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Objective To explore the molecular mechanisms of celecoxib-induced pancreatic cancer suppression in vivo and in vitro. Methods The anti-pancreatic cancer activities of celecoxib (0, 20, 60 and 100 μmol/L) were investigated by cell viability and migration of Panc-1 and Bxpc-3 cells in vitro. The expression of L1CAM in pancreatic cancer and adjacent tissues was compared using immunohistochemistry. The expressions of L1CAM, STAT3, p-STAT3, NF-κB, p-NF-κB were determined by western blotting, and cell invasive ability was determined by wound healing assay in L1CAM-silenced and over-expressed Panc-1and Bxpc-3 cells. Results The expression of L1CAM in pancreatic carcinoma was stronger than that in the adjacent tissues and L1CAM could increase the growth and invasion of pancreatic cancer cells. Over-expression of L1CAM activated the STAT3/NF-κB signaling pathway in Panc-1 and Bxpc-3 pancreatic cancer cells and celecoxib inhibited their viability and the expressions of STAT3, p-STAT3, NF-κB, p-NF-κB as well as full length L1CAM in a concentration dependent manner. Conclusions L1CAM was highly expressed in pancreatic cancer tissue and positively correlated with age, TNM staging and tumor differentiation. L1CAM activated the STAT/NF-κB signaling pathway and celecoxib could inhibit the activity of L1CAM, STAT3 and the NF-κB signaling pathway resulting in decreased growth and invasion of pancreatic cancer cells.

Objectives: Intrahepatic recurrence is the major cause of death among patients with hepatitis B virus (HBV)related hepatocellular carcinoma (HCC) after curative surgical resection. Several approaches have been reported to decrease the recurrence rate. The objective of our study was to compare the clinical effects of transcatheter arterial chemoembolization (TACE) combined with interferon-alpha (IFN-α) therapy on recurrence after hepatic resection in patients with HBV-related HCC with that of TACE chemotherapy alone. Methods: We retrospectively analyzed the data from 228 patients who were diagnosed with HBV-related HCC and underwent curative resection between January 2001 to December 2008. The patients were divided into TACE (n = 126) and TACE-IFN-α (n = 102) groups for postoperative chemotherapy. The TACE regimen consisted of 5-fluorouracil (5-FU), cisplatin (DDP) , and the emulsion mixed with mitomycin C (MMC) and lipiodol. The recurrence rates, disease-free survival (DFS), overall survival (OS), and risk of recurrence were evaluated. Results: The clinicopathological parameters and adverse effects were similar between the 2 groups (P > 0.05). The median OS for the TACEIFN-α group (36.3 months) was significantly longer than that of the TACE group (24.5 months, P < 0.05). The 3-and 5-year OS for the TACE-IFN-α group were significantly longer than those of the TACE group ( P < 0.05) and the recurrence rate was significantly lower ( P < 0.05). The TACE and IFN-α combination therapy, active hepatitis HBV infection, the number of tumor nodules, microvascular invasion, liver cirrhosis, and the BCLC stage were independent predictors of OS and DFS. Conclusions: The use of the TACE and IFN-α combination chemotherapy after curative hepatic resection safely and effectively improves OS and decreases recurrence in patients with HBV-related HCC who are at high risk. Our findings can serve as a guide for the selection of postoperative adjuvant chemotherapy for patients with HBV-related HCC who are at high risk of recurrence.

Simultaneous Localization and Mapping (SLAM) is a key component of mobile robot's navigation. In this paper, we present a vision-based system of mapping and localization. The system builds a map which contains 3D landmarks of environment. 3D landmarks are reconstructed based on sequential Harris Corner Features. In order to match regained features with landmarks in the map, we present a simple method, i.e. Inverse Projection. The method is relatively simple and effective. We also propose an efficient observe model that simplifies the Jacobian Matrix when apply the Extended Kalman Filter (EKF) framework. We conduct experiments in both simulation and real-time environment, and give error analysis for robot's and landmarks' position. Experiment results show that landmarks are localized accurately and robot trajectory is well estimated by matched landmarks.

