Telecommunication Systems

Recent research has focused on Continuous K-Nearest Neighbor (CKNN) query over moving objects in road networks. A CKNN query is to find among all moving objects the K-Nearest Neighbors (KNNs) of a moving query point within a given time interval. As the data objects move frequently and arbitrarily in road networks, the frequent updates of object locations make it complicated to process CKNN accurately and efficiently. In this paper, according to the relative moving situation between the moving objects and the query point, a Moving State of Object (MSO) model is presented to indicate the relative moving state of the object to the query point. With the help of this model, we propose a novel Object Candidate Processing (OCP) algorithm to highly reduce the repetitive query cost with pruning phase and refining phase. In the pruning phase, the data objects which cannot be the KNN query results are excluded within the given time interval. In the refining phase, the time subintervals of the given time interval are determined where the certain KNN query results are obtained. Comprehensive experiments are conducted and the results verify the effectiveness of the proposed methods.

A time-domain (TD) channel estimation (CE) technique is proposed in this paper to enhance the CE performance of IEEE 802.11p standards in cases when the pilot density is insufficient to accurately estimate channel states in rich-scattering environments. This technique is based on a least squares (LS) algorithm and assisted from Zadoff-Chu (ZC) sequences arranged into the symbol prefix and training preambles. The channel conditions tested in this paper were comprehensively simulated in urban, suburb and express-way scenarios.

The IEEE 802.16 system called WiMAX (Worldwide Interoperability for Microwave Access) provides quality of service (QoS) of several types for different service. The WiMAX is expected to support QoS for real time application, such as Voice over IP (VoIP). In this paper, when network congestion occurs, the VoIP bit-rate needs to be adapted to achieve the best speech quality. We propose a new scheme called Adaptive VoIP Level Coding (AVLC). VoIP is sensitive to delay and loss. According to different network conditions such as varied modulation, packet delay, packet loss, and residual time slot, we use G.722.2 codec to adapt each connectionpsilas data rate. Simulation experiments are conducted to test the performance (network delay, packet loss, and R-score) of the proposed mechanisms. In result, we increase the R-score about 40% to 50%.

Cellular network applications are growing drastically and this requires a fast and efficient transport method between the base station and the mobile switching center. One possible solution is to use ATM links. The low data rate and small-sized packets in the typical cellular applications imply that a significant amount of link bandwidth would be wasted if this small sized packet is carried by one ATM cell. For efficient operation for such cellular and low bit rate applications, a new type of ATM adaptation layer, AAL type 2, has been proposed. In this paper the principles of AAL type 2 are briefly described along with an introduction to other alternatives which have formed the basis for this new AAL. The result from the simulation to study the performance of the AAL type 2 is discussed from the packet delay and ATM cell use efficiency point of view. Due to the variable sizes of packets in this application, the fairness issue in serving variable sized packets is also discussed along with the effect of a fair queueing algorithm implemented at AAL type 2

The spread spectrum technique, the collision avoidance multiple access protocols and the controlled time of arrival scheme are combined to form a new set of medium access protocols for wireless networks. The request-to-send and clear-to-send message dialogue helps a transmitter detect whether an intended receiver is busy and whether any collision has occurred; thus speeding up the retransmission. The spreading code assignment avoids the disruption of any ongoing transmission by an intruder. Finally, the packets are sent under controlled time of arrival to further increase the channel throughput. Simulation results confirm that a higher channel throughput is achieved by the new protocols even in a dense network

This article proposes a solution for two issues in current communications. Firstly, that IMS suffers from client inexistence. Secondly, that mobile operators want to give the subscriber a possibility to access their VoIP network and efficiently cover special areas like airports. To address these problems, we have developed a novel service architecture, which allows 2G subscribers access to a SIP based VoIP network via WiFi complying security standards. User authentication and authorization is based on the EAP- SIM algorithm. Integrity and confidentiality is provided by IPSec connection established using parameters derived from authentication triplets. Consequently, all security-related issues can be performed exclusively using the subscriber identity module (SIM). We have tested and verified the proposed architecture with a mobile phone and have proven the correctness of our approach. The main drawback that remains is the difficulty of IPSec implementation that can be bypassed by a special application.

