Legacy congestion controls including TCP and its variants are known to perform poorly over cellular networks due to highly variable capacities over short time scales, self-inflicted packet delays, and packet losses unrelated to congestion. To cope with these challenges, we present Verus, an end-to-end congestion control protocol that uses delay measurements to react quickly to the capacity changes in cellular networks without explicitly attempting to predict the cellular channel dynamics. The key idea of Verus is to continuously learn a delay profile that captures the relationship between end-to-end packet delay and outstanding window size over short epochs and uses this relationship to increment or decrement the window size based on the observed short-term packet delay variations. While the delay-based control is primarily for congestion avoidance, Verus uses standard TCP features including multiplicative decrease upon packet loss and slow start. Through a combination of simulations, empirical evaluations using cellular network traces, and real-world evaluations against standard TCP flavors and state of the art protocols like Sprout, we show that Verus outperforms these protocols in cellular channels. In comparison to TCP Cubic, Verus achieves an order of magnitude (> 10x) reduction in delay over 3G and LTE networks while achieving comparable throughput (sometimes marginally higher). In comparison to Sprout, Verus achieves up to 30% higher throughput in rapidly changing cellular networks.

Content Centric Networking (CCN) is a new paradigm in networking and a future Internet architecture. Performance evaluations show that conventional CCN forwarding strategies which use replication of Interests (standard) or the shortest path (best-face) do not perform well under high bandwidth requirements and loaded networks. We have designed and evaluated the performance of an On-demand Multi-Path Interest Forwarding (OMP-IF) strategy which identifies a set of paths based on the disjointness of paths to content locations. Then, the discovered paths are used simultaneously to distribute (split) Interests based on the characteristics of the paths. We have evaluated OMP-IF strategy using a simulator with a large scale network scenario and a realistic traffic generation model. The results show improved performance in CCN networks considering download time, load balancing, content hit ratios, and others.

With the onset of multiple wireless technologies, the end-host devices of today are multi-homed. This has led to research in the simultaneous use of multiple paths between the multi-homed end devices. New multipath transport protocols such as Multipath TCP and Concurrent Multipath Transfer SCTP need to be fair to the existing transport protocols like TCP and SCTP in a congested network. One way of being fair is to couple the congestion control mechanisms of the protocol over its multiple utilized paths. This paper illustrates and compares the solutions that have been proposed to ascertain that Multipath TCP can yield an improved performance while being fair to standard TCP. In addition, this paper highlights improvements to one of the proposed solution - Dynamic Window Coupling. It is shown that the new solution performs best in most of the different scenarios of bottlenecks in a network with a higher preference given to being friendlier to TCP.

There has been a lot of research focusing on optimizing the Long Term Evolution (LTE) networks performance. A large part of this research focused on how to optimize the LTE radio interface to efficiently distribute the scarce radio resources among the LTE users. Since LTE offers a diverse amount of services, the Quality of Service (QoS) differentiation is a major challenge. QoS is handled in LTE at different levels; the most important one is the radio scheduler. The QoS aware radio scheduler is designed to guarantee different service requirements. However, if the LTE radio interface is fully optimized towards QoS guarantees, what happen when the LTE transport network runs into congestion? Congestion can happen at any point in the network, and normally congested LTE backhaul tends to create large unfairness among the users, even among the ones with the same QoS class. The LTE transport congestion can occur at any router in the backhaul network. In this paper, we present two novel algorithms, which focus on the LTE transport congestion. The first algorithm is a congestion detection mechanism (CD) that identifies the transport network congestion as early as possible, and triggers the congestion control function. The second algorithm preserves and optimizes the QoS performance of each user over the transport to match the radio scheduler service differentiation, i.e., achieving fairness among the users in the same QoS demand and guaranteeing the enforced radio QoS weights for different service classes.

QoS/QoE becomes increasingly important due to the growing usage of multimedia applications. In this paper, a solution is presented where Multipath TCP which supports the mobility on the transport layer is combined with QoS-enabled OSPF for performance-aware route selection. For the latter protocol, a novel extension called QoSxOSPF-LSFM (Load Sensitive Flow Management) is introduced which increases the stability of OSPFxQoS routing by evaluating the link utilization and reducing the frequency of route changes. For comparison, results are shown for the combination of QoSxOSPF-LSFM both with MPTCP as a multipath-capable transport-layer based approach as well as with Mobile IP which is a network layer based mobility approach. Simulation results prove the increase of performance compared to the case without LSFM extensions both for MPTCP and Mobile IP; however, due to the multipath capabilities, MPTCP in all cases outperforms Mobile IP.