This paper presents a new automatically quantitative facial features classification system for TCM spirit diagnosis based on facial features. Facial diagnosis is an important diagnostic method in TCM (Traditional Chinese Medicine) and has been used for a long time. However, this traditional diagnostic method is mainly based on observation by TCM doctors and their personal experience. To develop quantification methods for TCM is very important. First, we capture facial features from video images through some algorithms, and then we use some classification models to analyze them. Finally, IG (Information Gain) is used to analyse which kind of features has good performance. Experiment results show that our system has high classification accuracy.

Using the MPEG-4 advanced audio coder (AAC) music as an example of streaming applications, we investigate the improvement of error performance for the streaming service by link adaptation and power control techniques in an enhanced general packet radio services (EGPRS) cellular network. A low packet error rate and variability are essential in providing a short error-burst length so that error concealment techniques can be effectively applied to music packets. We study the effects of a combined link adaptation and power control scheme (referred to as the error-based scheme) for achieving a target error rate and reducing error variability. By simulation, we compare the error performance of the error-based scheme at both the EGPRS block and AAC frame level with another adaptation algorithm (referred to as the throughput-based scheme) with a goal of maximizing overall network throughput. It is found that when offered with a similar traffic load, the former scheme can provide noticeable improvement of music quality over the throughput-based scheme. To achieve a similar AAC frame error rate, our results also show that the error-based scheme can increase the link throughput over the throughput-based scheme by 66.7% in one of our examples. These results reveal that by aiming at required error targets and thus reducing error variability, the error-based scheme for link adaptation and power control are helpful in improving quality and capacity for streaming applications.

The Enhanced General Packet Radio Services is one of the proposals submitted to the IMT-2000 initiative of the ITU for third-generation wireless services. EGPRS is also the evolutionary path chosen by the Universal Wireless Communications Consortium, leading toward the convergence of GSM and IS-136 standards for their next-generation wireless systems. We discuss the feasibility of supporting integrated services, such as packet voice, Web browsing, and best-effort data, using EGPRS. We first describe the relevant capabilities offered by the currently proposed standard, and then outline additional capabilities at the radio link control/medium access control layer that are necessary to support integrated services in a spectrally efficient manner. In summary, these needed capabilities are: fast uplink access during an ongoing session; fast resource assignment for both uplink and downlink; and the ability to differentiate services at the base station subsystem. To realize these capabilities, we propose enhancements to the RLC/MAC layer. These enhancements include a new set of packet control channels (namely, a fast packet access channel in the uplink, and a fast packet access grant channel and a fast packet polling channel in the downlink); uplink access protocols that utilize these control channels; and a modified fast uplink access channel structure. These changes would enable EGPRS to offer a wide range of services through a single packet-based network. Similar ideas are also applicable to other third-generation systems.

Personal communications services (PCS) promise to provide a variety of information exchanges among users with any type of mobility, at any time, in any place, through any available device. To achieve this ambitious goal, two of the major challenges in the system design are: (i) to provide a high-speed wireless subsystem with large capacity and acceptable quality-of-service (QoS) and (ii) to design a network architecture capable of supporting multimedia traffic and various kinds of user mobility. A time-hopping spread-spectrum wireless communication system called ultra-wide bandwidth (UWB) radio is used to provide communications that are low power, high data rate, fade resistant, and relatively shadow free in a dense multipath environment. Receiver-signal processing of UWB radio is described, and performance of such communications systems, in terms of multiple-access capability, is estimated under ideal multiple-access channel conditions. A UWB-signal propagation experiment is performed using the bandwidth in excess of 1 GHz in a typical modern office building in order to characterize the UWB-signal propagation channel. The experimental results demonstrate the feasibility of the UWB radio and its robustness in a dense multipath environment. A ATM network is used as the backbone network due to its high bandwidth, fast switching capability, flexibility, and well-developed infrastructure. To minimize the impact caused by user mobility on the system performance, a hierarchical network-control architecture is postulated. A wireless virtual circuit (WVC) concept is proposed to improve the transmission efficiency and simplify the network control in the wireless subsystem. The key advantage of this network architecture and WVC concept is that the handoff can be done locally most of the time, due to the localized behavior of PCS users.

