Efficient user-space protocol implementations with QoS guaranteesusing real-time upcalls

Two important requirements for protocol implementations to be able to provide quality of service (QoS) guarantees within the endsystem are: (1) efficient processor scheduling for application and protocol processing and (2) efficient mechanisms for data movement. Scheduling is needed to guarantee that the application and protocol tasks involved in processing each stream execute in a timely manner and obtain their required share of the CPU. We have designed and implemented an operating system (OS) mechanism called the real-time upcall (RTU) to provide such guarantees to applications. The RTU mechanism provides a simple real-time concurrency model and has minimal overheads for concurrency control and context switching compared to thread-based approaches. To demonstrate its efficacy, we have built RTU-based transmission control protocol (TCP) and user datagram protocol (UDP) protocol implementations that combine high efficiency with guaranteed performance. For efficient data movement, we have implemented a number of techniques such as: (1) direct movement of data between the application and the network adapter; (2) batching of input-output (I/O) operations to reduce context switches; and (3) header-data splitting at the receiver to keep bulk data page aligned. Our RTU-based user-space TCP/Internet protocol (TCP/IP) implementation provides bandwidth guarantees for bulk data connections even with real-time and “best-effort” load competing for CPU on the endsystem. Maximum achievable throughput is higher than the NetBSD kernel implementation due to efficient data movement. Sporadic and small messages with low delay requirements are also supported using reactive RTUs that are scheduled with very low delay. We believe that ours is the first solution that combines good data path performance with application-level bandwidth and delay guarantees for standard protocols and OSs     

Rate control for real-time video network transmission on end-to-end rate-distortion and application-oriented QoS

In the merged multimedia packet-switched networks, the limited or no end-to-end Quality of Service (QoS) guarantees induce that rate control has been evolved to joint source-channel adjustment architecture based on application-oriented QoS. Based on the statistical analysis under the spatial intra and temporal inter prediction subject to a universal spatial-temporal coding framework and a general error concealment by the decoder, an end-to-end distortion estimation model is proposed. On the basis of the analytic model, this paper fulfills a picture quality parameters selection solution with rate-distortion (R-D) Lagrange optimization for a general coding engine including H.264/AVC, exploiting either the source-driven iterative prediction or feedback recursion. Further, a corresponding joint source-channel rate control strategy is proposed. For the real-time variable bit-rate (VBR) video transmission under a given time-varying network condition, the strategy could estimate an instantaneous available transmission rate on traffic smoothing and codec's buffer control, adopt the proposed an end-to-end distortion regressive model and a global optimal error control parameters selection, and address the consistent bit allocation in group of picture (GOP) level, picture level, and MB level. The extensive network simulation experiments show a better and more consistent end-to-end picture quality, in contrast with the locally optimal control strategy at MB level.     

Efficient channel borrowing strategy for real-time services inmultimedia wireless networks

An efficient resource sharing strategy is proposed for multimedia wireless networks. We assume the channel resource in a wireless system is partitioned into two sets: one for voice calls and one for video calls. In the proposed channel borrowing strategy, voice calls can borrow channels from those pre-allocated to video calls temporarily when all voice channels are busy. A threshold type decision policy is designed such that the channel borrowing request will be granted only if the quality of service (QoS) requirement on video call blocking will not be violated during the duration of channel lending. An analytical model is constructed for evaluating the performance of the channel borrowing strategy in a simplified wireless system and is verified by computer simulations. We found that the proposed channel borrowing scheme can significantly reduce the voice call blocking probability while the increase in video call blocking probability is insignificant     

无线信道的自适应视频质量保障技术研究

根据无线信道的质量变化提出一种基于达芬奇技术平台上H.264视频编码库的动态参数优化配置方案,实现了无线信道自适应的实时动态视频质量的保障。详细介绍了无线视频传输系统的编码器端软件平台搭建过程,重点介绍了H.264编码库的参数意义及其优化配置。最后在达芬奇技术平台TMS320DM3730上得以实现,通过对WCDMA无线信道的测试验证该方案设计的有效性和可靠性。     

