We study the data performance of WCDMA systems using the downlink shared channel (DSCH) by investigating the impact of loading, rate adaptation and power control for the typical urban (TU) channel. We conclude that, using DSCH, high speed data transmissions can be achieved if the offered traffic load is well controlled. We discuss a heuristic rate adaptation algorithm that adapts the transmission rate based upon the perceived user performance in previous frames. We show that the performance of the rate adaptation is significantly better than that for the fixed-rate allocation policy.

We investigate the performance of voice and best-effort data traffic over Enhanced General Packet Radio Services (EGPRS) using tight reuse configurations (e.g. 1/3 or 1/1), with the aim of identifying suitable bearer designs and optimal system configurations for these services. We find that voice and data have somewhat conflicting preferences: voice capacity is optimized using random frequency hopping and quality-driven power control: data throughput is optimized using no frequency hopping and no power control. These somewhat contradictory requirements impose a challenge in the system design. We investigate two simple traffic integration strategies: (1) segregating the resource for the two services and using the optimal configuration in each segment; and (2) integrating the traffic by allowing voice and data to share the same resource, and use a compromised configuration for the entire system. We find that if the majority of traffic is voice, traffic integration has an advantage since data can use any residual voice capacity. On the other hand, if the majority of traffic is data, segregation provides higher spectrum efficiency. Furthermore, the integrated traffic option becomes increasingly sub-optimal with increasing data traffic load. We conclude that the system design should depend upon the expected mix of traffic (including traffic types in addition to voice and best-effort data) and the desired engineering complexity, and that further work is needed to understand the possible design options. One possible design option is to soft segregate the resource, e.g., to allow data to use the voice segment on an overflow basis. Similar strategies may be good avenues for future work.

We propose a framework for network-assisted radio resource management in wireless data networks. This type of radio resource management techniques offer implementation and capacity benefits compared to conventional, interference-measurement based, dynamic channel assignment (DCA) algorithms. The basic idea is to use interbase signaling to shift most of the burden of the resource allocation from the air interface to the backbone infrastructure. By exchanging channel assignment as well as other relevant information in real time through the backbone network, each base can calculate the impact of a resource assignment on the system. As a result, rapid interference measurements, which are typically needed to implement DCA schemes, are replaced by a limited amount of path loss measurements and the computation of interference conditions by the base stations. This significantly reduces the measurement and over-the-air signaling requirements, and can also provide an opportunity for a better optimization of the system performance. We focus on two specific algorithms: network-assisted least-interference-based dynamic packet assignment (NA-LI-DPA) and network-assisted dynamic packet assignment with throughput optimization (NA-DPA). NA-LI-DPA closely resembles a least-interference-based dynamic channel assignment algorithm, and NA-DPA attempts to further improve the overall system throughput. The algorithms, as defined, are appropriate for a best-effort data service, where the primary goal is to provide a higher throughput. However, it will be clear from the discussion that it is also feasible to alter the algorithms to optimize performance metrics other than throughput, e.g., to ensure a certain quality of service. We show through simulation that, for a system like enhanced general packet radio service (EGPRS) system, NA-DPA can provide a throughput that is 50% higher than random packet assignment, and 25% higher than that obtained by conventional DCA algorithms.

In this paper, using the EDGE system as an example, we apply a Kalman filter power control method based on interference tracking and prediction to packet voice services in wireless networks. Our results reveal that power control significantly improves the spectral efficiency by enabling 1/3 frequency reuse while maintaining a stringent requirement of 2% packet loss probability for a voice service. More specifically, for allocated spectrum of 1.8, 3.6 and 5.4 MHz, the 1/3 reuse with the Kalman power control can yield 102.5%, 49.5% and 32.5% improvement in spectral efficiency, respectively, over 3/9 reuse (regardless of whether or not power control is used). We also find that the Kalman method provides 20% additional spectral efficiency when compared with a traditional SIR power control method, and the former method is more robust than the latter for increased power update period. The protocol requirements for the implementation of the Kalman method in the EDGE system are also discussed.