We propose a novel intercell interference management technique, called quasi-static resource allocation with interference avoidance (QRA-IA), for fixed broadband wireless systems with narrow sectors or equipped with fixed or adaptive array antennas. The basic idea of QRA-IA is for every base to periodically turn off each of its beams (sectors) for a certain amount of time. This periodic turn off introduces a predictable nonuniformity in a terminal's performance and therefore permits each terminal to identify a preferred time period for transmission. QRA-IA requires that each terminal sense the interference over time, select the preferred transmission periods, and report these to the base. The resource allocation algorithm at the base can then use this information for scheduling transmissions to the terminals. The base can also use this information for obtaining an appropriate beam-off sequence. A graph theory model is used to show that an acceptable beam-off sequence exists for each base. A simple, distributed, and measurement-based beam-switching algorithm is designed for the system to find these sequences autonomously. We demonstrate via simulations that a system with QRA-IA provides a marked improvement in the packet error rate performance of terminals over the same system without QRA-IA. This improvement translates directly to improved coverage and throughput and also reduces the burden on higher layer protocols to ensure fairness and quality of service.

We consider use of fixed broadband wireless networks to provide packet services for telecommuting and Internet access. Each cell is divided into multiple sectors, each of them served by a sector antenna colocated with the base station (BS), and user terminals also use directional antennas mounted on the rooftops of homes or small offices and pointed to their respective BS antennas. To support a target data rate of 10 Mb/s, a bandwidth of several MHz is required. Since radio spectrum is expensive, the bandwidth needs to be reused very aggressively. Thus, efficient strategies for frequency reuse and managing cochannel interference are critically important. We propose several algorithms for dynamic radio-resource allocation in the fixed wireless networks. In particular, a method to be referred to as the staggered resource allocation (SRA) method uses a distributed scheduling algorithm to avoid major sources of interference while allowing concurrent packet transmission and meeting signal-to-interference objectives. The performance of the method is studied by analytic approximations and detailed simulation. Our results show that the combination of directional antennas plus the SRA method is highly effective in controlling cochannel interference. For reasonable system parameters, the SRA method delivers a throughput in excess of 30% per sector while permitting a given frequency band to be reused in every sector of every cell. It also provides satisfactory probability of successful packet transmission. In addition, a simple control mechanism can be applied in the method to improve performance for harsh radio environments.

This paper presents a unified performance model for an integrated voice/data wireless system using reservation type multiple-access (MA) schemes. It is observed that although these schemes are different in the frame structure and resource assignment procedure, all of them can be described by several common state variables whose evolvement exhibits the Markovian property. Based on this observation, a general Markovian model is developed in this paper. Three performance measures, namely, voice-packet-loss probability, data throughput, and data delay are defined. As a special case, the performance evaluation model for the voice-only system is also presented. Numerical results are given and verified by simulation under both voice-only and integrated scenarios using packet-reservation MA (PRMA), dynamic time-division multiple access (D-TDMA), and resource auction multiple access (RAMA) as examples. It is found that our analytical model is quite accurate, especially in the region of interest. The impact of system parameters (such as the voice-permission probability, data-retransmission probability, maximum number of voice slots per frame, etc.) on the integrated system performance is also investigated for these three example systems.

In this paper, we study the interferencemanagement and resource allocation problem in apacket-switched, fixed, broadband, Time DivisionMultiple Access (TDMA) wireless system. These systemshave the potential to provide high-speed data access to the homeand therefore are of current interest. The systemcapacity is limited by the co-channel interference,which arises from both intercell and intracelltransmissions. The intracell interference management andresource allocation, which significantly impacts thesystem performance, is the focus of this study. The goalis to design efficient resource allocation schemes to combat intracell interference so as toimprove the system capacity and service quality. Weaddress the following issues: (1) estimating the optimalcapacity of the system; (2) proposing resourceallocation schemes with reasonable complexity that areclose to optimal; and (3) proposing a new class of powercontrol schemes that are suited to intracellinterference management. Two resource allocation schemes are proposed: Max-Min Scheduling Protocol(MMSP) and Revisited Max-Min Scheduling Protocol(RMMSP). We show via simulations that these schemescombined with the new power control provide close tooptimal performance.