The total amount of capacity that can be obtained from a network is an essential performance metric. However, in wireless ad hoc networks, the dynamic topology and stochastic variations in channel characteristics, makes it difficult to analytically determine this metric. In this paper we develop analytical closed form expression for the network capacity. First the cumulative distribution function (CDF) of the signal to interference power ratio (SIR) at each receiver is derived analytically. Then, the capacity of the network is studied and closed form expressions are determined. Also by examining the effect of the outage threshold, beta, it is shown that, less simultaneous communications, each with a higher rate can result in higher total capacity.

Greedy and Contention-based forwarding schemes perform data routing hop-by-hop, without discovering the end-to-end route to the destination. Accordingly, the neighboring node that satisfies specific criteria is selected as the next forwarder of the packet. Both schemes require the nodes participating in the selection process to be within the area that confronts the location of the destination. Therefore, the lifetime of links for such schemes is not only dependent on the transmission range, but also on the location parameters (position, speed and direction) of the sending node, the neighboring node and the destination. In this paper, we propose a new link lifetime prediction method for greedy and contention-based routing. The evaluation of the proposed method is conducted by the use of stability-based greedy routing algorithm, which selects the next hop node having the highest link lifetime.

ATM networks are today operational, both as backbones for existing LAN technologies and as commercial wide-area multiservice networks. Still, in the early deployment of multi-site ATM networks, a number of difficulties have arisen out of the differences between the service definitions in local ATM networks and long distance carrier networks. In particular, the adaptation of LAN emulation protocols relying on switched, best-effort connections to the first generation of ATM WAN services turned out to require the introduction of specific functions. In this paper, we study and discuss the nature of these adaptation functions which include peak rate shaping in order to comply with the traffic contract at the public UNI, efficient buffering and selective cell discarding to optimize the performance of end-to-end data protocols and fairness mechanisms to improve resource sharing. We describe a flexible hardware platform which enabled a quick prototyping of these functions. It provides on-board support for efficient cell processing and for associated functions (rate control, buffering, etc.). The use of a cell processor enables a software-only implementation of the ATM cell handling, yielding short development times and easy debugging while being compatible with an operation at line speeds up to 155 Mbit/s. We finally give examples and measurements of the use of the adapters in LAN/WAN interworking situations

In this paper, we propose and evaluate under realistic mobility patters an adaptive location management strategy that jointly considers location update and paging. It requires little involvement of the terminal, and a non excessive computational load in the network. The location update scheme is based on the definition of an adaptive macro-location area, adapted to the mobility pattern of the terminals. In addition, we propose a sequential paging strategy that takes advantage of the multi-register in the macro-location area to characterize the residence probabilities in each location area of a generic macro area. Results show the viability of the proposal and its applicability to new packet-based broadband cellular systems.

The role of call admission control (CAC) in high-speed networks is to maintain the network utilization at a high level, while ensuring that the quality of service (QoS) requirements of the individual calls are met. We propose a generic dynamic call admission scheme for VBR and ABR traffic whose aim is to reduce the blocking rate for VBR calls at the expense of a higher blocking rate for ABR calls. Our scheme is generic because it builds up on a pre-existing static scheme, e.g., one based on a simple notion of effective bandwidth

The transmission bandwidth consumed by delivering rich content, such as movie files, is enormous, so it is urgent for ISPs to design an efficient delivery system minimizing the amount of network resources consumed. To serve users rich content economically and efficiently, an ISP itself should provide servers with huge storage capacities at a limited number of locations within its network. Therefore, we have investigated the content deployment method and the content delivery process that are desirable for this ISP-operated content delivery network (CDN). We have also proposed an optimum cache server allocation method for an ISP-operated CDN. In this paper, we investigate the properties of the topological locations of nodes at which cache placement is effective using 31 network topologies of actual ISPs. We also classify the 31 networks into two types and evaluate the optimum cache count in each network type.

