We study the performance of a fixed frequency hopping (FH) wireless system that transmits traffic over a wireless link that is subject to random interference. The system consists of an access device that buffers traffic to transmission over N FH channels. Bit errors on the FH channels are controlled by a hybrid ARQ/FEC protocol. We analyze the effect of interference on traffic blocking and delay at the access device. To alleviate the effect of interference on traffic performance, we investigate the use of FEC as well as a simple dynamic frequency hopping (DFH) strategy that allows the transmitter to avoid hopping to the bad channels. We present numerical results that quantify the adverse effect of interference on traffic blocking and delay, as the hopping frequency is varied. In addition, the results indicate that FEC is effective in improving traffic performance, and the effect of interference on traffic performance can even be eliminated by a DFH strategy.

We consider an ATM switch that handles two types of traffic: voice and best-effort data. Voice is delay-sensitive while data is loss-sensitive. Contention at output ports is resolved by output queuing. The output buffer is organized into two virtual queues, one for voice and the other for data. To meet the delay QoS of voice, the channel is allocated to the respective queue with a probability that depends on the congestion level in that queue. We present a model and analysis of the output queues and calculate the waiting time and loss performance under two specific channel allocation strategies. The results indicate that the proposed channel allocation strategy is quite effective in maintaining a balance between the QoS objectives of the two traffic types.

ATM over xDSL allows integrated access of video, voice and data services into the Internet with QoS support. In this paper, we study the integration of MPEG video calls and high-speed data calls at an ATM DSL access multiplexer (DSLAM). The interactivity in video calls requires a stringent delay QoS (but relaxed cell loss). In contrast, data calls have a stringent cell loss QoS requirement (but relaxed delay). We study an access strategy which allocates separate buffers to the 2 types of traffic and assigns priority to video in accessing the channel. We first present a Markov-modulated fluid model for the video sources based on measurements performed on a large number of MPEG sequences. Then, we present a computationally efficient fluid analysis of multiplexed video calls. Finally, we provide a fluid analysis of the low-priority data buffer. We derive a model for the output fluid process of the video buffer and approximate it with a two-state model. We use both the full and approximate models to calculate the cell loss for data. We compare the performance of these two methods to a commonly-used approximate method. The study shows that: (1) our two-state approximation is fairly accurate while computationally very efficient; (2) the commonly used method is overly inaccurate compared to the first two methods.

Wireless ATM is a framework that extends a wired backbone network with the flexibility of wireless access and mobility support. The technology has been actively studied over the last couple of years by R&D groups as well as standards organizations worldwide. Standardization efforts of WATM have been lead by the ATM Forum and the European Telecommunications Standards Institute (ETSI). The first draft of WATM specification has been released by the ATM Forum. We present an update on the status of WATM standards. Then we address the main technical issues in the specification of a wireless ATM network, namely data link control, medium access control, wireless control, and mobility management.

We study an ATM access node which multiplexes on-off calls onto a common channel on a cell loss and call blocking basis. Calls arrive at a Poisson rate /spl lambda/ calls/sec and require an exponentially distributed holding time of mean 1//spl mu/ seconds. When on, a call generates traffic at a fluid rate R cells/sec and when off it generates no traffic. The on and off time intervals are exponentially distributed with means 1//spl alpha/ and 1//spl beta/ second, respectively. When arriving cells can't be transmitted immediately, they are queued in a common buffer of size B cells. Cells arriving when the buffer is full are assumed lost. Using a fluid model, we derive the cell loss and call blocking probabilities and apply them to provision the maximum call handling capacity, N, of the system. When N calls are already in progress, arriving calls are blocked. When call-level transition rates are much smaller than cell-level transitions, the input process is nearly completely decomposable (NCD). This NCD approximation greatly reduces the computational difficulty of the exact fluid model. Numerical results show that the NCD approximation is quite accurate over a wide range of traffic parameters.

We study the characterization of MPEG sources and their multiplexing onto a 25 Mbps port of an ATM LAN. The bit rate characteristics of a wide variety of MPEG sequences have been measured. Although long range dependence has been observed in sample autocorrelations, we have focused on the exponentially decaying short range dependence and constructed a 4-state Markov-modulated fluid source model based on ensemble measurements of various sequences. This MPEG source model is then used to study the statistical behavior of a common buffer that multiplexes multiple sources onto a 25 Mbps ATM LAN switch. We derive the cell loss performance versus buffer size when the number of multiplexed sources is increased. With an ensemble source average rate of 1.67 Mbps, our results show that a cell loss rate better than 10/sup -6/ is maintained by a 800-cell buffer if 10 sources are used, yielding a multiplexing gain of 2.

To be fully integrated, the Integrated Service Digital Network (ISDN) arena is preparing an evolution from its current narrowband structure (at basic and primary access rates) to a futuristic wideband structure where access rates will be capable of supporting services beyond 100 Mbits/s. A wideband channel structure for ISDN, supplementing the narrowband channel standards, is thus needed. This paper presents a channel structure in which the access bandwidth is organized in a hierarchy of channels of increasing speeds. Channels at a given speed level accommodate services with specific bandwidth and holding time requirements. However, to improve performance, the access interface uses a control strategy whereby a service request at a given level can overflow to a lower level for access at lower bit rate. The performance, in terms of access delay and access throughput, of this access scheme is analyzed. Numerical results are provided for the special case of an interface with two levels of channels. These results show that a significant improvement in access performance can be obtained with the overflow mechanism of the access control discipline used.

A common digital transmission facility in a wide-band integrated service digital network (ISDN) provides shared access to a community of heterogeneous users. Traffic demands from these users vary in their arrival rate, their service time, and their bit rate. In order for this type of communication system to handle its traffic demands with high efficiency and flexibility, a close control of access to the shared bandwidth is required. We model the system by a general multiserver queueing system where customers demand service from a random number of servers. If no waiting is allowed, this queueing model is readily analyzed, and various server allocation strategies can be studied. If the various access requests are queued for service, then the system calls for efficient strategies for allocating servers to waiting customers. In this case, exact analysis of the underlying queueing model becomes quite difficult. For this case, we present some analytic and simulation results of the performance of the system under several server allocation policies.

A common digital transmission facility provides service to a community of heterogeneous users generating traffic with differing intensity, message length, and bit rate. In order for this type of integrated communication system to handle its traffic demands with high efficiency and flexibility, close control of access and switching at the input node is required. We propose, analyze, and compare two different strategies for managing the access of two types of traffic, a blockable wide-band (WB) type of traffic and a queueable narrow-band (NB) type of traffic, sharing the transmission resource dynamically. The first strategy assigns preemptive priority to the WB traffic over the NB traffic, whereas the second strategy employs a wide-band to narrow-band bit rate compression mechanism. Exact analytic models are developed, and solution methods are presented and implemented. It is shown that the best combined performance is obtained when the two strategies are adaptively combined according to the offered load.