In Traditional Chinese Medicine TCM, lip diagnosis is an important diagnostic method to judge whether a person is healthy or not. Lip images can reflect the physical conditions of organs in the body. Lip diagnosis has a long history in China and the lips are analyzed by experienced doctors with their nude eyes. This method is not objective and efficient especially in the condition of handling many images. Developing an automatic way to split lips from an image is an important and necessary step. What's more, lip segmentation can provide improvement in the areas of speech recognition and speaker authentication. To segment lips and facial complexions, many methods are proposed which are based on color spaces such as RGB, HSV, Lab, etc. Other methods are based on different models such as snake, geometry model, etc. This paper proposes a lip segmentation method based on facial complexion template. A facial complexion template can be constructed when the face is detected. We construct the facial complexion template using Hue channel and Saturation channel of color information. By removing the skin similar to facial complexion template values an initial lip image can be got. Finally, by smoothing the lip contour an optimized lip segmentation result can be obtained.

This paper proposes a multiple access scheme for the forthcoming wireless ATM(Asynchronous Transfer Mode) system. Such ATM compatible wireless systems are motivated by the rapidly increasing demand for wireless extensions to broadband networks, which are expected to support mixed broadband services including Constant Bit Rate (CBR), Variable Bit Rate (VBR), and Available Bit Rate (ABR) traffic. Since these different traffics have very different performance requirements, the multiple access scheme design is very challenging. In this paper, we propose a multiple access scheme called Dynamic Time Division Multiple Access with Piggybacked Reservation (DTDMA/PR), attempting to achieve higher statistical multiplexing efficiency in the mixed VBR/CBR/ABRtraffic scenario. The basic idea is to exploit two levels of reservation. The first level deals with the isochronous nature of CBR and VBR traffic and the bursty nature of ABR traffic by using the ALOHA reservation procedure. The second level exploits the piggybacked reservation approach to cope with the dynamic feature of VBRtraffic in order to increase the multiplexing efficiency. An analytical model is also developed in this paper and verified by simulation. Numerical examples are given to gain some insight into the protocol itself.

To improve the spectrum efficiency of integrated voice and data services in Personal Communication System (PCS), several reservation-type multiple access schemes, such as Packet Reservation Multiple Access (PRMA), Dynamic Time Division Multiple Access (D-TDMA), Resource Auction Multiple Access (RAMA), etc., have been proposed. PRMA uses the data packet itself to make a channel reservation, and is inefficient in that each unsuccessful reservation wastes one slot. However, it does not have a fixed reservation overhead and offers shorter access delay. On the other hand, fixed reservation overhead is unavoidable in both RAMA and D-TDMA. Compared to D-TDMA and PRMA, RAMA is superior in the sense that its slot assignment is independent of the traffic load. But its implementation is difficult. With these observations, a new reservation protocol, called Dynamic Reservation Multiple Access (DRMA), is proposed in this paper. With this new protocol, the success probability of channel access is greatly improved at the expense of slightly increased system complexity. It solves the problem of inefficiency in PRMA, but without introducing the fixed reservation overhead as in D-TDMA and RAMA. In addition, it is more suited to the dynamic behavior of the integrated traffic because there is no fixed boundary between voice and data slots (which is mandatory in D-TDMA and RAMA). Our numerical results indicate that its performance is superior to the existing reservation protocols, especially in the integrated traffic scenario. Moreover, the soft capacity feature is exhibited when the traffic load increases.

An interactive video-on-demand (VoD) system allows users to access video services, such as movies, electronic encyclopedia, interactive games, and educational videos from video servers on a broadband network. This paper develops a performance evaluation tool for the system design. In particular, a user activity model is developed to describe the usage of system resources, i.e., network bandwidth and video server usage, by a user as it interacts with the service. In addition, we allow batching of user requests, and the effect of such batching is captured in a batching model. Our proposed queueing model integrates both the user activity and the batching model. This model can be used to determine the requirements of network bandwidth and video server and, hence, the trade-off in communication and storage costs for different system resource configurations.

