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Wireless Body Area Networks (WBANs) consist of tiny Biomedical Wireless Sensors (BWSs) and a Gate Way (GW) to connect to the external databases in the hospital and medical centres. The GW could connect the BWSs, to a range of wireless telecomnumication networks. These wireless telecommunication networks could be either a mobile phone network, a standard telephone network a dedicated medical centre or using public Wireless Local Area Networks (WLANs) nodes also known a Wi-Fi system. The electrocardiogram (ECG) signals are widely used in health care systems because they are non-invasive mechanisms to establish medical diagnosis of heart diseases. The current ECG systems suffer from important limitations: limited patient's mobility, limited energy, limited on wireless applications. The main drawback of current EGG systems is the location-specific nature of the systems due to the use of fixed/wired applications. That is why; there is a critical need to improve the current ECG systems to cover security handling and to achieve extended patient's mobility. With this in mind, Compressed Sensing (CS) procedure and the collaboration of Block Sparse Bayesian Learning (BSBL) framework is used to provide a robust low sampling-rate approach for normal and abnormal ECG signals. Advanced WBANs based on our approach will be able to deliver healthcare not only to patients in hospital and medical centres; but also in their homes and workplaces thus offering cost saving, and improving the quality of life. Our simulation results based on two proposed algorithms illustrate 15% incensement of Signal to Noise Ratio (SNR) and a good level of quality for the degree of incoherence between the random measurement and sparsity matrices.

In computing, a network generally denotes devices, often referred to as nodes, connected by links. Networks that are modeled with diagrams consist of hundreds of symbols, images, pictures, and icons, such as a computer, a server rack, or a cloud-based storage system. Network representations provide valuable insights into understanding the systems' underlying structures and mechanisms. Nevertheless, this unusually large number of superficial symbols and icons reflects a need for more systematic representations of the interiority of nodes. To give uniformity to this cascade of notions of basic units of nodes in network diagrams, the authors propose adoption of a new modeling methodology, called a thinging (abstract) machine (TM) (abstract machine of things) that represents all notions as a single diagrammatic machine. Because of the large number of network types, in this paper, they specifically and without loss of generality focus on IP telephone (internet protocol telephone) networks to exemplify communication networks. A real case study of IP telephone networks is modeled using TM.

This paper presents an analysis of voice traffic in telephone networks, based on machine learning algorithms to detect frauds made by callers. Starting from the raw data set that includes information about the call date, destination number, duration and caller's number, in our approach we were able to identify fraudulent calls in early stages. For balance, the data set was split in 2 parts: one for training and one for testing. To obtain mean’s values from dataset, a standardization technique was applied in order to scale the data before the dimensionality reduction using Principal Component Analysis. Then, the first two components were used as inputs for Logistic Regression and Random Forest models, having the caller as target. Finally, the target was moved on the destination file so as to identify the caller and the moment when the call has started based on a vector representation of words.

Telemonitoring of Parkinson’s Disease (PD) has attracted considerable research interest because of its potential to make a lasting, positive impact on the life of patients and their carers. Purpose-built devices have been developed that record various signals which can be associated with average PD symptom severity, as quantified on standard clinical metrics such as the Unified Parkinson’s Disease Rating Scale (UPDRS). Speech signals are particularly promising in this regard, because they can be easily recorded without the use of expensive, dedicated hardware. Previous studies have demonstrated replication of UPDRS to within less than 2 points of a clinical raters’ assessment of symptom severity, using high-quality speech signals collected using dedicated telemonitoring hardware. Here, we investigate the potential of using the standard voice-over-GSM (2G) or UMTS (3G) cellular mobile telephone networks for PD telemonitoring, networks that, together, have greater than 5 billion subscribers worldwide. We test the robustness of this approach using a simulated noisy mobile communication network over which speech signals are transmitted, and approximately 6000 recordings from 42 PD subjects. We show that UPDRS can be estimated to withinless than 3.5 points difference from the clinical raters’ assessment, which is clinically useful given that the inter-rater variability for UPDRS can be as high as 4-5 UPDRS points. This provides compelling evidence that the existing voice telephone network has potential towards facilitating inexpensive, mass-scale PDsymptom telemonitoring applications.

Voice over Internet Protocol (VoIP) has become mainstream in terms of the usability due to two key factors: flexibility and low cost, all of this towards classic PSTN (Public as well Service Telephone Network) based on several studies and the practical approach because your calls are no longer routed along a dedicated wire to a local exchange, but instead provider implements packet switching and the direct line needed is now performed by the VoIP provider. However, there are several concerns over security, [13] which is quite understandable. The rising issues have naturally come over the efficiency communication VoIP offers and with this being a main key, that’s where companies and businesses have invested, leaving things in the open over security. This study tries to exploit the vulnerabilities VoIP shows in its classic scenario and give a few suggestions on what to do regarding the problems raised during the way. All the work was based on previous knowledge regarding VoIP as studied in Computer Networks.