In this paper, we study joint channel estimation and maximum likelihood decoding method for coded orthogonal frequency division multiplexing (COFDM) systems. Using this method in conjunction with convolutional coding, robust and nearly optimal coherent detection can be achieved in rapid dispersive fading channels. Significant performance gain in packet data throughput is realized in a system with aggressive frequency reuse.

We present physical and MAC layer techniques for high-speed wireless access to the Internet based on OFDM. Using dynamic packet assignment (DPA) for interference avoidance based on two examples of resource definition with different time and frequency units, we demonstrate that OFDM has significant flexibility to provide high transmission rates on demand with different available bandwidths. Innovations in radio link design, such as joint channel estimation and maximum likelihood decoding as well as interference suppression can be readily applied to coded OFDM systems. Significant performance gain in packet data throughput is realized in a system with aggressive frequency reuse.

We investigate the possibility of supporting voice over EGPRS as a potential evolutionary path for 2/sup nd/ generation wireless TDMA systems. We discuss different voice bearer designs for EGPRS radio access networks. We demonstrate how the operating mode can be chosen at each layer in the protocol stack to form appropriate voice bearers, and quantify the performance of these bearers. Different system control options are examined, including frequency reuse factor, MAC layer operating mode, frequency hopping, power control, and dynamic channel assignment. We find that a baseline EGPRS system using random channel assignment and no power control can support up to 27 Erlang/Site/MHz, which is significantly higher than the capacity of current 2/sup nd/ generation TDMA systems. In addition, the packet-switched nature of EGPRS makes it a very flexible system. We conclude that EGPRS offers an excellent evolutionary path for 2/sup nd/ generation TDMA systems.

We propose a network-assisted radio resource allocation algorithm for wireless data networks with synchronized base stations, which support variable rate transmission. The algorithm, Soft-Bar, may be used in both mobile and fixed wireless systems to assign the channel, transmission mode and transmit power for each user, and may be implemented in a distributed manner. The basic idea of the algorithm is to use measurements at the base and the terminal, and limited, real-time inter-base signaling, to control the interference generated by the few dominant interferers for each user. We show that in an EDGE-like system using 1/3 reuse, compared to conventional random packet assignment, Soft-Bar can (1) increase the system throughput by more than 50% for a given delay constraint; or (2) increase the number of users by more than 30% for a given user outage requirement, while at the same time improving the user experience. In addition, it can facilitate QoS provisioning in a straightforward manner.

We examine the feasibility of supporting integrated services, such as packet voice, Web browsing, and best-effort data, using Enhanced General Packet Radio Services (EGPRS). Among many factors that need to be considered, we focus on enhancing the radio link control/medium access control (RLC/MAC) layer design to support such services. We first describe the relevant capabilities offered by the current standard, and then outline additional capabilities that are needed at RLC/MAC layer to support real-time services. In summary, these needed capabilities are: (1) fast uplink access during an ongoing session; (2) fast resource assignment for both uplink and downlink; and (3) the ability to differentiate services at the base station sub-system. To realize these capabilities, we propose enhancements to the RLC/MAC layer, including: (1) a new set of packet control channels; (2) uplink access protocols that utilize these control channels; and (3) a new access request identifier. These changes would enable EGPRS to offer a wide range of services through a single packet based network.

We describe a method to maintain quality for video transported over wireless channels. The method is built on three fundamental blocks. First, we use a transcoder that injects spatial and temporal resilience into an encoded bitstream. The amount of resilience is tailored to the content of the video and the prevailing error conditions, as characterized by bit error rate. Second, we derive analytical models that characterize how corruption propagates in a video that is compressed using motion-compensated encoding and subjected to bit errors. Third, we use rate distortion theory to compute the optimal allocation of bit rate among spatial resilience, temporal resilience, and source rate. Furthermore, we use the analytical models to generate the resilience rate distortion functions that are used to compute the optimal resilience. The transcoder then injects this optimal resilience into the bitstream. Simulation results show that using a transcoder to optimally adjust the resilience improves video quality in the presence of errors while maintaining the same input bit rate.