Modern mobile devices comprise multiple interfaces for heterogeneous network technologies. However, currently implemented mechanisms to decide which one(s) to use and distribute application flows accordingly (i.e., solving the multihomed flow management problem, MFM) are rather coarse and do not leverage the opportunities. A user-centric quality-aware (QA) MFM has been proposed which optimises network use based on user-perceivable metrics such as application quality as well as energy and monetary costs. This paper refines this approach by providing a single method for both real-time and elastic (i.e., TCP-based) traffic, and uses realistic available capacity estimation. We evaluate this method in OPNET-simulated LTE and WLAN mobile networks. We study the impact of methods used to trigger the decision algorithm, and investigate the influence of an increasing number of users employing the QA-MFM technique on both the user-perceivable metrics and the global network performance. We find that on-demand triggering performs better than a static periodic method. We also demonstrate that the proposed approach out-performs classical network selection techniques in terms of application quality. We also show that the QA-MFM is not too greedy as to not scale with a number of users, and has a positive effect on the network loads, by preemptively adapting applications parameters to match network conditions.

Multipath transport protocols such as Multipath TCP can concurrently use several subflows to transmit a TCP flow over potentially different paths. Since more than one subflow is used, an efficient multipath scheduling algorithm is needed at the sender. The objective of the scheduler is to identify the subflow over which the current data packet should be sent. This paper compares the most important types of schedulers for multipath transfers. We model their performance analytically and derive key metrics, most notably the resulting end-to-end delay over heterogeneous paths. Our results show that a scheduler minimizing the packet delivery delay yields the best overall performance, but it is complex to realize. An alternative scheduler based on the sender queue size is simpler and has sufficient performance for relatively small asymmetry between the multiple paths. Our model results are confirmed by measurements with a real multipath transport protocol.

Long Term Evolution (LTE) uses Single Carrier Frequency Division Multiple Access (SC-FDMA) as the uplink transmission scheme. The Quality of Service (QoS) provision is one of the primary objectives of the wireless network operators. In this paper, the end-to-end QoS performance of Bandwidth and QoS Aware (BQA) scheduler for LTE uplink is evaluated in heterogeneous traffic environment. The BQA scheduler is designed to provide efficient allocation of radio resources to users according to the QoS requirements of various traffic classes and the instantaneous channel conditions. The user QoS provision is ensured by using dynamic QoS weights. Additionally, the delay sensitive traffic is facilitated by employing delay thresholds. The BQA scheduler algorithm supports multi-bearer users. The end-to-end QoS performance of the scheduler is analyzed in several simulation scenarios. The results show that the proposed scheduler guarantees provision of QoS to users.

Mobile Network Virtualization (NV) is an emerging technique which has drawn increasingly research attention. Network Virtualization enables multiple network operators to share a common infrastructure (including core network, transport network and access network) so as to reduce the investment capital while improving the overall performance at the same time. This is achieved by exploring the multiplexing gain. Similarly, Load Balancing (LB) is a well-known mechanism used in mobile networks to offload excessive traffic from high-load cells (hot spots) to low-load ones within one network operator. This paper aims at investigating the potential gain of applying NV in LTE (Long Term Evolution) networks and compares it with the LB scheme gain. In this paper, we propose an LTE virtualization framework (that enables spectrum sharing) and a dynamic load balancing scheme for multi-eNB and multi-VO (Virtual Operator) systems. We compare the performance gain of both schemes for different applications, e.g. VoIP, video, HTTP and FTP. We also investigate the parameterization of both schemes, e.g. sharing intervals, LB intervals and safety margins, in order to find the optimal parameter settings. The presented results show that the LTE networks can benefit from both NV and LB techniques.

