VOIP动态缓出时延算法研究分析

在以包为单位进行数据传输合、语音应用程序(VOIP,Voice Over Internet Protocol)中,为了补偿数据包在网络传输中不可预知的网络传输延迟,在接收端首先必须把接收到的数据包缓存起来,缓存一定的时间再播放出来,以减少通话的抖动,得到比较满意的通话质量。文章主要研究动态缓出时延算法,力求使这个缓出时延尽可能小,同时尽可能减少包的丢失率。文章提出了一个有效动态缓出时延算法,该算法主要跟踪最近到达的数据包的网络传输时延求出其近似分布函数,并利用这些信息和延迟峰的侦测算法预测下一个语音峰的缓出时延。实验结果表明利用该算法可以在缓出时延和包丢失率之间达到最佳平衡,是一种理想、有效的算法。     

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.     

HMM delay prediction technique for VoIP

This paper proposes a new algorithm for predicting audio packet playout delay for voice conferencing applications that use silence suppression. The proposed algorithm uses a hidden Markov model (HMM) to predict the playout delay. Several existing algorithms are reviewed to show that the HMM technique is based on a combination of various desirable features of other algorithms. Voice over Internet protocol (VoIP) applications produce packets at a deterministic rate but various queuing delays are added to the packets by the network causing packet interarrival jitter. Playout delay prediction techniques schedule audio packets for playout and attempt to make a reasonable compromise between the number of lost packets, the one-way delay and the delay variation since these criteria cannot be optimized simultaneously. In particular, this paper will show that the proposed HMM technique makes a good compromise between the mean end-to-end delay, end-to-end delay standard deviation and average packet loss rate.     

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.     

Adaptive playout scheduling and loss concealment for voice communication over IP networks

The quality of service limitation of today's Internet is a major challenge for real-time voice communications. Excessive delay, packet loss, and high delay jitter all impair the communication quality. A new receiver-based playout scheduling scheme is proposed to improve the tradeoff between buffering delay and late loss for real-time voice communication over IP networks. In this scheme the network delay is estimated from past statistics and the playout time of the voice packets is adaptively adjusted. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-scale modification technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same time-scale modification technique is also used to conceal packet loss at very low delay, i.e., one packet time. Simulation results based on Internet measurements show that the tradeoff between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests, showing typical gains of 1 on a 5-point scale of the Mean Opinion Score.     

实时语音的自适应回放算法

由于丢包和延时抖动的引入而使网络传输的实时语音质量让人难以接受，目前对丢包和延时抖动提出了很多的解决方案．但是却很少把这两者结合在一起进行研究。本文提出了一种新的自适应回放算法，通过监测接收和回放队列，结合丢包的自适应恢复技术，达到语音高质量的连续回放。实验证明，该算法能在严格的平均回放延时条件下努力减小由于超时而引起的丢包，获得较好的重建语音质量。     

Dynamic playout scheduling algorithms for continuous multimedia streams

In this paper, we investigate a playout scheduling framework for supporting the continuous and synchronized presentations of multimedia streams in a distributed multimedia presentation system. We assume a situation in which the server and network transmissions provide sufficient support for the delivery of media objects. In this context, major issues regarding the enforcement of the smooth presentation of multimedia streams at client sites must be addressed to deal with rate variance of stream presentations and delay variance of networks. We develop various playout-scheduling algorithms that are adaptable to quality-of-service parameters. The proposed algorithms permit the local adjustment of unsynchronized presentations by gradually accelerating or retarding presentation components, rather than abruptly skipping or pausing the presentation materials. A comprehensive experimental analysis of the proposed algorithms demonstrates that our algorithms can effectively avoid playout gaps (or hiccups) in the presentations. This scheduling framework can be readily used to support customized multimedia presentations.     

Playout buffering of speech packets based on a quality maximization approach

To combat jitter problems in voice streaming over packet networks, playout buffering algorithms are used at the receiver side. Most of the proposed solutions rely on two main operations: prediction of delay statistics for future packets; setting of the end-to-end delay so as to limit or avoid packet losses. In recent years, a new approach has been presented, which is based on using a quality model to evaluate the impact of both packet loss and delay on the voice quality. Such a model is used to find the buffer setting that maximizes the expected quality. In this paper, we present a playout buffering algorithm whose main contribution is the extension of the new quality-based approach to the case of voice communications affected by bursty packet losses. This work is motivated by two main considerations: most of IP telephony applications are characterized by bursty losses instead of random ones; the human perception of the speech quality is significantly affected by the temporal correlation of losses. To this purpose, we make use of the extensions proposed in the ETSI Tiphon for the ITU-T E-Model so as to incorporate the effects of loss burstiness on the perceived quality. The resulting playout algorithm estimates the characteristics of the loss process varying the end-to-end delay, weights the loss and the delay effects on the perceived quality, and maximizes the overall quality to find the optimal setting for the playout buffer. The experimental results prove the effectiveness of the proposed technique.     

