ISDN Networks Research Articles

A methodological framework for end-of-life management of electronic products

A growing number of regulatory interventions such as the European Union's Extended Polluter Responsibility Policy hold manufacturers accountable for the damage inflicted to the natural environment by their products. However, not all components of an industrial product have an equitable role regarding their overall burden to the environment. In this context, it is of great interest to manufacturers to identify and rank those components with the highest potential value at the end of their useful life, employing an array of economical and environmental criteria. In order to assist manufacturers on their decision-making for the optimal end-of-life alternatives for their products, the “Multicriteria Matrix” methodology has been developed. The methodology relies on multicriteria analysis and takes into consideration the residual value, environmental burden, weight, quantity and ease of disassembly of each component. With the developed methodology only those components that do not have any residual value end up in landfills, while the majority of the components are either reused or recycled. The application of the developed methodology is demonstrated employing a real-world case study; that of an ISDN network terminal. Finally, the paper is concluded by presenting interesting managerial insights that were obtained.

Tool assisted H.323 dial‐plan design following formally defined strategies

AbstractThis paper considers H.323 v4 VoIP networks consisting of gateways and gatekeepers, addressing the task of terminating calls to multiple ISDN networks. Our goal is the maximal utilization of the gatekeeper's centralized routing mechanism for call termination to ISDN networks based on load balancing and resource availability indication (RAI), thus minimizing the need for usage of unwanted (second attempt) overflow mechanisms. We elaborate on the drawbacks of overflow mechanisms and give an overview on different solutions offered in practice by vendors. The discussion justifies the emphasis on load balancing, thus we present an algorithm and tool assisted approach for the systematic assignment of PRIs from ISDN networks to gateways. We follow strictly defined design criteria that lead to an optimized network dial‐plan. We thus offer a stand alone, off‐line solution without the need of any extra add‐ons or upgrades. This is intended to replace the purely intuitive approach followed by today's planners.Our scope is the design rather than the runtime environment; the latter continues to feature the simplicity of a single global RAI per gateway. We consider desirable and non‐desirable link distribution topologies between gateways and ISDN networks based on operational and economic evaluation criteria. These measure the number of ‘wasted’ gateway ports or ‘inappropriately’ allocated network interfaces. Our approach is illustrated through a worked out example showing key features of the proposed algorithm and revealing characteristic cases met in the ISDN interface allocation.Finally, we show how the presented methodology, which presently addresses a H.323 design issue, can provide performance benefits in the VoIP technologies of the immediate future. Copyright © 2005 John Wiley & Sons, Ltd.

A system employing high‐quality speech compression technology to produce voice on demand for visually handicapped persons

AbstractWe have developed a voice‐on‐demand system in which a voice reading the latest information such as newspapers and public bulletins or regional information is information‐compressed by a voice coding technique and stored on the disks of a PC server, and then distributed to receivers at the homes of visually handicapped persons via ISDN networks. In the compression of read speech, the low‐rate speech coding technique DualSpeech is developed and introduced. The information content is reduced to about 1/18 with almost no degradation of the quality of the read speech, so that it can be distributed at high speed. Moreover, the one‐megabyte memory embedded in the receiver can store about 60 minutes of the read speech compactly and the stored information content can be retrieved instantly by an index retrieval function. With these merits, the time required in distribution and communication charges are reduced, and a voice‐on‐demand service is realized which has the appeal of allowing repeated listening at any time to information which the visually handicapped persons would like to acquire. Evaluation of the performance of this voice‐on‐demand system indicates that the read speech data can be received in a short time, about 1/8 of the regeneration time, and regenerated speech can be obtained with a high quality equal to that of the personal handy‐phone system (PHS). Moreover, we have performed service experiments using the voice‐on‐demand system employing speech compression for about 50 visually handicapped persons. The following desirable features have become clear from the questionnaire results: enrichment of close‐at‐home information which the visually handicapped persons can select by themselves, increase of the update frequency of the information content, and further enhancement of the operability of the receiver. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 2, 85(11): 56–70, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjb.10095

Methods and systems for accurate as well as estimated evaluation of quality of speech signals transcoding in inter-network telecommunications link

Co-operation of different communications networks requires implementing specialized interfaces that make possible exchanging user’s information as well as control information. For that reason estimating information processing quality of a given interface is an important issue. The publication presents methodology for estimating quality of processing user’s information. User’s information is in a speech signal form. The methodology has been used for assessing interface between ISDN network and special network. Two methods have been employed – the (objective) method for automatic, though approximate, assessment, and the accurate method using intelligibility tests. The publication discusses the speech signal processing principle for the inter-network interface, measuring methods and systems as well as results of transcoding quality assessment for the developed interface. The above-mentioned research topics have been solved within the framework of the grant (No. 8 T11B 056 11) financed by the State Committee for Scientific Research.

