Class-Based Packet Scheduling to Improve QoS for IP Video

An important issue for video transmission over IP networks is the preservation of perceived video quality despite packet loss. Packet loss can be detrimental to compressed video. However, reducing packet loss to a very low level is difficult with current techniques. Furthermore, even a very low objective loss probability can still seriously distort perceived video quality. This paper presents a packet scheduling scheme at a router which addresses the loss issues of networked video. Experiments using real video data show that the proposed scheme can significantly improve the visual quality of video and network efficiency. Moreover, it can provide different classes of videos with different levels of loss guarantees, while maintaining service fairness among the same class of videos.     

Impact of playout buffer dynamics on the QoE of wireless adaptive HTTP progressive video

The quality of experience (QoE) of video streaming is degraded by playback interruptions, which can be mitigated by the playout buffers of end users. To analyze the impact of playout buffer dynamics on the QoE of wireless adaptive hypertext transfer protocol (HTTP) progressive video, we model the playout buffer as a G/D/1 queue with an arbitrary packet arrival rate and deterministic service time. Because all video packets within a block must be available in the playout buffer before that block is decoded, playback interruption can occur even when the playout buffer is non-empty. We analyze the queue length evolution of the playout buffer using diffusion approximation. Closed-form expressions for user-perceived video quality are derived in terms of the buffering delay, playback duration, and interruption probability for an infinite buffer size, the packet loss probability and re-buffering probability for a finite buffer size. Simulation results verify our theoretical analysis and reveal that the impact of playout buffer dynamics on QoE is content dependent, which can contribute to the design of QoE-driven wireless adaptive HTTP progressive video management.     

Video quality protection for real time video streams over wireless networks

Received video quality is dependent on the available link rate and the packet loss ratio, which are correlated in a busy network link. Even low packet loss ratios (PLRs) can significantly reduce the video quality. In this paper, a packet level parity Forward Error Correction (FEC) is applied to the video stream in order to reduce the video PLR. A constant gross data rate is assumed, such that adding a FEC leads to a decrease in effective video data rate. The FEC block is truncated at the end of each video VOP, such that there are no inter-VOP dependencies for FEC correction. An algorithm is proposed to optimize the FEC length, based on the Quality of Experience as modelled by the ITU-T R G.1070 standard. It is shown that the optimization algorithm can significantly increase the video quality, without increasing the gross data rate. The algorithm has been evaluated both analytically and through simulations, which confirm the very significant increases in subjective video quality.     

Adaptive Playout Buffer Algorithm for Enhancing Perceived Quality of Streaming Applications

An end-to-end packet delay in the Internet is an important performance parameter, because it heavily affects the quality of real-time applications. In the current Internet, however, because the packet transmission qualities (e.g., transmission delays, jitters, packet losses) may vary dynamically, it is not easy to handle a real-time traffic. In UDP-based real-time applications, a smoothing buffer (playout buffer) is typically used at a client host to compensate for variable delays. The issue of playout control has been studied by some previous works, and several algorithms controlling the playout buffer have been proposed. These studies have controlled the network parameters (e.g., packet loss ratio and playout delay), not considered the quality perceived by users. In this paper, we first clarify the relationship between Mean Opinion Score (MOS) of played audio and network parameters (e.g., packet loss, packet transmission delay, transmission rate). Next, utilizing the MOS function, we propose a new playout buffer algorithm considering user's perceived quality of real-time applications. Our simulation and implementation tests show that it can enhance the perceived quality, compared with existing algorithms.     

Efficient PFD-Based Networking and Buffering Models for Improving Video Quality over Congested Links

