Intelligent video smoother for multimedia communications

Multimedia communications often require intramedia synchronization for video data to prevent potential playout discontinuity resulting from network delay variation (jitter) while still achieving satisfactory playout throughput. In this paper, we propose a neural network (NN) based intravideo synchronization mechanism, called the intelligent video smoother (IVS), operating at the application layer of the receiving end system. The IVS is composed of an NN traffic predictor, an NN window determinator, and a window-based playout smoothing algorithm. The NN traffic predictor employs an on-line-trained back-propagation neural network (BPNN) to periodically predict the characteristics of traffic modeled by a generic interrupted Bernoulli process (IBP) over a future fixed time period. With the predicted traffic characteristics, the NN window determinator determines the corresponding optimal window by means of an off-line-trained BPNN in an effort to achieve a maximum of the playout quality (Q) value. The window-based playout smoothing algorithm then dynamically adopts various playout rates according to the window and the number of packets in the buffer. Finally, we show that via simulation results and live video scenes, compared to two other playout approaches, IVS achieves high-throughput and low-discontinuity playout under a mixture of IBP arrivals     

An efficient ATM voice service with flexible jitter and delayguarantees

A transmission and multiplexing strategy appropriate for voice over asynchronous transfer mode (ATM), called delayed frame queueing (DFQ), is proposed. This frame-based strategy has features in common with the synchronous transfer mode and is thus well suited to service synchronous applications such as voice, while retaining the statistical multiplexing capabilities of ATM. In particular, the DFQ service discipline can provide explicit and nontrivial bounds for queue delay and jitter, for both bursty as well as continuous traffic streams. Furthermore, the DFQ discipline can combine a wide range of delay and jitter bounds while also managing the distribution of quality of service violations among the traffic streams when congestion occurs. Jitter control is performed at the network periphery and thus does not negatively influence multiplexing gain at intermediate nodes. This efficient strategy has major implications in terms of the preferred alternatives chosen by clients when implementing source clock recovery for voice. DFQ allows the entire range of implementation alternatives for voice over ATM to be appropriately serviced, such as ATM adaptation layer types 1 and 2 (AAL1/2), adaptive playout, and immediate playout     

多媒体通信中智能化媒体内同步机制

本文提出了一种智能化视频流量的预测和同步机制（ＩＦＳＭ）,它由ＢＰ神经网络流量预测器（ＢＰＮＮ）、输出缓冲区和基于模糊神经网络（ＦＮＮ）的输出速率决策器所组成。ＢＰＮＮ采用一种在线训练的ＢＰ神经网络预测在将来的一定时间间隔（ＦＩ）内的平均分组速率,ＦＮＮ决策器根据预测的流量特性和缓冲区中的分组数动态地调节下一个分组输出的时间。仿真结果表明：与窗口机制相比,ＩＦＳＭ能够使视频流量取得较高的连续性和较低的时延,并且由于ＦＮＮ的学习能力,ＩＦＳＭ可以自适应地调节相应参数以满足不同的服务质量的要求。     

突发大时延下的IP语音缓冲算法

当网络中存在突发大时延时,就会出现极大丢包率或极大端到端时延的情况,从而难以获得好的语音质量。对于这个问题,提出针对突发大时延存在下的自适应语音缓冲算法。算法通过控制语音包在语音缓冲队列中的位置来控制语音包的播放时间,从而可以尽量减小语音裂缝(Gap)的出现。算法将突发大时延存在下的最大丢包率可以扩大到20％,而一般的预测算法只能容忍5-10％的最大丢包率。通过基于听觉模型的客观音质评价(PESQ)仿真计算,以及实际应用表明本文的算法对有突发大时延存在的网络中的语音通信质量有一定的改善作用。     

Design and performance evaluation of an RRA scheme for voice‐data channel access in outdoor microcellular environments

In PCS networks, the multiple access problem is characterized by spatially dispersed mobile source terminals sharing a radio channel connected to a fixed base station. In this paper, we design and evaluate a reservation random access (RRA) scheme that multiplexes voice traffic at the talkspurt level to efficiently integrate voice and data traffic in outdoor microcellular environments. The scheme involves partitioning the time frame into two request intervals (voice and data) and an information interval. Thus, any potential performance degradation caused by voice and data terminals competing for channel access is eliminated. We consider three random access algorithms for the transmission of voice request packets and one for the transmission of data request packets. We formulate an approximate Markov model and present analytical results for the steady state voice packet dropping probability, mean voice access delay and voice throughput. Simulations are used to investigate the steady state voice packet dropping distribution per talkspurt, and to illustrate preliminary voice-data integration considerations. This revised version was published online in June 2006 with corrections to the Cover Date.     

