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Issue 6 • Date Aug. 1997

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  • Guest Editorial Real-time Video Services In Multimedia Networks

    Publication Year: 1997 , Page(s): 961 - 964
    Cited by:  Papers (2)  |  Patents (1)
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    Freely Available from IEEE
  • The effect of multiple time scales and subexponentiality in MPEG video streams on queueing behavior

    Publication Year: 1997 , Page(s): 1052 - 1071
    Cited by:  Papers (40)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (524 KB)  

    Guided by the empirical observation that real-time MPEG video streams exhibit both multiple time scale and subexponential characteristics, we construct a video model that captures both of these characteristics and is amenable to queueing analysis. We investigate two fundamental approaches for extracting the model parameters: using sample path and second-order statistics-based methods. The model exhibits the following two canonical queueing behaviors. When strict stability conditions are satisfied, i.e., the conditional mean of each scene is smaller than the capacity of the server, precise modeling of the interscene dynamics (long-term dependency) is not essential for the accurate prediction of small to moderately large queue sizes. In this case, the queue length distribution is determined using quasistationary (perturbation theory) analysis. When weak stability conditions are satisfied, i.e., the conditional mean of at least one scene type is greater than the capacity of the server, the dominant effect for building a large queue size is the subexponential (long-tailed) scene length distribution. In this case, precise modeling of intrascene statistics is of secondary importance for predicting the large queueing behavior. A fluid model, whose arrival process is obtained from the video data by replacing scene statistics with their means, is shown to asymptotically converge to the exact queue distribution. Using the transition scenario of moving from one stability region to the other by a change in the value of the server capacity, we synthesize recent queueing theoretic advances and ad hoc results in video modeling, and unify a broad range of seemingly contradictory experimental observations found in the literature. As a word of caution for the widespread usage of second-order statistics modeling methods, we construct two processes with the same second-order statistics that produce distinctly different queueing behaviors View full abstract»

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  • Performance evaluation of ATM networks carrying constant and variable bit-rate video traffic

    Publication Year: 1997 , Page(s): 1115 - 1131
    Cited by:  Papers (21)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (544 KB)  

    In this paper, we present the performance of asynchronous transfer mode (ATM) networks supporting audio/video traffic. The performance evaluation is done by means of a computer simulation model driven by real video traffic generated by encoding video sequences. We examine the glitching that occurs when the video information is not delivered on time at the receiver; we characterize various glitching quantities, such as the glitch duration, total number of macroblocks unavailable per glitch, and the maximum unavailable area per glitch. For various types of video contents, we compare the maximum number of constant bit-rate (CBR) and constant-quality variable bit-rate (CQ-VBR) video streams that can be supported by the network while meeting the same end-to-end delay constraint, the same level of encoded video quality, and the same glitch rate constraint. We show that when the video content is highly variable, many more CQ-VBR streams than CBR streams can be supported under given quality and delay constraints, while for relatively uniform video contents (as in a videoconferencing session), the number of CBR and CQ-VBR streams supportable is about the same. We also compare the results with those obtained for a 100Base-T Ethernet segment. We then consider heterogeneous traffic scenarios, and show that when video streams with different content, encoding scheme, and encoder control schemes are mixed, the results are at intermediate points compared to the homogeneous cases, and the maximum number of supportable streams of a given type can be determined in the presence of other types of video traffic by considering an “effective bandwidth” for each of the stream types. We consider multihop ATM network scenarios as well, and show that the number of video streams that can be supported on a given network node is very weakly dependent on the number of hops that the video streams traverse. Finally, we also consider scenarios with mixtures of video streams and bursty traffic, and determine the effect of bursty traffic load and burst size on the video performance View full abstract»

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  • Videoconferencing over packet-based networks

    Publication Year: 1997 , Page(s): 1101 - 1114
    Cited by:  Papers (10)  |  Patents (40)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (172 KB)  

    The Internet explosion is driving the need for new collaboration tools which will enable two or more users to share data, audio, and video. The real-time packet-based solutions which are emerging differ considerably from the circuit-switch solutions which have existed for some time now. In this paper, we present one such packet-based approach, the Multimedia Multiparty Teleconferencing (MMT) system, which was fully implemented as a research prototype. Using MMT as an example, we address some of the fundamental issues related to videoconferencing systems in a packet-based environment, and discuss the differences with the traditional circuit-switch approaches, namely, the ITU H.320 standard. In particular, MMT is a distributed solution, while H.320 is centralized. The use of multicast and a novel video-mixing technique to facilitate the distributed solution are presented. Furthermore, MMT audio and video streams are susceptible to congestion and packet loss in the shared media packet-based environment, while H.320 streams use dedicated connections. As such, synchronization, error resilience, and dynamic rate control schemes for the packet-based system are presented View full abstract»

