he video delivered to some of participants will be of low quality (since packet losses and/or delays will be high on some branches of the tree). The second case is safest, but it results in disturbing the bulk of the participants. In practice, it is not clear how to choose an adequate value for this fraction. In IVS, we set the fraction to a few percent [2, 3]. Ideally, however, the source should be able to single out the parts of the multicast tree that experience congestion. In order not to disturb the bulk of participants in the tree, these branches should be treated separately. Two possible solutions include video gateways, and using some form of layered coding. Video gateways or layered coding schemes are not new ideas. Our contribution is to identify and discuss the issues associated with using these techniques in the Internet. We illustrate our points with the H.261[10] software coder of IVS. 2 Video gateways Video gateways take

The single class best effort service available in the current Internet does not provide the guarantees, typically expressed in terms of minimum bandwidth and/or maximum delay or loss, associated with real-time applications such as live video. One way to support such applications in best effort networks is to use control mechanisms that adapt the coding, transmission, reception, and decoding processes at the source and at the destination(s) depending on the state of the network. In this paper, we examine and report on our experience over the past several years with such mechanisms for videoconferencing software. We illustrate our points with results obtained with the IVS software developed at INRIA. We consider in particular rate and error control mechanisms. These mechanisms adapt the bandwidth requirements and the resilience to packet loss of the video stream sent by a source coder. We have found that they do prevent video sources from swamping the resources of the Internet, and that...

The recent advances in digital signal processing, communications, storage and integrated circuit design have caused an expansion of services and applications using digital video. We mention digital television broadcasting, video-ondemand, video-CDs, multipoint video conferencing, and multimedia communications with multiple windows [1, 2, 19, 28, 29, 47]. Among the various services and applications, a wide variety ofdigital video standards and communication channels with di erent bandwidth and reliability is considered for use. For instance, in advanced (high-de nition) television broadcasting a typical video format de nes a frame rate of 25 fps (frames per second) and a spatial frame dimension of 1152 lines and 1440 pixels per line. The bandwidth or bit rate of such a video signal, when PCM encoded at 8 bit per picture element (bpp) in the luminance-chrominance (YUV) color space, is 660 Mbs (Megabits per second). With appropriate video coding techniques, the required transmission bandwidth should be reduced to approximately 50 Mbs. This calls for a compression factor of 10{20. Advanced television broadcasting would typically take place over a terrestrial, satellite or ber-based ATM channel, each of which has speci c degradation characteristics such as delay, drop-out, and bit and burst error rate. Another example is the broadcasting or the storage of standard CCIR-601 resolution video signals (25 fps, 576 lines, 720 pixels per line, YUV representation). Straightforward PCM encoding at 8 bpp requires a bandwidth of 166 Mbs. The target bit rate for CD or magnetic tape storage, or for cable or ATM network transmission is in the range of 2 to 10 Mbs, requiring a compression factor of

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