Due to the coexistence of different generations of mobile radio technologies, more and more mobile network operators are keen on deploying multiple radio technologies on a single cell site and carry their traffic over a common transport network. This paper studies such a collocated multiradio system: collocated LTE and HSDPA. Firstly, we investigate the potential gain by using a shared transport network to carry different radio technologies. Furthermore, this paper proposes analytical models to dimension the shared transport network for elastic traffic (TCP-based traffic) from the different radio technologies, by taking traffic characteristics, QoS requirements of each radio technology and the deployed transport QoS function and packet scheduling into considerations. The proposal for modeling the shared transport network is to apply the M/G/R-PS model per QoS class while taking the potential multiplexing gain of bandwidth sharing among the different QoS classes into account. For validating the analytical models, a system simulation model is developed. The analytical results derived from the proposed dimensioning models are validated by comparing with the simulation results for various traffic and network scenarios. Presented results demonstrate that proposed analytical models are well suitable for dimensioning of the shared transport network. With the help of the proposed analytical models, we analyze the potential sharing gain of such a shared transport system under different traffic loads and mix of different radio technologies.

Choosing an adequate Wireless Sensor Network Platform for the intended application domain is of essential importance for successful deployments. Evaluating off-the-shelf WSN platforms by their datasheets can limit the available choices. The real performance however, cannot be judged by just evaluating datasheets. Measurements for crucial performance measures, like serial and radio performance, are necessary. In this work four available platforms have been evaluated. The platforms are compared among each other and to their respective datasheets. Several measurements show significant differences from the advertised performance numbers in the datasheet that can influence performance and successful operation of the protocol stack.

This paper proposes an analytical model for dimensioning transport bandwidths in the Long Term Evolution (LTE) access network. In this work the criterion for dimensioning is the transport network delay QoS (at the packet level). The presented analytical model takes into considerations the key features of the LTE radio interface and the use of Differentiated Service (DiffServ) QoS scheme in the LTE access transport network. Furthermore, the proposed dimensioning approach can be performed for a single transport link as well as for the entire transport network. For validating the analytical dimensioning models, a developed LTE system simulation model is used. The presented results demonstrate that the proposed analytical models can appropriately estimate the transport network delays for different QoS priorities and hence can be used for bandwidth dimensioning for various traffic and network scenarios.

High Speed Downlink Packet Access (HSDPA) is an extension of the Universal Mobile Telecommunication System (UMTS) technology of 3GPP Rel-99, with the objective of increasing the data rate and also reducing the latency in the downlink. The focus of this study is to develop an analytical model for HSDPA flow control (FC) and congestion control (CC) performance analysis. HSDPA flow control adapts transport network data flow to the time varying radio channel capacity of a MAC-d flow [4, 21, 22]. Often the Transport Network Layer (TNL) is congested when there is no sufficient transport capacity to fulfil the radio capacity demand for all connections giving rise to the degradation of the overall HSDPA performance [5, 17, 23, 22]. There are several simulation based investigations carried out so far for the performance analysis of flow control and congestion control algorithms [5, 17, 19, 22, 23]. Further an analytical model which is based on the Embedded Markov Chain with Multiple Departures (EMC- MD) was developed to analyse the effect of CC for the HSDPA performance [18]. However it does not include the flow control functionality which works in cascade with the congestion control algorithm. In order to overcome this issue a joint Markov model is developed which is the focus of this paper. The paper provides detailed modelling aspects of above two algorithms and also the comparison of results with simulation results. There is close agreement with analytical and simulation results.

The MAC scheduler is an important and crucial entity of the Long Term Evolution (LTE) and is responsible for efficiently allocating the radio resources among mobile users who have different QoS demands. The scheduler takes different considerations into account such as throughput and fairness when deciding the allocation of the scarce radio resources. LTE is an all IP packet system in which guaranteeing QoS is a real challenge. Therefore the LTE MAC scheduler should consider not only the throughput optimization but also the QoS differentiations in an effective manner. In this paper, we propose a novel LTE downlink MAC scheduling algorithm. The proposed scheduler differentiates between the different QoS classes and their requirements. Two different QoS classifications are considered: Guaranteed Bit Rate (GBR) and non Guaranteed Bit Rate (non-GBR). The proposed scheduler also considers the different users channel conditions and tries to create a balance between the QoS guarantees and the multi-user diversity in a proportional fair manner. The simulation analysis confirms that guaranteeing the different QoS requirements is possible.