Telephony technologies (mobile, VoIP, and fixed) have potentially improved the way we communicate in our daily life and have been widely adopted for business and personal communications. At the same time, scammers, criminals, and fraudsters have also find the telephony network an attractive and affordable medium to target end-users with the advertisement, marketing of legal and illegal products, and bombard them with the huge volume of unwanted calls. These calls would not only trick call recipients into disclosing their private information such as credit card numbers, PIN code which can be used for financial fraud but also causes a lot of displeasure because of continuous ringing. The fraudsters, political campaigners can also use telephony systems to spread malicious information (hate political or religious messages) in real-time through audio or text messages, which have serious political and social consequences if malicious callers are not mitigated in a quick time. In this context, the identification of malicious callers would not only minimize telephony fraud but would also bring peace to the lives of individuals. One way to classifies users as a spammer or legitimate is to get feedback from the call recipients about their recent interactions with the caller, but these systems not only bring inconvenience to callees but also require changes in the system design. The call detail records extensively log the activities of users and can be used to categorize them as the spammer and non-spammer. In this paper, we utilize the information from the call detailed records and proposed a spam detection framework for the telephone network that identifies malicious callers by utilizing the social behavioral features of users within the network. To this extent, we first model the behavior of the users as the directed social graph and then analyze different features of the social graph i.e. the Relationship Network and Call patterns of users towards their peers. We then used these features along with the decision tree to classify callers into three classes i.e. human, spammer and call center. We analyzed the call record data-set consisting of more than 2 million users. We have conducted a detailed evaluation of our framework which demonstrates its effectiveness by achieving acceptable detection accuracy and extremely low false-positive rate. The performance results show that the spammers and call center numbers not only have a large number of non-repetitive calls but also have a large number of short duration calls. Similarly, on the other hand, the legitimate callers have a good number of repetitive calls and most of them interacted for a relatively long duration.

Telephone network based on IMS technology has been widely applied inpower production and dispatching communication, especially in distributedpower stations. Analysis and positioning failure of IMS network is arduous,because it’s dependent on IP data communication network. In this paper,we first introduced IMS switching network architecture and distributed gen-eration communication network architecture, analyzed and summarized allkinds of network malfunction. Combining typical IMS network fault con-nection relations, we introduced an improved Petri net fault handling modeland reasoning method. The diagnosis and positioning results could reflectthe defects of equipment logic functions. This method on fault diagnosisand location of substation network has been proved to be effective throughpractical application.

Cloud computing is one of the leading options to store massive amount of data. The metaphor of the cloud was borrowed from old telecoms network schematics, in which the public telephone network was often represented as a cloud to denote that the just did not a matter it was just a cloud of stuff. However, cost and risk will increase inthe organization which are using cloud platforms. Hence this paper combines Cloud with blockchain technology to produce a secured file storage system. Blockchain is considered by many to be the safest technology. Here, a single file is splited and stored in blocks using the file manipulation and Advanced Encryption Standard (AES) algorithms. A high level, contract oriented language, Solidity is used for writing smart contracts. The system architecture which explains the cloud and blockchain processes is constructed. This system also ensures that it is free from attacks such as Brute force. Finally, the blockchain network in the cloud platform is implemented.

Latest technologies of voice over IP (VoIP) and mobile messaging for smartphones messengers such as WhatsApp, Viber, Skype, etc., offer free-of-charge facilities of worldwide SMS, MMS, and voice calls to their users, unlike the traditional and expensive cellular or telephone networks’ services. Customers of the formerly mentioned messengers are estimated in millions because of the attractive features offered by them. However, these messengers face many cyber security threats and the required security features are either not available at all or are insufficient for efficiently countering the threats. Professionals working in the domain of cyber security are challenged by the devastating effects of distributed denial of service (DDoS) attacks on all major platforms including Apple Macintosh, Windows, Unix, and Linux. In this paper, we demonstrate the effect of DDoS attack on the performance of an IRC server using a test bed. We use a game theoretic model to analyze the feasibility of DDoS attacks on the IRC platform, keeping in view the attacker’s objective. The analysis will help the security experts to propose appropriate countermeasures to reduce the attackers’ utility, thereby making it less attractive for those attackers to launch the attack.

The ideas of q-rung ortho pair fuzzy set (q-ROPFS) and interval-valued q-rung ortho pair fuzzy set (IVq-ROPFS) are two major recent developments in the field of fuzzy set theory. A q-ROPFS and IVq-ROPFS improved the limited structures of Pythagorean fuzzy set, intuitionistic fuzzy set as well as fuzzy set by improving the conditions that makes these concepts restricted. The goal of this research is to introduce a new notion of interval-valued q-rung ortho pair fuzzy graph (IVQ-ROPFG) and to study the related graphical terms such as subgraph, complement, degree of vertices and path etc. Each of the graphical concept is demonstrated with an example. Another valuable contribution of this manuscript is the modeling of some traffic networks, telephone networks and social networks using the concepts of IVQ-ROPFGs. First, the famous problem of finding a shortest path in a traffic network is studied using two different approaches. A study of social network describing the strength of co-authorship between different researchers from several countries is also established using the concept of IVq-ROPFGs. Finally, a telephone networking problem is demonstrated showing the calling ratios of incoming and outgoing calls among a group of people. Two engineering decision-making problems are also studied using some aggregation operators and the concepts of IVq-ROPFGs. Through comparative study, the advantages of working in the environment of IVq-ROPFG are specified.

