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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.

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.

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.

The robocall epidemic has caused millions of phone scam victims resulting in billions of financial loss in the USA. To address this issue, Federal Communication Commission (FCC) mandates all Internet Telephony Service Providers (ITSP) to implement Secure Telephony Identity Revisited (STIR) with the Signature-based Handling of Asserted information using toKENs (SHAKEN) to authenticate Voice of Internet Protocol (VoIP) calls on their networks. This chapter provides an analysis of the effectiveness of STIR/SHAKEN in protecting users from being victims of robocalls which are mostly scam calls with fake caller ID. Our analysis shows three majors issues that could have impact on the effectiveness of STIR/SHAKEN. These issues are (a) poorly protected enterprise IP-PBX, (b) unscrupulous service providers, and (c) lack of support of Q.1912.5, which is the interworking standard between IP and Public Switch Telephone Network (PSTN).

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.

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.

Tema članka je razvoj javne nepokretne i pokretne telefonije u Hrvatskoj pod okriljem novoosnovane nacionalne javne poštansko-telekomunikacijske tvrtke Hrvatska pošta i telekomunikacije od 1990. pa sve do njezina razdruživanja na dvije zasebne tvrtke: Hrvatska pošta d.d. i Hrvatske telekomunikacije d.d. početkom 1999. godine. Hrvatska je u razvoju javne telefonske mreže znatno zaostajala za razvijenim europskim zemljama, a izbijanje Domovinskoga rata i oružana agresija na Republiku Hrvatsku izložili su ju mogućnosti telekomunikacijske i informacijske blokade. Budući da su se građani Hrvatske opredijelili za tržišnu ekonomiju, politički pluralizam i demokratsko društveno uređenje, u načelu se podrazumijevala i poticala sloboda neposrednoga komuniciranja, a ona je, telekomunikacijski gledano, na razini pojedinca ponajviše bila izražena u dostupnosti domaćih i međunarodnih javnih telefonskih veza. U obrađenom su razdoblju na nacionalnom planu napravljeni ključni infrastrukturni zahvati i uvedena je suvremena tehnologija s novim telekomunikacijskim uslugama, što je omogućilo da Hrvatska prestane biti telekomunikacijski nerazvijena, a na međunarodnom planu postala je prepoznatljiv telekomunikacijski partner te dio tranzitnoga telekomunikacijskog mosta između istočne i zapadne te sjeverne i južne Europe., After the independence of the Republic of Croatia was declared in 1991, the newly-formed national postal and telecommunications company HPT (Croatian Post and Telecommunications) found itself in an unenviable position. Croatia was in the group of countries at the forefront of telecommunications development in Europe. In 1990, it had only 17 telephone connections per 100 inhabitants. However, in the next eight years it moved closer to the European average, doubling its capacity to 35 telephone lines per 100 inhabitants by 1999. It did so despite the devastation of the Homeland War and the enormous damage suffered by its public telephone network. Croatia managed to negate the telecommunications blockade by finding alternative telecommunication routes, and then set out to build an efficient telephony and transmission system based on the use of fibre-optic cables and modern switching equipment. By the end of 1993, international switching hubs in Zagreb, Split, Rijeka, and Đakovo using AXE 10 technology were built, and the following year cities on the Croatian Adriatic coast were connected with cities in continental Croatia by fibre-optic cable. During this process, Croatia received its one-millionth telephone subscriber. Thanks to its participation in the TEL (Trans Europe Line) project, it became part of the telecommunication bridge between Eastern and Western Europe. In addition to the fixed public telephone network, it simultaneously developed a mobile public telephone network, first an analogue mobile network aptly named MOBITEL in 1991, and then the CRONET digital mobile network in 1996. These steps resulted in a significant increase in telephone traffic and the widespread availability of fixed and mobile telephone connections, which completely revolutionised public telephony in Croatia.

The internet and other digital networks in the U.S. (and beyond) remain essentially an upgrade of the national telephone network built by AT&T over the first half of the twentieth century, and digi...

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.