As wireless IP-based networks become more popular and larger in size and coverage, simple mobility management protocols cannot deal with the demands. With the growing access to the Internet, mobility can be divided into macro/micro-mobility domains. Macro-mobility can be efficiently handled with Mobile IP and its derivatives but they do not scale that well for the micro-mobility case. An alternative for micro-mobility domains is to use a reactive route discovery mechanism. Ad-hoc routing protocols such as DSR and AODV are the well known protocols that can be used for this purpose. The disadvantage of the reactive approach is that the route discovery process becomes costly in terms of control overhead and latency of the connection setup when done frequently. In this paper, an optimization of the simple AODV route discovery is proposed, introducing a paging extension. The proposed mechanism is somewhat similar to other paging schemes like Cellular IP, but introduces modifications and enhancements for reducing the paging control overhead. Simulation results of AODV and the enhanced route discovery with paging are presented along with the performance evaluation of both schemes. The simulations were performed with the QualNet Network Simulator.

This paper proposes efficient analytical models to dimension the required transport bandwidths for the Long Term Evolution (LTE) access network for the elastic Internet traffic (which is carried by the TCP protocol). The dimensioning models are based on the use of Processor Sharing queuing theory to guarantee a desired end-to-end application QoS target. For validating the analytical dimensioning models, a developed LTE system simulation model is used. Extensive simulations are performed with various traffic and network scenarios. The analytical results derived from the proposed dimensioning models are compared against the simulation results. The presented results demonstrate that the proposed analytical models can appropriately estimate the application performances of different QoS priorities and thus be used for the link dimensioning for various traffic and network scenarios.

Due to the high scarcity and high costs of radio spectrum, more and more radio services are occupying unlicensed bands for their operation. Due to this, there is a high risk of destructive interference which degrades the performance and fails to support Quality of Service (QoS) for systems operating in these bands. IEEE 802.11 based wireless networks are already operating in unlicensed band. A new competitor for unlicensed bands is the IEEE 802.16 based wireless metropolitan area network. Therefore, spectrum sharing between coexisting competing wireless systems like 802.11 and 802.16 is an upcoming challenge. To understand the characteristics of interference in such a heterogeneous scenario, an analysis of possible interference is presented and the performance of the legacy systems is evaluated. Then a spectrum sharing concept is proposed which can generally be applied to both systems. In this paper, the proposed concept is adapted for coexisting 802.16 and 802.11e based systems, which is an extention of 802.11. In this case, the 802.11e Hybrid Coordination Function (HCF) Controlled Channel Access (HCCA) is extended to provide a protocol for airtime sharing. Simulation results are presented showing that the proposed algorithm provides excellent improvement of system performance in the context of capacity and channel utilization compared to the case without applying any spectrum sharing method.

Today spectrum demand for wireless communication is tremendous and even keeps growing. In contrast, there are no blank spots in the spectrum map. Additional spectrum opportunities are available in unlicensed bands and so-called non-exclusively licensed bands which are currently made available. This non-exclusive spectrum can be exploited whenever a license holder is not active. These spectrum opportunities cannot be exclusively accessed so that multiple systems need to coexist. Existing MAC protocols need to be enhanced to allow a system to meet own Quality of Service (QoS) demands while allowing other coexisting systems to meet their QoS requirements. In this work, methods for capacity calculation known from cellular networks are extended and applied to a scenario of coexisting IEEE 802.16 (WiMAX) systems. Developed methods are used to evaluate enhancements for interference mitigation which lead to increased capacity.

Wireless Sensor Networks (WSNs) have been used mainly to collect environmental data and send it to a base station. Routing protocols are needed to efficiently direct the information to the base station. Most of the available routing protocols are designed for conventional WSN applications, where small volume data transmission is required and low power consumption is the fundamental. Support to mobility is generally out of interest, since in most scenarios sensor nodes are static after being deployed. However, in fire fighting rescue scenarios, WSN routing protocols are expected to be able to adapt to change of network topology (caused by nodes failure and fire fighters' movement) rapidly. Moreover, routing protocols should be capable to transmit relatively large volume data (e.g. short voice message) without losing efficiency and reliability. In this paper we propose a broadcasting based communication protocol, which is dedicated to the fire fighting rescue scenarios under the framework of a European project wearIT@work. This protocol is applied to building up a virtual lifeline for fire fighters. The protocol is evaluated through experimental tests and simulations using variant scenarios. Performance metrics such as packet transmission time, jitter and packet loss rate are considered. Results show that our broadcasting based communication protocol is able to fulfill the requirements and is more suitable to fire fighting rescue scenarios.

