Call Admission Control and Traffic Policing Mechanisms for the Wireless Transmission of Layered Videoconference Traffic from MPEG-4 and H.263 Video Coders P. Koutsakis, M. Paterakis and S. Psychis Dept. of Electronics and Computer Engineering Information & Computer Networks Laboratory Technical University of Crete, 73100 Chania Greece e-mail: {polk ,pateraki,psycho}@telecom.tuc.gr ABSTRACT In this paper we explore, via an extensive simulation study, the performance of Call Admission Control (CAC) and Traffic policing mechanism proposed for transmitting multiple-layered videoconference movies over a wireless channel of high capacity, depending on the user’s needs and requests. We focus both on MPEG-4 and H-263 coded movies, and in the latter case our scheme achieves high aggregate channel throughput, while preserving the very strict Quality of Service (QoS) requirements of the video traffic. 1. INTRODUCTION-SYSTEM MODEL High-speed packet-switched network architectures have the ability to support a wide variety of multimedia services, the traffic streams of which have widely varying traffic characteristics (bit-rate, performance requirements). The main goal of wireless communication is to allow the user access to the capabilities of the global packet-switched network at any time without regard to location or mobility. In this work, we design and evaluate a scheme which multiplexes MPEG-4 and H.263 video streams (Variable Bit Rate, VBR), respectively, in high capacity picocellular systems with the picocell diameter of the order of a few dozen meters. We consider MPEG-4 movies with two coding layers (high and low bit rate), and H.263 movies with three coding layers (high, medium and low bit rate). Within the picocell, spatially dispersed source terminals share a radio channel that connects them to a fixed base station. The base station allocates channel resources, delivers feedback information and serves as an interface to the mobile switching center (MSC). The MSC provides access to the fixed network infrastructure. We focus on the uplink (wireless terminals to base station) channel, but the nature of our ideas can easily be implemented on the downlink channel as well. The uplink channel time is divided into time frames of equal length. Each frame has a duration of 12 ms (in this study we investigate the case where video traffic is the only traffic in the system. Nevertheless, high capacity wireless channels are often used for integrating voice and video traffic, and the frame duration is chosen to be equal to the time a voice terminal needs to generate a new voice packet [5,8]. Assuming that the speech codec rate is 32 Kbps and that the packet length is equal to the size of an ATM cell, yields the the frame duration of 12 ms), and accommodates 256 information slots. The channel rate is 9.045 Mbps. Each information slot accommodates exactly one, fixed length, packet that contains video information and a header. We consider the channel to be error-free and without capture. 1.1 MPEG-4 and H-263 Streams The MPEG group initiated the new MPEG-4 standards in 1993 with the goal of developing algorithms and tools for high efficiency coding and representation of audio and video data to meet the challenges of video conferencing applications. The MPEG-4 standards differ from the MPEG- 1 and MPEG-2 standards in that they are not optimized for a particular application but integrate the encoding, multiplexing, and presentation tools required to support a wide range of multimedia information and applications. In addition to providing efficient audio and video encoding, the MPEG-4 standards include such features as the ability to represent audio, video, images, graphics, text, etc. as separate objects, and the ability to multiplex and synchronize these objects to form scenes. Support is also included for error resilience over wireless links, the coding of arbitrary shaped video objects, and content-based interactivity such as the ability to randomly access and manipulate objects in a video scene. [2] In our study, we use the trace statistics of actual MPEG-4 streams from [3]. The video streams have been extracted and analyzed from a camera showing the events happening within an office. We have used two coding versions of the movie: a) the high quality version, which has a mean bit rate of 400 Kbps, a peak rate of 2 Mbps, and a standard deviation of the bit rate equal to 434 Kbps, and b) the low quality version, which has a mean bit rate of 90 Kbps, a peak rate of 1 Mbps, and a standard deviation of the bit rate equal to 261 Kbps. New video frames arrive every 40 msecs. We have set the maximum transmission delay for video packets to 40 msecs, with packets being dropped when this deadline is reached. That is, all video packets of a video frame (VF) must be delivered before the next VF arrives. The allowed video packet dropping probability is set to 0.0001 [1,5]. H.263 is a video standard that can be used for compressing the moving picture component of audio-visual services at