Control of perceived quality of service in multimedia retrieval services: prediction-based mechanism vs. compensation buffers

In multimedia systems end-to-end delay jitter has a great impact on the continuity of information playback. Therefore, it is necessary to introduce appropriate mechanisms to compensate for delay variations, so that the intramedia and intermedia temporal relationships can be preserved. In this paper, two methods for compensation of the network delay jitter in a distributed multimedia retrieval service are compared: the first is based on prediction of the network delay jitter suffered by each information unit and retrieval time modification at the source site; the second is based on a compensation buffer at the destination site. Comparison is made by assuming a master/slave relationship between the monomedia streams composing the multimedia data flow.     

Dejittering in the transport of MPEG-2 and MPEG-4 video

When multimedia information is transported over a packet-switched network, the quality of presentation can be degraded due to network delay variation or jitter. This paper presents a dejittering scheme that can be used in the transport of MPEG-4 and MPEG-2 video to absorb any introduced network jitter, thus preserving the presentation quality of transported media streams. The dejittering scheme is based on the statistical approximation of delay variation in the arrival times of video packets carrying encoded clock reference values and a filtering and re-stamping mechanism. In addition, a brief overview of the MPEG-4 system is presented.     

Transporting QoS adaptive flows

Distributed audio and video applications need to adapt to fluctuations in delivered quality of service (QoS). By trading off temporal and spatial quality to available bandwidth, or manipulating the playout time of continuous media in response to variation in delay, audio and video flows can be made to adapt to fluctuating QoS with minimal perceptual distortion. In this paper, we extend our previous work on a QoS Architecture (QoS-A) by populating the QoS management planes of our architecture with a framework for the control and management of multilayer coded flows operating in heterogeneous multimedia networking environments. Two key techniques are proposed: i) an end-to-end rate-shaping scheme which adapts the rate of MPEG-coded flows to the available network resources while minimizing the distortion observed at the receiver; and ii) an adaptive network service, which offers “hard” guarantees to the base layer of multilayer coded flows and “fairness” guarantees to the enhancement layers based on a bandwidth allocation technique called Weighted Fair Sharing.     

Experiments with video transmission over an Asynchronous Transfer Mode (ATM) network

We present end-to-end performance of digital coded video (JPEG, MPEG-1, and MPEG-2) over a local asynchronuous transfer mode (ATM) network. We discuss performance in terms of both delay (jitter) and frame loss as a function of load. The experimental data reveal that the burstiness of the variable bit-rate-coded video streams degrades the performance significantly when the hosts and the network are stressed. Our results show that traffic smoothing decreases frame loss significantly while maintaining acceptable jitter and loss bounds. We also discuss requirements for system components, such as the network interface and switch, which are necessary to support video services efficiently.     

KNX/EIB通信协议的实时性分析与改进

KNX/EIB通信协议在其链路层使用CSMA/CA机制解决通信冲突问题,虽然提高了KNX/EIB网络防冲突能力,却造成同等级命令帧数据传输时延抖动非常大,传输实时性大打折扣。提出对KNX/EIB实时性的改进方法KNX/EIB-A,在不改变原有的通信协议栈的基础上,将一个调度程序应用于KNX/EIB通信协议的应用层与用户应用程序之间;将原有的分布式平等结构划分为分级主从结构,对数据命令帧的发送进行调度。最后通过对KNX/EIB-A进行分析及原型实现,证明了该方法通过有效地减轻传输时延的抖动从而改进了协议的实时性。     

Adaptive multimedia synchronization in a teleconference system

In this paper, we present an adaptive buffering scheme for implementing intra-stream and inter-stream synchronization in real-time multimedia applications. The essence of the proposed scheme is to dynamically enforce equalized delays to incoming media streams, in order to piece-wise smooth the network delay variations and to synchronize the streams at the sink. An adaptive control mechanism based on an event-counting algorithm is employed to calibrate the PlayOut Clocks (POCs), which manages the presentations of multimedia data. The algorithm does not rely on globally synchronized clock and makes minimal assumption on underlying network delay distribution. Also, the user defined quality of service (QoS) specifications can be directly incorporated into the design parameters of the synchronization algorithm. The proposed synchronization scheme has been experimentally implemented in a teleconference system which consists of separately controllable audio, video, and data channels. The modular structure of the synchronization control provides the flexibility to maintain an arbitrary synchronization group in conjunction with a distributed conference management scheme. This paper also shows the experimental results of the test implementation and the suitability of the proposed scheme with respect to the multimedia traffic across an FDDI/Ethernet network.     