Multistage interconnection networks are used as a medium for interconnecting processors and memories in multiprocessor systems. Multistage interconnection networks are an effective substitute for crossbar switches, used in telephone network systems and parallel computers, due to their low cost, better performance and easy maintainability. In this paper, we propose new fault-tolerant hybrid multistage interconnection networks obtained from shuffle exchange gamma interconnection network (SEGIN) and logical neighborhood network named as shuffle exchange gamma logical neighborhood interconnection network (SEGLNIN). We evaluated the performance of SEGLNIN in terms of disjoint paths, reliability and hardware cost and compared it with some well-known MINs like shuffle exchange network (SEN), shuffle exchange network with one extra stage (SEN +) and shuffle exchange network with two extra stages (SEN + 2), SEGIN-1, and SEGIN-2. The results illustrate that the performance of the proposed SEGLNIN is better than that of the compared networks in terms of reliability and disjoint paths.

This article analyzes how software pirates who formed the ‘warez scene’ appropriated telephone network infrastructures and Bulletin Board System (or BBS) technology to circulate computer so...

This paper introduces the concept of Phone of Things (PoT), a novel approach to extend the connectivity options for IoT systems through the pervasive telephone network infrastructure. This is a paradigm shift that integrates traditional telephones to monitor and control IoT-enabled devices via VoIP protocols and public switched telephone network (PSTN). This integration democratizes IoT through the promotion of a Do-It-Yourself (DIY) IoT system that takes advantage of the freedom the comes with open-source technologies and leverages the emerging cloud-hosted services to build customized IoT solutions while accentuating affordability, accessibility, performance, ease of use, and security. Nevertheless, the proposed system inhibits a modular design philosophy that allows the system to be tailored and scaled according to the application needs with minimum hardware and software modification requirements. The proposed paradigm provides an arbitrarily shared data space between IoT systems which would coalesce into a social network of things (SNT) and support assistive services in public sensing.

Malicious phone calls, including spam and scams, caused millions of global financial losses every year and was a difficult problem over many years. This work introduces the solution based on machine learning for telecommunications without underlying the telephone network infrastructure. The major obstacle of this ten years old problem is building efficient functions without access to the telephony network infrastructures. The previous Spam Call data set is collected first. The dataset includes several labelbased features to predict malicious calling. We primarily focus on using Recurrent Neural Network (RNN) algorithm to detect the malicious calls. With the proposed features, we review different state of the art methods of machine-learning, and it is inferred that the most optimal approach can minimize malicious calls to 90% while keeping over 90% of the binary call accuracy. The outcomes also show that without significant overhead latency the models can be implemented effectively with the help of an evaluation analysis.

Many articles have used voice analysis to detect Parkinson's disease (PD), but few have focused on the early stages of the disease and the gender effect. In this article, we have adapted the latest speaker recognition system, called x-vectors, in order to detect an early stage of PD from voice analysis. X-vectors are embeddings extracted from a deep neural network, which provide robust speaker representations and improve speaker recognition when large amounts of training data are used. Our goal was to assess whether, in the context of early PD detection, this technique would outperform the more standard classifier MFCC-GMM (Mel-Frequency Cepstral Coefficients - Gaussian Mixture Model) and, if so, under which conditions. We recorded 221 French speakers (including recently diagnosed PD subjects and healthy controls) with a high-quality microphone and with their own telephone. Men and women were analyzed separately in order to have more precise models and to assess a possible gender effect. Several experimental and methodological aspects were tested in order to analyze their impacts on classification performance. We assessed the impact of audio segment duration, data augmentation, type of dataset used for the neural network training, kind of speech tasks, and back-end analyses. X-vectors technique provided better classification performances than MFCC-GMM for text-independent tasks, and seemed to be particularly suited for the early detection of PD in women (7 to 15% improvement). This result was observed for both recording types (high-quality microphone and telephone)., Submitted

The term “SDN” represents a significant evolution in networking technology and draws attention and support from network operators, vendors, researchers and industry regulators. Contrary to popular belief that SDN is a recent invention, or is related only to IP technologies, the concept of network programmability and control/data plane separation has its roots in the 1960s and 1970s when the telephone network started its transition to digital. Many iterations followed, leading to applications in packet networks and today’s Internet. In this article, we review a number of early works on network programmability that illustrate how several features of SDN emerged progressively over several decades. Our review starts from the early concepts of network control in the telephone network and continues to examine a prolific period of research advancements in the 1990s and early 2000s that led to a number of startup companies that followed, IEEE’s own efforts in standardizing network programmability, and finally the arrival of the OpenFlow standard. We study the importance of this architectural transformation and its influences on modern cloud computing and next-generation networking.