A real-time video quality estimator for emerging wireless multimedia systems

Wireless Mesh Networks (WMNs) are increasingly deployed to enable thousands of users to share, create, and access live video streaming with different characteristics and content, such as video surveillance and football matches. In this context, there is a need for new mechanisms for assessing the quality level of videos because operators are seeking to control their delivery process and optimize their network resources, while increasing the user’s satisfaction. However, the development of in-service and non-intrusive Quality of Experience assessment schemes for real-time Internet videos with different complexity and motion levels, Group of Picture lengths, and characteristics, remains a significant challenge. To address this issue, this article proposes a non-intrusive parametric real-time video quality estimator, called MultiQoE that correlates wireless networks’ impairments, videos’ characteristics, and users’ perception into a predicted Mean Opinion Score. An instance of MultiQoE was implemented in WMNs and performance evaluation results demonstrate the efficiency and accuracy of MultiQoE in predicting the user’s perception of live video streaming services when compared to subjective, objective, and well-known parametric solutions.     

Efficient transmit diversity scheme for synchronisation channel in OFDM cellular systems

Transmit diversity is a well-known technique that improves receiving performance by mitigating the amplitude variation of received signal. A number of diversity techniques have been investigated for data and control channels in OFDM cellular systems [1]. However, the diversity schemes for the synchronisation channel have not yet been fully investigated. Because of the inherent characteristics of a synchronisation channel, FSTD and TSTD can be considered as diversity schemes [2]. For data and control channels, it is important to increase the average detection probability for reliable system operation. For the synchronisation channel, however, the searching time is more important than the average detection probability. It implies that the transmission scheme increasing the average detection probability after multiple detection trials can be more beneficial even if it degrades the average detection performance at a first detection trial.     

End-user traffic policing for QoS assurance in polyservice RINA networks

Telecommunication Systems - Looking at the ever-increasing amount of heterogeneous distributed applications supported on current data transport networks, it seems evident that best-effort packet...     

Efficient MIMO channel estimation for linear precoded OFDMA uplink systems

A novel and simple uplink MIMO channel estimation technique based on spread pilots well suited to LP OFDMA schemes is proposed. The combination with Alamouti code is studied and two different approaches are compared. The performance evaluated over a realistic MIMO channel shows the efficiency of this novel scheme, which appears to be a promising channel estimation for the uplink of future wideband wireless networks.     

Maximum-throughput delivery of SVC-based video over MIMO systems with time-varying channel capacity

In this paper, we present a novel scalable video transmission strategy over multi-input multi-output (MIMO) wireless systems with time-varying channel capacity. It is a great challenge to simultaneously guarantee the QoS for video delivery and maximize the system throughput over time-varying MIMO channel. We demonstrate that, by making full use of estimated channel state information (CSI) through feedback, a cascade of adaptive operations can be designed to satisfy maximum throughput for scalable video over MIMO systems. These operations include power allocation based on water-filling (WF), adaptive channel selection (ACS), and novel throughput maximizing power reallocation (PR). The proposed ACS transmission scheme enables overall increase in data throughput among enhancement layers by adaptively launching base layer bit-stream to proper sub-channel. Then, after initial power allocation with WF and proper adaptive mode selection, we obtain the surplus power across enhancement layer sub-channels which can be reallocated to some sub-channels by the proposed PR scheme. With such power reallocation, certain enhancement layers will be able to reach new level of QAM modulation through PR so as to maximize the system data throughput. We present in this paper some detailed analysis on these adaptive operations. We also present some simulation results to demonstrate that maximum throughput video transmission over MIMO wireless systems indeed can be achieved based on scalable video coding (SVC) and a sequence of appropriately designed adaptive operations.     

