suffer from low throughput (under high-error rates) and large waiting-times leading to serious degradation of video playback quality. In this paper, we develop an analytical framework for video communication which captures the behavior of real-time video traffic at the wireless link-layer while taking into consideration both reliability and latency conditions. Using this framework, we introduce a delay constraint packet embedded error control (DC-PEEC) protocol for wireless link-layer. DC-PEEC ensures reliable and rapid delivery of video packets by employing various channel codes to minimize fluctuations in throughput and provide timely arrival of video. In addition to theoretically analyzing DC-PEEC, the performance of the proposed scheme is analyzed by simulating real-time video communication over “real” channel traces collected on 802.11b WLANs using H.264/AVC JM14.0 video codec. The experimental results demonstrate performance gains of 5–10 dB for different real-time video scenarios.      

Providing MAC QoS for multimedia traffic in 802.11e based multi-hop ad hoc wireless networks

Ad hoc wireless networks with their widespread deployment, now need to support applications that generate multimedia and real-time traffic. Video, audio, real-time voice over IP, and other multimedia applications require the network to provide guarantees on the Quality of Service (QoS) of the connection. The 802.11e Medium Access Control (MAC) protocol was proposed with the aim of providing QoS support at the MAC layer. The 802.11e performs well in wireless LANs due to the presence of Access Points (APs), but in ad hoc networks, especially multi-hop ones, it is still incapable of supporting multimedia traffic.One of the most important QoS parameters for multimedia and real-time traffic is delay. Our primary goal is to reduce the end-to-end delay, thereby improving the Packet Delivery Ratio of multimedia traffic, that is, the proportion of packets that reach the destination within the deadline, in 802.11e based multi-hop ad hoc wireless networks.Our contribution is threefold: first we propose dynamic ReAllocative Priority (ReAP) scheme, wherein the priorities of packets in the MAC queues are not fixed, but keep changing dynamically. We use the laxity and the hop length information to decide the priority of the packet. ReAP improves the PDR by over 28% in comparison with 802.11e, especially under heavy loads. Second, we introduce Adaptive-TXOP (A-TXOP), where transmission opportunity (TXOP) is the time interval during which a node has the right to initiate transmissions. This scheme reduces the delay of video traffic by reducing the number of channel accesses required to transmit large video frames. It involves modifying the TXOP interval dynamically based on the packets in the queue, so that fragments of the same packet are sent in the same TXOP interval. A-TXOP is implemented over ReAP to further improve the performance of video traffic. ReAP with A-TXOP helps in reducing the delay of video traffic by over 27% and further improves the quality of video in comparison with ReAP without A-TXOP. Finally, we have TXOP-sharing, which is aimed at reducing the delay of voice traffic. It involves using the TXOP to transmit to multiple receivers, in order to utilize the TXOP interval fully. It reduces the number of contentions to the channel and thereby reduces the delay of voice traffic by over 14%. A-TXOP is implemented over ReAP to further improve the performance of voice traffic. The three schemes (ReAP, A-TXOP, and TXOP-sharing) work together to improve the performance of multimedia traffic in 802.11e based multi-hop ad hoc wireless networks.     

A network coding based hybrid ARQ algorithm for wireless video broadcast

This paper proposes a network coding (NC) based hybrid ARQ (HARQ) algorithm for video broad- cast over wireless networks. The sender applies NC technique to combine the lost packets of different terminals, so that multiple terminals can recover their lost packets via per transmission from the sender. The proposed algorithm combines the advantage of FEC scheme and NC based ARQ scheme so as to maximize not only wireless throughput but also video quality for broadcast communication. Simulation results show tha...     

一种用于无线局域网可伸缩视频传输的跨层结构与SBED算法*

为了增强802.11e无线局域网上的视频传输质量,提出了一种跨层结构与自适应映射算法。将H.264可伸缩视频编码（SVC）的层次信息与802.11e MAC层的访问类（AC）相结合,基于重要性早期检测（SBED）策略将SVC数据包动态映射到合适的AC上,在基础层损失率与平均重要性损失度间实现良好的平衡。仿真表明,该方案的PSNR性能明显优于现有的静态与随机映射方案。     

Priority and delay aware packet management framework for real-time video transport over 802.11e WLANs

The rigid delay constraint is one of the most challenging issues in real-time video delivery over wireless networks. The expired video packets will become useless for the decoding and display even if they are received correctly at the receiver. Because the significance of each video packet is different, the schedulers have to take into account not only the urgency of the packet but also its importance in the real-time video applications. However, the existing QoS-based IEEE 802.11e MAC protocol leaves the urgency and the importance of video packets out of consideration. This paper proposes a Priority and Delay Aware Packet Management Framework (PDA-PMF) to improve the transmission quality of real-time video streaming over IEEE 802.11e WLANs. In the MAC layer, this framework estimates the delay of each video packet. Subsequently, video packets are sent or dropped according to both the significance of the video packets and the estimation value of the delay. Simulation results show that the proposed scheme can not only reduce the packet losses, but also protect the more important video packets, so as to improve the received video quality effectively.     

