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.     

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。     

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.     

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.     

Delay reduction techniques for playout buffering

Receiver synchronization of continuous media streams is required to deal with delay differences and variations resulting from delivery over packet networks such as the Internet. This function is commonly provided using per-stream playout buffers which introduce additional delay in order to produce a playout schedule which meets the synchronization requirements. Packets which arrive after their scheduled playout time are considered late and are discarded. In this paper, we present the Concord algorithm, which provides a delay-sensitive solution for playout buffering. It records historical information and uses it to make short-term predictions about network delay with the aim of not reacting too quickly to short-lived delay variations. This allows an application-controlled tradeoff of packet lateness against buffering delay, suitable for applications which demand low delay but can tolerate or conceal a small amount of late packets. We present a selection of results from an extensive evaluation of Concord using Internet traffic traces. We explore the use of aging techniques to improve the effectiveness of the historical information and hence, the delay predictions. The results show that Concord can produce significant reductions in buffering delay and delay variations at the expense of packet lateness values of less than 1%     

Smooth Control of Adaptive Media Playout for Video Streaming

Client-side data buffering is a common technique to deal with media playout interruptions of streaming video caused by network jitters and packet losses of best-effort networks. However, stronger playout interruption protection inevitably amounts to larger data buffering and results in more memory requirements and longer playout delay. Adaptive media playout (AMP), also a client-side technique, can reduce the buffer requirement and avoid buffer outage but at the expense of visual quality degradation because of the fluctuation of playout speed. In this paper, we propose a novel AMP scheme to keep the video playout as smooth as possible while adapting to the channel condition. The triggering of the playout control is based on buffer variation rather than buffer fullness. Experimental results show that our AMP scheme surpasses conventional schemes in unfriendly network conditions. Unlike previous schemes that are tuned for a specific range of packet loss and network instability, the proposed AMP scheme maintains consistent performance across a wide range of network conditions.     

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。     

Switching between fixed and call-adaptive playout: a per-call playout algorithm

Adaptive playout algorithms provide a popular way to calculate voice-over-IP (VoIP) packets' playout delay - the difference between the playout time at the receiver and the packet-generation time at the sender. The authors' proposed per-call adaptive algorithm uses network delays received from the VoIP gatekeeper to switch between fixed and call-adaptive playout. Their approach also reduces loss rates while increasing playout delay only slightly.     

A Garch-based adaptive playout delay algorithm for VoIP

Network delay, packet loss and network delay variability (jitter) are important factors that impact on perceived voice quality in VoIP networks. An adaptive playout buffer is used in a VoIP terminal to overcome jitter. Such a buffer-control must operate a trade-off between the buffer-induced delay and any additional packet loss rate. In this paper, a Garch-based adaptive playout algorithm is proposed which is capable of operating in both inter-talkspurt and intra-talkspurt modes. The proposed new model is based on a Garch model approach; an ARMA model is used to model changes in the mean and the variance. In addition, a parameter estimation procedure is proposed, termed Direct Garch whose cost function is designed to implement a desired packet loss rate whilst minimising the probability of consecutive packet losses occurring. Simulations were carried out to evaluate the performance of the proposed algorithm using recorded VoIP traces. The main result is as follows; given a target Packet Loss Rate (PLR) the Direct Garch algorithm produces parameter estimates which result in a PLR closer than other algorithms. In addition, the proposed Direct Garch algorithm offers the best trade-off between additional buffering delay and Packet Loss Rate (PLR) compared with other traditional algorithms.     

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 client-driven media synchronization mechanism for RTP packet-based video streaming

Media synchronization is used to correctly playback a video stream with its associated audio. To support synchronization between video and audio streams transported over IP networks, an RTP/RTCP protocol suite is usually employed. In conventional server-driven media synchronization, the server needs to periodically transmit an RTCP sender report (SR) packet to provide the client with a UTC time in NTP format corresponding to the RTP timestamp carried by each RTP packet. In this paper, we propose a precise client-driven media synchronization mechanism for an RTP packet-based multimedia streaming service. In the proposed method, the server does not need to send any RTCP SR packets for synchronization. Instead, the client device derives the precise normal play time (NPT) for each video and audio stream from the received RTP packets containing an RTP timestamp. Simulations show that the proposed client-driven synchronization method can provide accurate media synchronization without employing an RTCP SR packet and accordingly reduce the initial synchronization delay, the processing complexity at the client device, the number of required user datagram protocol ports, and the amount of control traffic injected into the network.     