Video traffic over the Internet becomes increasingly popular and is expected to comprise the largest proportion of the traffic carried by wired and wireless networks. On the other hand, videos are usually compressed by exploiting spatial and temporal redundancy for the reason of increasing the number of video streams that can be simultaneously carried over links. Unfortunately, receiving high-quality video streaming over the Internet remains a challenge due to the packet loss encountered in the congested wired and wireless links. In addition, the problem is more apparent in wireless links due to not only employing limited system capacity, but also some of the major drawbacks of wireless networks, out of which the bandwidth limitations and link asymmetry which refers to the situation where the forward and reverse paths of a transmission have different channel capacities. Therefore, the wireless hops may be congested which result in dropping many video frames. Additionally, as a result of compressing videos, dependencies among frames and within a frame arise. Consequently, the overall video quality tends to be degraded dramatically. The main challenge is to support the growth of video traffic while keeping the perceived quality of the delivered videos high. In this paper, we extend our previous work concerning improving video traffic over wireless networks through professionally studying the dependencies between video frames and their implications on the overall network performance. In other words, we propose very efficient network and buffer models proportionately to novel algorithms that aim to minimize the cost of aforementioned possible losses by selectively discarding frames based on their contribution to picture quality, namely, partial and selective partial frame discarding policies considering the dependencies between video frames. The performance metrics that are employed to evaluate the performance of the proposed algorithms include the rate of non-decodable frames, peak signal-to-noise ratio, frameput, average buffer occupancy, average packet delay, as well as jitter. Our results are so promising and show significant improvements in the perceived video quality over what is relevant in the current literature. We do not end up to this extent, but rather the effect of producing different bit-stream rates by the FFMPEG codecs on aforementioned performance metrics has been extensively studied.     

视频监视系统中视频质量优化策略研究

为了对视频监视系统中监视质量的进行优化,提出了3种优化控制策略：零拷贝缓冲区策略、网络拥塞抑制策略、编解码速率协调策略。零拷贝缓冲区策略降低了终端负载,提高了系统处理能力,网络拥塞抑制策略有效地减少了丢包率,编解码速率协调策略平衡了系统延时与流畅性。实验测试结果显示,随着监视时间的增加,优化后系统时延基本稳定、丢包率显著减少、视频播放流畅,系统性能满足一般应用需要。     

Scalable video transmission over multi-hop wireless networks with enhanced quality of experience using swarm intelligence

In this work, we take the advantages of the particle swarm optimization method which belongs to the family of swarm intelligence algorithms to find improved solutions for delivering digital video content with enhanced quality of experience to the end users over error-prone multi-hop wireless networks. In video transmission over such wireless networks, many network-based (packet loss, delay, etc.) and source-based (encoding quantization level, etc.) parameters can impair the perceived video quality. The main contributions of the proposed work are twofold. At first, an optimal bandwidth allocation framework is being developed based on the particle swarm optimization algorithm in which by incorporating an accurate video quality metric, the total weighted quality of experience of some competing video sources is being optimized. Secondly, these optimal rates have been used for differentiated quality of experience enforcement between multiple competing scalable video sources. The resulting optimal rates can be used as rate-feedbacks for on-line rate adaptation of a moderate scalable video encoder such as H.264/MPEG4 AVC. The aforementioned weight parameters are selected based on the importance of each video sequence's quality and can be associated with some previous service level agreement based prices. Some guidelines about the practical implementation of the proposed algorithm are given. Numerical analysis has been performed to validate the theoretical results and to verify the claims.     

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.     

Power Efficient Video Multipath Transmission over Wireless Multimedia Sensor Networks

This paper proposes a power efficient multipath video packet scheduling scheme for minimum video distortion transmission (optimised Video QoS) over wireless multimedia sensor networks. The transmission of video packets over multiple paths in a wireless sensor network improves the aggregate data rate of the network and minimizes the traffic load handled by each node. However, due to the lossy behavior of the wireless channel the aggregate transmission rate cannot always support the requested video source data rate. In such cases a packet scheduling algorithm is applied that can selectively drop combinations of video packets prior to transmission to adapt the source requirements to the channel capacity. The scheduling algorithm selects the less important video packets to drop using a recursive distortion prediction model. This model predicts accurately the resulting video distortion in case of isolated errors, burst of errors and errors separated by a lag. Two scheduling algorithms are proposed in this paper. The Baseline scheme is a simplified scheduler that can only decide upon which packet can be dropped prior to transmission based on the packet’s impact on the video distortion. This algorithm is compared against the Power aware packet scheduling that is an extension of the Baseline capable of estimating the power that will be consumed by each node in every available path depending on its traffic load, during the transmission. The proposed Power aware packet scheduling is able to identify the available paths connecting the video source to the receiver and schedule the packet transmission among the selected paths according to the perceived video QoS (Peak Signal to Noise Ratio—PSNR) and the energy efficiency of the participating wireless video sensor nodes, by dropping packets if necessary based on the distortion prediction model. The simulation results indicate that the proposed Power aware video packet scheduling can achieve energy efficiency in the wireless multimedia sensor network by minimizing the power dissipation across all nodes, while the perceived video quality is kept to very high levels even at extreme network conditions (many sensor nodes dropped due to power consumption and high background noise in the channel).     