Self-similarity of compressed video traffic in ATM networks

Data, voice, and compressed moving pictures flowing in ATM broadband ISDN networks are expected to be high autocorrelated time series traffics which are different from conventional voice and packet traffics. This paper first describes traffic modeling by a self-similar process for time series data. Second, it computes the value of the Hurst parameter which shows self-similarity of the traffic using the method of R/S analysis for moving picture data recorded on a laser disk with MPEG2 compression with variable bit length. As a result of this analysis, we found a high degree of self-similarity, that is, long memory property of the traffic, because MPEG2 generates I frame periodically. © 2000 Scripta Technica, Electron Comm Jpn Pt 1, 83(6): 108–116, 2000

The COVEN Project: Exploring Applicative, Technical, and Usage Dimensions of Collaborative Virtual Environments

COVEN (Collaborative Virtual Environments) is a European project that seeks to develop a comprehensive approach to the issues in the development of collaborative virtual environment (CVE) technology. COVEN brings together twelve academic and industrial partners with a wide range of expertise in CSCW, networked VR, computer graphics, human factors, HCI, and telecommunications infrastructures. After two years of work, we are presenting the main features of our approach and results, our driving applications, the main components of our technical investigations, and our experimental activities. With different citizen and professional application scenarios as driving forces, COVEN is exploring the requirements and supporting techniques for collaborative interaction in scalable CVEs. Technical results are being integrated in an enriched networked VR platform based on the dVS and DIVE systems. Taking advantage of a dedicated Europe-wide ISDN and ATM network infrastructure, a large component of the project is a trial and experimentation activity that should allow a comprehensive understanding of the network requirements of these systems as well as their usability issues and human factors aspects.

Dynamic bandwidth control in ATM networks

The Asynchronous Transfer Mode (ATM) principle has been recommended by the ITU as the transport vehicle for future Broadband ISDN Networks. One of the most important objectives of ATM-based networks is to support multimedia traffic services. The multimedia traffic is a diverse mixture of several types of traffic (e.g., video, voice, image, data) that have different correlations, burstiness characteristics and quality of services (QoS) requirements form the network. Efficient bandwidth allocation and access control algorithms are required to ensure (and maintain) specific QoS requirements for each type of traffic. In this article, we propose and analyze a dynamic bandwidth allocation and control scheme based upon the virtual path (VP) principle. The scheme exploits the statistical multiplexing gain principle in order to increase the bandwidth utilization per VP. It allocates a certain amount of bandwidth to each VP that is necessary to meet the QoS requirements of the traffic carried by that VP. Each VP carries only homogeneous traffic (i.e., traffic with similar characteristics and QoS requirements are supported by a single VP). To dynamically control the allocated path-bandwidth a Bandwidth Control Period (BCP) rule is proposed to control the scheduling of different types of traffic that are supported by separate VPs. It is shown that with proper choice of the BCP, a path-bandwidth can be allocated to guarantee QoS required by the traffic. The scheme is dynamic in the sense that the bandwidth allocated, per VP, depends upon the number of accepted calls and variations in the traffic characteristics. Further, it is shown that access flow control is essential in order to minimize the cell loss rate and enhance the statistical multiplexing gain per VP.

ITU-T standardization activities for interactive multimedia communications on packet-based networks: H.323 and related recommendations

The Telecommunication Sector of the International Telecommunication Union (ITU-T) has developed a series of recommendations together comprising the H.323 system that provides for multimedia communications in packet-based (inter)networks. This series of recommendations describe the types and functions of H.323 terminals and other H.323 devices as well as their interactions. The H.323 series of recommendations includes audio, video and data streams, but an H.323 system minimally requires only an audio stream to be supported. Motivated by straightforward interoperability with the ISDN and PSTN networks and a variety of other protocols, the recommendation H.323 has been accepted as being the standard for IP telephony, developed by the ITU-T and broadly backed by the industry—which is also adopted by both the Voice over IP (VoIP) forum and the European Telecommunication Standards Institute (ETSI). This paper presents an overview of the H.323 system architecture with all its functional components and protocols and points out all the related specifications.

