Public Switched Telephone Network Research Articles

VoIP (Voice over Internet Protocol) Based on Asterisk Using Trixbox at SMKN 1 Bangkinang

The rapid advancement of informatics technology in the telecommunications sector has remarkably mitigated the challenges of distance and time, rendering them inconsequential in today`s dynamic landscape. Presently, long-distance conversations are effortlessly facilitated through telephone media, predominantly employing the Public Switch Telephone Network (PSTN) network technology. However, at SMKN 1 Bangkinang Kota, conventional communication practices persist, leading to inefficiencies, particularly in locating teachers, staff, and employees within the school premises. This search process, whether for determining their whereabouts or utilizing smartphones, often necessitates additional time and may incur expenses related to data quotas or credits. In response to these challenges, the primary objective of this research is to conceptualize and establish a VoIP communication system, strategically designed to alleviate operational expenditures associated with communication. The study is conducted within the framework of an existing internet network established on a local scale, involving the integration of wireless and LAN cable technologies. Notably, two distinct operating systems are evaluated, and their performance is compared to identify the optimal choice for serving as a VoIP server. Through this comparative analysis, the study seeks to determine the most suitable operating system that aligns seamlessly with the project's objectives. Central to the research is the successful deployment of a VoIP server utilizing the available local network infrastructure, a feat achieved through the implementation of the Trixbox Linux server. This innovative approach harnesses the existing network facilities to facilitate cost-free communication, effectively bypassing the need for additional expenses typically associated with conventional communication tools like smartphones and PCs. A notable aspect of the system is the pivotal role played by the VoIP server, particularly the Trixbox variant, in managing Session Initiation Protocol (SIP) calls emanating from the diverse client base registered on the Trixbox server. In summation, this research showcases the transformative potential of VoIP technology within educational institutions such as SMKN 1 Bangkinang Kota. By adopting innovative strategies like deploying the Trixbox Linux server, the study underscores the capability of VoIP to streamline communication processes, optimize resource utilization, and ultimately enhance operational efficiency.

A Framework for Smart Homes for Elderly People Using Labview®

This paper deals with home automation systems that are essential for safe and independent living of elderly people. These individuals must be able to perform their Activities of Daily Living (ADLs) without help from caretakers. They must be confident enough that help is just a few minutes with the systems in place. The focus will be on the medical and emergency systems, which are most essential for independent living. Zigbee based Wireless Sensor Networks (WSN) are used to collect the data throughout the home. Smart shirts and smart phones monitor different parameters of the individual and transmit the data to the home network. A PC running LabVIEW is used as the central control unit for the entire house. The house must have a fail proof communication framework for connecting with emergency services, doctors, helpers and relatives. The home network is connected to the Public Switched Telephone Network (PSTN) and internet. SMS, email, prerecorded voice-based telephone calls are appropriately used to report emergencies. A GUI (Graphical User Interface) was built in LabVIEW in order to control and simulate an entire home. Ambient Intelligence is built into the system to make it adaptive and dynamic as far as possible.

Математические модели построения систем WFM телекоммуникационных операторов

The article is devoted to the problems of advanced workforce management systems WFM (Workforce Management) of post-NGN telecommunication operators, a formalized approach to the description of extended WFM functionality and mathematical WFM models are discussed. The proposed approach allows to develop new WFM tools for telecommunications operators to increase their performance, to manage more effectively the work schedules of engineering employees of the company, visit customer sites, and control the workload of engineers in real-time. Workforce Management systems (WFM) of Next Generation Networks (NGN) are becoming more and more important. Compared to the routine distribution of requests between telephone technicians in the repair bureau of the old Public Switched Telephone Network (PSTN) in the last century, the modern WFM system operates with a disproportionately large set of functions, a wide range of professional competencies and key performance indicators (KPIs). In partnership with WFM, other new information technology tools (IT-landscape) of the telecommunications operator, elements of BSS (Business Support Systems), human resource accounting (HR) software, enterprise resource planning (ERP) systems, other workforce planning, and management tools employee vacations, the timing of basic engineering operations at the operator's and customer's premises, and other support tools to optimize the performance of the required work by personnel, increase productivity and reduce costs.