多媒体播放中的动态双门限自适应调整算法

针对无线网络存在的自相似特性会影响视频流的播放质量问题,提出了基于滑动窗口的接收端播放缓存调整算法,根据网络流量的变化,动态地调整双门限,并利用播放缓存的占用率来控制视频流的播放速度,平滑时延抖动.仿真实验证明,无论网络流量处于平稳状态还是处于突发状态,本文设计的算法都能够较好地保证视频流的连续播放,提高视频流的播放质量,为用户提供良好的视觉效果.     

Neural and fuzzy computation techniques for playout delay adaptation in VoIP networks

Playout delay adaptation algorithms are often used in real time voice communication over packet-switched networks to counteract the effects of network jitter at the receiver. Whilst the conventional algorithms developed for silence-suppressed speech transmission focused on preserving the relative temporal structure of speech frames/packets within a talkspurt (intertalkspurt adaptation), more recently developed algorithms strive to achieve better quality by allowing for playout delay adaptation within a talkspurt (intratalkspurt adaptation). The adaptation algorithms, both intertalkspurt and intratalkspurt based, rely on short term estimations of the characteristics of network delay that would be experienced by up-coming voice packets. The use of novel neural networks and fuzzy systems as estimators of network delay characteristics are presented in this paper. Their performance is analyzed in comparison with a number of traditional techniques for both inter and intratalkspurt adaptation paradigms. The design of a novel fuzzy trend analyzer system (FTAS) for network delay trend analysis and its usage in intratalkspurt playout delay adaptation are presented in greater detail. The performance of the proposed mechanism is analyzed based on measured Internet delays.     

Dynamic Video Playout Smoothing Method for Multimedia Applications

Multimedia applications including video data require the smoothing of video playout to prevent potential discontinuity. In this paper, we propose a dynamic video playout smoothing method, called the Video Smoother, which dynamically adopts various playout rates in an attempt to compensate for high delay variance of networks. Specifically, if the number of frames in the buffer exceeds a given threshold (TH), the Smoother employs a maximum playout rate. Otherwise, the Smoother uses proportionally reduced rates in an effort to eliminate playout pauses resulting from the emptiness of the playout buffer. To determine THs under various loads, we present an analytic model assuming the Interrupted Poisson Process (IPP) arrival. Based on the analytic results, we establish a paradigm of determining THs and playout rates for achieving different playout qualities under various loads of networks. Finally, to demonstrate the viability of the Video Smoother, we have implemented a prototyping system including a multimedia teleconferencing application and the Video Smoother performing as part of the transport layer. The prototyping results show that the Video Smoother achieves smooth playout incurring only unnoticeable delays.     

An Optimal Hybrid Learning Approach for Attack Detection in Linear Networked Control Systems

A novel learning-based attack detection and estimation scheme is proposed for linear networked control systems (NCS), wherein the attacks on the communication network in the feedback loop are expected to increase network induced delays and packet losses, thus changing the physical system dynamics. First, the network traffic flow is modeled as a linear system with uncertain state matrix and an optimal Q-learning based control scheme over finite-horizon is utilized to stabilize the flow. Next, an adaptive observer is proposed to generate the detection residual, which is subsequently used to determine the onset of an attack when it exceeds a predefined threshold, followed by an estimation scheme for the signal injected by the attacker. A stochastic linear system after incorporating network-induced random delays and packet losses is considered as the uncertain physical system dynamics. The attack detection scheme at the physical system uses the magnitude of the state vector to detect attacks both on the sensor and the actuator. The maximum tolerable delay that the physical system can tolerate due to networked induced delays and packet losses is also derived. Simulations have been performed to demonstrate the effectiveness of the proposed schemes.      