Before focusing our interest on telecommunication networks, it is important to take a brief look at the history of telecommunications, referring to the most important steps, which are at the basis of modern transmissions of signals at distance.

Telephones are one of the most common telecommunication devices in the protected premises. The telephone includes various acoustic transducers (speaker, microphone), as a result of which these devices have a microphone effect and can cause leakage of speech information into the telephone network. This paper describes the model of the piezoelectric speaker of the Gigaset DA210 telephone connected to the buzzer circuit using simulations in COMSOL Multiphysics. The piezoelectric (Lead Zirconate Titanate 8) actuator is investigated, a comparison based on the obtained results of acoustic transducers is made with the parameters specified by the piezoelectric speaker designers. The paper studies the characteristics of this piezoelectric speaker as a sensor under the influence of an acoustic field, as a result of which a technical channel of speech information leakage channel. Based on the data obtained, the speech intelligibility assessment in the telephone network is made using the developed model. The results can be used in the development of secure telephones and the educational process in the training of specialists in the area of information security.

The 5G network phasing system is a remote phasing system that uses 5G network timing technology, wireless transmission technology, telephone network, ultrahigh voltage spark discharge processing, and phasing recognition to be applied to power systems. It can achieve ultra-high voltage and remote phasing in a few seconds. Measurement of distance, phase difference between power systems in different places, determination of phase, and calculation of phase angle. This system is mainly composed of server (platform layer) base station group (sensing layer), handheld terminal (sensing layer and application layer) and network layer (LAN/WIFI/5G network or electric power network). After testing, the system has been successfully applied in the power supply company's operation, maintenance, and technological transformation projects, and the application situation is explained in detail.

Being part of the so-called “Pacific Ring of Fire” and “Belt of Low Pressure,” Ecuador is located in one area most likely to suffer seismic, volcanic, and hydrometeorological threats, which make it possible to catalog as a country with a high vulnerability. For this reason, the use of alert systems such as Emergency Warning Broadcasting System EWBS is necessary for the face of one of these threats, which can be implemented in analog and digital broadcast signals such as Digital Terrestrial Television. In an emergency, the receivers compatible with the EWBS are currently the decoders or digital televisions that include this system. They turn on automatically and emit a visual and audible alert signal that gives time to the population to act more quickly to an event. This article presents the design proposal of a receiver that replicates the warning emergency alert EWBS for digital terrestrial television with the ISDB-Tb standard, through an institutional telephone PBX, implementing an IP telephony server that receives the EWBS system and replicates it to landlines, infocast systems, and mobile phones connected to it, thus avoiding a percentage of losses in the economic and human fields.

The division of a frequency band into dedicated channels by modulation, called frequency-division multiplexing (FDM), has a long history. Prominent examples are the analog telephone networks until the 1970s. They were replaced by digital telephony, operating with time-division multiplex (TDM), in which dedicated time slot are allocated to each application. Moreover, the analog and later the digital television broadcasting networks—terrestrial, satellite and cable based—operate with FDM. Most mobile and cellular networks today utilize combinations of FDM and TDM.

Encryption plays a major role in achieving privacy, especially in communications and information systems such as military systems, banks, telecommunications and individuals, and in mobile phone networks in particular. The (GSM) standard provides a level of privacy that may not be sufficient for some users, as this study is presented a proposed system for applying audio coding in cellular communication networks from the beginning (mobile) to the end (mobile (via the standard mobile communications system) GSM), and in this standard is used the A5 algorithm for audio coding is only for transmitting stations, while for the rest of the network it is without transmitting coding the data as it is, so we in this study proposed the application of RC4 algorithm with A5 algorithm through encryption keys controlled by the sender and receiver in audio coding from start to finish to ensure complete encryption and privacy in network phone calls and the tool used in this the research is a matlab program to create a simulated environment for the transmission between the sender and the receiver to add the RC4 algorithm for these stages and one of the results of the study using the RC4 algorithm in the transmission model in the telephone network can achieve greater privacy in telephone calls communication without affecting the sound quality through the two encryption keys controlled by both ends connection.

Voice over Internet Protocol (VoIP) is a technology that allows people to use the Internet, rather than a traditional telephone network. VoIP is becoming more and more popular due to its significant advantages. VoIP telephony has become the most developed and frequently used cellular service. It also created new threats to caller’s privacy. Due to the high degree of interest in ensuring anonymization of calls in the GSM network. Previous research on the development of algorithms to increase anonymization in IP-telephony and their implementation. In this work, we developed a new mechanism for achieving anonymity in the Global System for Mobile Communications (GSM - Global System for Mobile Communications). The lowest probability of anonymity disclosure in this algorithm is achieved by redirecting calls from one mobile operator to another. In addition, GPSS-Studio implements a model for redirecting incoming calls using multi-channel modeling. Probabilistic-temporal characteristics of the distribution of incoming calls to specific nodes.