Optimized software-defined multimedia framework: networking and computing resource management

Today, the multimedia over IP (MoIP) network has become a cost-effective and efficient alternative to the public switched telephone network (PSTN). Free applications for multimedia transmission over the Internet have become increasingly popular, gaining considerable popularity around the world. This communication consists of two phases, i.e., signaling phase and media exchange phase. The SIP protocol is responsible for the MoIP network signaling to provide services such as VoIP, voice and video conferencing, video over demand (VoD), and instant messaging. This application layer protocol has been standardized by the IETF for initiating, managing, and tearing down multimedia sessions and has been widely used as the main signaling protocol on the Internet. The signaling and media are handled by SIP proxies and network switches, respectively. One of the most critical challenges in MoIP is the overloading of SIP proxies and network switches. Because of these challenges, a wide range of network users experiences a sharp drop in service quality. Overload occurs when there are not enough processing resources and memory to process all the messages due to the lack of proper routing. This study aims to model the routing problem in MoIP by providing a framework based on software-defined networking (SDN) technology and a convex mathematical programming model to prevent overload. The proposed framework is simulated and implemented using various scenarios and network topologies. The results show that throughput, latency, message retransmission rate, and resource consumption have improved using the proposed approach.

Hybrid Transform Based Speech Band Width Enhancement Using Data Hiding

The limited narrow band (LNB) speech signal spread in the range of 300 to 3400Hz used in public switched telephone networks results in poor-quality telephony speech. Bandwidth extension techniques are performed to expand the frequency range from LNB speech to a clear wideband (CWB) speech signal range of 50Hz-7000Hz over existing public telephone networks. In this paper, a novel robust speech bandwidth extension algorithm by Discrete Wavelet Transform- Discrete Cosine Transform- Based Data Hiding (DWT-DCT-DH) Hybrid transform model was used to spread the out-of-band (3400Hz to 7000Hz) speech frequencies over the LNB speech. In this proposed technique the out-of-band speech frequencies are embedded in LNB speech and imperceptibly spread over the network. These Embedded out-of-band speech frequencies are recovered steadily at the receiver end to generate a restored CWB telephony speech of considerably better quality. The proposed technique simulation results show more intelligible and better-quality telephony speech generated compared to the other bandwidth extension techniques.

IMPLEMENTASI VIRTUAL PRIVATE NETWOR MENGGUNAKAN POINT-TO-POINT TUNNELING PROTOCOL

Today's technological development is very rapid, especially in the field of network security. The field of network security is a matter that will be a very important part because the network security system is used to guarantee the confidentiality, theft, or burglary of data on a company. In its implementation, a VPN network (Virtual Private Network) uses a site-to-site Point to Point Tunel Protocol (PPTP) method to connect between two places that are located far apart. The main facility of using PPTP is being able to use a public-switched telephone network (PSTN) to build a VPN. At PT Indosis Integrasi Jakarta, the place where the author conducts research there are several problems that the author found, namely, when sending data between the head office and branch offices in Bandung, there is no security that ensures that the data is safe. With the current state of the network, uninterested parties will easily access or steal confidential company data.

A Strategy to Strengthen and Enhance the Telecommunication Network in Sri Lanka by Using Concepts of Graph Theory and Linear Programming Models

The telecommunication sector in Sri Lanka is one of the most dynamic sectors, contributing significantly, both directly and indirectly, to investment, employment, productivity, and overall economic growth. According to the transformation of the telecommunication network of Sri Lanka, this study proposes a structure to increase the average upload and download speeds by using some concepts in pure Mathematics. The research design has aligned into two categories as interpretation and optimization. They established by using the concept of Graph Theory and linear programming models respective. We forecast the public switched telephone network (PSTN) model for the local network. Then we ranked carrier systems of Sri Lanka according to their services without violating privacy. Next, we created the minimum traffic flow, local model, based on the traffic flows. Then we proposed cable systems around Sri Lanka. After that, we calculated the required signal towers for the uniform coverage by considering the population density. Linear programming models had generated to optimize the traffic flows. We could obtain six hundred twenty-five linear programming models due to different coefficients in each case. So, we have separated those models into three sectors: low, moderate, and high. Three examples had discussed by considering them. The solutions had evaluated by using the PHP Simplex calculator.