突发业务环境下CBR业务性能的实验分析

CBR业务是一类极为重要的实时业务,能否有效地传递这类业务关系到从现有网络向ATM的过渡,因此CBR业务的服务质量是一个值得研究的重要课题。本文利用计算机仿真的方法,全面地分析了突发业务环境下,影响CBR业务服务质量的各种因素,指出复接器占有率、缓冲区容量、背景强度、背景流的自相关特征对CBR业务的时延及时延抖动有很大的影响,尤其是背景流具有长时相关性时,CBR业务的服务质量将严重恶化,必须设法加以控制。     

A New Buffer Algorithm for Speech Quality Improvement in VoIP Systems

Jitter buffer plays an important role in Voice over IP (VoIP) applications because it provides a key mechanism for achieving good speech quality to meet technical and commercial requirements. The main objective of this paper is to propose a new, simple-to-use jitter buffer algorithm as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance, in terms of enhanced user-perceived speech quality and reduced end-to-end delay. Supported by signal processing features, the new algorithm, the so-called Play Late Algorithm, alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. The results show that the new algorithm achieves the best performance under different network conditions when compared to conventional static and adaptive jitter buffer algorithms. The results reported here are based on live tests and emulated network conditions on real mobile phone prototypes. The mobile phone prototypes use AMR codec and support full IP/UDP/RTP stack with IPSec function in some of the tests. The method for perceived speech quality measurement is based on the ITU-T standard for speech quality evaluation (PESQ).

Packetized voice/data integrated transmission on a token passingring local area network

The performance of a token-passing ring network with packetized voice/data mixed traffic is investigated through extensive simulations. Both data and voice users are modeled in the simulations. Data users produce bursty traffic. Voice traffic is modeled as having alternating talkspurts and silences, with generation of voice packets at a constant rate during talkspurts and no packet generation during silence periods. Token passing ring local area networks are shown to effectively handle both voice and data traffic. The effects of system parameters (e.g. voice packet length, talkspurt/silence lengths, data traffic intensity, and limited exhaustive service disciplines) on network performance are discussed     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

Design issues relevant to developing an integrated voice/datamobile radio system

The various design issues related to developing an integrated voice/data mobile radio system, including high speed digital radio frequency modulation in a mobile environment, statistics for the talkspurt/silence gap composition of speech, switching schemes for voice/data integration, encoding techniques, and voice and data traffic statistics are discussed. A performance analysis is conducted for a typical design, showing that a voice-only mobile radio system can be upgraded to an integrated voice/data system capable of carrying the full voice and data loads without requiring additional radio channels and without compromising voice performance. Data traffic is only minimally delayed (46.2 ms mean delay) for a fully loaded system     

Improving voice over IP subjective call quality

Adaptive playout algorithms rely on estimates of network delays to calculate playout times of voice packets. Typically, network estimators are either able to react quickly to delay variations or to ignore transient noise conditions, but cannot do both. In our solution, the weighting factor that controls the estimation process is dynamically adjusted according to the observed delay variations. This results in higher quality estimates of network delays. Experimental results show that our algorithm can achieve higher subjective call quality than the basic adaptive algorithm, as measured by the ITU-T E-Model methodology.     