Розвиток Інтернет технологій характеризується широкими можливостями впровадження різноманітних телекомунікаційних послуг, зокрема IP-телефонії, що надає користувачам нову технологію передачі мультимедійного трафіку з використанням Інтернет мереж. Для застосування цієї технології активно проваджуються спеціальні протоколи передачі даних – SIP та H.323, що надає можливість організовувати приватні та корпоративні мережі передачі даних. У статті проведений аналіз застосування даних протоколів на базі IP-станції Panasonic KX-NS500, що дало змогу надати рекомендації щодо підвищення ефективності функціонування внутрішньої телефонної мережі для передачі голосового трафіку при різних типах кодеків. Авторами розглядається задача підвищення ефективності функціонування мережі голосового зв’язку на базі IP-станції із застосуванням протоколів SIP та H.323. Запропоновано модель зв’язку, у якій використовуються три аналогові телефонні апарати та три IPтелефони, обслуговуванням яких займається ІР-станція. Телефонна лінія використовує протокол SIP. Для дослідження характеристик роботи запропонованої моделі мережі використовується програмне середовище OPNET Modeler. Визначено оптимальний тип голосового кодеку для протоколу SIP та можливості використання застосовуваного протоколу в організації внутрішньої телефонної мережі з використанням IP-станції. Проведено моделювання сегменту телекомунікаційної мережі із застосуванням протоколу SIP та проведено аналіз характеристик двох кодеків зв’язку, що найчастіше використовуються у технології VoIP. На основі проведеного моделювання здійснено оцінку якості мовного сигналу за рейтингом MOS, діапазону затримки мовного сигналу та вибір кодеку для відтворення голосового сигналу для запропонованої моделі мережі. The development of Internet technologies is characterized by wide possibilities for introducing various telecommunication services, in particular IP telephony, which provides users with a new technology for multimedia traffic transmittion using Internet networks. Special data transfer protocols (SIP and H.323) are being actively developed to apply this technology, which enables the organization of private and corporate data networks. The paper analyzes the use of these protocols based on the Panasonic KX-NS500 IP station, which has made it possible to give recommendations on improving the operation of internal telephone network for the transmission of voice traffic for various types of codecs. The article discusses the problem of improving the efficiency of IP based voice communications using SIP and H.323 protocols. A communication model, that uses three analog phones and three IP phones serviced by an IP station, is proposed. The telephone line uses SIP protocol. The OPNET Modeler software environment is used to investigate the performance of the proposed network model. The optimal type of voice codec for SIP protocol and the possibilities of using the protocol in the organization of internal telephone network using the IP station are determined. The telecommunication network segment using SIP protocol has been simulated and the characteristics of two communication codecs most commonly used in VoIP technology have been analyzed. Based on the simulation, the assessment of the quality of the speech signal by the MOS rating, the range of delay of the speech signal and the choice of a codec to play a voice signal for the proposed network model have been made

Classic public switched telephone networks (PSTN) are often a black box for VoIP network providers, as they have no access to performance indicators, such as delay or packet loss. Only the degraded output speech signal can be used to monitor the speech quality of these networks. However, the current state-of-the-art speech quality models are not reliable enough to be used for live monitoring. One of the reasons for this is that PSTN distortions can be unique depending on the provider and country, which makes it difficult to train a model that generalizes well for different PSTN networks. In this paper, we present a new open-source PSTN speech quality test set with over 1000 crowdsourced real phone calls. Our proposed no-reference model outperforms the full-reference POLQA and no-reference P.563 on the validation and test set. Further, we analyzed the influence of file cropping on the perceived speech quality and the influence of the number of ratings and training size on the model accuracy.

While the experienced quality of a conversation transmitted over a telephone network is dependent on the parameters of the network and possible degradations, it has been shown that the sensitivity to these degradations is influenced by the type of the conversation and its contents. Especially for delayed speech transmission the conversational scenario and the adaption of turn-taking behavior of the interlocutors factor into the conversational quality rating of a specific conversation. Parametric Conversation Analysis has been proven to be a good method to extract parameters from a recorded conversation that are representable of the interactivity. While the narrowband version of a popular and standardized quality prediction model, the E-model, uses the interactivity to predict a conversational quality MOS for the given conditions, there are no attempts to predict the conversational quality of individual conversations so far. In this paper, we propose a model to predict the quality of a conversation under the influence of transmission delay based on the interactivity parameters extracted from that conversation. We evaluate which parameters are most suited for such a prediction, and compare our results to the parameter-based predictions of the E-model.