Transmitter diversity and down-link beamforming can be used in high-rate wireless networks with OFDM for capacity improvement. In this paper, we compare the performance of delay, permutation and space-time coding transmitter diversity for OFDM systems, and its impact on high-rate packet data wireless networks, where relatively high block error rates, such as, 10%, on the wireless links are acceptable using ARQ. As an alternative, we also consider using the same number of transmitter antennas for down-link beamforming in the direction of the desired mobiles. Our investigation indicates that the delay transmitter diversity with QPSK modulation and adaptive antenna arrays can provide good quality of service (QoS) with a low retransmission probability while space-time coding transmitter diversity can provide high peak-rate. However, down-link beamforming together with adaptive antenna arrays, if it is applicable, appears to be a better alternative to transmitter diversity.

In this study, we propose a new link adaptation algorithm for wireless data communications. Compared to the existing algorithms, the most distinguishing feature of this algorithm is the implementation of a no-transmission mode, a so called mode 0, for active users. In particular, transmissions to or from a user are not permitted while the estimated signal quality of the user, expressed as signal-to-interference ratio (SIR), is lower than a cutoff threshold. Unlike the traditional admission control, which is performed only once upon admitting a user, mode 0 is an integral part of an ongoing link adaptation process for admitted users. This mode automatically controls the system load, reduces unnecessary retransmissions, and therefore, improves the overall throughput. Our simulation results show that for the system under study, a throughput improvement of up to 75% can be achieved.

The rapid growth of wireless voice subscribers; the growth of the Internet; and the increasing use of portable computing devices suggest that wireless Internet access will rise rapidly over the next few years. Transmission rates are currently only about 10 kb/s for large cell systems. Third generation wireless access and the evolution of second-generation systems will provide bit rates of 50 to 384 kb/s in macrocellular systems. This paper discusses packet data transmission rates of 2 to 5 Mb/s in macrocellular environments and up to 10 Mb/s in microcellular and indoor environments as a complementary service to evolving second and third generation wireless systems. Dynamic packet assignment for high efficiency resource management and packet admission; OFDM at the physical layer with interference suppression, space-time coding and frequency diversity; as well as smart antennas to obtain good power and spectral efficiency are discussed.

Link adaptation and incremental redundancy are two link level techniques proposed to enhance data rate and increase throughput in wireless data networks. Link adaptation pro-actively adjusts the modulation and coding scheme based upon estimated channel conditions. Incremental redundancy, on the other hand, adjusts the code rate to actual channel conditions by incrementally transmitting redundancy information until decoding is successful. In this study, we investigate the system performance of these two techniques. Our results show that incremental redundancy achieves significantly higher throughput compared to link adaptation, especially under high traffic loading conditions. This gain is achieved at the expense of higher terminal and base station complexities, as well as a higher packet delay. On the other hand, in contrast to link adaptation, incremental redundancy does not require any channel quality measurements, and can operate based solely upon ACKs and NACKs.

Transmitter diversity and down-link beamforming can be used in high-rate data wireless networks with orthogonal frequency division multiplexing (OFDM) for capacity improvement. We compare the performance of delay, permutation and space-time coding transmitter diversity for high-rate packet data wireless networks using OFDM modulation. For these systems, relatively high block error rates, such as 10%, are acceptable assuming the use of effective automatic retransmission request (ARQ). As an alternative, we also consider using the same number of transmitter antennas for down-link beamforming as we consider for transmitter diversity. The investigation indicates that delay transmitter diversity with quaternary phase-shift keying (QPSK) modulation and adaptive antenna arrays provides a good quality of service (QoS) with low retransmission probability, while space-time coding transmitter diversity provides high peak data rates. Down-link beamforming together with adaptive antenna arrays, however, provides a higher capacity than transmitter diversity for typical mobile environments.

In this paper, we introduce a simple geometric approach that is based on signal-space concepts to efficiently evaluate the performance of M-ary phase-shift keying (M-PSK) and M-ary quadrature amplitude modulation (M-QAM) schemes over an additive white Gaussian noise channel. In particular, new bit error rate approximations are derived and shown to be in excellent agreement with Monte Carlo simulation results.