A pricing mechanism for scalable video delivery

Many video applications tolerate continuous media (CM) scaling. Scaling is acceptable due to human tolerance to degradation in picture quality, frame loss, and end-to-end delay. CM scaling enables the network to utilize its resources efficiently for supporting additional customers and to increase its revenue. However, due to quality degradation, users will not be willing to tolerate scaling unless it is coupled with monetary or availability incentives. We propose a pricing policy and a corresponding admission control scheme for scalable video applications. The pricing policy is two-tiered, based on a connection set-up component and a scalable component. Connections that are more scalable are charged less, but are more liable to be degraded. The proposed policy trades off performance degradation with monetary incentives to improve user benefit and network revenue and to decrease the blocking probability of connection requests. We demonstrate by means of simulation that this policy encourages users to specify the scalability of an application to the network.     

Design and analysis of a video-on-demand server

The availability of high-speed networks, fast computers and improved storage technology is stimulating interest in the development of video on-demand services that provide facilities similar to a video cassette player (VCP). In this paper, we present a design of a video-on-demand (VOD) server, capable of supporting a large number of video requests with complete functionality of a remote control (as used in VCPs), for each request. In the proposed design, we have used an interleaved storage method with constrained allocation of video and audio blocks on the disk to provide continuous retrieval. Our storage scheme interleaves a movie with itself (while satisfying the constraints on video and audio block allocation. This approach minimizes the starting delay and the buffer requirement at the user end, while ensuring a jitter-free display for every request. In order to minimize the starting delay and to support more non-concurrent requests, we have proposed the use of multiple disks for the same movie. Since a disk needs to hold only one movie, an array of inexpensive disks can be used, which reduces the overall cost of the proposed system. A scheme supported by our disk storage method to provide all the functions of a remote control such as “fast-forwarding”, “rewinding” (with play “on” or “off”), “pause” and “play” has also been discussed. This scheme handles a user request independent of others and satisfies it without degrading the quality of service to other users. The server design presented in this paper achieves the multiple goals of high disk utilization, global buffer optimization, cost-effectiveness and high-quality service to the users.     

Motion compensated film restoration

Motion picture films are susceptible to local degradations such as dust spots. Other deteriorations are global such as intensity and spatial jitter. It is obvious that motion needs to be compensated for before the detection/correction of such local and dynamic defects. Therefore, we propose a hierarchical motion estimation method ideally suited for high resolution film sequences. This recursive block-based motion estimator relies on an adaptive search strategy and Radon projections to improve processing speed. The localization of dust particles then becomes straightforward. Thus, it is achieved by simple inter-frame differences between the current image and motion compensated successive and preceding frames. However, the detection of spatial and intensity jitter requires a specific process taking advantage of the high temporal correlation in the image sequence. In this paper, we present our motion compensation-based algorithms for removing dust spots, spatial and intensity jitter in degraded motion pictures. Experimental results are presented showing the usefulness of our motion estimator for film restoration at reasonable computational costs. Received: 9 July 2000 / Accepted: 13 January 2002 Correspondence to:S. Boukir     

RCSP and stop-and-go: a comparison of two non-work-conserving disciplines for supporting multimedia communication

To support emerging real-time applications, high-speed integrated services networks must provide end-to-end performance guarantees on a per-connection basis in a networking environment. Resource management algorithms must accommodate traffic that may get burstier as it traverses the network due to complex interactions among packet streams at each switch. To address this problem, several non-work-conserving packet-service disciplines have been proposed. Non-work-conserving servers may be idle and hold packets under certain conditions, to reconstruct, fully or partially, the traffic pattern of the original source inside the network and prevent the traffic from becoming burstier. We compare two non-work-conserving service disciplines. Stop-and-go uses a multilevel framing strategy to allocate resources in a single switch and to ensure traffic smoothness throughout the network. Rate controlled static priority (RCSP) decouples the server functions with two components: (1) a regulator to control traffic distortion introduced by multiplexing effects and load fluctuations in previous servers, and 2) a static priority scheduler to multiplex the regulated traffic. We compare the two service disciplines in terms of traffic specification, scheduling mechanism, buffer space requirement, end-to-end delay characteristics, connection admission-control algorithms, and achievable network utilization. The comparison is first done analytically, and then empirically by using two 10-min traces of MPEG compressed video.     