One of the most frequently encountered auxiliary technical devices installed in a protected room is a telephone that created acoustoelectric leakage channels of the acoustic speech information. In the NI Multisim simulation environment, typical models of passive telephone protection devices have been designed. The device protects a telephone set of the speech information leakage arising from the acoustoelectric effect and «high-frequency injection». The study was carried out of the main characteristics of the developed protection device and an assessment of the performance when connected to a telephone network using the virtual laboratory bench.

When a disaster strikes, many people make calls to their loved ones in the affected area. As a result, telephone networks become congested, making it difficult for people to contact each other. This congestion continues in the aftermath of a large-scale disaster. The sooner the congestion is eased, the sooner and easier people can contact each other. We accomplish this by reducing communication demand without directly restricting call duration. We propose a congestion control method, called the road space rationing (RSR) method (RSR was originally designed for transportation and restricts access to congested areas on the basis of the last digit of a vehicle’s license plate or license number). Our RSR method only restricts the period in which to make a call on the basis of the last digits of calling parties’ phone numbers (e.g., only people with phone numbers ending in 1 can make calls between XX:06 and XX:12). It not only avoids restricting the overall number of calling parties but also prompts people to reduce their call duration consciously despite it being unrestricted. It thus has a mechanism to change human communication behavior without directly restricting call duration. This consciously willed reduction is effective against disaster congestion because there are no redials to the same called parties. This reduction in call duration in turn decreases congestion. Experimental results revealed that the RSR method reduced call duration by 30%.

Evaluation the disruptions of telephone network uses a method fault tree analysis (FTA) and failure mode and effect analysis (FMEA). By using FTA, the fault events are identified, which contribute causing top level event of jarlokat isolation, jarlokaf isolation, and devices can’t function jarlokar. The results of FTA; 8 basic events for jarlokat isolation, 7 basic events for jarlokaf isolation, and 6 basic events for jarlokar devices can’t function, that is smallest events. These basic events are analyzed by FMEA it based on value of the risk priority number by knowing severity from failure effect scale, occurrence from causes scale, and detection from control scale. The result FMEA is priorities problem solving in the following order: the drop wire 1x2 broken, the short circuit of housing wire, the short circuit of distribution point, the trans radio broken, the telephone instrument damaged, the primary wire broken, and the secondary wire broken.

This work describes the process of collection and organization of a multi-style database for speaker recognition. The multi-style database organization is based on three different categories of speaker recognition: voice-password, text-dependent and text-independent framework. Three Indian institutes collaborated for the collection of the database at respective sites. The database is collected over an online telephone network that is deployed for speech based student attendance system. This enables the collection of data for a longer period from different speakers having session variabilities, which is useful for speaker verification (SV) studies in practical scenario. The database contains data of 923 speakers for the three different modes of SV and hence termed as multi-style speaker recognition database. This database is useful for session variability, multi-style speaker recognition and short utterance based SV studies. Initial results are reported over the database for the three different modes of SV. A copy of the database can be obtained by contacting the authors.

Hearing loss is an invisible but significant barrier in daily life, including telephone conversations. Internet Protocol Caption Telephone Services (IP CTS) is a telecommunications relay service for an individual who can speak, but who has difficulty hearing over the telephone. An individual can use a telephone and an IP-enabled device to listen to the other party and simultaneously read transcriptions of the other party’s words. This article presents the results from a usability assessment of IP CTS devices and alternative speech recognition technologies to provide qualitative and quantitative measures for device and caption performance. The results demonstrate that there is at least one automatic Speech-To-Text (STT) engine that is equivalent or better than three of the four currently available IP CTS devices at producing an accurate, expedient, and usable service.

The paper provides an overview of development and use of Internet technologies in improving business by integrating different types of communication. The advantages of using the standard open IP - Internet Protocol in establishing integrated communication are presented as some of the potential possibilities of upgrading the existing conventional telephone network. The paper presents a concrete implementation of internet telephony in business premises in order to reduce operating costs, as well as the analysis of the possibilities of applying new multimedia applications and services by using a single integrated network. Comparison of performance of the original, sent speech signal and the received speech signal, after passing through the IP network, for different values of packet loss rate is analyzed.

The constant growth of the number of Internet users places new demands on the bandwidth of modern communication. The World Wide Web has led to the emergence of various types of traffic, Graphics information, voice data, as well as various video applications demanding their special requirements for such networks. To meet all the requests, one increase of the network is not enough. As the number of Internet users and the various networks annex offices in-creases with each passing day, network needs a means of control that would ensure the support of both existing and new applications and services. There is still a need to address a number of issues before it announces the VoIP as the ultimate replacement of the traditional telephone network. This work provides an overview of the major benefits of VoIP network, as well as the presentation of a new way around the problem of ensuring the high quality of service for VoIP protocol on the basis of neural network model.