Personal communication systems (PCS) represent a rapidly growing and increasingly important segment of the telecommunication industry. The goal of PCS is to provide truly personal, cost-efficient communication services to users through portable handsets. In this paper, we present a survey on the research and development in PCS, emphasizing several important aspects such as the PCS concept, service requirements, system architecture, operation, and management. Some ongoing field trials are described as well. We focus on the wireless and the mobility-related features of PCS, discuss their impact on the system design and performance, and provide an overview of different technology choices. >

We propose a time slot reuse partitioning (TSRP) scheme for resource assignment in packet switched, fixed, wireless networks, especially for applications where it is necessary to provide quality of service (QoS) guarantees to subscribers. The TSRP scheme is based upon two key ideas. The first one is for the system to have more than one coexisting reuse pattern in the time domain. The second is to match a terminal or application to an appropriate reuse pattern, so as to guarantee the QoS. This technique seems particularly suited to a fixed wireless system. We define and then simulate a specific implementation of a TSRP scheme for an integrated voice and broadband data system. The results show that this scheme provides good performance for both voice and data.

We consider use of fixed broadband wireless networks to provide packet services for telecommuting and Internet access. Each cell in such networks is divided into multiple sectors, each of them served by a sector antenna co-located with the base station (BS), and user terminals also use directional antennas mounted on the roof top of homes or small offices and pointed to their respective BS antennas. To support a target data rate of 10 Mb/s, a bandwidth of several MHz is required. Since radio spectrum is expensive, the bandwidth need to be reused very aggressively. Thus, efficient strategies for frequency reuse and managing co-channel interference are critically important. We propose here a dynamic radio-resource allocation method, to be referred to as staggered resource allocation (SRA) method, that uses a distributed scheduling algorithm to avoid major sources of interference, while allowing concurrent packet transmission and meeting a specified signal-to-interference objective. The performance of the method is studied by analytic approximations and detailed simulation.

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.

The use of microcells is an attractive option for providing high data rate access for fixed and low mobility users. We propose and analyze a wireless backhaul network for connecting microcells to the wireline network. In the proposed network, several microcellular radio ports are connected to a so called hub through radio links. These hubs are then connected to the wireline network using cable or fiber. For ease of deployment, we propose that omnidirectional antennas be used at the radio ports and sectored antennas be used at the hubs. Our results show that for 95% of the links, a link reliability of 99% may be achieved with an effective frequency reuse of 3 by using site-selection, flexible channel assignment, power control and two-branch minimum mean square error combining. For the remaining 5% of the more vulnerable links, a direct wireline connection may be provided. We also find that power control plays a critical role in reducing the variation of the link performance over time. In addition, the best performance is achieved by jointly optimizing the receiver, i.e., the diversity combining weights, and the transmit power settings. We show that this joint optimization may be implemented in a distributed manner, making it attractive for use in real systems.

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.

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.

This paper describes a broadband wireless data access system using a microcellular infrastructure. This system can be envisioned as an overlay data system which shares the same network infrastructure as a microcellular voice system, such as the Personal Access Communications System (PACS). Unlike microcellular voice systems, however, the proposed system provides a high peak data rate of 2 Mbps, can operate with a group frequency reuse factor of one, only requires handoffs between the group areas of sixteen radio ports, and can start with a minimum of 1 MHz spectrum in each direction. We quantify the performance of the proposed system under different air interface assumptions, each of which has a different required complexity and achievable performance. The proposed system can provide adequate performance (i.e., 1% packet error rate) with only the use of either half rate channel coding or dynamic time slot assignment, for a 95% target coverage. Further performance improvement is possible with the use of SINR-based feedback power control.