On protective buffer policies

Studies buffering policies which provide different loss priorities to packets/cells, while preserving packet ordering (space priority disciplines). These policies are motivated by the possible presence, within the same connection, of packets with different loss probability requirements or guarantees, e.g., voice and video coders or rate control mechanisms. The main contribution of the paper is the identification and evaluation of buffering policies which preserve packet ordering and guarantee high priority packets performance (loss probability), irrespective of the traffic intensity and arrival patterns of low priority packets. Such policies are termed protective policies. The need for such policies arises from the difficulty to accurately characterize and size low priority traffic, which can generate large and unpredictable traffic variations over short periods of time. The authors review previously proposed buffer admission policies and determine if they satisfy such “protection” requirements. Furthermore, they also identify and design new policies, which for a given level of protection maximize low priority throughput     

A framework for end-to-end video quality prediction of MPEG video

This paper proposes, describes and evaluates a novel framework for video quality prediction of MPEG-based video services, considering the perceptual degradation that is introduced by the encoding process and the provision of the encoded signal over an error-prone wireless or wire-line network. The concept of video quality prediction is considered in this work, according to which the encoding parameters of the video service and the network QoS conditions are used for performing an estimation/prediction of the video quality level at the user side, without further processing of the actual encoded and transmitted video content. The proposed prediction framework consists of two discrete models: (i) a model for predicting the video quality of an encoded signal at a pre-encoding stage by correlating the spatiotemporal content dynamics to the bit rate that satisfies a specific level of user satisfaction; and (ii) a model that predicts primarily the undecodable frames (and subsequently the perceived quality degradation caused by them) based on the monitored averaged packet loss ratio of the network. The proposed framework is experimentally tested and validated with video signals encoded according to MPEG-4 standard.     

The performance of layered video over an IP network

We evaluate the quality of full-motion video, when transmitted over IP via a two-layer codec and CBQ routers in the presence of TCP traffic. We investigate the effects of packet size and distribution of I-frames. We derive guidelines for the cost-effective divisions of bandwidth between base and enhancement layers, so as to maintain perceived quality in the presence of known amounts of packet loss.     

User-oriented QoS in packet video delivery

We focus on packet video delivery, with an emphasis on the quality of service perceived by the end user. A video signal passes through several subsystems, such as the source coder, the network (ATM or Internet), and the decoder. Each of these can impair the information, either by data loss or by introducing delay. We describe how each of the subsystems can be tuned to optimize the quality of the delivered signal, for a given available bit rate in the network. The assessment of end-user quality is not trivial. We present research results, which rely on a model of the human visual system     

AAL2分组话音复接器缓冲器队列容量的确定方法

该文研究AAL2分组话音复接器缓冲器队列容量的确定方法。提出并从理论上证明用话音分组的最大排队时延为9ms作为确定缓冲器队列容量的标准,可很好地满足分组话音业务服务质量要求的结论,并推导出缓冲器队列容量及门限值的计算公式。仿真结果表明:按作者提出的方法确定缓冲器队列容量及门限值,可获得较低的分组丢弃概率和较小的平均分组排队时延;在满足分组话音业务服务质量要求的前提下,减少了话音分组缓冲器队列的容量,是一种很好的确定缓冲器队列容量和门限值的方法。     