THE CALCULATING METHOD OF NGN ACCESS NODE CHARACERISICS WHEN SERVICING VOICE AND DATA TRAFFIC

Access nodes of NGN networks provide traffic aggregation to various network services and its transfer to the transport layer, which allows calling such networks multiservice.1;2. These networks are used to simultaneously transmit voice, video, and data presented in the from of standard packets. At the same time, network services provide an unlimited set of telecommunication services with specified characteristics of quality of service.1;2 The selected technology of information distribution in the NGN determines the degree of complexity of access nodes and, of course, affects the quality of information exchange service between user terminals. One of the ways to ensure a given level of quality of service for each of the services is the introduction of priority service for certain traffic flows. Introducing priorities for packets is an effective way to manage the size of queues and the time spent in it. When a high-priority packet is received, a packet with a lower priority is either interrupted (absolute priority) or a high priority packet becomes the beginning of the packets queue (relative priority). In this paper, we study the characteristics of NGN access nodes as a two priority queuing system with a limited expectation.5. Formulas for calculating the probability of loss of speech packets, as well as data packets, and other characteristics of the access node have been proposed. The NGN's access nodes provide aggregation of traffic to various services directed by the Public Switched Telephone Network (PSTN) and its transportation to a higher level, which allows you to call such networks multiservice ones. The new generation networks allow simultaneous transmission of mixed streams (voice, video, and data) represented in the form of standard packets. One of the ways to ensure the specified quality of service is to introduce a priority method for servicing traffic flows 5.

Single Board Computer Applications as Multi-Server VoIP

Telecommunication technology is developing along with information technology and several innovations in several audio and data transmission and reception techniques. Innovation and communication technology are hoped to be able to create efficiencies in regards to time, equipment, and cost. The Public Switched Telephone Network (PSTN) telephone technology has experienced integration towards communication using Internet Protocol (IP) networks, better known as Voice over Internet Protocol (VoIP). VoIP Technology transmits conversations digitally through IP-based networks, such as internet networks, Wide Area Networks (WAN), and Local Area Networks (LAN). However, the VoIP cannot fully replace PSTN due to several weaknesses, such as delay, jitter, packet loss, as well as security and echo. Telephones calls using VoIP technology are executed using terminals in the form of computer devices or existing analogue telephones. The benefit of VoIP is that it can be set in all ethernet and IP addresses. Prefixes can be applied for inter-server placements as inter-building telephone networks without the addition of inefficient new cables on single board computers with Elastix installed. Prefix and non-prefix analysis on servers from single board computers can be tested using QoS for bandwidth, jitter, and packet loss codec. The installation of 6 clients, or 3 simultaneous calls resulted in a packet loss value in the prefix Speex codex of 2.34%. The bandwidth in the prefix PCMU codec has an average value of 82.3Kbps, and a non-prefix value of 79.3Kbps, in accordance to the codec standards in the VoIP. The lowest jitter was found in the non-prefix PCMU codec with an average of 51.05ms, with the highest jitter for the prefix Speex codec being 314.65ms.

Optimizing VoIP server resources using linear programming model and autoscaling technique: An SDN approach

SummaryNowadays, the Voice Over IP (VoIP) technology is an important component of the communications industry as well as a low‐cost alternative to Public Switched Telephone Networks. Communication in VoIP networks consists of two main phases, for example, signaling and media exchange. VoIP servers are responsible for signaling exchange using the Session Initiation Protocol (SIP) as the signaling protocol. The saturation of SIP server resources is one of the issues with the VoIP network, which causes problems such as overload or loss of energy. Resource saturation occurs mainly due to a lack of integrated server resource management. In the traditional VoIP networks, management and routing are distributed among all equipment, including servers. These servers are overloaded during peak times and face energy loss during idle times. Given the importance of this issue, this paper introduces a framework based on Software‐Defined Networking technology for SIP server resource management. The advantage of this framework is to have a global view of all the server resources. In this framework, the resource allocation optimization problem and resource autoscaling are presented to deal with the problems posed. The goal is to maximize total throughput and minimize energy consumption. In this regard, we seek to strike a balance between efficiency and energy. The proposed framework is implemented in the actual testbed. The results show that the proposed framework has succeeded in achieving these goals.