In this paper, we propose an inter-cell interference management technique, called dynamic resource allocation with interference avoidance (DRA-IA), for fixed broadband wireless systems with narrow sectors or equipped with fixed or adaptive array antennas. The basic idea of DRA-IA is for every base to periodically "turn-off" each of its beams (sectors) for a certain amount of time. This periodic turn-off introduces a predictable non-uniformity in a terminal's performance, and therefore permits each terminal to identify a preferred time period for transmission. DRA-IA requires and that each user sense the interference over time, select the preferred transmission periods, and report these to the base. The resource allocation algorithm at the base can then use this information for obtaining the appropriate "beam-turn-off" sequence and for scheduling transmissions. We demonstrate that DRA-IA provides a marked improvement in the packet error rate performance of "disadvantaged" terminals. This improvement translates directly to improved coverage and throughput, and also reduces the burden on higher layer protocols to ensure fairness and quality of service.

We investigate the interactions between the transmission control protocol (TCP) and radio link protocols (RLP) of various reliability, when used in a spectrally efficient microcellular system similar to the EDGE cellular system. The study is done through detailed downlink simulations of a large cellular system with numerous simultaneous users. Either fixed-load or WWW-like packet traffic is generated for each link, and RENO TCP is run. The results show that the choice of RLP retransmission parameters impacts the throughput loss and application packet delay significantly, and that the range of 3-10 RLP block retransmissions (60-200ms persistency) offers the smallest performance degradation caused by TCP/RLP interactions. We conclude that variable RLP persistency based on the knowledge of radio conditions for individual users, is highly desirable.

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.

Traditionally, packet scheduling has been considered to be the primary technique for providing quality of service in communication networks. In a previous paper (see IEEE Globecom '00 Proc., vol.3, p.1515-21, Nov. 2000) we explored different packet scheduling techniques for providing differentiated services (premium vs. basic service offerings) in the 3G EGPRS (enhanced general packet radio services) system. A premium service is defined here as one with a higher priority, and therefore resulting in a better user experience, than a basic service. This paper continues the investigation. In particular, we assess algorithms that allocate multiple time slots for premium users. The results show that multi-time slot assignment for premium users is very effective in supporting differentiated service offerings. It provides a better user experience to premium users with little performance degradation to the basic users.

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 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.

This paper reports on studies of the effects of the TCP/IP protocol header on the system performance of a spectrally efficient cellular system. Integrated cellular system and TCP-protocol simulations are used to examine the degradation in the effective data throughput and packet delay due to the header overhead. Performance improvements achievable by header compression are investigated. Simulations have been designed for an EDGE-like cellular system. System-wide performance is examined using cumulative distributions for data throughput, delay and packet success. For example, the data throughput of a micro-cellular system with Web traffic is improved about 10% when header compression from 40 to 2 bytes is implemented. More aggressive compression yielded minimal further improvements.

This paper discusses the feasibility of offering multiple service classes, e.g., premium and basic, in the 3G EGPRS (Enhanced General Packet Radio Services) system using different scheduling mechanisms. We investigate weighted round-robin type scheduling for differentiating the service classes together with two types of packet schedulers: radio-aware packet schedulers at the radio access network, and conventional packet schedulers at the edge of the core network. The results show that for the EGPRS-like system simulated: (i) incorporating radio link conditions and radio resource management into the scheduling can improve the overall delay performance by more than 20%; (ii) a weighted round-robin scheduling to differentiate service classes may not be suitable for an interference limited environment. Our ongoing work will consider using multiple time-slot assignments for providing differentiated services.

We investigate the feasibility of transmitting MPEG-2 AAC (Advanced Audio Coder) music over packet-switched bearers of an Enhanced General Packet Radio Services cellular network. Streaming data applications like AAC music differ from conversational voice in that they permit a larger transmission delay and delay jitter than the former. In addition, AAC music can tolerate a higher frame error rate than voice by using error concealment techniques at the receiver. This creates the opportunity to transmit AAC music over data-optimized, high-throughput and nominally high-error rate, wireless networks, by using appropriate performance enhancing techniques. In particular, we investigate the performance with the following techniques: (i) link-layer retransmissions; (ii) dynamic packet assignment; and (iii) packet interleaving or shuffling. We find that all the above techniques are desirable in order to simultaneously achieve a high spectral efficiency and the required quality. While this paper is focused on one application, the conclusions and insights are applicable to most streaming audio and video applications.