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     

Efficient video transmission over correlated Nakagami fadingchannels for IS-95 CDMA systems

This paper deals with the problem of efficient transmission of video signals over generalized fading channels in direct sequence-spread spectrum (DS-SS) code division multiple access (CDMA) systems. We first propose a modified version of the H.263 video codec incorporating a selective forward error correction (FEC) coding scheme combined with a forced intra-frame update mechanism. The modified codec results in the improvement of the average video and frame-to-frame performance. We further consider a coherent DS-CDMA system for the forward link (base-to-mobile) in both single-cell and multiple-cell environment. We provide performance evaluation results by both analysis, employing the Gaussian approximation, and computer simulations, using Monte Carlo error counting techniques. By integrating the proposed video codec with a coherent DS-CDMA system based upon the IS-95 standard, we investigate the performance of the video transmission over frequency-selective, correlated Nakagami fading forward-link channels employing a RAKE receiver. To simulate the fading channel, we have implemented in software a correlated Nakagami fading simulator based upon the principle of superposition of complex partial waves, an approach which replicates the wave propagation process in actual physical situations. A variety of performance evaluation results, both in single-cell and multiple-cell environment, are presented for a different number of resolving paths, cell user capacity, signal propagation characteristics, as well as for the presence of channel estimation errors. Heuristic explanations and interpretations of the trend of the obtained results are also given     

Delivering scalable video with QoS to the home

User satisfaction is a key factor in the success of novel multimedia services. Yet, to enable service providers and network operators to control and maximize the quality (QoS, QoE) of delivered video streams, quite some challenges remain. In this paper, we particularly focus on three of them. First of all, objectively measuring video quality requires appropriate quality metrics and methods of assessing them in a real-time fashion. Secondly, the recent Scalable Video Coding (SVC) format opens opportunities for adapting video to the available (network) resources, yet the appropriate configuration of video encoding as well as real-time streaming adaptation are largely unaddressed research areas. Thirdly, while bandwidth reservation mechanisms in access/core networks do exist, service providers lack a means for guaranteeing QoS in the increasingly complex home networks (which they are not in full control of). In this paper we offer a broad view on these interrelated issues, by presenting the developments originating in a Flemish research project (including proof-of-concept demonstrations). From a developmental perspective, we propose an architecture combining a real-time video quality monitoring platform, on-the-fly adaptation (optimizing the video quality) and QoS reservation in a heterogeneous home network based on UPnP QoS?v3. From a research perspective, we propose a new subjective test procedure that revealed user preference for temporal scalability over quality scalability. In addition, an extensive study on optimizing HD SVC encoding in IPTV scenarios with fluctuating bandwidth showed that under certain bandwidth constraints (prohibiting sufficient fidelity) spatial scalability is a better option than quality scalability.     

视频流传输的网络质量保障

首先从视频流传输角度对当前研究的各种网络质量保障机制进行了介绍和分析,然后对当前研究的最新动身进行了阐述,最后介绍了基于区分服务体系结构的MPEG视频传输的研究和实验成果。     

基于能效和服务质量保障的动态资源分配机制

以最小化平均消耗功率为目标,提出了一种具有服务质量保障的用户调度和功率分配机制。每个用户维持一个用于存储随机到达业务的数据队列,用户的服务质量要求被刻画成平均排队时延。基于无线信道和数据队列长度的动态变化,将用户调度和功率分配刻画成一个带有约束条件的马尔可夫决策问题。为了应对系统难以精确获取信道和数据到达过程分布参数的情况,采用Q学习算法求解马尔科夫决策问题,进而提出了一种在线学习的用户调度和功率控制算法。系统通过在线学习进行用户调度和功率分配,以实现平均消耗功率的最小化目标。     

Efficient handoff based on link quality prediction for video streaming in urban transport systems