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  • Smoothing, statistical multiplexing, and call admission control for stored video

    Publication Year: 1997 , Page(s): 1148 - 1166
    Cited by:  Papers (55)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (424 KB)  

    Variable bit-rate (VBR) compressed video is known to exhibit significant, multiple-time-scale rate variability. A number of researchers have considered transmitting stored video from server to a client using smoothing algorithms to reduce this rate variability. These algorithms exploit client buffering capabilities and determine a “smooth” rate transmission schedule, while ensuring that a client buffer neither overflows nor underflows. We investigate how video smoothing impacts the statistical multiplexing gains available with such traffic, and we show that a significant amount of statistical multiplexing gains can still be achieved. We then examine the implication of these results on network resource management and call admission control when transmitting smoothed stored video using VBR service with statistical quality-of-service (QoS) guarantees. Specifically, we present a uniform call admission control scheme based on a Chernoff bound method that uses a simple, novel traffic model requiring only a few parameters. This scheme provides an easy and flexible mechanism for supporting multiple VBR service classes with different QoS requirements. We evaluate the efficacy of the call admission control scheme over a set of MPEG-1 coded video tracts View full abstract»

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  • Call admission for prerecorded sources with packet loss

    Publication Year: 1997 , Page(s): 1167 - 1180
    Cited by:  Papers (24)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (408 KB)  

    We develop call admission policies for statistically multiplexing prerecorded sources over a bufferless transmission link. Our model is appropriate for video on demand, as well as other on-demand multimedia applications. In particular, we allow users to specify when the sources begin transmission; we also allow the user to invoke VCR actions such as pause and temporal jumps. We suppose that the quality of service (QoS) requirement allows for a small amount of packet loss. We develop a stochastic model which captures the random phases of the sources. We then apply large deviation theory to our model to develop global admission rules. The accuracy of the large deviation approximation is verified with simulation experiments employing importance sampling techniques. We also propose a refined admission rule which combines the global test and a myopic test. Numerical results are presented for the Star Wars trace; we find that the statistical multiplexing gain is potentially high and often insensitive to the QoS parameter. Finally, we develop efficient schemes for the real-time implementation of our global test. In particular, we demonstrate that the Taylor series expansion of the logarithmic moment generating function of the frame size distribution allows for fast and accurate admission decisions View full abstract»

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  • Optimal bit allocation for coding of video signals over ATM networks

    Publication Year: 1997 , Page(s): 1002 - 1015
    Cited by:  Papers (40)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (448 KB)  

    We consider optimal encoding of video sequences for ATM networks. Two cases are investigated. In one, the video units are coded independently (e.g., motion JPEG), while in the other, the coding quality of a later picture may depend on that of an earlier picture (e.g., H.26x and MPEGx). The aggregate distortion-rate relationship for the latter case exhibits a tree structure, and its solution commands a higher degree of complexity than the former. For independent coding, we develop an algorithm which employs multiple Lagrange multipliers to find the constrained bit allocation. This algorithm is optimal up to a convex-hull approximation of the distortion-rate relations in the case of CBR (constant bit-rate) transmission. It is suboptimal in the case of VBR (variable bit-rate) transmission by the use of a suboptimal transmission rate control mechanism for simplicity. For dependent coding, the Lagrange-multiplier approach becomes rather unwieldy, and a constrained tree search method is used. The solution is optimal for both CBR and VBR transmission if the full constrained tree is searched. Simulation results are presented which confirm the superiority in coding quality of the encoding algorithms. We also compare the coded video quality and other characteristics of VBR and CBR transmission View full abstract»

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  • Low-complexity video coding for receiver-driven layered multicast

    Publication Year: 1997 , Page(s): 983 - 1001
    Cited by:  Papers (324)  |  Patents (26)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (276 KB)  

    The “Internet Multicast Backbone,” or MBone, has risen from a small, research curiosity to a large-scale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. We describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an efficient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Internet). Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a “real” application-the UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone View full abstract»

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  • Rate control for VBR video coders in broad-band networks

    Publication Year: 1997 , Page(s): 1040 - 1051
    Cited by:  Papers (38)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (284 KB)  

    We present a rate control algorithm adapted to MPEG video coders ensuring that output conforms to the parameters of a leaky-bucket network access controller. The algorithm avoids unpredictable rate variations without the rigidity and systematic coding delay of constant bit-rate (CBR) coders, and makes possible resource provision for guaranteed quality of service. A relatively large burst tolerance parameter allows considerable scope for variation at GoP scale, and only restricts the natural rate when necessary to avoid long-term overloads. Possible multiplexing schemes are discussed distinguishing buffer provision for cell-scale and burst-scale congestion View full abstract»