This paper looks at the development of Colmar city's metropolitan area network (MAN) called OASICE. This MAN uses ATM technology to provide a range of business, academic and community tele-services such as tele-teaching, tele-information, tele-documentation, tele-management and tele-surveillance of property, high definition teleconference and VOD (video on demand)

The CORBA standard as published by the OMG (object management group) defines among other things the structure of a middleware platform called an object request broker (ORB) in order to achieve interoperability of applications in distributed and heterogeneous environments. The standard does not prescribe a specific technology for building this middleware platform. In this paper we show how to integrate ATM technology into a CORBA compliant implementation. Making use of the advantages offered by ATM requires the modification of the ORB API. A prototype is based on the freely available CORBA implementation called MICO. We show how to make use of MICO's micro-kernel architecture in order to achieve a seamless integration of a new transport layer

An approximate model is considered for characterizing aggregated cell arrivals from heterogeneous sources which may generate cells in a highly time correlated manner. The model consists of two types of on-off sources and provides enough degrees of freedom to match a number of moments of cell arrival rates as well as the time correlation of the aggregated cell arrivals. Numerical examples are presented to illustrate the accuracy of the queue length distributions obtained using the model compared with the exact results

Many distributed multimedia applications involve data delivery from a source to multiple destinations, the participating nodes forming a multicast group. In the naive solution, separate connections can be established from each source to other group members. However a tree can be established for each source with the participants as the leaf nodes or just have one tree spanning all the participants. In this paper, we introduce a data forwarding model to support such shared multicast trees over the ATM networks called CRAM (cell re-labeling at merge-points for ATM multicast). Our work allows VC merging in the MPLS architecture and supports recently proposed wide area multicast protocols (like CBT and PIM) in ATM networks

This paper presents a comparative study of the different techniques aimed at transmitting MPEG-2 streams over ATM networks, with emphasis on the presentation of practical results achieved on an experimental platform available at LSI, a laboratory of the University of Sao Paulo, Brazil. This study determines some tuning parameters for optimizing the implementation of distributed multimedia applications on different network technologies. The focus of this experimental study has been on non native ATM techniques like “classical IP over ATM” and “LAN emulation”, as well as native ATM techniques based on direct access service primitives to AAL 5. The influence of different parameters on the resulting throughput, like packet length, buffer size, CPU speed, has been studied. Furthermore, a testing tool is being developed for assessing the transmission of MPEG-2 streams based on the practical experience achieved so far. These results will permit the evaluation of different network technologies and also help in the development of distributed multimedia applications like video-on-demand, videoconferencing and telemedicine

The Internet has traditionally relied on end-to-end congestion control performed at the transport layer, where sources reduce their offered traffic only after congestion sets in. By then, network resources have been wasted and effective throughput compromised. In next generation Internets, the delay-bandwidth product is large and bandwidth is a precious resource. Hence, in this paper we present a link layer flow control for Internet backbones over ATM using the ABR service and flow control. We describe a backpressure mechanism that reduces packet losses and promotes effective utilization of allocated and unused resources along backbone links. We then show how to perform dynamic renegotiation of allocated backbone resources based on our flow control approach and on class based queueing

Two very simple and very tight lower bounds on average delay are derived for packet satellite protocols with contention-free and contention-based control channels. The derivation indicates that, for minimum delay, a protocol should maintain a balance between transmitting packets immediately and making reservations before transmissions. In addition, whenever possible, all unreserved slots should be filled with new transmissions at a rate of one packet per slot. For a small number of users, TDMA (time division multiple access) may be a better alternative than contention-based protocols. The exact conditions on the number of users and system throughput for selecting either protocol are also derived

This paper presents a new group key management protocol, CKC (Code for Key Calculation). This protocol is based on logical key hierarchy. When a new member joins the group, server sends only the group key for that member. Then, current members and the new member calculate the necessary keys using node codes and one-way hash function. Node code is a code which is assigned to each node of the key tree. Again at leave, server just sends the new group key to remaining members. By this key, members calculate necessary keys using node codes and one-way hash function. The security of the keys is based on one-wayness of hash function. The results show that CKC reduces computational and communication overhead and message size at join while it keeps minimized the overhead of key update at leave.