We propose an implementation of dynamic frequency hopping (DFH) for cellular systems with synchronized bases. The implementation uses network assisted resource allocation (NARA), which is based upon "real-time" inter-base signaling. We review the concept of DFH and propose a supporting network and signaling structure for DFH with NARA. We find that the resulting system complexity, in terms of the mobile measurements, over the air signaling, and inter-base signaling, is not high. Using voice traffic as an example, we find that for the system considered, DFH has twice the user capacity compared to traditional random frequency hopping schemes.

We propose a dynamic packet assignment algorithm, network-assisted dynamic packet assignment (NA-DPA), for wireless data networks with frame synchronized base stations. The basic idea of NA-DPA 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 path loss and channel assignment information in real time, each base can calculate the effect of a resource assignment on the overall system. As a result, rapid interference measurements, which are typically needed to implement dynamic channel assignment (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. It also provides an opportunity for a better optimization of the system performance, than that permitted by other DCA algorithms. We show NA-DPA can provide a throughput that is 50% higher than that obtained by other proposed DCA algorithms.

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.

Using the MPEG-4 Advanced Audio Coder (AAC) music as an example, we investigate the improvement of error performance for the streaming service by link-adaptation and power-control techniques in an Enhanced General Packet Radio Services (EGPRS) cellular network. In particular, we study the effects of a combined link adaptation and power control scheme (referred to as the error-based scheme) for achieving a target error rate, which is non-zero but low enough so that error concealment techniques can be effective. By simulation, we compare the error performance of the error-based scheme at both the EGPRS block and AAC frame level with another adaptation algorithm with a goal of maximizing overall network throughput (referred to as the throughput-based scheme). Our results reveal that when offered with similar traffic load, the former scheme can provide noticeable improvement of music quality over the throughput-based scheme. To achieve similar AAC frame error rate, our results also show that the error-based scheme can increase the link throughput over the throughput-based scheme by 66.7% in one of our examples. These results reveal that link adaptation and power control to achieve required error targets are helpful in improving quality and capacity for streaming data applications.

Wireless broadband networking imposes significant challenges for frequency reuse. This paper introduces methods for improving link adaptation, which is essential for achieving high throughput in TDMA wireless packet communications. We present simulation results based on the EDGE (Enhanced Data-rate for GSM Evolution) system with low reuse factor (1/3) but the techniques are applicable for any packet-based frequency-reuse wireless systems. Our proposed enhancement, termed "mode 0," is a non-transmission mode in addition to the regular transmission modes available in link adaptation. Further enhancement is achieved by using multi-slot assignment with mode 0 as the back off mechanism to control system loading with an acceptable interference level. Simulation results indicate that this simple enhancement, which requires no standard modifications, results in improved throughput even for the disadvantaged users.

We investigate the ability of multiple descriptions (MD) and layered coding to improve the quality of video transmitted over EGPRS networks. One-layer video sent over a single channel on such a network has a fairly sharp quality transition, depending on a user's location. Either the video can be transmitted reliably (if the video rate is less than or equal to what the channel can sustain), or it is subjected to many lost packets. In this system, MD and layered video may offer two ways to improve the video quality beyond that of the one-layer video. First, each sub-stream can be sent on a separate channel, essentially doubling the assigned bandwidth and increasing the video quality. Second, MD and layered video are more error resilient than one-layer video, potentially improving the video quality seen by users in poor locations. We find that for the system scenarios considered, one and two-layer coding outperform MD coding, depending upon the number of wireless channels used for the video transport.

We investigate the transmission of H.263-based streaming video over packet-switched bearers of an enhanced general packet radio services (EGPRS) cellular network. We assume a limited amount of interaction between the mobile video client and the server, consisting of video rate and resilience selection at the beginning of each session. We find that there is a large variability of perceived video quality among users. We also find that the following features are important to improving the perceived video quality: (i) using dynamic channel assignment and link-layer retransmissions in EGPRS; and (ii) choosing the video rate and resilience on a user-by-user basis. We propose a simple algorithm for the rate selection, and find that this is quite effective.