In this paper, we investigate the issue of how multiparty-to-multiparty communication should be realized. In order to quantify the different techniques, we examine the network-layer (bandwidth and buffer) allocations of four models for realizing multiparty-to-multiparty communication. An example application considered in this paper, where such communication will be needed, is video conferencing based on MPEG video streams. Analytical comparisons are carried out based on actual data obtained from empirical measurements of the source characteristics of MPEG video (determined to be a 40 state Markov modulated fluid source). Our results show that in order to minimize bandwidth and buffer (network-layer resources) allocations, we must exploit the effect of statistical multiplexing by using shared buffers for such multiparty-to-multiparty communication. Using these results we identify the best technique for multiparty-to-multiparty communications.

Increasingly, there is a need for context-aware networking. For example, in multimedia networks with heterogeneous end equipment, if two or more communicating nodes do not have the same application-layer interface, the network should be aware of the "context" of the communication in order to select an appropriate converter through which to route connections. In this paper, we motivate the need for logical connections, where a logical connection consists of one or more network-layer connections routed through application-layer resources, such as bridges and/or format/rate converters. We then describe the application-aware routing protocol (AARP) for managing logical connections.

The performance of error control based on forward error correction (FEC) for MPEG-2 video transmission in an indoor wireless ATM LAN is studied. A multipath fading model is used to investigate the effect of errors on video transport. Combined source and channel coding techniques that employ single layer and scalable MPEG-2 coding to combat channel errors are compared. Simulation results indicate that FEC-based error control in combination with 2-layer video coding techniques can lead to acceptable quality for indoor wireless ATM video.

Wireless ATM LANs have the potential to support multi-Mb/s bandwidths to mobile users with guaranteed quality of service. However, the lossy nature of the wireless medium will pose problems for loss-sensitive applications. Techniques to minimize the effect of these losses will therefore be required. In this paper, we examine the use of combined source and channel coding for MPEG video transport in a wireless ATM environment. Although forward error correction (FEC) provides protection against channel bit errors, the bandwidth overhead can become a significant drawback in a fixed bandwidth scenario. Additional protection against losses can be realized by using two-layer video coding. In this work, we compare the performance of a 1-layer main profile and 2-layer data partitioning and SNR scalable MPEG-2 encoders in a system with random channel errors and forward error correction. The results indicate that if the channel bit error rate is known an optimum FEC level can be chosen for the 1-layer case; However, at this fixed FEC level, if a critical bit error rate is exceeded then video quality degrades dramatically. The 2-layer cases appear to lead to more graceful degradation in quality at this critical bit error rates. In particular, SNR scalability may lead to better video quality over a larger range of bit error rates than a 1-layer approach.

Successful exploitation of prioritized ATM networks requires matching the characteristics of the coded source with the unique properties of the network. This is particularly true for video which is very bandwidth intensive. The authors describe and investigate the performance of three prioritization schemes in terms of the traffic characteristics, SNR, and visual quality. These schemes are further designed such that the high priority data stream is MPEG-I compliant and only a single pass through the encoder is required. A technique for choosing parameter values which achieve a specified high priority traffic bandwidth is presented for two of these schemes. The technique is demonstrated to be remarkably accurate for different video sequences.>

The performance of variable bandwidth allocation schemes that estimate the required bandwidth for a variable-bit-rate (VBR) MPEG video source using a simple prediction algorithm is studied. The advantages of using a layered MPEG coder over the standard MPEG coder are discussed. A method for controlling a layered MPEG coder in order to reduce the effect of cell losses when compared to a standard MPEG coder is demonstrated.>

In this paper, we describe MPEGTool, an X window-based tool that can be used to generate a motion picture expert group (MPEG) encoded bit stream for video sequences and to study the statistical properties of the encoded data. The tool was designed to study the characteristics of variable bit rate video sources for transmission over asynchronous transfer mode (ATM) based broadband integrated services digital network (B-ISDN). The tool, which has a window-based graphic user interface, allows a user to specify several of the MPEG parameters such as the intraframe-to-interframe ratio and the quantizer scale. The tool also includes a statistical package that allows the user to plot graphs of various statistics including bit distributions, ATM cell distributions, time se ries, autocorrelation functions and cell interarrival times.

The complex traffic characteristics of variable-bit-rate (VBR) video sources makes them difficult to accommodate in asynchronous-transfer-mode (ATM) networks. To efficiently transport these services will require both an understanding of the source traffic and novel network control schemes. The performance of variable-bandwidth-allocation schemes that estimate the required bandwidth for a VBR Motion Picture Experts Group (MPEG) video source is studied using a simple prediction algorithm. The advantages of using a layered MPEG coder over the standard MPEG coder are discussed. A method for controlling a layered MPEG coder to reduce the effect of cell losses when compared to a standard MPEG coder is demonstrated.>

The MPEG video coding standard for the transmission of variable-bit-rate video on asynchronous transfer mode (ATM)-based broadband ISDN is examined. The focus is on its use for real-time transmission of broadcast-quality video. The impact of two key parameters, the intraframe to interframe picture ratio and the quantization index that are defined in the standard, on the bit rates per frame was studied. These parameters can be used to control video sources depending on the state of the network. Also, as opposed to previous work which looks only at bit rates per frame, the bits generated per macroblock are studied. This is the basic MPEG coding unit. By packetizing these bits, insight was obtained into the cell arrival process to a network for a MPEG video source.>