This paper presents dimensioning of an IP-based UMTS Terrestrial Radio Access Network (UTRAN) with multiple Node Bs connected to one RNC, where the applied QoS architecture is based on Differentiated Service (DiffServ) with an integrated Weighted Fair Queue (WFQ) and Strict Priority (SP) scheduling. To provide an appropriate network dimensioning for guaranteeing a desired end-to-end QoS, a general analytical approach is proposed in this paper to dimension the individual links of the Iub interface (between the Node B and the RNC) and furthermore to derive over-booking for the backbone link for elastic traffic. The analytical approach is validated by simulations. Furthermore, based on the proposed analytical approach, important dimensioning rules are summarized.

The Proxy Mobile IPv6 (PMIPv6) is a network based mobility management protocol standard that was ratified recently by the Network-based Localized Mobility Management (NetLMM) working group of the Internet Engineering Task Force (IETF). PMIPv6 is a protocol that uses the same concepts as used in Mobile IPv6 (MIPv6), but modified to operate in the network part only instead of involving the Mobile Node (MN) as well. PMPv6 is claimed to posses a number of advantages over the host based mobility management protocols in use today, above all MIPv6. The main advantage of using PMIPv6 is the freeing up of the mobile host in doing any mobility related activities and thereby saving its resources. The saving of resources may result in their usage for other purposes or even enable otherwise capabilities restricted devices to operate in the PMIPv6 domains. Other advantages include reduced signaling traffic volume and no tunneled packets in the access network. These aspects become very important since the access networks in mobile networks usually are air interfaces. Further, PMIPv6 is also becoming a very attractive mobility management protocol for mobile network operators as seen by its inclusion in current 3rdGeneration Partnership Project (3GPP) standardization as a possible alternative mobility management protocol for the Long Term Evolution (LTE) technologies. In addition to qualitative analyses and comparisons, the work attempts to quantify these advantages to show the achieved benefits. The quantifications are done through measurements in a real test-bed which is installed with a PMIPv6 implementation developed as part of this work.

High Speed Uplink Packet Access (HSUPA) is an extension of the Universal Mobile Telecommunications System (UMTS) technology of 3GPP Rel-99, with the objective increasing the data rate (up to 5.76 Mbps) and reducing the latency of the uplink. 3GPP UMTS standard Rel-6 introduces a new "Enhanced Dedicated Channel" (E-DCH) for the uplink. The main focus of the work presented in this paper is to analyse the effect of TNL congestion control on the HSUPA performance. The congestion control function works together with the scheduler and HARQ functionality in order to control the offered load to the TNL network. In this manner, the data flow over the transport network (TNL) is adequately controlled and limited to the available TNL capacity by avoiding congestion in the transport network. The TNL congestion control consists of two independent parts: congestion detection and congestion control which have been implemented, tested and validated within the focus of this paper. The simulation results presented in this paper confirm that the congestion in the transport network can be controlled in such a way that the available TNL capacity can be effectively utilised and hence the performance of HSUPA network can be significantly improved in all aspects.

HSUPA (High Speed Uplink Packet Access) is introduced by the 3GPP Release 6 to enhance the UMTS uplink with higher data rates, reduced latency and increased capacity. This paper discusses the dimensioning of the backhaul resources on the ATM-based Iub interface in UMTS HSUPA networks. The main focuses of this paper is to (1) analyze the important factors that have influence on the transport network dimensioning and further investigate their impacts; (2) estimate the required backhaul bandwidth as a function of user QoS requirements; (3) discuss the impact when including HSDPA traffic on the downlink. In addition, this paper also provides insights of HSUPA air interface given diverse user applications as well as different number of users within one cell. The investigations and dimensioning are based on simulations.