The continuously increasing interest in developing efficient vehicle‐roadside communication networks to provide on‐board connectivity has recently brought to the definition of several solutions based on different wireless technologies. Among them, wireless local area network‐based solutions emerged as an attracting alternative to guarantee high‐quality connectivity enabling services, like video streaming, that require stringent quality of service guarantees. In this work, we propose a handoff procedure based on a forecasting model of the link quality for mobile routers operating in vehicle‐roadside wireless local area network‐based networks. First, a preliminary set of experiments is performed in a realistic environment to study the behavior of the wireless channel when mobility in urban environment is considered. Then, considering the hands‐on experience gained from the initial set of experiments, a novel handoff procedure is designed, which exploits a forecasting technique to predict link channel quality. The proposed procedure is then exploited in a cross‐layer manner to proactively reduce the number of transmitted layers during handoff in the case of real‐time video traffic based on H.264/SVC encoding. The proposal is assessed by means of simulation and compared with existing solutions. Results demonstrate that our proposal guarantees performance comparable with other algorithms. The advantage of predicting the handoff point is demonstrated by means of simulations employing a realistic video streaming traffic model, showing how the quality of experience perceived by end‐users can be improved through the adaptation of the traffic load. Copyright © 2016 John Wiley & Sons, Ltd.     

Uncoordinated real-time video transmission in wireless multicode CDMA systems: An SMPT-based approach

We consider the real-time transmission of encoded video from distributed, uncoordinated wireless terminals to a central base station in a multicode CDMA system. Our approach is to employ the recently proposed simultaneous MAC packet transmission (SMPT) approach at the data link layer (in conjunction with UDP at the transport layer). We consider the real-time transmission of both video encoded in an open loop (i.e., without rate control) and video encoded in a closed loop (i.e., with rate control). We conduct extensive simulations and study quantitatively the trade-off between video quality, transmission delay (and jitter), and number of supported video streams (capacity). We find that the simple-to-deploy SMPT approach achieves significantly higher video quality and smaller delays than the conventional sequential transmission approach, while ensuring high capacity. In typical scenarios, with SMPT the probability of in-time video frame delivery is more than twice as large as with sequential transmission (for given delay bounds). Our results provide guidelines for the design and dimensioning of cellular wireless systems as well as ad hoc wireless systems.     

UMTS-to-IP QoS mapping for voice and video telephony services

Quality of service mapping between UMTS services and IP transport is crucial for maintaining a suitable end-to-end delay for emerging UMTS multimedia telephony. However, due to incompatibilities in QoS classifications within these two technologies, straightforward mapping is impossible and current proposals within the 3GPP could lead to unpredictable and undesirable behavior for certain services. In this article we focus on two very important UMTS services, voice and video telephony, and establish the QoS issues that exist for these services. We then propose a refined QoS mapping that differentiates between the transmission of voice and video-telephony and a weighted fair queuing scheduler to schedule the transmissions. Through a simulation study, we show the effect on the queuing delays of both traffic types when their WFQ weights vary and then derive an optimal weight that provides the best overall delays for multimedia telephony services.     

分布式Web服务QoS注册中的高效负载均衡方法

该文提出了一种分布式Web服务QoS注册系统的负载均衡方法。提出了节点负载状态划分的概念。根据负载状态采用不同的负载信息散布策略,极大地减少了网络消耗。提出了基于简单协商的负载均衡方法。负载均衡用数据复制的方式,主要特点是在进行实际负载均衡操作以前即与复制目标节点进行协商,提出合理的复制需求,然后根据对方提供的资源情况,发起复制。提高了负载均衡的效率,降低了复制的盲目性。该方法在分布式Web服务QoS注册原型系统中进行了实验,达到了较好的效果。     

通过有效的端系统动态资源管理实现QoS控制

本文试图采用全局观点，基于连续媒体活动模型，利用数学工具详细分析在保持端到端媒体服务质量前提下，建立高层的细粒度QoS控制技术－端系统动态资源管理来实现QoS控制，及其策略与算法，最后通过实验验证了管理策略的可行性。