Realistic modelling of blocked packets for accurate performance evaluation of ATM switches

Multistage switches have been used as ATM switching fabrics in broadband ISDN networks, and also to connect processors to memories in massively parallel multiprocessor systems. Previous performance models for multistage switches have been neither accurate enough nor based on realistic assumptions regarding modelling the correlation of the blocked packets in successive stages of the switch. Two new analytical models for accurate analysis of multistage switches are proposed. The new models reflect the realistic behaviour of blocked packets, and take into account the fact that a blocked packet always hunts for the same output link in successive clock cycles. The results obtained from the models are more accurate than those available in the literature.

Teleconsultation

Teleconsultation is a consultation between two or more physicians about the diagnostic work-up and therapeutic strategy in the treatment of an individual case by means of modern telematics. Due to more complex therapeutic strategies and legally defined formal requirements, the need for teleconsultation will increase significantly in the future. Rapid technical improvements in telematics will progressively facilitate the practical performance of teleconsultation (based upon an ISDN network in the beginning, later on by the use of a national health network). The medicolegal aspects of teleconsultation have already been defined sufficiently for use in surgery. However, the question of adequate financial compensation for this type of medical service is still unclear.

Dual-state augmentation for minimizing conformance test costs

This paper discusses the dual-state augmentation method to generate the minimum-length test sequences using the multiple UIO sequences of minimal and nonminimal lengths. In dual-state augmentation, all UIO sequences (minimum-length or not) can be used to minimize the test sequence length. The resulting test sequences are significantly shorter than the test sequences generated by the existing methods which only use the minimum-length UIO sequences. As a case study, the dual-state augmentation technique is applied to a subset of the ISDN Network Layer protocol for supplementary voice services. Significant savings on test sequence length are observed.

Concept of a multistandard receiver for digital broadcast and communication services

The article describes several aspects of realisation of a so called multistandard receiver, i.e. a device, which combines receiving facilities for several digital (DAB, DMB, DVB) and, at least for a transitional period, analogue broadcasting services (FM, PAL/SECAM). Possible structures and realisation aspects are considered. Furthermore, the connection to an ISDN or GSM network and to an indoor communication system is discussed. The systems to be integrated in the receiver are characterised by their main parameters and modes. From that, requirements to the architecture of the multistandard receiver are derived. The A/D conversion, signal identification and synchronisation, channel estimation and correction, channel decoding, program and service selection and frame demultiplexing are discussed in more detail. Furthermore, basic differences of multistandard receivers for mobile, portable and stationary receiving conditions concerning the services to be implemented and the receiver architecture are considered. Further discussions on the use of such a receiver as an office or entertainment device are presented and the requirements resulting from that are derived.

Image Bank, Diagnostic Codification and Telediagnosis in Hematology

Recent technological improvements which permit transmission and discussion of color images with good resolution and in real time may become very useful for hematological and histopathological applications. The workstations used a PC Pentium, 16 Mo RAM 1,6 Go of hard disk, including a laser-reader for CD-ROM and network cards. Applications are developed under Windows 95. The system allows to digitalize, compress and decompress any kind of colored or black and white images according to the International Norms, JPEG. The storage is managed by a specific software according to specific classification for diagnostic nomenclature. It can store typical, less typical and atypical images of examples of pathology which can contribute to the constitution of a digitized bank of images and CD-ROM. The system used have been launched to allow pathology image transmission on ISDN networks for diagnosis assessment. Standardization of nomenclature which has been achieved both for immunophenotyping and cytogenetics is necessary for the cyto-anatomical lesions in the specialized field of hematology. In using the major published recommendations (FAB, EGIL, REAL etc.), a comprehensive approach of nomenclature for our image data base is proposed through a 500 item Thesaurus, which have been written simultaneously as a multilangage version.

Traffic and video quality with adaptive neural compression

Video sequences are major sources of traffic for broadband ISDN networks, and video compression is fundamental to the efficient use of such networks. We present a novel neural method to achieve real-time adaptive compression of video. This tends to maintain a target quality of the decompressed image specified by the user. The method uses a set of compression/decompression neural networks of different levels of compression, as well as a simple motion-detection procedure. We describe the method and present experimental data concerning its performance and traffic characteristics with real video sequences. The impact of this compression method on ATM-cell traffic is also investigated and measurement data are provided.