This paper discusses the use of Carrier Ethernet for the transport of UMTS radio access network as a alternative solution for the gradual migration towards pure IP-based RAN. By means of Pseudo-Wire technique, the ATM service is emulated over the underlying Ethernet network. Within this work, the performance of such Carrier Ethernet based UTRAN is evaluated and compared to the ATM-based UTRAN of UMTS Release 99, in particular the transport efficiency, the delay and packet losses of the Iub. Another contribution of this paper is to investigate the parameter settings to provide a guideline for the optimum configurations.

High speed downlink packet access (HSDPA) is an extension of the universal mobile telecommunication system (UMTS) technology of 3GPP Rel-99, with the objective of increasing the data rate and also reduce the latency in the downlink. The focus of this study is to develop an analytical model for HSDPA congestion control (CC) performance analysis. Often the transport network layer (TNL) is congested in HSDPA networks which degrades the overall HSDPA performance. The congestion control and detection algorithms are implemented in the Node B and RNC in order to minimize system congestions. There are several simulation based investigations carried out so far for the performance analysis of such algorithms. An analytical model which is based on the Embedded Markov chain with multiple departures (EMC-MD) is developed to analyse the effect of CC for the HSDPA performance. This analytical model includes all the important functionalities of the congestion control at TNL level. This paper also shows that the results from the analytical model are in agreement with the simulation results and that the analytical model can be effectively used to analyse the effect of parameters on performance of the HSDPA network and can be used dimensioning of the parameters.

High speed downlink packet access (HSDPA) is an extension of the Universal Mobile Telecommunications System (UMTS) technology of 3GPP Rel-99, with the objective to increase the data rate and reduce the latency in the downlink. The main focus of the work presented is to analyse the effect of congestion at the Iub interface on the HSDPA performance. The data flows should be adequately controlled in order to avoid congestion in the transport network. The 3GPP (3rd Generation Partnership Project) Rel. 5 specifications highlight two congestion detection mechanisms which are based on the frame sequence number (FSN) and the delay reference time (DRT) fields of HSDPA data frame. In addition to these, a third congestion detection mechanism based of Checksum of HSDPA data frame is considered. This paper discusses a congestion control scheme deploying all three congestion detection methods. It is shown, that a congestion control algorithm can effectively work using these congestion detection triggers and can control the offered load to the transport network. The simulation results presented in this paper confirm that the congestion in the transport network can be avoided, and hence the performance of HSDPA network can be significantly improved in all aspects.

The use of relay stations in the cellular network is a good candidate method to enhance the cell edge data rate, and to provide homogeneous QoS experience for users in different locations. We discuss the decode-and-forward type relay station within the context of the 3GPP RLC/MAC layers. The emphasis is on proper RLC layer operation for the relay-enhanced-cell (REC). RLC layer evolvement in the future high data rate radio access network will first be discussed, then its performance in REC will be investigated, with or without it being implemented on the relay station. Simulations are conducted on the downlink for the future OFDMA-based relay network, from which we finally draw significant conclusions.

In this paper, a novel approach of HSUPA modeling for static simulation is proposed. This approach is based on the simulation results from time-based simulation and combines the well-known snapshot procedure for R99 with corresponding enhancements for the specific features of HSUPA. The detailed simulation structure and required deduction of multi-cell system equation group are described, respectively. The simulation results show that the new static approach can yield significant performance evaluation for a network much faster than the time- based approach and have comparable quality of the results at the same time. The presented approach is as well applicable for detailed system-level simulations as for incorporation in network dimensioning and planning tools.

An ad-hoc network is a network whose particular characteristics include absence of any kind of infrastructure, dynamic topology and limited resources. Up to day, several routing protocols have been proposed to face such a hostile environment. These solutions usually provide to the sources of data flows with single valid paths towards their destinations. However, it has been proven that maintaining multiple paths can be favorable to the network, either using the paths at the same time (for load balancing) or using an alternative path, when the primary one fails. It is also a way to meet the Quality of Service requirements demanded by some particular applications. In this paper, we propose a novel routing mechanism based on Dynamic MANET On-demand (DYMO) routing protocol, in order to enable the establishment of more than one node disjoint routes towards the same destination. Simulation results show that there is a significant gain in end-to-end delay for data packets and a reduction in the number of the transmitted routing packets.