Developments in the EU telecommunications markets require a recurrent redesign of the regulatory framework for telecommunications services. In this regard, the analysis of the substitution effects between different types of telephony is the cornerstone of market definition and therefore of effective regulation. This paper explores the access substitution between fixed-lines, mobiles, and managed VoIP in a unified EU cross-country framework. We employ a half-yearly dataset for 20 EU countries for the 2008–2011 period and apply dynamic panel data methods. Our analysis demonstrates strong access substitution between fixed-lines and mobiles and provides indicative evidence on the substitution between fixed-lines and VoIP. Overall, we find evidence in favor of access substitution and therefore of joint market definition. Regulatory obligations imposed on the market for access to fixed telephone networks might therefore be redundant.

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.

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.

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.

The global telephone network is relied upon by billions every day. Central to its operation is the Signaling System 7 (SS7) protocol, which is used for setting up calls, managing mobility, and facilitating many other network services. This protocol was originally built on the assumption that only a small number of trusted parties would be able to directly communicate with its core infrastructure. As a result, SS7 — as a feature — allows all parties with core access to redirect and intercept calls for any subscriber anywhere in the world. Unfortunately, increased interconnectivity with the SS7 network has led to a growing number of illicit call redirection attacks. We address such attacks with Sonar, a system that detects the presence of SS7 redirection attacks by securely measuring call audio round-trip times between telephony devices. This approach works because redirection attacks force calls to travel longer physical distances than usual, thereby creating longer end-to-end delay. We design and implement a distance bounding-inspired protocol that allows us to securely characterize the round-trip time between the two endpoints. We then use custom hardware deployed in 10 locations across the United States and a redirection testbed to characterize how distance affects round trip time in phone networks. We develop a model using this testbed and show Sonar is able to detect 70.9% of redirected calls between call endpoints of varying attacker proximity (300–7100 miles) with low false positive rates (0.3%). Finally, we ethically perform actual SS7 redirection attacks on our own devices with the help of an industry partner to demonstrate that Sonar detects 100% of such redirections in a real network (with no false positives). As such, we demonstrate that telephone users can reliably detect SS7 redirection attacks and protect the integrity of their calls.

Telephone networks first appeared more than a hundred years ago, long beforetransistors were invented. They, therefore, form the oldest large scale networkthat has grown to touch over 7 billion people. Telephony is now merging manycomplex technologies and because numerous services enabled by these technologiescan be monetized, telephony attracts a lot of fraud. In 2015, a telecom fraudassociation study estimated that the loss of revenue due to global telecom fraudwas worth 38 billion US dollars per year. Because of the convergence oftelephony with the Internet, fraud in telephony networks can also have anegative impact on security of online services. However, there is littleacademic work on this topic, in part because of the complexity of such networksand their closed nature. This paper aims to systematically explorefraud in telephony networks. Our taxonomy differentiates the root causes, thevulnerabilities, the exploitation techniques, the fraud types and finally theway fraud benefits fraudsters. We present an overview of eachof these and use CAller NAMe (CNAM) revenue share fraud as aconcrete example to illustrate how our taxonomy helps in better understandingthis fraud and to mitigate it.

SummaryBackground: Discovery of clinical workflows to target for redesign using methods such as Lean and Six Sigma is difficult. VoIP telephone call pattern analysis may complement direct observation and EMR-based tools in understanding clinical workflows at the enterprise level by allowing visualization of institutional telecommunications activity.Objective: To build an analytic framework mapping repetitive and high-volume telephone call patterns in a large medical center to their associated clinical units using an enterprise unified communications server log file and to support visualization of specific call patterns using graphical networks.Methods: Consecutive call detail records from the medical center’s unified communications server were parsed to cross-correlate telephone call patterns and map associated phone numbers to a cost center dictionary. Hashed data structures were built to allow construction of edge and node files representing high volume call patterns for display with an open source graph network tool.Results: Summary statistics for an analysis of exactly one week’s call detail records at a large academic medical center showed that 912,386 calls were placed with a total duration of 23,186 hours. Approximately half of all calling called number pairs had an average call duration under 60 seconds and of these the average call duration was 27 seconds.Conclusions: Cross-correlation of phone calls identified by clinical cost center can be used to generate graphical displays of clinical enterprise communications. Many calls are short. The compact data transfers within short calls may serve as automation or re-design targets. The large absolute amount of time medical center employees were engaged in VoIP telecommunications suggests that analysis of telephone call patterns may offer additional insights into core clinical workflows.Citation: Rucker DW. Using telephony data to facilitate discovery of clinical workflows. Appl Clin Inform 2017; 8: 381–395 https://doi.org/10.4338/ACI-2016-11-RA-0191