Narrowband ISDN and broadband ISDN service and network interworking

For a significant period of time, B-ISDN will coexist with the 64-kb/s-based (narrowband) ISDN. The specification of a concept and the mapping functions employed for interworking between the two networks, in particular for voice services, is a major factor for the success of B-ISDN.

Architecture for interworking data over PCS

This article presents a new approach for wireless service providers to offer data services by taking advantage of the existing infrastructure for voice services and interworking with existing wireline-based data services. The article presents a framework: for interworking between any wireless radio system and any data application on the wireline network. The interworking is provided by a common interworking function that uses a generic interworking control protocol (ICP). Any radio system capable of using the ISDN-based C-interface and implementing ICP can take advantage of the proposed approach. ICP is a generic protocol and can be implemented using different networks. The article considers an ISDN network for the lower-layer transport of ICP. Though the article focuses on interworking with the PSTN and the Internet, the architecture also allows access to other data networks.

Performance of Buffered Multistage Interconnection Networks in a Nonuniform Traffic Environment

Multistage interconnection networks (MIN) are used to connect processors to memories in shared-memory multiprocessor systems. They have also been proposed as switching fabrics in ATM switches in Broadband ISDN networks. A multistage network consists of several stages of switching elements (SE) connected by a permutation function. Since routing conflicts among packets at the SEs cause blocking and reduce the performance of the network, analytical models have been developed and simulation studies carried out for performance evaluation of MINs in the presence of uniform traffic in the network, Buffers are used at SEs to increase the performance of the network, The suitability of two existing models for possible use in the presence of a hot spot is investigated in this paper, Results show that errors from the models are large. In this paper, we propose a new and efficient Markov chain model for performance evaluation of a single-buffered Omega network in the presence of a hot spot. Results obtained from the existing models and the proposed model for the uniform and the hot spot traffic patterns are presented. It is shown that the proposed model is better than existing models in the presence of hot spot traffic, While the existing models produce small errors in the case of low blocking in the network, the errors become prominent with increased blocking. The proposed model produces accurate results in the more general case of a hot spot traffic, whereas the existing models produce accurate results only in the special case of uniform traffic.

Revenue optimization of B-ISDN networks

We consider the optimization of routing in a broadband ISDN network through the maximization of the revenue generated by the network. Two types of calls are offered to the network with different holding times and bandwidths. The routing is alternative routing and load sharing. First we state the Kuhn-Tucker conditions and show how they provide a simple routing rule based on equal marginal net revenue. We then proposed two-stage decomposition algorithm for optimizing the routing which requires a flow-deviation stage combined with the solution of a linear system of equations. We obtain simple expressions for the net path revenues and the corresponding optimal routing rule. We summarize numerical results related to the convergence of the algorithm and the structure of the optimal routing.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

The interworking of Internet and ISDN networks for multimedia conferencing

This report focuses on the interworking between Internet based and ISDN based conferences, and shows how both the Internet based multicast multimedia conferencing environment and the ISDN based multimedia conferencing environment can be connected. In the MICE project a Conference Management and Multiplexing Center (CMMC) is used to provide facilities for users who require special facilities. A description of the components of the CMMC, which let it act as a gateway between the multicast environment and the ISDN environment, are presented

Bandwidth allocation and access control in high-speed networks

The problem of bandwidth allocation and access regulation arises in the congestion control of Broadband ISDN networks. This paper assumes that a single user, described by an on-off fluid model, is connected to the network via a leaky bucket access control mechanism. The bandwidth allocated to this user and the leaky bucket parameters are to be selected so as to guarantee a negotiated level of delay probability at the access point and packet loss probability in the network which is modelled as an output buffer. The design problem is to minimize the allocated bandwidth subject to service guarantees and stability conditions for the input and output buffers. We provide a desirable feasible solution to the design problem. The paper studies the effect of non-conforming users on the network performance using the leaky bucket access control corresponding to this feasible solution. We provide expressions that quantify the impact of the leaky bucket parameters in access regulation and the worst-case queueing behavior at the output buffer. Finally, we discuss the extension of this methodology to the multiple leaky buckets case.