延迟抖动在TFRC拥塞控制策略中的应用

研究在多媒体实时业务中得到广泛应用的基于速率控制的TCP友好性拥塞控制机制TFRC,分析其基本工作流程、吞吐量模型和关键参数的计算。针对实时业务对于网络传输的要求,提出将延迟抖动作为网络拥塞的预警信号应用于TFRC的拥塞控制中,并采用自适应抖动阈值和调整因子来改进TFRC的速率调整机制。仿真实验结果表明,该方法在实时业务的网络传输中能够取得较好的效果,其友好性和平滑性均得到一定程度的改善。     

An efficient joint channel assignment and QoS routing protocol for IEEE 802.11 multi-radio multi-channel wireless mesh networks

With the emerging of video, voice over IP (VoIP) and other real-time multimedia services, more and more people pay attention to quality of service (QoS) issues in terms of the bandwidth, delay and jitter, etc. As one effective way of broadband wireless access, it has become imperative for wireless mesh networks (WMNs) to provide QoS guarantee. Existing works mostly modify QoS architecture dedicated for ad hoc or sensor networks, and focus on single radio and single channel case. Meanwhile, they study the QoS routing or MAC protocol from view of isolated layer. In this paper, we propose a novel cross-layer QoS-aware routing protocol on OLSR (CLQ-OLSR) to support real-time multimedia communication by efficiently exploiting multi-radio and multi-channel method. By constructing multi-layer virtual logical mapping over physical topology, we implement two sets of routing mechanisms, physical modified OLSR protocol (M-OLSR) and logical routing, to accommodate network traffic. The proposed CLQ-OLSR is based on a distributed bandwidth estimation scheme, implemented at each node for estimating the available bandwidth on each associated channel. By piggybacking the bandwidth information in HELLO and topology control (TC) messages, each node disseminates information of topology and available bandwidth to other nodes in the whole network in an efficient way. From topology and bandwidth information, the optimized path can be identified. Finally, we conduct extensive simulation to verify the performance of CLQ-OLSR in different scenarios on QualNet platform. The results demonstrate that our proposed CLQ-OLSR outperforms single radio OLSR, multi-radio OLSR and OLSR with differentiated services (DiffServ) in terms of network aggregate throughput, end-to-end packet delivery ratio, delay and delay jitter with reasonable message overheads and hardware costs. In particular, the network aggregate throughput for CLQ-OLSR can almost be improved by 300% compared with the single radio case.     

Supporting QoS in IEEE 802.11e wireless LANs over fading channel

Transferring real-time traffic such as voice and video over wireless LAN networks (WLAN) requires stringent delay and jitter requirements. Recently IEEE 802.11e standard has been emerged to support QoS in WLAN. One of the methods to provide QoS in this standard is Enhanced Distributed Channel Access (EDCA) which benefits form the concept of traffic categories. However, EDCA is a contention based method; therefore it can not guarantee strict QoS required by real-time services without proper network control mechanisms. In this paper, we analyze the effect of loss and delay caused by fading channel on EDCA performance. Then, we propose a modification to the media access scheme, called CAFD (Collision Avoidance with Fading Detection) to elevate the performance against channel failures. Moreover an adjustment for the maximum number of retransmissions is proposed to maintain the delay and jitter requirements of the real-time traffic. The performances of the proposed methods are evaluated by simulations.     

MIPv6切换性能测量和瓶颈分析

MIPv6(mobile IPv6)是IETF(Internet Engineering Task Force)工作组提出的IP层移动解决方案.切换是影响MIPv6性能的关键因素.从网络层、传输层和应用层3个层次测量分析MIPv6切换性能,确定协议层次性能相互影响与切换性能瓶颈.根据网络层切换过程,改进其测量移动检测时延的方法,测量MIPv6各个阶段的切换时延并提出减少各阶段时延的建议,分析发现切换性能瓶颈.进一步完成传输层性能测量,分析移动切换对TCP滑动窗口的影响,发现TCP的特性将影响切换过程中上层应用的性能;以FTP应用为例,测量并分析了移动切换对上层应用的影响.相关结论对设计高效的移动切换协议提供了研究基础.     