BACKGROUND Telecommunication is limited or even impossible for more than one-thirds of all cochlear implant (CI) users. OBJECTIVE We sought therefore to study the impact of voice quality on speech perception with voice over Internet protocol (VoIP) under real and adverse network conditions. METHODS Telephone speech perception was assessed in 19 CI users (15-69 years, average 42 years), using the German HSM (Hochmair-Schulz-Moser) sentence test comparing Skype and conventional telephone (public switched telephone networks, PSTN) transmission using a personal computer (PC) and a digital enhanced cordless telecommunications (DECT) telephone dual device. Five different Internet transmission quality modes and four accessories (PC speakers, headphones, 3.5 mm jack audio cable, and induction loop) were compared. As a secondary outcome, the subjective perceived voice quality was assessed using the mean opinion score (MOS). RESULTS Speech telephone perception was significantly better (median 91.6%, P 15%) were not superior to conventional telephony. In addition, there were no significant differences between the tested accessories (P>.05) using a PC. Coupling a Skype DECT phone device with an audio cable to the CI, however, resulted in higher speech perception (median 65%) and subjective MOS scores (3.2) than using PSTN (median 7.5%, P

A fixed telephone cluster management system was investigated in this paper. To meet the development demands of fixed telephone service and management, we designed a fixed telephone cluster management system based on browser/server structure which is developed by the MVC (Model-View-Controller) design pattern. This system attains independent inquiry and management for personal fixed-phone information and it has advantages over scalability, portability and maintainability. As a result, this system can applied in fixed telephone management for the corporation, the school, and so on.

The telephone is the most important means of communication in business. The manner in which employees speak to their clients and customers plays an important part in building an image of a company. A telephone conversation is a complex process with special challenges. When all parts of the communication process work effectively together, telephone communication is clear and useful. It is imperative to know all the rules while talking on a phone for business purposes.

The intelligent telephone alarm system is designed with a single chip microcomputer as the core of system,and the intelligent system is controlled by AT89C51. A telephone network is used as transmission of voice digital information in the system. The telephone alarm is composed of a single-chip microcomputer control, sensor and signal input circuit, and DTMF send and receive circuit, voice controlling and receiving circuit, alarm telephone circuit. According to the different sensor output signal, it can send out different alarm information to realize remote monitoring function.

The Telecommunications Sector Assessment Note in the Palestinian territories is a knowledge product prepared by the World Bank in response to a specific request from the Palestinian Ministry of Telecommunications and Information Technology (MTIT) to assess the performance of the telecommunications sector, identify specific issues and make recommendations for further development and reform.

Fixed and mobile telephony with other telecommunication services are moving to a fundamentally different infrastructure as the transition proceeds from the public telephone network to the Internet. There are many opportunities to make significant progress on major developments such as separation of operator independent functions, modular design, and the integration of other desirable features of sustainable Internet solutions. This progress also means that the impacts of some shortcomings of currently competitive telephony services will decline—another benefit of the transition. However, because telephony services are essentially being “emulated” over the Internet infrastructure, other aspects of these services will be changed in ways that could affect the public good. In particular, trust-related issues such as authentication and validation, along with their business implications, present significant challenges. This paper explores these issues, including the opportunity that industry and the regulators now have to address them, and suggests approaches to sustainable solutions that can benefit all the stakeholders in this transition.

In this paper, based on the traditional fixed telephony features to increase the short message processing module. To meet the voice communication between users, the fixed telephone can realize mutual development between the SMS function .The main telephone lines in the completion of the CAS signal detection, and traditional communications in FSK and DTMF signal reception and transmission.

CTI is a new technology combines communication technology with Internet. The Telephone Voice System is typical application of CTI. This paper presents the design and realization of a Telephone Voice System, utilizing D081A voice boards of DongJin Company. System structure, module di-vision and realization procedures are demonstrated. The system code is compiled with VC++, and database adopts SQLite. The system realizes recording, playback and other basic functions, and network calls function is extended. Users can operate the system with telephone, which is simple and convenient.