We describe a method to maintain an acceptable quality for video transported over wireless networks under time-varying conditions. We use a transcoder to modify the resilience of the encoded bitstream by using source coding techniques to provide the appropriate level of resilience for the prevailing channel conditions. We develop a statistical model for image loss versus resilience and wireless conditions. Simulation results indicate that using a transcoder to adjust the resilience can improve video quality when errors occur without significantly sacrificing quality when there are no errors. Also, simulation results compare favorably to the analytical model.

This paper investigates the use of a maximum a posteriori probability (MAP) algorithm to realize soft-output equalization in a concatenated equalization and trellis-coded modulation (TCM) decoding-based wireless communication system. Specifically, we first begin with a general MAP algorithm and then focus on studying Bahl's (1974) MAP and Lee's (1974) MAP algorithms. We then propose a modified version of Lee's MAP algorithm which is much simpler than the original, in terms of complexity, and is more practical. In particular, a very simple channel estimation method which employs orthogonal training sequences is proposed. In order to improve the system performance, equal-gain combining and selection diversity are also considered. Finally, we compare the performance of the MAP algorithm-based equalization with our previously proposed equalization scheme, which combines decision feedback equalization and TCM.

The advanced cellular Internet service (ACIS) is targeted for applications such as Web browsing with a peak downlink transmission rate on the order of 1-2 Mbits/s using a wide-area cellular infrastructure. In order to provide bandwidth on demand using scarce radio spectrum, the medium-access control (MAC) protocol must: 1) handle dynamic and diverse traffic with high throughput, and 2) efficiently reuse limited spectrum with high peak rates and good quality. Most of the existing approaches do not sufficiently address the second aspect. This paper proposes a dynamic packet assignment (DPA) scheme which, without coordinating base stations, allocates spectrum on demand with no collisions and low interference to provide high downlink throughput. Interference sensing and priority ordering are employed to reduce interference probability. A staggered frame assignment schedule is also proposed to prevent adjacent base stations from allocating the same channel to multiple mobiles at the same time. Simulation results based on a packet data traffic model derived from wide-area network traffic statistics, which exhibit a "self-similar" property when aggregating multiple sources, confirm that this method is able to reuse spectrum efficiently in a large cellular system having many users with short active periods. Distributed iterative power control further enhances spectrum efficiency such that the same channel can be simultaneously reused in every base station.

Future wireless personal communications systems are expected to support a wide range of high-quality services that require high data rates. Communicating at high transmission rates over wireless environments, however, creates many difficult problems such as intersymbol interference (ISI) and cochannel interference. In this paper, we present a method which employs an adaptive decision feedback equalization (DFE), interleaving, space diversity and trellis-coded modulation (TCM) to combat both ISI and cochannel interference in cellular mobile radio environments. The feedback filter of DFE uses final decision symbols of Viterbi decoding of TCM to cancel tail ISI with a square root Kalman algorithm. Some simulation results which illustrate the potential of the proposed system are given. In particular, a performance comparison between the proposed method and QPSK system is also undertaken.

This paper investigates the use of soft-output equalization methods that are based on Bahl's (1974) MAP algorithm along with a modified version of such algorithm which does not require knowledge of the noise variance. We use both algorithms to realize soft-output detection in a concatenated equalization and trellis-coded modulation (TCM) decoding based wireless communication system. Further, the effect of using different detection block lengths in the modified MAP algorithm is studied and the use of two-branch equal-gain combining and selection diversity is also considered. Finally, we compare the performance of the MAP algorithm based equalization with our previously proposed equalization scheme which combines decision feedback equalization (DFE) and TCM.

Advanced cellular Internet services (ACIS) are targeted for applications such as web browsing with a peak downlink rate on the order of 1-2 Mb/s using the wide-area cellular infrastructure. PCS or new spectrum allocations by the FCC could be used for ACIS. In order to provide bandwidth on demand using scarce spectrum, the medium access control (MAC) protocol must: (1) handle dynamic and diverse traffic with high throughput and (2) efficiently reuse limited spectrum with high peak rates and good quality. Most of the existing approaches do not address the second aspect. In this paper, we first outline the problems for media access for ACIS and review some existing approaches, followed by a discussion of the proposed methodology. The remaining part focuses on a channel assignment method which combines the concepts of interference sensing and priority ordering to dynamically reuse spectrum with low interference probability. Preliminary simulation results based on a packetized voice model indicate that this method is able to reuse spectrum efficiently in a large cellular system having many users with short active periods.