Graph clustering is a technique for grouping vertices having similar characteristics into the same cluster. It is widely used to analyze graph data and identify its characteristics. Recently, a large-capacity large-scale graph data is being generated in a variety of applications such as a social network service, a world wide web, and a telephone network. Therefore, the importance of clustering technique for efficiently processing large capacity graph data is increasing. In this paper, we propose a clustering algorithm that efficiently generates clusters of large capacity graph data. Our proposed method efficiently estimates the similarity between clusters in the graph using Min-Hash and generates clusters according to the calculated similarity. In the experiment using real world data, we show the efficiency of the proposed method compared with the proposed method and existing graph clustering methods.

Online security and security of personal information is an important topic in information technology. Due to the recent high-profile security breaches as a result of authentication manipulation, there has been a growing need for enhanced methods of authentication as attackers have become more skilled in defeating existing authentication mechanisms. Attackers have resulted to phishing, social engineering, token interception and other highly sophisticated attackers all to gain access to the accounts of user. These attacks have resulted in serious damages for individuals and organizations. As a result, methods, such as, Two-Factor Authentication (2FA) have been introduced to. Nevertheless, loopholes major vulnerabilities in the existing two-factor authentication implementation. In this paper, This paper analyzes the. The are listed and analyzed, and the methods of manipulation explained. This paper finally suggests a solution of a web platform service which utilizes the mobile phones of users as a form of secure confirmation of the identity of the user through a dedicated mobile application. This solution hopes to eliminate the need for communication of security parameters of the traditional telephone network.

A Short Message Service (SMS) is a tool for personal communication systems using ubiquitous telephone networks. SMS also has been used for sharing information between intelligent devices, but the transferable payload size of a single SMS message is limited and can constrain its application. For example, when sharing context information between devices, a message size of more than 140 bytes needs be divided by the sender and reassembled by the receiver to fit into SMS messages. Although efforts have been made to reduce the context size using compression algorithms, their effectiveness is unreliable and their size reduction is, in many cases, not enough. We present SEMS (Size Efficient Messaging System) to reduce the footprint of context information to overcome this limitation. Using a probabilistic data structure, SEMS can significantly increase context representation size. SEMS also uses SMS to deliver the encoded message with far less overhead. Despite a slight potential decrease in data quality, our system and findings are shown to facilitate the sharing of context information in a highly size-efficient manner.

This document extends PASSporT, which is a token object that conveys cryptographically-signed information about the participants involved in communications. The extension is defined, corresponding to the SHAKEN specification, to provide both a specific set of levels-of- confidence in the correctness of the originating identity for a SIP based Communication Service Provider (CSP) telephone network originated call as well as an identifier that allows the CSP to uniquely identify the origin of the call within its network.

Modern stage of development of information and communication networks requires solving of crucial tasks of network traffic statistical analysis and traffic simulation modeling. The predominance of non-Poisson traffic leads to the impossibility of analyzing multichannel communication systems by the methods of queuing theory that used to describe telephone networks. The last decade has paid much attention to research on traffic that has signs of self-similarity. The main purpose of this work is a statistical analysis of non-Poissonian traffic, represented by multimodal non-standard Pascal (negative binomial) and Rice distributions. As a result, a study of the self-similarity degree has been performed by the R/S analysis and the aggregation method. In addition, we propose to use EM-algorithm with an algorithm for determining an optimal number of clusters for an approximation of non-typical multimodal distributions.

A huge number of calls are generated as a means of confirming the safety of others or communicating with the police, fire department, etc. in an emergency, such as an earthquake or fire. Therefore, phone congestion occurs and it is difficult to connect all calls over a local telephone network. As countermeasures against congestion, to increase the number of calls accommodated in an emergency over a circuit-switched network, the trunk reservation control method for reserving certain lines exclusively for priority calls used by public institutions and the holding time control method for general calls have been proposed. These effectiveness of these methods has been demonstrated in the past. However, when using these controls, the call blocking probability for general calls is high, meaning it is necessary to further reduce call loss probability. In this paper, we propose a trunk reservation control method that facilitates call waiting to reduce call loss probability. In comparison to the conventional method without a waiting queue, the proposed method decreases call loss probability for general calls while maintaining a low call loss probability for emergency calls.

A telecommunication system is vital for the operation and control of large electric power systems such as present and future multi-terminal HVDC networks. The latency in data transmission, and the variation in latency, known as jitter, are one of the most important features, in addition to reliability and security, to be considered in the modelling of telecommunication networks when assessing the overall performance of electric power systems. Further, in switched packet telecommunication systems, such as the internet, data packets arriving out of order might occur. Based on the behaviour of switched telecommunication networks and considering the latency as the main component to be represented, in this paper, a telecommunication model is developed and a user defined component is built in PSCAD/ EMTDC using the FORTRAN 90 language. Such a model is suitable to represent telecommunication networks with variable and constant latency as well as switched packet telecommunication systems, i.e. the internet and telephone networks. Results obtained from the application of the telecommunication model to the control a multi-terminal VSC HVDC network, interfacing offshore wind power generation, are then presented and discussed in the paper.