Performance analysis of a packet-switched synchronous voice/datatransmission system

The queuing behavior of voice and data traffic for a proposed voice/data multiplexing system is investigated. For synchronization of packetized voice in the packet-switched network, the master frame format of time-division multiplexing (TDM) is utilized. This scheme has discrete delay characteristics for packetized voice and provides a simple play-out method for voice signals. The performance of voice and data traffic is investigated by obtaining the cumulative distribution of delay of voice packets and the mean delay time of voice and data packets. The performances of the proposed system and the circuit-switched system are compared to time-assignment speech interpolation (TASI) in terms of the loss probability of voice traffic and the maximum allowable number of input voice channels for a given trunk capacity. The proposed system has been found to be more efficient than the circuit-switched system with TASI for integrating multiple voice channels in a single link. For a given trunk capacity, the system can accommodate about twice the number of input voice channels as the circuit-switched system     

Per-stream jitter analysis in CBR ATM multiplexors

Constant bit-rate (CBR) traffic is expected to be a major source of traffic in high-speed networks. Such sources may have stringent delay and loss requirements and, in many cases, they should be delivered exactly as they were generated. A simple delay priority scheme will bound the cell delay and jitter for CBR streams so that in the network switches CBR traffic will only compete with other CBR traffic in the networks, In this paper we consider a multiplexor in such an environment. We provide an exact analysis of the jitter process in the homogeneous case. In this case we obtain the complete characterization of the jitter process showing the inaccuracies of the existing results. Our results indicate that jitter variance is bounded and never exceeds the constant 2/3 slot. It is also shown that the per-stream successive cell interdeparture times are negatively correlated with the lag 1 correlation of -1/2. Higher order correlation coefficients are shown to be zero. Simple asymptotic results on per-stream behavior are also provided when the number of CBR streams is considered large. In the heterogeneous case we bound the jitter distribution and moments. Simple results are provided for the computation of the bound on the jitter variance for any mix of CBR streams in this case. It is shown that streams with a low rate (large period) do experience little jitter variance. However, the jitter variance for the high-rate streams could be quite substantial     

保障流媒体时间结构的网络时延分析

流媒体同步对端到端时延和时延抖动提出了确定的要求,而终端抖动缓存一方面能消除时延抖动的影响,一方面却增加了端到端时延,流媒体同步保障对网络时延的要求不明确。论文从概率保障流媒体同步的角度,确定了保障流媒体同步的抖动缓存容量范围,提出了流媒体同步网络保障的充分条件,针对基于Internet VoIP(Voice over IP)业务的实际网络测试结果,给出了应用流媒体同步网络保障充分条件进行同步保障评价的应用实例并验证了其正确性。     

A distributed MAC scheme supporting voice services in mobile ad hoc networks

Future mobile ad hoc networks are expected to support voice traffic. The requirement for small delay and jitter of voice traffic poses a significant challenge for medium access control (MAC) in such networks. User mobility presents unique difficulties in this context due to the associated dynamic path attenuation. In this paper, a MAC scheme for mobile ad hoc networks supporting voice traffic is proposed. With the aid of a low‐power probe prior to DATA transmissions, resource reservation is achieved in a distributed manner, thus leading to small packet transmission delay and jitter. The proposed scheme can automatically adapt to dynamic path attenuation in a mobile environment. Statistical multiplexing of on/off voice traffic can also be achieved by partial resource reservation for off voice flows. Simulation results demonstrate the effectiveness of the proposed scheme. Copyright © 2009 John Wiley & Sons, Ltd.     

Optimal smoothing for guaranteed service

We consider the transmission of variable bit rate (VBR) video over a network offering a guaranteed service such as ATM VBR or the guaranteed service of the IETF. The guaranteed service requires that the flow accepted by the network has to be conforming with a traffic envelope σ; in return, it receives a service guarantee expressed by a network service curve β. Functions α and β are derived from the parameters used for setting up the reservation, for example, from the T-SPEC and R-SPEC fields used with the resource reservation protocol (RSVP). In order to satisfy the traffic envelope constraint, the output of the encoder is fed to a smoother, possibly with some look-ahead. The resulting stream is transported by the network; at the destination, the decoder waits for an initial playback delay and reads the stream from the receive buffer. We consider the problem of whether there exists one optimal strategy at the smoother which minimizes the playback delay and the receive buffer size, given the traffic envelope α and the service curve β. We show that there does exist such an optimal smoothing, and give an explicit representation for it. We also obtain a simple expression for the smallest playback delay and playback buffer size which can be achieved over all possible smoothing and playback strategies. We show that the computation of optimal smoothing and minimum playback delay do not depend on the past. We show that separate delay equalization is optimal in the constant bit rate (CBR) case, but not otherwise. We also apply the theory to the analysis of which T-SPEC should be requested by a source-destination pair, given some playback delay and buffer constraint, and given the path characteristics advertised in RSVP PATH messages     

