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.     

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.     

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.     

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.     

Analysis of playout strategies for voice transmission using packet switching techniques

Due to variations in network delay, a stream of voice packets with deterministic interarrival times to a data network may not have deterministic interdeparture times at the destination. Two playout schemes which are designed to remove such variations in delay are considered. Analytic results for the performance of these two schemes are obtained. Numerical examples showing the effect of coefficient of variation of interdeparture time on performance are presented.     

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。     

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.     

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。     

Performance considerations for integrated voice and data networks

Performance considerations, particularly network delays, for integrated voice and data networks are reviewed. The nature of the delay problem is discussed, followed by a review of concepts, objectives and advances in enhanced circuit, packet and hybrid switching techniques, including fast circuit switching (FCS), virtual circuit switching (VCS), buffered speech interpolation (SI), packetized virtual circuit (PVC), cut-through switching (CTS), composite packets and various frame-management strategies for hybrid switching.In particular, the concept of introducing delay to resolve contention in SI is emphasized and, when applied to both voice talkspurts and data messages, this forms a basis for a relatively new approach to network design called transparent message switching (TMS). This approach and its potential performance advantages are reviewed in terms of various architectural aspects of integrated services networks, such as packet structure, multiplexing scheme, server structure and queuing performance, network topology and network protocols.A number of traffic-management strategies and their grade-of-service implications for voice service are discussed. These strategies include voice call and data session blocking, voice talkspurt and data message buffering, speech loss and data integrity and speech processing techniques, including variable quality, rate, speed and entropy coding. Emphasis is placed on the impact of variable delays on voice traffic, especially the importance of generating and preserving appropriate length speech talkspurts in order to mitigate the effects of variable network delay.     

Performance analysis of an integrated voice/data cut-through switching network

In this paper, the performance of an integrated voice/data cut-through switching network is studied. We first derive cut-through probabilities of voice and data packets at intermediate nodes. Then, the Laplace transform for the network delay is obtained. According to our numerical results, the cut-through switching method is superior in its delay characteristics to the conventional packet switching3for voice and data in integrated voice/data networks.     

Performance of a distributed control LAN access protocol supporting voice and data traffic

This paper describes a local area network access protocol for integrated voice and data communications in a token passing ring. The transmission scheme is decentralized and suitable for both voice and data services. In this system access delay for voice packets can be bounded to a desired upper limit, even at heavy network load. The quality of speech remains unaffected at any level of offered data load and all voice packets are served within the specified delay limit. The protocol provides equal priority to pure data at normal network load while voice is given priority over pure data at heavy network load. The delay-throughput characteristic of both voice and data traffic has been evaluated at a transmission speed of 10 Mb s^-1 and it is shown that the mean delay for voice packets, at heavy data load, can be limited to a value equal to half the threshold token rotation time.     

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.     

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.     

Backbone networks using rotation counters

We use simulations to evaluate the performance of an RCP ring network for network interconnection. The RCP (Rotation Counter Protocol) is a token ring protocol that has in its token a special field for counting down of the number of packets transmitted in one cycle. By correlating the queuing performance of each station through the counter, the delay of the high-priority data can be made almost independent and the delay jitters made small, Thus, RCP can be utilized in real-time control environment or integrated services environment. The performance of the backbone networks is investigated, using voice and data integration; the performance measures are mean voice delay, delay jitters, and mean data delay. Various buffering schemes for voice packets are compared. The effect of asymmetric data load is examined, and RCP is found to be capable of minimizing the loading effect of one station to the other     

Subjective Quality of Speech Over Packet Networks as a Function of Packet Loss, Delay and Delay Variation

An expanding proportion of voice traffic is being carried by packet networks. Speech quality can be impaired in qualitatively new ways in packet networks when packets are lost or the spacing between them is distorted. Three parameters that characterize the performance of packet networks were examined for their relative impact on speech quality as judged by human observers: network delay or latency, packet loss, and packet delay variation or jitter. We manipulated these variables via a network emulator made available by NIST. This report summarizes five laboratory experiments that examined the variables in a variety of experimental procedures for presenting and judging speech. The experiments agreed in showing that the relative importance of the variables for affecting speech quality was, in decreasing order: packet loss, jitter, delay. The effect on speech quality of 200 ms of network delay was shown to be equivalent to the effect of one percentage point of packet loss. Many consumers also traded off some speech quality for a free, added feature, unified messaging.     

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.     

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%     

VoWLAN中QoS延迟性能分析与改进

VoWLAN（VoIP over WLAN）技术是一种利用无线局域网（WLAN）来传输VoIP帧的技术。VoWLAN中的语音业务是对延时较为敏感的实时业务,因此对其作延迟分析尤为重要。重点介绍了VoWLAN系统中AP节点处延迟产生过程,并采用一种新的具有优先级的M/G/1排队分析模型对AP节点处的延迟进行详细分析,该模型比以往所采用的单一的基于泊松过程的分析模型更精确;探讨了减少延迟的方法,提出新的点协调功能（PCF）方式下站点轮询算法,该算法可以更好的支持VoWLAN系统的VoIP等实时业务;最后用OPNET仿真工具进行了验证。     

Novel approaches for online playout delay prediction in VoIP applications using time series models

Voice over IP (VoIP) applications requires a buffer at the receiver to minimize the packet loss due to late arrival. Several algorithms are available in the literature to estimate the playout buffer delay. Classic estimation algorithms are non-adaptive, i.e. they differ from more recent approaches basically due to the absence of learning mechanisms. This paper introduces two new formulations of adaptive algorithms for online learning and prediction of the playout buffer delay, the first one being based on the standard Box-Jenkins autoregressive model, while the second one being based on the feedforward and recurrent neural networks. The obtained results indicate that the proposed algorithms present better overall performance than the classic ones.     

Performance evaluation of two multiple-access protocols

A real-time, microprocessor-based simulator was designed to study the packet transmission of voice on a broadcast type local area network, based on the CSMA/CD and Hymap multiple-access protocols. The speech quality is evaluated subjectively. A packetization-frozen protocol is used to eliminate the successive collisions due to possible synchronization of packet generation among stations. The variance of the network delay is bounded by discarding packets which have not been transmitted within a certain amount of time. Smooth speech output can be obtained by introducing additional buffer delay at the receiver.