Cross-Layer Error Control for Multimedia Streaming in Wireless/Wireline Packet Networks

In this paper, we propose a cross-layer error control framework for robust and low delay multimedia streaming in tandem-connected IEEE 802.11 wireless LANs and the Internet. For this network configuration, we model the end-to-end delay and packet loss rate as a function of the automatic repeat request (ARQ) and forward error correction (FEC) error control mechanisms that are employed at the application and wireless link layers. The analytical model is used as the basis of a delay-constrained error control algorithm that adapts the protection level at the application and link layers so that the end-to-end packet loss rate is minimized. With extensive simulations, we validate the efficiency of the proposed cross-layer error control methodology for delay-sensitive pre-compressed video streaming.      

On Channel Capacity Estimation and Prediction for Rate-Adaptive Wireless Video

Packet drops caused by residue errors (MAC-layer errors) can severely deteriorate the wireless video quality. Prior studies have shown that this loss of quality can be circumvented by using forward error correction (FEC) to recover information from the corrupted packets. The performance of FEC encoded video streaming is critically dependent upon the choice of source and channel coding rates. In practice, the wireless channel conditions can vary significantly, thus altering the optimal rate choices. Thus, it is essential to develop an architecture which can estimate the channel capacity and utilize this estimate for rate allocation. In this paper we develop such a framework. Our contributions consist of two parts. In the first part we develop a prediction framework that leverages the received packets' signal to silence ratio (SSR) indications and MAC-layer checksum as side information to predict the operational channel capacity. In the second part, we use this prediction framework for rate allocation. The optimal rate allocation is dependent upon the channel capacity, the distribution of the (capacity) prediction error and the rate-distortion (RD) characteristics of the video source. Consequently, we propose a framework that utilizes the aforementioned statistics for RD optimal rate adaptation. We exhibit the efficacy of the proposed scheme by simulations using actual 802.11b wireless traces, an RD model for the video source and an ideal FEC model. Simulations using source RD models derived from five different popular video codecs (including H.264), show that the proposed framework provides up-to 5-dB improvements in peak signal-to-noise ratio (PSNR) when compared with conventional rate-adaptive schemes.      

Efficient multimedia transmission using adaptive packet bursting for wireless LANs

In addition to an unprecedented 600 Mb/s physical data rate in upcoming standards with greater than 1 Gb/s being considered for future systems, 802.11 has evolved from its earlier incarnations to become a ubiquitous, high-throughput wireless access technology utilized in heterogeneous networks. Although capable of advanced QoS provisioning, commercial 802.11 implementations often support only a subset of QoS specifications, utilize manufacturer-specific QoS enhancements, or use sub-optimal MAC reference specifications with limited capability in achieving optimal system throughput and QoS provisioning.Although more efficient ARQ modes have been defined in the 802.11e and imminent 802.11n amendment standards, opportunities exist within this framework for further optimization through dynamic adaptation of key ARQ-related parameters, which is out of scope of current 802.11 standard specifications. Considering such opportunities, this article presents a novel adaptive ARQ scheme designed to improve the quality and reliability of multimedia transmission through the real-time adaptation of the maximum packet burst size and actual ARQ mode employed. Comprehensive simulation studies show that this scheme can potentially improve the QoS and throughput performance of multimedia traffic in both existing and future 802.11 wireless LANs.     

单向信道的信息可靠传输机制研究

提出了两级FEC的机制来解决单向信道的信息可靠传输的问题。数据包级FEC（第一级）通过添加冗余数据包来恢复传输过程中数据包的丢失。比特级FEC（第二级）通过添加冗余比特来纠正数据在传输过程中比特差错。鉴于包头信息是数据包级FEC的关键,提出数据包包头FEC机制来尽量保证包头信息的正确。提出的方案能很好地恢复丢失的数据包,特别是连续数据包的丢失,能很好地纠正比特差错。     

Adaptive unequal protection for wireless video transmission over IEEE 802.11e networks

Packet loss of video streams cannot be avoided at wireless links for limited wireless bandwidth and frequently changed environments. To provide differentiated Quality of Service (QoS) guarantees between multimedia and data services, IEEE 802.11e was proposed. However, its performance and flexibility need to be further improved. In this paper, after a survey on various modifications of IEEE 802.11e, we formulate the problem of video transmission over IEEE 802.11e networks to help scheme design and performance analysis. Then accompanied with in-depth analysis, an adaptive unequal protection schema is proposed, which is composed of three mechanisms: (1) Insert each video packet into the access category (AC) with the minimum relative queuing delay; (2) Assign each packet dynamically to a proper AC based on several parameters to guarantee the transmission of high priority frames; (3) Apply fuzzy logic controllers to adjust parameters so as to reply quickly to the variation of video data rate, coding structure and network load. Finally, regarding MPEG-4 codec as the example, we perform extensive evaluations and validate the effectiveness and flexibility of proposed scheme. Simulations are divided into WLAN and multihop parts, involving different video sequences and various traffic modes of data streams. Beside performance comparison between proposed scheme and other ones, influence of parameter setting and combination with routing algorithms are also evaluated.     