VoIP Security: A Performance and Cost-benefit Analysis

Voice over IP (VoIP) has become the standard technology for telephony and has replaced the old Public Switched Telephone Network (PSTN). This research focuses on the security aspect of VoIP systems. Unsecured VoIP systems are vulnerable to malicious attacks. However, the overhead of the security techniques hampers the performance of VoIP systems. This research analyzes how a VoIP system performs with different security techniques. The performance of the VoIP system is analyzed on different types of data networks such as IPv4, IPv6, and IPv4/IPv6 mixed networks, and in scenarios such as with and without network traffic. Additionally, the research includes a cost-benefit analysis of the security techniques, to determine their cost effectiveness. Based on the performance analysis and cost-benefit analysis, this research proposes three security techniques that can be applied to VoIP systems deployed on IPv4, IPv6, and IPv4/IPv6 networks.

Integrating Voice over IP Solution in IPv6 and IPv4 Networks to Increase Employee Productivity: A Case Study of Cameroon Telecommunications (Camtel), North-West

Telecommunications organizations have to follow the rapid innovation of technology if they want to face challenges raised by competition. The challenge to respond to the huge market demand of updated products and services from customers requires that the organization‘s working environment be equipped with tools and communication facilities that contribute to ameliorating productivity. Cameroon Telecommunications (Camtel) is facing a digital telephony and Internet Protocol strategic management challenge. Successful implementation cannot be achieved if the employees are still depending on the ageing public switched telephone network (PSTN) as their primary communication system, despite the frequent loss of dial tone experience in a day which can last up to a week, with serious repercussions on business activities and revenues. This study is designed to provide a solution to the telecommunications challenge. The fundamental question is how to integrate a digital telephony system that will provide telephony services in the existing IPv4 data network while prioritizing IPv6 traffic forwarding. This study proposes and implements solutions that integrate a Voice over IP solution with IPv6 as an alternative communication system that relies on the existing IPv4 data network. VoIP is deemed as one of the driving forces behind the adoption of IPv6. The purpose is to offer to workers an option that will free them from the poor Quality of Service (QoS) of their existing PSTN based solution, hopefully enhancing the overall productivity. This paper follows two research methodologies: Qualitative Research in Applied Situations and Engineering design process. The first part of this study reports the results of the evaluation of how much such a solution can enhance workers’ productivity. As it is important to provide an environment where IPv4 and IPv6 networks and applications/devices can interoperate in the context of VoIP; the second part describes practically a simulation environment where various configurations of network entities are done following a Dual-Stack transition approach. Document and records were used to gather information related to the structure, operations, and topological update of the Camtel’s existing IP data network. The findings demonstrated that VoIP can be an effective communication solution for Camtel and its implementation with IPv6 will be preferable. However, for this to be efficient there must be a provision of sufficient bandwidth and usage of types of equipment and transmission mediums that minimizes processing and propagation delays. Findings also reveal that better productivity will be achieved if workers are fully trained for the exploitation. This research article tries to highlight, discuss a required transition roadmap and extend the local knowledge and practice on IPv6. Future expansion of this research work will consist of deploying Dual-Stack VoIP in the remaining 9 regional offices for full integration in the corporate communication system of Camtel.

Using Analytical Hierarchy Process (AHP) in selection of suitable platform to present communication services to subscribers

Before emerging internet and data network, remote communication had a clear meaning: Telephone (telegraph before it) was the only technology that allowed people to communicate by voice from remote distance. After 1960, fax and data were presented in Public Switched Telephone Network (PSTN). Recently, remote communications have been developed and they include data transmission, video conference, email, instant messages, web browsing and potential for transmission of multimedia content. Today, consumers think about two categories of products and services relating to communications. Tendency of consumers to purchase communicative applications or equipment has growingly become pervasive as products and services and the used technology for remote communications has changed to great extent during 50 years ago. Using semiconductors and digital electronic devices in telecommunication industry could replace analog sound with images and video. Then, switching circuit was completed and it should be more unlikely substituted with packet switching. All these changes indicate a new definition of remote communication: The communication is a group of technology, devices, equipment, installations, networks and software programs that support telecommunication. It has been tried in this paper to identify important parameters and criteria in communication field by utilization from Analytical Hierarchy Process (AHP) technique to interpret suitable platform for presentation of communicative services for subscribers and by recognition and dividing platforms in which there is potential for giving services to the subscribers.

