M. El Zarki

SIR-based call admission control for DS-CDMA cellular systems

Signal-to-interference ratio (SIR)-based call admission control (CAC) algorithms are proposed and studied in a DS-CDMA cellular system. Residual capacity is introduced as the additional number of initial calls a base station can accept such that system-wide outage probability will be guaranteed to remain below a certain level. The residual capacity at each cell is updated dynamically according to the reverse-link SIR measurements at the base station. A 2/sup k/ factorial experimental design and analysis via computer simulations is used to study the impact of the parameters used in the algorithms. The influence of these parameters on system performance, namely blocking probability and outage probability, is then examined via simulation. The performance of the algorithms is compared together with that of a fixed call admission control scheme (fixed CAC) under both homogeneous and hot spot traffic loading. The results show that SIR-based CAC always outperforms fixed CAC even under overload situations, which is not the case in FDMA/TDMA cellular systems. The primary benefit of SIR-based CAC in DS-CDMA cellular systems, however, lies in improving the system performance under hot spot traffic. >

MPEG coding for variable bit rate video transmission

For real-time transmission of broadcast-quality video on ATM-based B-ISDN, the intraframe to interframe ratio and the quantizer scale are two key parameters that can be used to control a video source in a network environment. Their impact on the traffic characteristics of the coder provides insights into the cell arrival process for an MPEG source. The authors obtain video source models for coders that utilize a standard algorithm that can be applied to a multitude of video services. The output stream of a video coder, which complies with the Motion Pictures Expert Group (MPEG) coding standard, is studied with an National Television Systems Committee (NTSC) quality video sequence as the input. Because the MPEG video coding algorithm has been proposed for a variety of applications, they also investigate the effect of changing the coding parameters on the statistics of interest.<<ETX>>

Performance of H.263 video transmission over wireless channels using hybrid ARQ

This paper proposes a hybrid ARQ error control scheme based on the concatenation of a Reed-Solomon (RS) code and a rate compatible punctured convolutional (RCPC) code for low-bit-rate video transmission over wireless channels. The concatenated hybrid ARQ scheme we propose combines the advantages of both type-I and type-II hybrid ARQ schemes. Certain error correction capability is provided in each (re)transmitted packet, and the information can be recovered from each transmission or retransmission alone if the errors are within the error correction capability (similar to type-I hybrid ARQ). The retransmitted packet contains redundancy bits which, when combined with the previous transmission, result in a more powerful RS/convolutional concatenated code to recover information if error correction fails for the individual transmissions (similar to type-II hybrid ARQ). Bit-error rate (BER) or signal-to-noise ratio (SNR) of a radio channel changes over time due to mobile movement and fading. The channel quality at any instant depends on the previous channel conditions. For the accurate analysis of the performance of the hybrid ARQ scheme, we use a multistate Markov chain (MSMC) to model the radio channel at the data packet level. We propose a method to partition the range of the received SNR into a set of states for constructing the model so that the difference between the error rate of the real radio channel and that of the MSMC model is minimized. Based on the model, we analyze the performance of the concatenated hybrid ARQ scheme. The results give valuable insight into the effects of the error protection capability in each packet, the mobile speed, and the number of retransmissions. Finally, the transmission of H.263 coded video over a wireless channel with error protection provided by the concatenated hybrid ARQ scheme is studied by means of simulations.

Bandwidth-allocation schemes for variable-bit-rate MPEG sources in ATM networks

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. >

Performance analysis of DS-CDMA with slotted ALOHA random access for packet PCNs

In this paper, we present a discrete time Markov chain based analytical framework for the study of Direct-Sequence Code-Division-Multiple-Access (DS-CDMA) with slotted ALOHA random access protocols (DS-CDMA-S-ALOHA) for packet Personal Communications Networks (PCNs). It incorporates both the random access and the random errors associated with DS-CDMA-S-ALOHA protocols into a unified framework. The key feature is that it distinguishes between the two stages in the transmission process, namely the access stage and the reception stage, which characterize the random access and the random errors associated with DS-CDMA-S-ALOHA protocols respectively. Two DS-CDMA-S-ALOHA protocols are presented and analyzed. The performance of the protocols and the effects of the design parameters, namely the packet retransmission probability and the forward error correction code rate of the Bose-Chaudhuri-Hocquenghem (BCH) block codes are evaluated numerically and compared with a bandwidth equivalent conventional multi-channel slotted ALOHA system. The results show that, by proper design, the DS-CDMA-S-ALOHA protocols can double the throughput with respect to that of a bandwidth equivalent conventional multi-channel slotted ALOHA system.

Bandwidth requirements of variable bit rate MPEG sources in ATM networks

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.<<ETX>>

A look at the MPEG video coding standard for variable bit rate video transmission

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.<<ETX>>

Quality control for VBR video over ATM networks

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.

Prioritized transmission of variable bit rate MPEG video

The traffic characteristics of an MPEG coder which generates a high-priority and a low-priority bit stream are studied. The prioritization scheme for the coder assigns components of the data stream to each priority level based on the value of a parameter B. Statistics of both high- and low-priority cells for a pure intraframe and a mixed interframe/intraframe coder are presented. In particular, the probability mass functions on a cells per slice basis of the high-priority and low-priority component for various values of B are shown. Statistics gathered from this type of prioritization scheme can be applied in several ways for network design and operation.<<ETX>>

Delay and synchronization control middleware to support real-time multimedia services over wireless PCS networks

This paper presents a discussion of several middleware design issues related to the support of real-time multimedia communications over wireless personal communication services (PCS) networks. Specific interests are given to error recovery and synchronization mechanisms. A hybrid automatic repeat request (ARQ) scheme is employed for error control in the proposed system because it can efficiently adapt to nonstationary wireless channels and yield high throughput and reliability. In particular, delay and delay jitter control related to retransmissions in the error control module are addressed. An adaptive source rate control mechanism is used to handle the fluctuation of the effective channel data rate due to retransmissions. An adaptive synchronization scheme is developed to compensate for long-term delay variation caused by large-scale fading so that synchronization is preserved and end-to-end delay is kept low. Simulation results from the performance evaluation of the system are presented.