In recent years, Internet Protocol (IP) telephony has been a real alternative to the traditional Public Switched Telephone Networks (PSTN). IP telephony offers more flexibility in the implementation of new features and services. The Session Initiation Protocol (SIP) is becoming a popular signalling protocol for Voice over IP (VoIP) based applications. The SIP proxy server is a software application that provides call routing services by parsing and forwarding all the incoming SIP packets in an IP telephony network. The efficiency of this process can create large scale, highly reliable packet voice networks for service providers and enterprises. We established that the efficient design and implementation of the SIP proxy server architecture can enhance the performance characteristics of a SIP proxy server significantly. Since SIP proxy server performance can be characterised by its transaction states of each SIP session, we emulated the M/M/1 performance model of the SIP proxy server and studied some of the key performance benchmarks such as average response time to process the SIP calls, and mean number of SIP calls in the system. We showed its limitations, and then studied an alternative M/M/c based SIP proxy server performance model with enhanced performance model and studied additional key performance characteristics such as server utilisation, queue size and memory utilisation. Provided the comparative results between the predicted results with the experimental results conducted in a lab environment.

The early growth of telephone networks offered business a powerful new tool, and had the potential to reshape the working environments of offices and business districts in major cities. From the 1880s, the telephone became a favoured means of communication for firms conducting large numbers of routine transactions, especially in trading, shipping and related intermediary activities. However, senior businessmen continued to prefer face-to-face contact with their peers, and insisted on having offices close to trading exchanges, banks and other traditional focal points where they could meet, negotiate and network in person. In many cases, they also delegated telephone use to their subordinates. Early business telephony therefore tended to reinforce rather than undermine office hierarchies and business clustering alike.

Internet telephony is promising for long-distance calls. To serve the general public, a service provider can operate a telephone gateway in each servicing city to bridge the local telephone network and the Internet, so that users can use telephones or mobile phones to access these telephone gateways for long-distance calls. The service provider should serve many cities to attain good service coverage but it is costly to operate telephone gateways in all of these cities. In this paper, we propose a sparse telephone gateway configuration to serve many cities cost-effectively. Using this configuration, if the service provider serves N cities, it operates telephone gateways in only M cities where M

The Power Utility of Serbia Company intends to migrate portions of its time-division-multiplexing- based telephony network to IP. The high-level, capital project requirements include: (a) building a new VoIP infrastructure; (b) exchanging approximately one-third of the existing TDM-based PBXs with new VoIP based and hybrid PBXs; and (c) integrating these new PBXs with the remaining existing legacy PBXs to create a new hybrid network. Telephony service availability should be five nines (99.999 percent), a requirement that created a derived requirement for network element redundancy. One of the most important elements is the softswitch. The softswitch is the equipment responsible for call control of IP phones and voice gateways at the level of the Power Utility of Serbia Company telephone network. Softswitch redundancy in a failover configuration proved to be most challenging because of a performance requirement to achieve a switchover time of six seconds or less, whereas commercially available softswitches typically support switchover times of a few minutes. Another significant challenge was integrating the legacy PBXs and new (VoIP) PBXs. Although industry marketing promotes SIP trunking as the preferred integration/interconnection method, we found that with the exception of bearer traffic (voice media) pathing, service interoperability between the equipment of different vendors was often problematic. Vendors typically suggest Q.Sig as an interconnection alternative when SIP trunking is infeasible, but some vendor interoperability problems persisted. This article presents technical solutions to these major issues that can benefit similar TDM-to-IP migration and hybridization projects.

Recently, real-life data have revealed that the number of calls originating from or received by a telephone number in a network follows a power-law distribution. They show that a few telephone numbers make or receive a very large number of calls, whereas a large number of telephone numbers make or receive very few calls. The data have overthrown the general assumption that all telephone numbers are similar in generating telephone traffic. The first objective of this paper is to therefore construct a telephone call network (TCN) with connection properties following power-law distributions. With a more realistic TCN, researchers and engineers will be able to evaluate the telephone traffic behavior more accurately. Having constructed the aforementioned TCNs, we then consider the scenario when there is a sudden surge in the number of telephone calls, for example, during natural or man-made disasters. Under such a condition, the telephone network is usually overloaded and cannot operate properly. To mitigate the problem, we propose a preferential call blocking (PCB) scheme, aiming at blocking calls to target telephone numbers which have large numbers of incoming calls (in-strengths). We will investigate the effect on the carried traffic intensity when the PCB scheme is applied. We will compare the results with a benchmark, which corresponds to the case when all calls are blocked with equal probability. For the sake of completeness, we will also study the effectiveness of the blocking schemes when applied to a traditional TCN, in which all telephone numbers can call one another with equal probability.