Design and Experimental Evaluation of an Adaptive Playout Delay Control Mechanism for Packetized Audio for Use over the Internet

We describe the design and the experimental evaluation of a playout delay control mechanism we have developed in order to support unicast, voice-based audio communications over the Internet. The proposed mechanism was designed to dynamically adjust the talkspurt playout delays to the traffic conditions of the underlying network without assuming either the existence of an external mechanism for maintaining an accurate clock synchronization between the sender and the receiver during the audio communication, or a specific distribution of the audio packet transmission delays. Performance figures derived from several experiments are reported that illustrate the adequacy of the proposed mechanism in dynamically adjusting the audio packet playout delay to the network traffic conditions while maintaining a small percentage of packet loss.     

A synchronization algorithm for distributed multimedia environments

Network synchronization plays a significant role in transmitting multimedia objects over computer networks. Even packets from a single channel must be synchronized due to the problems in a packet switching environment, such as network jitter, frequency, and time offsets. We present an algorithm that determines the set of packets generated periodically by various participants arriving at a node. The basic advantage of the proposed algorithm is that the receiver estimates the reference times (expected arrival times of the packets) and achieves synchronization, without knowledge of the packet delays. The accuracy is improved and the complexity is reduced by predicting the time/frequency offsets between the clocks at the source and the mixer. The error is calculated by the Chernoff bound, demonstrated by simulation, and shown to be acceptable in practical applications.     

Adaptive playout scheduling algorithm tailored for real-time packet-based voice conversations over wireless ad-hoc networks

The effective provision of real-time, packet-based voice conversations over multi-hop wireless ad-hoc networks faces several stringent constraints not found in conventional packet-based networks. Indeed, MANETs (mobile ad-hoc networks) are characterized by mobility of all nodes, bandwidth-limited channel, unreliable wireless transmission medium, etc. This environment will surely induce a high delay variation and packet loss rate impairing dramatically the user experienced quality of conversational services such as VoIP. Indeed, such services require the reception of each media unit before its deadline to guarantee a synchronous playback process. This requirement is typically achieved by artificially delaying received packets inside a de-jitter buffer. To enhance the perceptual quality the buffering delay should be adjusted dynamically throughout the vocal conversation.In this work, we describe the design of a playout algorithm tailored for real-time, packet-based voice conversations delivered over multi-hop wireless ad-hoc networks. The designed playout algorithm, which is denoted MAPA (mobility aware playout algorithm), adjusts the playout delay according to node mobility, which characterizes mobile ad-hoc networks, and talk-spurt, which is an intrinsic feature of voice signals. The detection of mobility is done in service passively at the receiver using several metrics gathered at the application layer. The perceptual quality is estimated using an augmented assessment approach relying on the ITU-T E-Model paradigm while including the time varying impairments observed by users throughout a packet-based voice conversation. Simulation results show that the tailored playout algorithm significantly outperforms conventional playout algorithms, specifically over a MANET with a high degree of mobility.     

Detecting global predicates in distributed systems with clocks

Summary. This paper proposes a framework for detecting global state predicates in systems of processes with approximately-synchronized real-time clocks. Timestamps from these clocks are used to define two orderings on events: “definitely occurred before” and “possibly occurred before”. These orderings lead naturally to definitions of 3 distinct detection modalities, i.e., 3 meanings of “predicate held during a computation”, namely: (“ possibly held”), (“ definitely held”), and (“ definitely held in a specific global state”). This paper defines these modalities and gives efficient algorithms for detecting them. The algorithms are based on algorithms of Garg and Waldecker, Alagar and Venkatesan, Cooper and Marzullo, and Fromentin and Raynal. Complexity analysis shows that under reasonable assumptions, these real-time-clock-based detection algorithms are less expensive than detection algorithms based on Lamport's happened-before ordering. Sample applications are given to illustrate the benefits of this approach. Received: January 1999 / Accepted: November 1999     

Deadlock detection in distributed database systems: a new algorithm and a comparative performance analysis

This paper attempts a comprehensive study of deadlock detection in distributed database systems. First, the two predominant deadlock models in these systems and the four different distributed deadlock detection approaches are discussed. Afterwards, a new deadlock detection algorithm is presented. The algorithm is based on dynamically creating deadlock detection agents (DDAs), each being responsible for detecting deadlocks in one connected component of the global wait-for-graph (WFG). The DDA scheme is a “self-tuning” system: after an initial warm-up phase, dedicated DDAs will be formed for “centers of locality”, i.e., parts of the system where many conflicts occur. A dynamic shift in locality of the distributed system will be responded to by automatically creating new DDAs while the obsolete ones terminate. In this paper, we also compare the most competitive representative of each class of algorithms suitable for distributed database systems based on a simulation model, and point out their relative strengths and weaknesses. The extensive experiments we carried out indicate that our newly proposed deadlock detection algorithm outperforms the other algorithms in the vast majority of configurations and workloads and, in contrast to all other algorithms, is very robust with respect to differing load and access profiles. Received December 4, 1997 / Accepted February 2, 1999     

Adaptive VoIP playout scheduling: assessing user satisfaction

Delay and packet loss dramatically affect the quality of voice-over-IP (VoIP) calls and depend on the playout buffer scheme implemented at the receiver. The choice of playout algorithm can't be based on statistical metrics without considering the perceived end-to-end conversational speech quality. The authors present a method for evaluating various playout algorithms that extends the E-model concept by estimating user satisfaction from time-varying transmission impairments. This article evaluates several playout algorithms and shows a correspondence between the authors' results and those obtained via statistical loss and delay metrics.     