Multimedia network protocols: where are we?

Protocols for multimedia communication are needed to integrate into a single network services intended to satisfy the different requirements of multiple types of traffic. An essential prerequisite for designing these protocols is that the services to be offered by the network must be selected and specified in detail. We present the service models proposed, or being developed, by the Internet community, by the ATM community, and by the Tenet Group. We compare their common characteristics, which reveal the characteristics of the first integrated services networks are likely to offer. The services referred to in this paper are those at the network and transport layers, which support the services to be offered to the system's end users.     

Online handwriting recognition: the NPen++ recognizer

This paper presents the online handwriting recognition system NPen++ developed at the University of Karlsruhe and Carnegie Mellon University. The NPen++ recognition engine is based on a multi-state time delay neural network and yields recognition rates from 96% for a 5,000 word dictionary to 93.4% on a 20,000 word dictionary and 91.2% for a 50,000 word dictionary. The proposed tree search and pruning technique reduces the search space considerably without losing too much recognition performance compared to an exhaustive search. This enables the NPen++ recognizer to be run in real-time with large dictionaries. Initial recognition rates for whole sentences are promising and show that the MS-TDNN architecture is suited to recognizing handwritten data ranging from single characters to whole sentences. Received September 3, 2000 / Revised October 9, 2000     

Packet audio playout delay adjustment: performance bounds and algorithms

In packet audio applications, packets are buffered at a receiving site and their playout delayed in order to compensate for variable network delays. In this paper, we consider the problem of adaptively adjusting the playout delay in order to keep this delay as small as possible, while at the same time avoiding excessive “loss” due to the arrival of packets at the receiver after their playout time has already passed. The contributions of this paper are twofold. First, given a trace of packet audio receptions at a receiver, we present efficient algorithms for computing a bound on the achievable performance of any playout delay adjustment algorithm. More precisely, we compute upper and lower bounds (which are shown to be tight for the range of loss and delay values of interest) on the optimum (minimum) average playout delay for a given number of packet losses (due to late arrivals) at the receiver for that trace. Second, we present a new adaptive delay adjustment algorithm that tracks the network delay of recently received packets and efficiently maintains delay percentile information. This information, together with a “delay spike” detection algorithm based on (but extending) our earlier work, is used to dynamically adjust talkspurt playout delay. We show that this algorithm outperforms existing delay adjustment algorithms over a number of measured audio delay traces and performs close to the theoretical optimum over a range of parameter values of interest.     

改进的实时流媒体中流量自适应控制机制

为了提高实时多媒体通信的服务质量,在综合考虑网络延迟和网络抖动对实时流媒体的影响下,定义了基于RTT的综合标志量。在此基础上,提出了一种改进的实时流量自适应控制机制。仿真结果表明,与基于丢包率和仅考虑延迟的RTT算法相比,该机制有效提高了数据流的平稳性和带宽的利用率,有更高的自适应性。     

Fast techniques for the optimal smoothing of stored video

Work-ahead smoothing is a technique whereby a server, transmitting stored compressed video to a client, utilizes client buffer space to reduce the rate variability of the transmitted stream. The technique requires the server to compute a schedule of transfer under the constraints that the client buffer neither overflows nor underflows. Recent work established an optimal off-line algorithm (which minimizes peak, variance and rate variability of the transmitted stream) under the assumptions of fixed client buffer size, known worst case network jitter, and strict playback of the client video. In this paper, we examine the practical considerations of heterogeneous and dynamically variable client buffer sizes, variable worst case network jitter estimates, and client interactivity. These conditions require on-line computation of the optimal transfer schedule. We focus on techniques for reducing on-line computation time. Specifically, (i) we present an algorithm for precomputing and storing the optimal schedules for all possible client buffer sizes in a compact manner; (ii) we show that it is theoretically possible to precompute and store compactly the optimal schedules for all possible estimates of worst case network jitter; (iii) in the context of playback resumption after client interactivity, we show convergence of the recomputed schedule with the original schedule, implying greatly reduced on-line computation time; and (iv) we propose and empirically evaluate an “approximation scheme” that produces a schedule close to optimal but takes much less computation time.