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  • Optimal nonlinear adaptive prediction and modeling of MPEG video in ATM networks using pipelined recurrent neural networks

    Publication Year: 1997 , Page(s): 1087 - 1100
    Cited by:  Papers (41)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (456 KB)  

    This paper investigates the application of a pipelined recurrent neural network (PRNN) to the adaptive traffic prediction of MPEG video signal via dynamic ATM networks. The traffic signal of each picture type (I, P, and B) of MPEG video is characterized by a general nonlinear autoregressive moving average (NARMA) process. Moreover, a minimum mean-squared error predictor based on the NARMA model is developed to provide the best prediction for the video traffic signal. However, the explicit functional expression of the best mean-squared error predictor is actually unknown. To tackle this difficulty, a PRNN that consists of a number of simpler small-scale recurrent neural network (RNN) modules with less computational complexity is conducted to introduce the best nonlinear approximation capability into the minimum mean-squared error predictor model in order to accurately predict the future behavior of MPEG video traffic in a relatively short time period based on adaptive learning for each module from previous measurement data, in order to provide faster and more accurate control action to avoid the effects of excessive load situation. Since those modules of PRNN can be performed simultaneously in a pipelined parallelism fashion, this would lead to a significant improvement in the total computational efficiency of PRNN. In order to further improve the convergence performance of the adaptive algorithm for PRNN, a learning-rate annealing schedule is proposed to accelerate the adaptive learning process. Another advantage of the PRNN-based predictor is its generalization from learning that is useful for learning a dynamic environment for MPEG video traffic prediction in ATM networks where observations may be incomplete, delayed, or partially available. The PRNN-based predictor presented in this paper is shown to be promising and practically feasible in obtaining the best adaptive prediction of real-time MPEG video traffic View full abstract»

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  • Probabilistic burstiness-curve-based connection control for real-time multimedia services in ATM networks

    Publication Year: 1997 , Page(s): 1072 - 1086
    Cited by:  Papers (8)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (600 KB)  

    In this paper we present a method to establish real-time connections with guaranteed quality of service (QOS), based on a per-session probabilistic burstiness curve (PBC). Under two distinctive service disciplines, role proportional processor sharing and fixed rate processor sharing, we derive useful probabilistic bounds on per-session end-to-end loss which is caused by either buffer overflow in the path or excessive delay to the destination. One remarkable feature of the bounding solutions is that they are solely determined by the PBC of each session itself, independent of the network environment and other connections. To improve network resource utilization, our method is extended to allow statistical sharing of buffer resources. The admission control scheme presented in this paper has a great flexibility in connection management since bandwidth and buffer allocations can be adaptively adjusted among incoming and existing sessions according to present network resource availability. We also present a novel method to compute the PBC of multimedia traffic based on the measurement of two important statistics (rate histogram and power spectrum). Our study of MPEG/JPEG video sequences reveals the fundamental interrelationship among the PBC, the traffic statistics, and the QOS guarantee, and also provides many engineering aspects of the PBC approach to real-time multimedia services in ATM networks View full abstract»

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  • Joint selection of source and channel rate for VBR video transmission under ATM policing constraints

    Publication Year: 1997 , Page(s): 1016 - 1028
    Cited by:  Papers (76)  |  Patents (16)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (408 KB)  

    Variable bit-rate (VBR) transmission of video over ATM networks has long been said to provide substantial benefits, both in terms of network utilization and video quality, when compared with conventional constant bit-rate (CBR) approaches. However, realistic VBR transmission environments will certainly impose constraints on the rate that each source can submit to the network. We formalize the problem of optimizing the quality of the transmitted video by jointly selecting the source rate (number of bits used for a given frame) and the channel rate (number of bits transmitted during a given frame interval). This selection is subject to two sets of constraints, namely, (1) the end-to-end delay has to be constant to allow for real-time video display and (2) the transmission rate has to be consistent with the traffic parameters negotiated by user and network. For a general class of constraints, including such popular ones as the leaky bucket, we introduce an algorithm to find the optimal solution to this problem. This algorithm allows us to compare VBR and CBR under the same end-to-end delay constraints. Our results indicate that variable-rate transmission can increase the quality of the decoded sequences without increases in the end-to-end delay. Finally, we show that for the leaky-bucket channel, the channel constraints can be combined with the buffer constraints, such that the system is identical to CBR transmission with an additional, infrequently imposed constraint. Therefore, video quality with a leaky-bucket channel can achieve the same quality of a CBR channel with larger physical buffers, without adding to the physical delay in the system View full abstract»

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  • Multirate scheduling of VBR video traffic in ATM networks