In this paper, the validity of recommendation ITU-R P.1546 in a short-range terrestrial environment is analyzed. Its three versions (P.1546, P.1546-1, and P.1546-2) are compared against simple models and evaluated using measurement results that were obtained by utilizing the pilot signal of a commercial code-division multiple-access mobile telephone network. Measurement results show that P.1546-2, on average, underestimates the field strength by more than 10 dB for typical Australian rural areas. However, it improves the error standard deviation compared to previous versions. The causes of these effects and the suggestions for further development of the Recommendation are discussed and evaluated.

This article seeks to develop a richer understanding of the suitability of IP telephony in the Public Switched Telephony Network (PSTN) with an actual local exchange carrier (LEC)’s case and network simulation. We also performed a simple real options analysis to evaluate a telecommunications network. The underlying network and associated data were derived from our studies of an actual LEC. The preliminary result shows that an IP telephony network would save about 73% of the total link capacity of a circuit switched network and it could also carry some integrated services traffic at low incremental cost. The value of the IP telephony network was shown as positive under real options and as negative using net present value (NPV). We measured deferrable and irreversible value as well as the uncertainty value of an IP telephony network. Through this experiment, we can get inference that IP has potential for a voice carriage.

In this article, we share our experience of a trial that attempted to gain practical experience in migrating telephony services from the public service telephone network (PSTN) to the IP multimedia system (IMS) domain. We point out some key challenges on the migration path to IMS and identify many issues for successful service deployment in an IMS-based next generation network. Some regulatory concerns and non-technical issues also are addressed.

The rapid growth of smart phones and recent advances in telephony have triggered an enormous increase in the number of telephony subscribers. Some people have more than one calling identity for different purposes, such as for business and personal use. Discovering multiple identities of an individual can be important for fraud detection, detection of coordinated criminals, collaborative recommendations, and analyses of complete social graph of an individual. To address the challenge of finding identities that belong to a single individual, this paper describes ID-CONNECT, an approach for linking multiple identities of an individual based on that individual's social call graph and call behavior. The approach is based on the assumption that similarity between identities can be better estimated by using both call network and call behavior of identities towards their friends and that the more similar the weighted call graph of two identities, the higher is the similarity between them. ID-CONNECT is a two-step approach: 1) it estimates similarity between identities by incorporating call rate and duration, and 2) it generates sets of candidate identities for each target identity. We experimentally evaluate our approach on different random graph models and compare its performance against other approaches. Our evaluation results show that our approach significantly reduces candidate set size (up to 70%) while achieving comparable true positive rate.

In the design, the MCU AT89C52 is the core of the system. The system can send and receive the DTMF signal, which can identify the city telephone number or the long-distance telephone number. According the number, it can billing and show call duration and call cost. It can realize automatically the conversion between the full price and the discount price. It can billing accurately and conveniently query ticket information. It is configurable product and resets rate by pressing a button. It is low cost and low power consumption.

Internet telephony is promising for long-distance calls because of its low service charge and value-added functions. To provide Internet telephony to the general public, a service provider can operate a telephone gateway in each servicing city to bridge the local telephone network and the Internet, so that users can use telephones or fax machines to access this gateway for services. In this paper, we propose a dynamic bandwidth allocation scheme for two purposes: (1) each telephone gateway can fully utilize the available bandwidth to serve more telephone and fax sessions and (2) it can respond to the changing environments. We exploit three properties for dynamic bandwidth allocation. First, in a telephone session, each user usually alternates between speaking and listening. When a user is not speaking, she does not send any voice stream and hence the bandwidth can be dynamically released from this session for the other sessions. Second, voice traffic is elastic because it can be further compressed at the cost of a lower quality. Third, fax traffic is flexible because it can be temporarily delayed. We exploit these three properties to allocate bandwidth to telephone and fax sessions dynamically. When a telephone gateway adopts dynamic bandwidth allocation, it can serve more telephone and fax sessions while providing acceptably good quality-of-service (QoS), and it can give more stable QoS when the available bandwidth varies.