Exploring innovative cellular architectures to achieve enhanced system capacity and good coverage has become a critical issue towards realizing the next generation of wireless communications. In this context, this paper proposes a novel concept of Universal Intelligent Small Cell (UnISCell) for enabling the densification of the next generation of cellular networks. The proposed novel concept envisions an integrated platform of providing a strong linkage between different stakeholders such as\ud street lighting networks, landline telephone networks and future wireless networks, and is universal in nature being independent of the operating frequency bands and traffic types. The main motivating factors for the proposed small cell concept are the need of public infrastructure re-engineering, and the recent advances in several enabling technologies. First, we highlight the main concepts of the proposed UnISCell platform. Subsequently, we present two deployment scenarios for the proposed UnISCell concept considering infrastructure sharing and service sharing as important aspects. We then describe the key future\ud technologies for enabling the proposed UnISCell concept and present a use case example with the help of numerical results. Finally, we conclude this article by providing some interesting future recommendations.

La telefonía es un servicio global, y por ello las redes telefónicas han sido un objetivo codiciado de los ciberdelincuentes. Ahora que la voz se puede transportar a través del protocolo IP y que múltiples servicios se integran en un modelo convergente mediante Internet, los incentivos para atacar los sistemas telefónicos y los atacantes son, sin duda, más numerosos. Además, el desarrollo de aplicaciones telefónicas basadas en software libre ha permitido la masificación del uso de telefonía IP, sin que la conciencia sobre los riesgos de seguridad inherentes se haya incrementado. En vista de la vigente e intensiva adopción de sistemas de telefonía IP en el Ecuador, se realizó una exploración basada en información pública para determinar estadísticas sobre los sistemas telefónicos conectados a Internet en Ecuador. En base a estos datos y a los recopilados por un prototipo de telefonía IP deliberadamente vulnerable, se realizó un análisis preliminar de las vulnerabilidades y amenazas de estos sistemas telefónicos. Se encontraron cientos de sistemas telefónicos públicamente disponibles en Internet, muchos con versiones desactualizadas y, por tanto vulnerables, de soluciones de telefonía IP basadas en Asterisk. En pocos días, se identificaron miles de interacciones maliciosas sobre el sistema de telefonía IP publicado en Internet en Ecuador.

This work presents the development of a multi-level speech based person authentication system with attendance as an application. The multi-level system consists of three different modules of speaker verification, namely voice-password, text-dependent and text-independent speaker verification. The three speaker verification modules are combined in a sequential manner to develop a multi-level framework which is ported over a telephone network through interactive voice response (IVR) system for aiding remote authentication. The users call from a fixed set of mobile handsets to verify their claim against their respective models, which is then authenticated in a multi-level mode using the above stated three modules. An analysis over a period of two months is shown on the performance of the multi-level system in attendance marking. The multi-level framework having combination of the three modules helps in achieving better performance than that of the individual modules, which shows its potential for practical deployment.

over internet protocol (VoIP) is a method of providing phone services over dedicated public IP networks. It allows significant cost savings over traditional Public Switched Telephone Networks (PSTN). Speech quality, as perceived by the users of VoIP telephony, is critically important. Signal quality of the VoIP system is degraded by various network layer problems, which include delay, packet loss and jitter. The implementation of signal through various coding algorithms and digital signal processor can improve the quality of degraded VoIP signal. The present work deals with comparative study and analysis of VoIP Codecs (G.711, G.729, AMR, AMR-WB etc.). The VoIP simulations are conducted for G.711, G.729 and AMR-WB speech coders for different network conditions. The coding algorithms are implemented on VoIP speech signal. The results are validated through the measurement of enhancement signal using perceptual evaluation of speech quality (PESQ)/MOS measurement.

I used a subscriber identity module card reader to copy the lists of saved contacts from 170 mobile phones in Fiji. This approach has both advantages and disadvantages compared to other techniques for collecting telephone network data. Copying phone contacts avoids recall biases associated with survey-based name generators. It also obviates the need to obtain calling records from network service providers or to provide research subjects with specialized phones. In markets with multiple service providers, or in contexts in which mobile phones are shared extensively, the contents of contact lists may represent personal networks more accurately than actual calling records do. However, owners’ varying durations of phone ownership and varying tendencies to save and delete contacts are sources of error. Also, copying contacts from phones relies on self-reports of tie strength, which are frequently inaccurate. I analyze the data I collected in rural Fiji to evaluate these sources of error.

A mobile telephone system is designed to facilitate better communication among 8000 users in the centre of Roma. The network performance prediction tool was used to determine the best location for deployment of the base station. It was based on some parameters or predictions which were obtained by calculating needed data to construct the base station. Theoretically call planning, cell splitting, frequency re-use, offered traffic etc. techniques are used to obtain a good coverage for a certain city. In order to find the best cell cluster, many calculations were also implemented. The objective achieved was designing a network system in city centre of Roma. The 7 cell cluster and 12 cell cluster was selected for its overall efficiency. In addition to this, co-channel interference, best server, maximum received power etc. were examined for deployment of the base station with help of Winprop.