Understanding the Data Gap in Emergency Response: Evidence from U.S. 911 Agencies

Using a combination of data analysis, live field testing, and interviews with 911 centers (hereafter referred to as Public Safety Answering Points, or PSAPs), this paper studies the interaction between alarm systems and first responders using both publicly available data as well as in-person data collection at PSAPs across the country. This data was augmented with a survey of 145 telecommunicators as well as qualitative interviews from PSAP directors. The results demonstrate a significant information gap between increasingly sophisticated alarm systems and the emergency services responding to their signals for help. Specifically, our testing and survey indicated that nearly all alarms are communicated via phone calls over the public switched telephone network (PSTN), limiting information flow to a verbal conversation between 911 telecommunicators and alarm monitoring center operators. The limitations of this legacy infrastructure often prevent 911 telecommunicators from acquiring enough information to triage and prioritize alarm calls. This lack of information leads to long PSAP processing time (6-13 minutes) and a proliferation of false alarms which clog first responder systems. Various studies have demonstrated that 90 to 99% of all security alarms are false depending on the municipality. We estimate the total yearly impact on the US emergency response system to be approximately 62 million false alarms at a cost of $3.1B annually. Unsurprisingly, the lack of information and tendency for false alarms causes most public safety agencies to deprioritize central station alarm calls. This challenge is magnified by the use of 10-digit non-emergency lines for incident reporting. These factors converge to drive a high degree of variability in alarm response times and exacerbate the first responders’ challenges in responding effectively to real alarm emergencies, given limited resources.

Technical Specification for Effective Next Generation Network Interconnection in Indonesia

This paper proposes technical specifications for effectively implementing the interconnection of Next Generation Network (NGN) in Indonesia. NGN is one of the current telecommunication infrastructure network technology trends. It provides a simpler concept with only two layers of service and transport. The NGN IP-based transport system can connect with various types of networks, which leads to low management costs by offering different kinds of services. Meanwhile, there are currently various types of telecommunication networks depending on the services provided, such as Public Switched Telephone Network (PSTN), IPv4 Internet as well as Public Switched Data Network (PSDN). PSTN is a circuit-switched voice communications network, and PSDN is a network for data-based communications with International Telecommunication Union (ITU)-T X.121 standards. Another type is a packet-switched based network that uses IPv4 addressing systems. Each network has its customers. One of the problems that arise, however, is how to transform the current network system to the NGN network effectively. The effectiveness of network transformation in the service provision for users is determined by the technical aspects used. Some of the technical aspects issues on implementation of NGN networks in Indonesia are the use of a protocol for signaling, coding standards, Quality of Service (QoS), numbering and addressing, and security. This paper proposes technical specifications for the effectiveness of NGN network implementation in Indonesia. Through the technical specification model, we propose, the risks that will arise in the implementation of NGN networks in Indonesia can be managed. Appropriate technical specifications have an essential role in the effectiveness of NGN network implementation in Indonesia.

Analisis Peningkatan Kualitas Layanan Penyelesaian Gangguan Pstn Dengan Metode Six Sigma

In the era of industrialization increasingly competitive currently, every business actors that wants to winthe competition in the industry will give full attention to the quality. Six Sigma provides a perfect solution aboutbreakthroughs or steps that should be done to produce a dramatic improvement to the level of failures = 0 (zerodefect). PT. X is a company engaged in the business of telecommunications services with one of its products is thePSTN (Public Switch Telephone Network). The author will examine the improvement of service quality controlproblems using the five phases of Six Sigma, namely: 1) Define, The completion of repair that have been madeduring the year 2010 as many as 26 421 with the failure of 23, 2) Measure, cause of the failure that occurred atthe IKR 39.13%, DP 34.78%, 21.73% MDF and RK 4.34%, 3) Analyze, the overall repair process disruption incontrol limits, indicated by the absence of data that passes through the UCL and LCL.The process capability hasbeen which achieved is 5:01 sigma, and the dominant cause of any failure lies in the material factors 43.54%, 4)improve, target to be achieved is 3.4 failures per million opportunities DPMO (Defect per million opportunities)with the proposed improvements on Human Factors, Methods, Machinery/equipment, material and theenvironment, 5) Control, not discussed in this paper because the analysis is only a proposal to the Company.