Jitter and clock recovery for periodic traffic in broadband packetnetworks

The transparent transport of real-time periodic traffic through a broadband packet network requires the recovery of source clock frequency at the destination. The arriving packet stream at the destination buffer contains the source clock frequency information, which can be recovered by monitoring the buffer level, and using it to adjust the destination clock frequency without buffer under- or overflow. However, the inherently stochastic nature of packet transport in a broadband packet network makes the recovery difficult. The authors illustrate the packet jitter phenomenon, and develop algorithms to construct the destination clock frequency such that it asymptotically follows the source clock frequency with minimal variation. They provide a systematic design procedure for the implementation of these algorithms, and apply them to a computer model of a single stage FCFS multiplexer to demonstrate their effectiveness and different design trade-offs. They also discuss the application of the algorithms to the audio and video services     

Dynamic buffer management using per‐queue thresholds

Shared buffer switches consist of a memory pool completely shared among output ports of a switch. Shared buffer switches achieve low packet loss performance as buffer space is allocated in a flexible manner. However, this type of buffered switches suffers from high packet losses when the input traffic is imbalanced and bursty. Heavily loaded output ports dominate the usage of shared memory and lightly loaded ports cannot have access to these buffers. To regulate the lengths of very active queues and avoid performance degradations, threshold‐based dynamic buffer management policy, decay function threshold, is proposed in this paper. Decay function threshold is a per‐queue threshold scheme that uses a tailored threshold for each output port queue. This scheme suggests that buffer space occupied by an output port decays as the queue size of this port increases and/or empty buffer space decreases. Results have shown that decay function threshold policy is as good as well‐known dynamic thresholds scheme, and more robust when multicast traffic is used. The main advantage of using this policy is that besides best‐effort traffic it provides support to quality of service (QoS) traffic by using an integrated buffer management and scheduling framework. Copyright © 2006 John Wiley & Sons, Ltd.     

Radio Link Buffer Management and Scheduling for Wireless Video Streaming

In this paper we compare strategies for joint radio link buffer management and scheduling for wireless video streaming. Based on previous work in this area [8], we search for an optimal combination of scheduler and drop strategy for different end-to-end streaming options including timestamp-based streaming and ahead-of-time streaming, both with variable initial playout delay. We will show that a performance gain versus the two best drop strategies in Liebl et al. [8], i.e. drop the HOL packet or drop the packet with the lowest priority starting from HOL, is possible: Provided that some basic side-information on the structure of the incoming video stream is available, a more sophisticated drop strategy removes packets from an HOL group of packets in such a way that the temporal dependencies usually present in video streams are not violated. This advanced buffer management scheme yields significant improvements for almost all investigated scheduling algorithms and streaming options. In addition, we will demonstrate the importance of fairness among users when selecting a suitable scheduler, especially if ahead-of-time streaming is to be applied: Given a reasonable initial playout delay at the streaming media client, both the overall achievable quality averaged over all users, as well as the individual quality of users with bad channel conditions can be increased significantly by trading off fairness with maximum throughput of the system.     

New video discarding policies for improving UDP performance over wired/wireless networks

Multimedia communication over wired and wireless networks becomes a compulsory need for many recent applications. To effectively react to the tremendous demand of video streaming over the Internet, videos are usually compressed by utilizing spatial and temporal redundancy. It is noteworthy to mention that compressing videos may degrade their quality if it is not investigated properly. In other words, as a consequence of exploiting redundancies, frame dependencies emanate, which make discarding frames, because of occupying the whole capacity of network elements, have severe implications on the video quality. Furthermore, transmitting videos over capacity‐limited links owing to error‐prone channels, power constraints and bandwidth variations will severely affect the video quality. Additionally, as the current coding schemes are characterized by being able to afford high compression efficiency, sensitivity to packet losses becomes untolerated. Therefore, insuring the perceived quality of the delivered videos to be always high in spite of aforementioned challenges is the primary focus of current researchers. In this paper, we propose efficient and novel video discarding policies that mainly aim to reduce the number of frames being lost through substitution of those frames that are very difficult or even impossible to decode at the receiver side. This is accomplished by controlling and maintaining the buffer occupancy of network elements. Our proposed policies are evaluated in terms of frameput, rate of non‐decodable frames, peak signal‐to‐noise ratio, structural similarity index and average buffer occupancy. Our proposed policies behave very well and achieve a remarkable enhancement over what is closely connected in the literature. Copyright © 2015 John Wiley & Sons, Ltd.