    Publication Year: 1997 , Page(s): 1132 - 1147
    Cited by:  Papers (8)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (464 KB)  

    One of the major attractions of asynchronous transfer mode (ATM) networks for transporting bursty video traffic is its ability to exploit the multiplexing gains of packet switching while providing quality of service guarantees. Unfortunately, most of the multiplexing mechanisms proposed in the literature fail to exploit the multiplexing gains of ATM. We propose a multirate service mechanism that allows a session to be served at different rates at different times. Applications generating bursty data, such as variable bit-rate (VBR) video, can take advantage of multirate service by requesting a high rate of service for brief periods of bursty arrivals and a much lower rate of service for all other times. Consequently, the applications can improve their delay performance without reserving a high bandwidth for the entire duration of the sessions. Furthermore, the scheduler can multiplex the peaks and the lulls in service rates of different sessions and improve the utilization of the system. Using MPEG video traces from a number of applications, we show that multirate servers outperform single-rate PGPS (packet-by-packet generalized processor sharing) servers and CBR (constant bit-rate) servers in terms of number of connections admitted, while providing the same level of service guarantees. We also investigate the performance of multirate service when service quality need not be guaranteed. We refer to this as predictive service. We propose a measurement-based admission control procedure for predictive service, and show that it helps increase the size of the admissible region even further View full abstract»

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  • ITU-T standardization of audiovisual communication systems in ATM and LAN environments

    Publication Year: 1997 , Page(s): 965 - 982
    Cited by:  Papers (12)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (300 KB)  

    This paper presents the ITU-T Study Group 15 development of H-series Recommendations that allow interworking between different audiovisual communication terminals manufactured by different equipment providers. The paper focuses on H.310 and H.321 systems for broad-band ATM environments and H.322 and H.323 systems for LAN environments where the quality of service may or may not be guaranteed. The paper first lists the Recommendations developed by the ITU-T for audiovisual communication systems and the network environments in which they may be used. It then describes the design philosophy, the network specific characteristics, and hardware trials for each system. Then it describes the communication control protocol defined in H.245 which is used commonly by different audiovisual communication systems. Finally, the paper discusses interworking scenarios for communication between the different types of terminal on different networks View full abstract»

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  • Quality control for VBR video over ATM networks

    Publication Year: 1997 , Page(s): 1029 - 1039
    Cited by:  Papers (25)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (260 KB)  

    Uncontrolled variable-bit-rate (VBR) coded video yields consistent picture quality, but the traffic stream is very bursty. When sent over ATM networks, cell losses may be incurred due to limited buffer capacity at the switches; this could cause severe picture quality degradation. Source rate control can be implemented to generate a controlled VBR bit stream which conforms to specified bit rate bounds and buffer constraints. However, source rate control could result in picture quality degradation too. Hence, for real-time video services, an important issue to address is whether the picture quality degradation incurred by source rate control is within acceptable levels or how to choose the appropriate coding parameters to make it so. We establish quantitatively the relationship between picture quality and source rate control for the case of guaranteed service with different combinations of allocated bandwidth, buffer size, and other key video-coding parameters of MPEG-2. In addition, quality control in the context of two-layered scalable video service (basic and enhanced quality) is also considered. Our study reveals that, in order to maximize both the basic and the enhanced quality, source rate control should be implemented on both layers. The relationships between the two types of quality and different combinations of allocated bandwidths, buffer sizes, and some key coding parameters are also established quantitatively for MPEG-2 SNR scalability View full abstract»

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  • Lossless aggregation: a scheme for transmitting multiple stored VBR video streams over a shared communications channel without loss of image quality

    Publication Year: 1997 , Page(s): 1181 - 1189
    Cited by:  Papers (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (256 KB)  

    This paper introduces a new concept called lossless aggregation for the transmission of video information. It is a scheme for the delivery of variable bit-rate (VBR) video streams from a video server to a group of users over a shared channel. No data are dropped at the source during the adaptation process that reshapes the VBR video traffic to conform to the channel bit-rate characteristics. The transmission schedules of individual video streams evolve in a dynamic way that depends on their relative traffic characteristics. Receiver buffer underflow and overflow are prevented. Therefore, the data delivery process does not cause any loss of image quality. We show that very significant receiver-buffer reduction can be achieved with aggregation compared with the independent transmission of individual video streams over separate channels. Several bandwidth allocation methods for aggregation are studied extensively. The frame equalization algorithm stands out in terms of its simplicity and optimality View full abstract»

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Aims & Scope

IEEE Journal on Selected Areas in Communications focuses on all telecommunications, including telephone, telegraphy, facsimile, and point-to-point television, by electromagnetic propagation.

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Meet Our Editors

Editor-in-Chief
Muriel Médard
MIT