The restricted audio quality of today's telephone networks is mainly due to the narrowband (NB) limitation to the frequency range from about 300 Hz to 3.4 kHz. Meanwhile, codecs for wideband (WB) telephony (50 Hz to 7 kHz) exist with significantly improved speech intelligibility and naturalness. However, the broad introduction of wideband speech coding requires strong efforts of both network operators and their customers because many elements of the networks (i.e., terminals and network nodes) have to be modified. An intermediate step to overcome the narrowband limitation can be achieved by applying artificial bandwidth extension (BWE) in the receiver. In this article we review the basic principles of bandwidth extension, and discuss several application scenarios in which both wideband coding and BWE complement each other. The introduction of BWE methods in terminals and networks may help to speed up the introduction of true wideband speech coding in the near future

Many law enforcement wiretap systems are vulnerable to simple, unilateral countermeasures that exploit the unprotected in-band signals passed between the telephone network and the collection system. This article describes the problem as well as some remedies and workarounds.

In this paper, we consider the evolution of telephone networks from time-division multiplexing circuit switching to packet switching and, in particular, to packet switching-based on Internet Protocol (IP-supported telephony). We analyze IP-supported telephony design solutions by proposing a layered reference model in which each layer is associated to a subset of the functions that support telephony. We use the reference model to establish a terminology and a framework for the comparison of the design solutions. We group the design solutions in scenarios and compare them in terms of the reference model proposed. We then focus on IP telephony, in which IP is used in telephone company networks, and on Internet telephony, in which the Internet is used to support telephony. We show that they both can be seen as implementations of the same architecture, which consists of a set of components, associated to functions, and of the interactions among these components. We then consider the issue of voice-data integration and analyze the variety of design solutions that can be adopted to integrate voice and data.

As mobile operators worldwide migrate to third-generation (3G) networks, conversational video-telephony services are becoming a key differentiator between new 3G offerings and existing 2G/2.5G services. Although it's possible to have limited video-based services - such as a multimedia messaging service - that deliver pictures and video clips over 2.5G services, these are delay-insensitive applications that could run over a packet-based wireless network like general packet radio service (GPRS) or code division multiple access (CDMA)'s 1XRTT. For delay-sensitive applications such as conversational video telephony, present 3G packet bearers are inadequate, and the Third Generation Partnership Project (3GPP; http://www.3gpp.org) mandates using the 3G bandwidth-guaranteed circuit-switched bearer and the 3G-324M system. The 3G-324M system is a derivative of the International Telecommunication Union (ITU) H.324 protocol standard for low-bitrate multimedia communication, which ITU-T developed for the public switched telephone network (PSTN). This article describes the 3G-324M system, which has been adopted by both 3GPP and 3GPP2 (htpp://www.3gpp2.org), as well as its H.324 roots.

With the rapidly spreading use of IP networks such as intranets and the Internet, there have been efforts to develop voice-over IP (VoIP) devices and services by utilizing them. In particular, in the commercial user environment, wide use is likely to be made of services with a high added value in which the telephone network function and the IP network connection function are combined by fully utilizing a high-level user interface provided by an information processing terminal. This paper considers a voice conference service as the value-added service in which the telephone function terminals on the telephone network and IP network are interconnected. The construction of a system for providing a voice conference service is investigated, considering requirements such as flexibility, scalability, and quality. A method of construction is proposed. The effectiveness of the proposed method is demonstrated by performance evaluation of an experimental system based on the method. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 2, 87(4): 53–64, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjb.20071

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.

The telecommunications industry today is facing an increasing degree of complexity due to the proliferation of available technologies and separate implementations of voice and data networks and services. Recent trends in the industry toward a convergence of telephony/computer technologies, networks, and services create an opportunity for reducing complexity in the converged voice/data network. The converged telecommunications network will be packet based, providing integrated voice/data/video services with third-party programmability to any subscriber, anywhere, over any access media. This paper describes an architectural approach to meet these challenges. A set of guiding principles is formulated and used for developing the converged network and service architectures. These principles, which provide an overall framework for partitioning network and service functions into components, are illustrated using the planned 7R/E™ Packet Solutions system and service architectures to build component-based converged voice and data services with interfaces to operations, administration, maintenance, and provisioning (OAM&P) frameworks for industry-grade reliability, availability, and security.