Nowadays the image and video compression standards employ variable length coding (VLC). One disadvantage of using VLC is that whenever an error occurs, it propagates until the next resynchronization can be achieved. Error propagation under fading conditions makes severe destruction in a bundle of blocks within an image. In such situations, error concealment may not perform well due to the missing information in consecutive blocks. In this paper, block shuffling is proposed for the purpose of randomizing the blocks' location so as to minimize the chance of getting groups of error blocks. Image transmissions under AWGN and fading channels are simulated and the enhancement of the error concealment performance is investigated.

We present a strategy to equalize the short burst without using a training sequence. This method is suitable for low-overhead TDMA systems such as the Personal Access Communications System (PACS) whose frame is very short. The blind equalization algorithms are discussed. The simulation results indicate that the proposed method is feasible.

We have proposed and studied the performance of a measurement based MAC protocol for wireless packet data communications in the presence of co-channel interference. A fuzzy logic based algorithm is used to control the permission probability of each TDMA slot based on the interference and traffic conditions. This algorithm provides an adaptive access mechanism to achieve a balance between traffic throughput and signal quality. We performed computer simulations to study the protocol sensitivity to various system parameters. Performance measures includes average delay, protocol efficiency and channel utilization. Compared with the unmodified PRMA, this protocol can provide significant improvement since it is adaptive to traffic intensity and interference conditions. This method may also be suitable for multimedia wireless access.

The diverse environments emerging for wireless communication applications could render the centralized prediction-based channel assignment methodology, conventionally employed in cellular radio networks, impractical. The distributed measurement-based approach seems to be a more practical solution. We evaluate and compare several distributed measurement-based algorithms for dynamic channel assignment (DCA). Their performance is also compared with a centralized prediction-based algorithm. It is found that a simple aggressive algorithm with the use of a threshold, known as the least interference algorithm (LIA), performs the best.

We combine autonomous algorithms for dynamic channel assignment (DCA) and power control in a TDMA/FDMA wireless system as a medium access control (MAC) protocol. The DCA algorithm determines paired radio channels that experience the least interference and are least likely to cause interference. The power control algorithm uses local estimations of signal to interference ratio (SIR) at a receiver to iteratively command power adjustment on the desired transmitter. A common control frequency, which is frame-synchronized among base stations, provides all necessary information for DCA without blind slots. Computer simulations are used to evaluate system performance. Results from computer simulations demonstrate good spectrum efficiency and robustness. Although studied under a specific set of parameters, this type of MAC protocols can be applied in different wireless communications environments.>

A new approach to indoor propagation prediction which is based on the Rayleigh-Gans approximation is proposed. The approach has been motivated by the observation that the indoor environment is different from the outdoor environment in that many of the structures within buildings are approximately the size of a wavelength making propagation prediction based on high frequency techniques not strictly applicable. In the paper details of the Rayleigh-Gans approximation are provided. Actual indoor propagation measurements are also provided to allow preliminary discussions on the accuracy of the Rayleigh-Gans approximation to be given. Comparisons with simulations of empirical and ray tracing models are also provided. The results indicate that further development would be worthwhile.>

We combine autonomous algorithms for dynamic channel assignment (DCA) and power control in a TDMA/FDMA wireless system as a medium access control (MAC) protocol. The DCA algorithm determines paired radio channels that experience the least interference and are least likely to cause interference. The power control algorithm uses local estimations of signal to interference ratio (SIR) at a receiver to iteratively command power adjustment on the desired transmitter. A common control frequency, which is frame-synchronized among base stations, provides all necessary information for DCA without blind slots. Computer simulations are used to evaluate system performance. Results from computer simulations demonstrate good spectrum efficiency and robustness. Although studied under a specific set of parameters, this type of MAC protocol can be applied in different wireless communications environments.>