Performance evaluation of a packet voice system

A study is made of statistical multiplexing of voice packets from a number of packetized voice sources onto a single channel. Each source alternates between talkspurt (active period) and silence, and packets are generated during active periods only. The packets are buffered (in a finite size buffer) when transmission capacity is not available. An embedded Markov chain model is adopted to analyze the system and a numerical technique is presented to compute system performance. Simulation results validate the analysis     

Expedited forwarding end‐to‐end delay and jitter in DiffServ

The scheduling disciplines and active buffer management represent the main components employed in the differentiated services (DiffServ) data plane, which provide qualitative per‐hop behaviors corresponding to the QoS required by supported traffic classes. In the first part of this paper, we compute the per‐hop delay bound that should be guaranteed by the different multiservice scheduling disciplines, so that the end‐to‐end (e2e) delay required by expedited forwarding (EF) traffic can be guaranteed. Consequently, we derive the e2e delay bound of EF traffic served by priority queuing–weighted fair queuing (PQ–WFQ) at every hop along its routing path. Although real‐time flows are principally offered EF service class, some simulations on DiffServ‐enabled network show that these flows suffer from delay jitter and they are negatively impacted by lower priority traffic. In the second part of this paper, we clarify the passive impact of delay jitter on EF traffic, where EF flows are represented by renewal periodic ON–OFF flows, and the background (BG) flows are characterized by the Poisson process. We analyze through different scenarios the jitter effects of these BG flows on EF flow patterns when they are served by a single class scheduling discipline, such as first‐input first‐output, and a multiclass or multiservice scheduling discipline, such as static priority service discipline. As a result, we have found out that the EF per‐hop behaviors (PHBs) configuration according to RFCs 2598 and 3246 (IETF RFC 2598, June 1999; RFC 3246, IETF, March 2002) cannot stand alone in guaranteeing the delay jitter required by EF flows. Therefore, playout buffers must be added to DiffServ‐enabled networks for handling delay jitter problem that suffers from EF flows. Copyright © 2008 John Wiley & Sons, Ltd.     

Delay Related Issues in Integrated Voice and Data Networks

This paper focuses on network delays as they apply to voice traffic. First the nature of the delay problem is discussed and this is followed by a review of enhanced circuit, packet, and hybrid switching techniques: these include 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, 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 packet structure, multiplexing scheme, network topology, and network protocols. The paper then deals more specifically with the impact of variable delays on voice traffic. In this regard the importance of generating and preserving appropriate length speech talkspurts in order to mitigate the effects of variable network delay is emphasized. The results indicate that a desirable length of talkspurt "hangover" of about 200 ms will accomplish this without unduly affecting speech activity, and that, under these circumstances, the perceptable threshold of variable talkspurt delay can be as high as about 200 ms average. As such, the results provide a useful guideline for integrated services system designers. Finally, suggestions are made for further studies on performance analysis and subjective evaluation of advanced integrated services systems.     

An adaptive reservation scheme for VBR voice traffic in wireless ATM networks

A reservation scheme, named dynamic hybrid partitioning, is proposed for the Medium Access Control (MAC) protocol of wireless ATM (WATM) networks operating in Time Division Duplex (TDD) mode. The goal is to improve the performance of the real-time Variable Bit Rate (VBR) voice traffic in networks with mixed voice/data traffic. In most proposed MAC protocols for WATM networks, the reservation phase treats all traffic equally, whether delay-sensitive or not. Hence, delay-sensitive VBR traffic sources have to compete for reservation each time they wake up from idle mode. This causes large and variable channel access delays, and increases the delay and delay variation (jitter) experienced by ATM cells of VBR traffic. In the proposed scheme, the reservation phase of the MAC protocol is dynamically divided into a contention-free partition for delay-sensitive idle VBR traffic, and a contention partition for other traffic. Adaptive algorithms dynamically adjust the partition sizes to minimize the channel bandwidth overhead. Simulation results show that the delay performance of delay-sensitive VBR traffic is improved while minimizing the overhead.