In this paper, a QoS-aware cross-layer scheduler for wireless LANs is introduced which allows the control of a number of data flows according to QoS requirements of the respective applications. It is based on a two-stage concept: The MAC layer scheduler uses an importance function to calculate the priority of a particular flow, whereas the PHY layer scheduler considers the channel conditions of a MIMO-TDMA transmission. It is shown that with the concept presented here, throughput and delay requirements of time-critical applications can be met.

In the OFDMA-based downlink of wireless communication networks, in order to counteract the strong influence of multi-cell interference on the cell edge users, the so-called fractional power loading strategy was proposed. The key idea is to divide the whole frequency band into two parts, one part has lower transmission power, the other has higher transmission power and is orthogonal in the neighboring cells. With such power allocation, a natural thinking is to schedule the cell edge users onto the high power band, and to schedule the cell center users onto the other. Such strategy leads to the partitioning of users into the cell center/edge user groups. This paper proposes a way to do dynamic user grouping, and compares the cell performance for elastic traffic, with and without the user grouping. Extensive system level simulations with multi-channel proportional fair scheduler are conducted to reveal the effect of user grouping on the scheduler performance and QoS fulfillment.

In order to improve the effective utilization of the radio resources, bit rate adaptation (BRA) is applied in the UMTS system, specifically for the best effort and interactive packet traffic. This paper presents investigation results on the optimization of bit rate adaptation (BRA) scheme for an efficient data support in the UMTS radio access network.

In this paper, Transmit Power Control algorithms are discussed to support the Spectrum Management required by the Wireless LAN standard IEEE 802.1lh. A centralised algorithm for the infrastructure mode and an algorithm working on a link-by-link basis for ad-hoc networks are discussed. The performance of the algorithms, in particular the reduction of the transmit power and the enhancements of the QoS parameters delay and throughput are considered. It is shown that both algorithms significantly reduce the transmit power and improve the quality-of-service characteristics of wireless networks.

The intelligent network (IN) is a worldwide concept which specifies the functions, interfaces, and service-independent building blocks for the implementation of value-added services in digital telecommunication networks. One example is UPT (universal personal telecommunication), a service for fixed network mobility, which is presently being standardized. This paper describes the evaluation and implementation of an advanced UPT service, which allows a reachability control for ISDN based on the signaling system no. 7 (SS.7) and the intelligent network, which depends on the user's present social role (work, meeting, travel, vacation, etc.). The paper applies models of the users' communication habits, the components, and network services, develops simulation and queuing models for the traffic analysis and evaluation; and determines the call setup time for the new services as well as the quantitative load on the signaling network. Different scenarios for the distribution of the functionality of the intelligent network are studied. Two prototypes of the advanced UPT service are described: an integrated implementation placed in the network, and a standalone implementation outside the network. These prototypes have been displayed at several conferences and fairs. Services of this type are of interest in view of new competitive value-added services.

The method used for analyzing the combined shortest remaining processing time first/round robin (SRPT/RR) strategy is briefly introduced, and it is shown that this combined strategy preserves the essential advantages of the SRPT principle. In data communications, this strategy potentially may be applied to the packet switching process. The packet length corresponds to the time slice, and overhead corresponds mainly to the packet frame because most of the operations needed to organize the queue can be performed in parallel. In networks with virtual or static routing, this strategy selects the next packet of the message with the least number of remaining packets; for an equal number of remaining packets, the message with the shortest last packet is chosen. The first in, first out (FIFO) rule is used to resolve any remaining ambiguity. Local area networks are a special area of application for this strategy.>

Considering queueing systems in data networks, one can identify a special feature of the communication system: the service time, which is proportional to message length, is known in advance. This is a precondition for queueing strategies such as SPT (shortest processing time first) or SRPT (shortest remaining processing time first), which offers the shortest mean delay time among all conceivable strategies. For a practical evaluation of these strategies it is essential to include the influence of overhead time, especially for the Preemptive SRPT strategy. This paper analyzes SRPT with constant overhead time CV modified by a constant preemption gapCP \gcong CVforM/G/1systems. The comparison of strategies for FIFO, SPT, SRPT, and RR (round robin) using different service time distributions with coefficients of variation >1 and overhead shows, for SRPT and for a combination of SRPT and RR, a considerable reduction of the mean delay time\overline{TD}, especially for service time distributions with high coefficients of variation. This result indicates the potential advantages of these strategies in data network applications.