Recent research on statistical multiplexing has provided many new insights into the achievable multiplexing gain in QoS networks, however, generally only in terms of the gain experienced at a single switch. Evaluating the statistical multiplexing gain in a general network remains a difficult challenge. In this paper we describe two distinct network designs for statistical end-to-end delay guarantees, referred to as class-level aggregation and path-level aggregation, and compare the achievable statistical multiplexing gain. Each of the designs presents a particular trade-off between the attainable statistical multiplexing gain and the ability to support delay guarantees. The key characteristic of both designs is that they do not require, and instead, intentionally avoid, consideration of the correlation between flows at multiplexing points inside the network. Numerical examples are presented for a comparison of the two designs. The presented class-level aggregation design is shown to yield very high achievable link utilizations while simultaneously achieving the desired statistical guarantees on delay

The ATM service category UBR is intended for non-real-time applications that do not require guaranteed QoS commitments. With additional, relatively inexpensive control functions such as packet discard schemes, UBR could become a cost-effective alternative for the transmission of data traffic; offering a straightforward and flexible solution as opposed to nrt-VBR and GFR that applies stricter traffic specifications as well as ABR with its sophisticated and complex rate-control protocol. This paper presents the results obtained from a comprehensive set of experiments with TCP over UBR, comprising of measurements taken on different protocol layers. The goal is to experimentally investigate the performance of UBR to carry TCP traffic, to evaluate the performance gain achievable by packet discard schemes and TCP parameter tuning, to study the influence of the TCP implementation, and in a final step, to relate the measurements to simulation results

Unified modeling language (UML) sequence diagrams combined with the UML profile for modeling and analysis of real-time and embedded (MARTE) systems are used to represent the systems' requirements. To enhance formal analysis, sequence diagrams annotated with MARTE stereotypes are mapped into timed colored Petri nets with inhibitor arcs (TCPNIA). The mapping rules for the fragments of sequence diagrams and MARTE stereotypes are proposed respectively. The mapping rules for general ordering between messages are proposed to cope with complicated interactions. The data related issues are handled through colored properties in TCPNIA models, guard functions and operational functions. A mapping rule for state invariant is proposed based on data related information. Through state invariant, complicated control relations can be expressed. Formal definitions for morphing and substitution in TCPNIA models are given. They provide modular and hierarchical modeling methods for TCPINA models. A case study shows the applicability and feasibility of our method.

The ATM Forum's PNNI (private network node interface) standard specifies a flooding mechanism to distribute topological state information amongst nodes participating in a PNNI network. While the flooding mechanism provides robust topology distribution, we show by simulation, that it can disproportionately overload lower-bandwidth links and, due to its inherent redundancy, can generate considerable computational overheads due to the processing of redundant topology updates. To address these issues, we introduce the hybrid spanning tree algorithm, a spanning tree-based topology distribution mechanism that has low computational maintenance and can support policy that restricts topology distribution control traffic from being carried over lower-bandwidth links. Unlike other spanning tree proposals, the hybrid spanning tree algorithm provides a simple and practical migration path to smoothly transition PNNI nodes executing the flooding algorithm to that of the hybrid spanning tree algorithm

When developing a telephone switch, it is useful to know how long it will take to process the various tasks associated with call processing. The model developed in this paper gives expected sojourn times for those tasks. It is a priority queueing model with a modified first-come first-served (FCFS) service discipline, which mimics the treatment of tasks in actual system software. The model is an M/G/1 queueing model with preemption (preemptive resume). It consists of multiple queues, one for each distinct priority, where each task has been preassigned a constant priority. Within each priority queue, the tasks are further grouped by type. An arriving task will join the back of the group of tasks of its type, regardless of where this group is positioned in the queue. Upon completion of a task, multiple subsequent tasks can be given ready-for-service status, and enter the priority queues. This is referred to its forking. Call processing involves many ordered sets of tasks (jobs), some of which will contain forks. The model produces results that compare favorably with those obtained by simulation