Rationing the main parameters of the Quality of Network services

The European Telecommunications Standards Institute (ETSI) recommends that IP-telephony networks be divided into four classes according to the quality of service (quality of network services) QoS, the main indicator of which is packet delay (Y.1541). ITU-T Recommendation G.114 for public switched telephone networks provides delays close to ETSI gradations that correspond to different types of communication. The quality of the service from the user's point of view can be expressed by a set of indicators. These metrics are described in terms that are understandable to both the user and the service, and are independent of the network structure. Quality of service indicators are focused primarily on the effect perceived by the user, must be guaranteed to the user by the service and be objectively measured at the point of access to the service (ITU-T Recommendation I.350). ITU-T Recommendation E.862 provides possible approaches to accounting for the economic losses of the operator (in planning, design, operation and telecommunication network) and the user associated with technical failures. Operators of networks, working in market conditions, are interested in assessing possible losses due to failures and in comparing them with the costs of increasing the reliability of their funds.

Performance of Various VoIP Vocoders using Wireshark with Asterisk PBX

A private branch exchange (PBX) is implemented by an Asterisk software. In conjunction with appropriate telecommunication hardware interfaces and network applications, Asterisk is employed to establish and manage the telephone calls between telecommunication endpoints, like customary telephone sets, destinations on the general public switched telephone network (PSTN), and devices or services on voice over internet Protocol (VoIP) networks. A Vocoder may be a system of coders and encoders that is employed to scaleback the bandwidth over the restricted use of bandwidth necessities and restricted capability channels in real time needs. This paper presents Performance of Various VOIP Vocoders using wireshark with Asterisk PBX.

Unified call admission control in corporate domains

Call Admission Control is a central mechanism for assurance of quality of service in telephony. While CAC is integrated into Public Switched Telephony Network (PSTN), its application to voice over IP in a corporate environment is challenging not only due to the heterogeneity of technologies, but also because of the difficulty of implementation into commercial VoIP terminals or Access Points.We present a novel framework that unifies call admission control for VoIP telephony corporate users despite their access network (i.e., WiFi or Ethernet) under a single corporate management domain. Our Unified CAC (U-CAC) system can be implemented in a VoIP Gateway/Proxy and uses only standard protocols already present in commercial off-the-shelf devices, avoiding the need to modify the firmware of existing APs or VoIP terminals. We define two variants of the decision algorithm: basic and advanced. In the basic mode of operation, the admission of new calls is based on the availability of spare circuits and the impact of the new call in the speech quality of VoWiFi calls in progress. In the advanced mode of operation, the traffic load in affected APs is proactively reduced by reconfiguring ongoing calls before rejecting the new call. Simulation results show that the number of simultaneous VoWiFi calls under guaranteed quality increases with our unified call admission control scheme. When using the advanced mode of operation, the number of simultaneous calls under guaranteed quality can be doubled when compared to the standard mode of operation.

An Integrated Emergency Call System based on Public Switched Telephone Network for Elevators

Today, most of elevators have an emergency call facility for emergency situations. However, if the network installed in the elevator is also out of power, it cannot be used for the elevator remote monitoring and management. So, we develop an integrated and unified emergency call system, which can transmit not only telephone call but also data signals using PSTN(Public Switched Telephone Network) in order to remote monitoring and management of elevators, even though a power outage occurs. The proposed integrated emergency call system to process multiple data such as voice and operational information is a multi-channel board system which is composed of an emergency phone signal processing module and an operational information processing module in the control box of elevator. In addition, the RMS(remote management server) systems based on the Web consist of a dial-up server and a remote monitoring server where manages the elevator's operating information, status records, and operational faults received via the proposed integrated and unified emergency call system in real time. So even if there's a catastrophic emergency, the proposed RMS systems shall ensure and maintain the safety of passengers inside the elevator. Also, remote control of the elevator by this system should be more efficient and secure. In near future, all elevator emergency call system need to support multifunctional capabilities to transmit operational data as well as phone calls for the safety of passengers. In addition, for safer elevators, it is necessary to improve them more efficiently by combining them with high-tech technologies such as the Internet of Things and artificial intelligence.