Optimization of a packet video receiver under different levels of delay jitter: an analytical approach

This paper studies the problem of analyzing and designing optimal playout adaptation policies for packet video receivers (PVRs) that operate in a delay jitter inducing best-effort network, like the current Internet. The developed system model is built around the E_k/D_i/1/N phase-type queue and allows for the effective modeling of key design and system parameters, such as: the level of delay jitter, the performance metrics and the employed playout policy. The optimal playout policy is derived under k-Erlang interarrivals by formulating and solving an optimization problem. The (theoretical) optimal solution is transformed into an approximately optimal one that utilizes observable information and it is, thus, feasible. Numerical results are derived under the optimal policy and compared against those under the optimal policy that assumes a fixed level of jitter as determined by Poisson arrivals, as well as against the deterministic service that applies no playout adaptation. Based on this work, a PVR is proposed that adapts to varying network delay jitter and tries to induce a performance that approximates the derived theoretical optimal one.     

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.     

A robust method for estimating synchronization and delay of audio and video for communication services

One of the main contributions to the quality of experience in streaming services or in two-way communication of audio and video applications is synchronization. This has been shown in several studies and experiments but methods to measure synchronization are less frequent, especially for situations without internal access to the application and independent of platform and device. In this paper we present a method for measuring synchronization skewness as well as delay for audio and video. The solution incorporates audio and video reference streams, where audio and video frames are marked with frame numbers which are decoded on the receiver side to enable calculation of synchronization and delay. The method has been verified in a two-way communication application in a transparent network with and without inserting known delays, as well as in a network with 5 and 10 % packet loss levels. The method can be used for both streaming and two-way communication services, both with and without access to the internal structures, and enables measurements of applications running on e.g. smartphones, tablets, and laptops under various conditions.     

Queuing-Based Dynamic Channel Selection for Heterogeneous Multimedia Applications Over Cognitive Radio Networks

In this paper, we propose a dynamic channel-selection solution for autonomous wireless users transmitting delay-sensitive multimedia applications over cognitive radio networks. Unlike prior works that seldom consider the requirement of the application layer, our solution explicitly considers various rate requirements and delay deadlines of heterogeneous multimedia users. Note that the users usually possess private utility functions, application requirements, and distinct channel conditions in different frequency channels. To efficiently manage available spectrum resources in a decentralized manner, information exchange among users is necessary. Hence, we propose a novel priority virtual queue interface that determines the required information exchanges and evaluates the expected delays experienced by various priority traffics. Such expected delays are important for multimedia users due to their delay-sensitivity nature. Based on the exchanged information, the interface evaluates the expected delays using priority queuing analysis that considers the wireless environment, traffic characteristics, and the competing users' behaviors in the same frequency channel. We propose a dynamic strategy learning (DSL) algorithm deployed at each user that exploits the expected delay and dynamically adapts the channel selection strategies to maximize the user's utility function. We simulate multiple video users sharing the cognitive radio network and show that our proposed solution significantly reduces the packet loss rate and outperforms the conventional single-channel dynamic resource allocation by almost 2 dB in terms of video quality.      

Impact of Network Dynamics on User's Video Quality: Analytical Framework and QoS Provision

We develop an analytical framework to investigate the impacts of network dynamics on the user perceived video quality. Our investigation stands from the end user's perspective by analyzing the receiver playout buffer. In specific, we model the playback buffer at the receiver by a G/G/1/? and G/G/1/N queue, respectively, with arbitrary patterns of packet arrival and playback. We then examine the transient queue length of the buffer using the diffusion approximation. We obtain the closed-form expressions of the video quality in terms of the start-up delay, fluency of video playback and packet loss, and represent them by the network statistics, i.e., the average network throughput and delay jitter. Based on the analytical framework, we propose adaptive playout buffer management schemes to optimally manage the threshold of video playback towards the maximal user utility, according to different quality-of-service requirements of end users. The proposed framework is validated by extensive simulations.