Broadband access will not only change the character of data transmission and user platforms, it will have a profound effect on voice features. In the converged network, the feature developer has new tools to control configuration, activation, and content delivery to the end user. These new resources, available from converged telephony endpoints, include Web pages, electronic forms, interactive graphics, bookmarks, and Java∗ applets. The Internet and Broadband Enhanced Telephony project, sponsored by Lucent Technologies' Switching Solutions Group, has established a demonstration environment that shows how to develop these new interactive features. The goals are to establish frameworks that will speed feature development, demonstrate the feasibility of Internet-enabled telephony services in an open-source environment, and integrate these services with video and high-speed Internet access in a broadband endpoint. This paper discusses the demonstration environment, the new user platform hardware and software, and the roles of converged data and programmable switches. The paper also presents an overview of one of the initial features developed — a conference portal.

Application of the Internet protocol (IP) as a network technology for telephony has the power to revolutionize the telecommunications industry. The bandwidth is flexible and almost free, and new services can easily be introduced. This abundance of services and bandwidth is the promise of IP telephony. Before this promise can be fulfilled, however, IP telephony needs to mature and bridge the gap to current telephone systems: the public switched telephone network (PSTN), the integrated services digital network (ISDN), and the wireless systems. Transferring telephone service to the IP domain is not a simple task. Solutions used in switched telephone networks may not be appropriate for the data world. On the other hand, the data world creates its own problems - addressing, reliability, and security are some examples. Initially, these problems may seem best solved by reusing protocols and ideas from the switched telephone world and adapting them to IP. However, tomorrow's problems may be created by such solutions. For example, should an IP telephony service be encumbered with legacy issues such as in-band tone signaling? In this paper, we give an overview of interesting issues in this area and provide a framework in which solutions to small problems can contribute to reaching the final goal of an IP-based telephony service.

The purpose of this paper is to focus on the local echo-canceling problem of full-duplex scrambled speech communications over a two-wire telephone network when the scrambling transformation is located between the handset and body of a telephone. Such a design makes possible very efficient protection against electromagnetic compromising emanation, which in turn substantially enhances the overall security of a protected communication. We propose a new adaptive FIR filter algorithm for local echo cancellation in such applications. The proposed algorithm differs from the conventional one by the construction of input signals in an optimal way using the D-optimal experiment design. In this way, at each step, we generate a new sample of the D-optimal pilot sequence for the filter parameter estimation. Consequently, the adaptation of the local echo canceler is defined as an initialization process in the first phase of each protected telephone call. The advantage in using the proposed adaptive FIR echo canceler is demonstrated through simulation results.

In 1960 the transmission technology of the public switched telephone network was ordinary cable pairs, carrier systems on both paired and coaxial cable, and point-to-point microwave radio systems. Voice signals were frequency multiplexed for efficient transmission, but remained in analog form from end to end of a connection. Electromechanical switching systems provided circuit-switched connections, and about half of the customers could directly dial long distance calls. The following decade saw the beginnings of electronic switching and direct distance dialing on a global basis. Meanwhile, applying digital technology to achieve lower costs and better quality for voice transport began in earnest, and the interconnection of data terminals and computer centers became an increasingly important communications need. The initial data communications approach, still used with ever increasing sophistication and speed, was to convert data to speech-like form with data modems. Later, the direct connection of digital data to digital network facilities became the method of choice for building computer communications networks and the backbone of the Internet. It now appears likely that one day all signals, including voice, will be transported by unified high-speed digital networks with a common set of protocols.

High availability is the main reason for building corporate telephone network of an electric power utility. In this paper is described a novel method for improving the availability, using pre-alarm detector based on the analysis and statistics of telephone traffic. Basic parameters for describing its efficiency are introduced: probability of false pre-alarm detection, probability of a detection miss, and mean time to detection. Detection is proposed in one, two and multiple number of steps, and their basic parameters are presented and compared.

Since 2007, high quality IP telephone service has been provided over a Next Generation Network (NGN) developed by Nippon Telegraph and Telephone Corporation (NTT) Group of Japan. Prior to the commencement of service provision, a wideband telephone handset was developed on an experimental basis and tested in a field trial to verify service validity. The handset demonstrated improved characteristics over those of conventional telephone handsets.