Using dynamic channel assignment (DCA) algorithms to select communications channels as needed, time-division multiple access (TDMA) or frequency-division multiple access (FDMA) systems can serve dynamic and nonuniform traffic demands without frequency planning as long as quality is sufficient and equipment is available. Here, performance issues and algorithms for DCA in a TDMA portable radio system are considered. A fixed number of traffic servers (time slots) per radio port is assumed: therefore, the system capacity is hard-limited by the equipment availability, and the DCA efficiency is compared mainly through the signal-to-interference ratio in both the uplink and downlink directions.>

A low-overhead burst coherent demodulation method that jointly estimates symbol timing and carrier frequency offset and then performs diversity selection is studied. It coherently demodulates individual bursts of TDMA (time division multiple access) symbols by operating solely on random data within the burst without requiring training sequences. Its performance is robust against frequency offset between transmitter and receiver, thereby eliminating the need for a highly stable frequency reference. The performance of this demodulation method in a fading channel can be further improved by using a diversity selection technique based on a quality measure derived as part of the joint timing/frequency offset estimation process. Simulations and experiments have confirmed that two-branch diversity using this method can provide reliable speech communication using TDMA with a transmission rate of 450 kb/s for a portable radio channel with an RMS delay spread of 555 ns or less.>

The advantage of coherent detection over differential detection for quadrature phase-shift keying (QPSK) and binary phase-shift keying (BPSK) in a flat-fading channel is evaluated based on receiver-sensitivity analysis using three performance criteria. QPSK is concentrated upon to perform simulations for a frequency-selective fading channel. Spectral efficiency improvement due to better frequency reuse for the more sensitive coherent detection is evaluated with a combination of analysis and simulation. It is found that coherent detection with perfect carrier recovery is superior to differential detection by at least 2.4 dB, which converts to about a 30% improvement in spectral efficiency in a two-dimensional, frequency reuse system with a fourth-power propagation exponent. This advantage is preserved even if diversity selection is used to mitigate fading.>

Simulation and experimental results for a hardware implementation of a low-overhead digital coherent burst demodulator for time-division multiple-access (TDMA) radio systems using short bursts are analyzed. This implementation is based on digitizing a receiving signal after a low-frequency IF amplifier and performing all demodulation functions using digital signal processing techniques. Demodulation with very low overhead for TDMA is made possible by storing a burst in memory. A novel forward loop plus backward loop structure performs carrier recovery. Symbol-timing and carrier-frequency-offset estimations are performed by block processes, using the error signal resulting from differential demodulation. Bursts as short as 32 symbols can be demodulated without overhead. Since the demodulator is entirely digital and has the potential for low-power VLSI implementation, it is suitable for a portable radio system. Other advantages of this implementation include the availability of a channel quality measure derived from symbol timing estimation and the possibility of providing differential detection when it is desirable.>

It is shown that the degradation in the performance of wideband digital transmission systems due to intersymbol interference can be reduced by utilizing diversity selection using coding. A BCH (31, 21, 5) double-error-correcting code and QPSK (quadrature-phase-shift-keying) modulation with coherent detection are used for this study. The probability of word error is taken to be the measure of system performance. Analytic expressions are derived for the word error probability; these expressions simplify the simulation processes. It is found that a diversity selection algorithm using coding can reduce the floor of word error probability, and hence increase the dynamic range of received power before the word error probability reaches that floor.>

Frequency-selective fading caused by multipath time delay spread degrades digital communication channels by causing intersymbol interference, thus resulting in an irreducible BER and imposing a upper limit on the data symbol rate. In this paper, a frequencyselective, slowly fading channel is studied by computer simulation. The unfiltered BPSK, QPSK, OQPSK, and MSK modulations are considered first to illustrate the physical insights and the error mechanisms. Two classes of modulation with spectral-shaping filtering are studied next to assess the tradeoff between spectral occupancy and the performance under the influence of time delay spread. The simulation is very flexible so that different channel parameters can be studied and optimized either individually or collectively. The irreducible BER averaged over fading samples with a given delay profile is used to compare different modulation/detection methods, while the cumulative distribution of short-term BER is employed to show allowable data symbol rates for given values of delay spread. It is found that both GMSK and QPSK with a raised-cosine Nyquist pulse are suitable for a TDM/TDMA digital portable communications channel.