Networks on which hot-potato routing does not livelock

Summary. Hot-potato routing is a form of synchronous routing which makes no use of buffers at intermediate nodes. Packets must move at every time step, until they reach their destination. If contention prevents a packet from taking its preferred outgoing edge, it is deflected on a different edge. Two simple design principles for hot potato routing algorithms are minimum advance, that advances at least one packet towards its destination from every nonempty node (and possibly deflects all other packets), and maximum advance, that advances the maximum possible number of packets. Livelock is a situation in which packets keep moving indefinitely in the network without any packet ever reaching its destination. It is known that even maximum advance algorithms might livelock on some networks. We show that minimum advance algorithms never livelock on tree networks, and that maximum advance algorithms never livelock on triangulated networks. Received: March 1999 / Accepted: August 1999     

Motion of disturbances: detection and tracking of multi-body non-rigid motion

We present a new approach to the tracking of very non-rigid patterns of motion, such as water flowing down a stream. The algorithm is based on a “disturbance map”, which is obtained by linearly subtracting the temporal average of the previous frames from the new frame. Every local motion creates a disturbance having the form of a wave, with a “head” at the present position of the motion and a historical “tail” that indicates the previous locations of that motion. These disturbances serve as loci of attraction for “tracking particles” that are scattered throughout the image. The algorithm is very fast and can be performed in real time. We provide excellent tracking results on various complex sequences, using both stabilized and moving cameras, showing a busy ant column, waterfalls, rapids and flowing streams, shoppers in a mall, and cars in a traffic intersection. Received: 24 June 1997 / Accepted: 30 July 1998     

Wavelet-based video indexing and querying

We present several algorithms suitable for analysis of broadcast video. First, we show how wavelet analysis of frames of video can be used to detect transitions between shots in a video stream, thereby dividing the stream into segments. Next we describe how each segment can be inserted into a video database using an indexing scheme that involves a wavelet-based “signature.” Finally, we show that during a subsequent broadcast of a similar or identical video clip, the segment can be found in the database by quickly searching for the relevant signature. The method is robust against noise and typical variations in the video stream, even global changes in brightness that can fool histogram-based techniques. In the paper, we compare experimentally our shot transition mechanism to a color histogram implementation, and also evaluate the effectiveness of our database-searching scheme. Our algorithms are very efficient and run in realtime on a desktop computer. We describe how this technology could be employed to construct a “smart VCR” that was capable of alerting the viewer to the beginning of a specific program or identifying     

Quantifiable data mining using ratio rules

Association Rule Mining algorithms operate on a data matrix (e.g., customers products) to derive association rules [AIS93b, SA96]. We propose a new paradigm, namely, Ratio Rules, which are quantifiable in that we can measure the “goodness” of a set of discovered rules. We also propose the “guessing error” as a measure of the “goodness”, that is, the root-mean-square error of the reconstructed values of the cells of the given matrix, when we pretend that they are unknown. Another contribution is a novel method to guess missing/hidden values from the Ratio Rules that our method derives. For example, if somebody bought $10 of milk and $3 of bread, our rules can “guess” the amount spent on butter. Thus, unlike association rules, Ratio Rules can perform a variety of important tasks such as forecasting, answering “what-if” scenarios, detecting outliers, and visualizing the data. Moreover, we show that we can compute Ratio Rules in a single pass over the data set with small memory requirements (a few small matrices), in contrast to association rule mining methods which require multiple passes and/or large memory. Experiments on several real data sets (e.g., basketball and baseball statistics, biological data) demonstrate that the proposed method: (a) leads to rules that make sense; (b) can find large itemsets in binary matrices, even in the presence of noise; and (c) consistently achieves a “guessing error” of up to 5 times less than using straightforward column averages. Received: March 15, 1999 / Accepted: November 1, 1999     

WIRE3: Driving Around the Information Super-Highway

Interactive voice browsers offer an alternative paradigm that affords ubiquitous mobile access to the WWW using a wide range of consumer devices. This technology can facilitate a safe, “hands-free” browsing environment that is of importance both to car drivers and various mobile and technical professionals. This paper describes the challenges of architecting an interactive voice browser that combines digital audio with the features of a speech synthesizer to make structural elements of the document explicit to the listener. The aesthetics of the audio rendition can simultaneously help reduce the monotony factor and enhance comprehension. The evolution of the voice browser gave rise to a new conceptual model of the HTML document structure and its mapping to a 3D audio space. A number of novel features are discussed for improving both the user’s comprehension of the HTML document structure and their orientation within it. These factors, in turn, can improve the effectiveness of the browsing experience.