A robust method for soft IP handover

Multimedia streaming over wireless networks - often called mobile multimedia streaming lets users access music, movie, and news services at any time, regardless of location. Given that multimedia streaming is a key goal of third-generation and future wireless networks, vendors will soon deploy streaming clients in advanced mobile terminals. Current mobile terminals, however, fail to adequately support mobile multimedia communication because wireless networks have high packet-loss rates. To eliminate packet loss during handover, we use a packet path diversity scheme and an end-to-end bicasting mechanism that enables soft IP handover. To offset wireless errors, we use a forward error correction (FEC) scheme and embed it in the bicasting mechanism. Our bicasting method encodes the data stream and then splits it, providing more effective diversity than general bicasting, which sends the same data down both paths.' To support our method, we propose the mobile multimedia streaming protocol (MMSP), a new transport-layer protocol that supports multihoming and bicasting in combination with FEC.     

Concurrent multipath transmission combining forward error correction and path interleaving for video streaming

Video streaming is a popular application on next generation networks (NGNs). However, video streaming over NGNs has many challenges due to the high bit error rates of these networks. Forward error correction (FEC) is often applied to improve the quality of video streaming. However, continuous lost packets decrease the recovery performance of FEC protection in NGNs. To disperse continuous lost packets to different FEC blocks, we propose a concurrent multipath transmission that combines FEC with path interleaving. Our proposed control scheme adaptively adjusts the FEC block length and concurrently sends data interleaved over multiple paths. Experimental results with our approach show improved packet loss and signal to noise ratio performance.     

自适应前向纠错增强TCP在无线链路上性能研究

针对无线链路引发的TCP性能缺陷,提出一种适用于TCP端对端前向纠错的自适应算法,并在此基础上利用Reed Solomon码实现一种新的TCP ARS。TCP ARS利用包一级的前向纠错在传输层恢复丢失的分组,并能自适应地根据网络丢包率、往返时延等参数调整前向纠错的冗余信息度,较好地解决了TCP在无线环境中的性能缺陷问题。     

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.     

存在隐藏终端的IEEE 802.11e多跳无线网络模型分析*

首次将802.11e的接入机制放入多跳环境中进行仿真分析和定量研究。提出一种新的多跳环境下802.11e网络模型,结合M/G/1/K排队模型,定量分析在隐藏终端影响下802.11e网络的MAC层吞吐率、MAC时延和帧丢失率,研究802.11e在多跳环境下性能表现的内在原因。经过仿真实验结果与数值分析结果的对比,验证了分析模型的准确性;通过有（无）隐藏终端影响下MAC层性能的对比分析,指出了802.11e在多跳无线网络中支持QoS的局限性：受隐藏终端的影响,802.11e对不同接入等级的业务提供QoS区分的性能明显降级。因而有必要研究隐藏终端问题的解决方案,提高802.11e在多跳环境下的性能。     

存在隐藏终端的IEEE802.11e多跳无线网络模型分析

首次将802.11e的接入机制放入多跳环境中进行仿真分析和定量研究。提出一种新的多跳环境下802.11e网络模型,结合M/G/1/K排队模型,定量分析在隐藏终端影响下802.11e网络的MAC层吞吐率、MAC时延和帧丢失率,研究802.11e在多跳环境下性能表现的内在原因。经过仿真实验结果与数值分析结果的对比,验证了分析模型的准确性;通过有(无)隐藏终端影响下MAC层性能的对比分析,指出了802.11e在多跳无线网络中支持QoS的局限性:受隐藏终端的影响,802.11e对不同接入等级的业务提供QoS区分的性能明显降级。因而有必要研究隐藏终端问题的解决方案,提高802.11e在多跳环境下的性能。     

Generic forward error correction of short frames for IP streaming applications

If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The generic forward error correction (FEC) mechanisms proposed in the literature for RTP transmission do not perform optimally in terms of stability when the RTP payload consists of several individual data elements of equal priority. In this paper, we present a novel approach for generating FEC packets optimized for applications packing multiple individually decodable media frames in each RTP payload. In the proposed method, a set of frames and its corresponding FEC data are spread among multiple packets so that the experienced frame loss rate does not vary greatly under different packet loss patterns. We verify the performance improvement gained against traditional generic FEC by analyzing and comparing the variance of the residual frame loss rate in the proposed packetization scheme and in the baseline generic FEC.