The problem of error control and concealment in video communication is becoming increasingly important because of the growing interest in video delivery over unreliable channels such as wireless networks and the Internet. This paper reviews the techniques that have been developed for error control and concealment in the past ten to fifteen years. These techniques are described in three categories according to the roles that the encoder and decoder play in the underlying approaches. Forward error concealment includes methods that add redundancy at the source end to enhance error resilience of the coded bit streams. Error concealment by postprocessing refers to operations at the decoder to recover the damaged areas based on characteristics of image and video signals. Finally, interactive error concealment covers techniques that are dependent on a dialog between the source and destination. Both current research activities and practice in international standards are covered.

A new retransmission-based error control technique is presented that does not incur any additional latency in frame playout times, and hence are suitable for interactive applications. It takes advantage of the motion prediction loop employed in most motion compensationbased codecs. By correcting errors in a reference frame caused by earlier packet loss, it prevents error propagation. The technique rearranges the temporal dependency of frames so that a displayed frame is referenced for the decoding of its succeeding dependent frames much later than its display time. Thus, the delay in repairing lost packets can be effectively masked out. The developed technique is combined with layered video coding to maintain consistently good video quality even under heavy packet loss. Through the results of extensive Internet experiments, the paper shows that layered coding can be very effective when combined with the retransmissionbased error control technique for low-bit rate transmission over best...

This paper gives an introduction to the issues involved in asynchronous video transfers. Brief overviews of video coding, rate control, multiplexing, as well as delay, error and loss control are given. Section 1. Background A collection of the world's telecommunication operators and their equipment suppliers are currently standardizing the asynchronous transfer mode (ATM) within the International Telecommunication Union (ITU) as a basis for broadband integrated services digital networks. In parallel with the ITU, the ATM Forum is furthering the development process by including computer communication manufacturers, as well as end--users in the standardization process. Although the representations give different perspectives on the proposed network solutions, the overall goal of building networks capable of transferring multimedia efficiently is shared between the ITU and the ATM Forum. In addition to the work on ATM, new developments for internet protocols lean towards supporting full--fledged multimedia communications. The simple internet protocol plus (RFC 1710) has been selected by the Internet Engineering Task Force as a basis for the next generation internet protocol (version 6, often denoted IPng; see RFC 1752). It features connection--oriented services, denoted flows, and signaling by means of the resource reservation protocol (RSVP) [103]. An implied assumption in the development of both ATM and IPng is that network users will get access to copious amounts of transmission capacity at affordable prices, even in the local loop. This capacity is needed to lower the latency of bulk data transfers and to enable audio and video communications. The provision of the latter has received considerable attention among researchers and is the topic of this paper. Asynchronous ...

Visual communications over wireless networks require the efficient and robust coding of video signals for transmission over wireless links having time-varying channel capacity. We compare several schemes for encoding video data into two priority streams, thereby enabling the transmission of video data over wireless links to be switched between two bit rates. An H.261 (p \Theta 64) algorithm is modified to implement each candidate scheme. The algorithms are evaluated for a microcellular wireless environment and a clear-channel bit rate of 65 kb/s. Our results show that by combining layering with automatic-repeat request (wireless-)link control, almost-wireline visual quality can be achieved. 1. INTRODUCTION We consider how to achieve efficient and reliable interactive video communications over microcellular wireless networks. Existing video coding schemes amenable to videophone or teleconferencing applications (such as the CCITT H.261 standard) are suitable for channels that are relat...

In this paper efficient modeling of VBR MPEG coded video sources is proposed by appropriately combining properties of frame and GOP layer signals. In particular, a Markovchain is presented for modeling the video activity, the states of which correspond to correlated AR models responsible for generating the I, P and B frames. Furthermore, an adaptive implementation of the AR coefficients is accomplished in cases that we are interested in video traffic prediction.

Transmission of MPEG2-encoded video is one of the most demanding applications in terms of network resources and QoS requirement. It needs high bandwidth with stringent transmission delays. It can not tolerate large variations on delays and it requires low error and loss data rates. Therefore, in order to design efficient integrated video communication systems over ATM networks, we propose in this paper to analyze the effects of errors and delays on both video signal and network performance. 1. INTRODUCTION MPEG2 and ATM have been adopted as the key technologies for the deployment of broadcast and interactive video services. The aim of these two international standards is to provide all the advantages of transmitting variable bit rate video over packet networks, i.e. better video quality, less delay, more connections, and lower cost. However asynchronous transfer of video requires careful integration between the network and the video end systems. Numerous considerations and tra...

Any performance evaluation of broadband networks requires modeling of the actual network traffic. Since multimedia services and especially MPEG coded video streams are expected to be a major traffic component over these networks, modeling of such services and traffic prediction are useful for the reliable operation of the broadband based an Asynchronous Transfer Mode (ATM) networks. In this paper, a Recursive implementation of a Non linear AutoRegressive model (RNAR) is presented for on line traffic prediction of Variable Bit Rate (VBR) MPEG-2 video sources. This is accomplished by using an efficient weight adaptation algorithm so that the network provide good performance even in case of highly fluctuated traffic rates. In particular, the network weights are adapted so that the output is approximately equal to the current data while preserving the former knowledge of the network. Experimental results are presented to show the good performance of the proposed scheme. Furthermore, comparison with other linear or non linear techniques is presented to show that the adopted method yields better results than the other ones.

This paper outlines a study of MPEG compressed video sequences and simulation of multiplexed video traffic in the ATM environment. A number of statistical characteristics including autocorrelation and variance of MPEG-1 compressed video sequences are used to characterize the 16 sample traces used in this study. From these measurements, a preliminary model is developed which utilizes basic measurements of the individual component video sequences to predict bandwidth requirements and cell loss of the multiplexed video traffic. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1 2. Proposed Video Distribution . . . . . . . . . . . . . . . . . . . . . . . . . 1 3. MPEG Image Coding Basics . . . . . . . . . . . . . . . . . . . . . . . . . 2 4. GOPs and Frames as Network Traffic . . . . . . . . . . . . . . . . . . 3 5. GOP Size Distribution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 6. Development of Sample Set ...

A new approach for bandwidth allocation and congestion control is reported in this paper, which is of the Rate Controlled admission with Priority Scheduling service type. It is called Dynamic Time Sharing (DTS), because of the dynamic nature of the procedure for resource partitioning to allocate and guarantee a required bandwidth for every traffic class. This approach is based on guaranteeing specific traffic parameters (bandwidth requirements) through a policing unit, and then optimizing the bandwidth assignment within the network for specific parameters of interest (like delay or jitter, and loss). The optimization process is based on the parameters guaranteed by the policing unit. A batch admission policy is used at the edges of the network according to a specific framing strategy to follow the traffic characteristics (e.g., the traffic constraint function) of different traffic classes. On the other hand, another framing (congestion control) strategy is used within the network, whic...

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