VoIP Research Articles

Возможности перераспределения функций городской поликлиники при открытии подразделения – Дистанционный консультативный центр

Goal. To evaluate the interim results of a pilot project on the organization of an independent unit of a city clinic that provides remote support to citizens – the Distant Consultation Center of the Government Autonomous Organization Healthcare “City Clinic No. 8” (hereinafter referred to as the DCC). M e t h o d s o f w o r k . To assess the effectiveness of the pilot project, a number of indicators of the work of the medical organization were analyzed before the start of the project and 12 months after the start of its implementation. The analysis was carried out in two way of work of the Distant Consultation Center: 1. the process of providing medical care to patients with acute respiratory viral infections / Influenza / COVID‑19; 2. the process of providing medicines to the Subsidized category of citizens. The volumes of work that can be transferred to the Distant Consultation Center from the department of primary care physicians care and from the emergency medical care are analyzed. The source of the data was the medical information system of the medical organization “URMO TO” (modules Registry, Control of realization, doctor’s ARM, Disability sheets, Additional processing). The period of work of the medical organization from 01.01.2022 to 31.12.2022 is analyzed. Statistical processing of the results was carried out using the Statistica 6.0 application software package. Distance consultation was carried out via IP telephony, as well as using specialized Telemedicine software. R e s u l t s o f t h e w o r k . Before the organization of the DCC, the median indicator of the actual workload per hour of the covid team was 2,5 [2,3–2,7] patients, p<0,07. While the doctor of DCC carried out an average of 56,9 [44–68] remote consultations, which is an average of 8,8 patients per hour, p< 0,07. The median value of patients for prescriptions for Subsidized medicines before the start of the project was 4 [3–6] patients per day (p<0,07), on average 22.2% time of the initial consultation falls on patients requiring a Subsidized medicine. The median value of applications for the purpose of providing Subsidized categories of citizens with medicines when discharged by a medical assistant at the DCC was 1 [0–3] (p<0,08). Thus, this section began to occupy 0,06% in the structure of reasons for the initial appointment to the primary care physician. C o n c l u s i o n s . The organization of the work of the DCC allowed us to develop approaches, approve and adapt them to the processes of filling out the schedule of primary care doctors in order to increase accessibility to the primary consultations of the primary care physician’s department. Scaling up the organizational model and processes of the Distance consultation center of the Municipal Polyclinic No. 8 can serve as an effective tool in achieving the goals announced within the framework of the pilot project of the Ministry of Health of the Russian Federation “Incident 38” launched in 2022. T h e s c o p e o f t h e r e s u l t s . The presented experience shows the possibility of managing patients with respiratory viral diseases in outpatient polyclinic conditions on an ongoing basis, as well as the telemedicine format of care can be extended and adapted to non-infectious nosologies.

Improving VoIP Bandwidth Utilization Using the PldE Technique

Abstract The use of Voice over Internet Protocol (VoIP) innovation is rising due to its various merits. Nevertheless, the ineffective use of bandwidth is a key dilemma that restricts the fast-rising use of VoIP innovation. The main factor behind this ineffective use of the bandwidth is the sizable VoIP packet preamble. This research creates a technique to address this dilemma of packet preamble. The created technique is known as payload elimination (PldE). The fundamental concept of the PldE technique is to exploit the information (elements) of the VoIP packet preamble that is superfluous for point-to-point calls. In general, these elements are utilized to transport the payload of VoIP packets. Consequently, the payload size of VoIP packet will be lowered or removed, preserving the available bandwidth. The performance test of the PldE technique indicated an improvement of up to 41.6% in the exploitation of IP network bandwidth. So, the PldE technique is showing signs that it could help solve the problem of the IP network's inefficient use of bandwidth.

Evaluating Latency in Fifth-Generation Vehicle-to-Everything Communications using Adaptive Neuro-Fuzzy Inference System Model

The fifth-generation new radio technology (5G NR) introduces improved functions to the air interface. In addition, the 5G NR non-standalone (NSA) will operate with long-term evolution, enabling vehicle-to-everything communications (V2X) for improved infotainment services. V2X includes four main classes of communications: vehicle-to-vehicle, vehicle-to-infrastructure, vehicle-to-pedestrian devices, and vehicle-to-network. However, the stringent transmission frequency, latency, and throughput requirements of infotainment applications constrain the transmitting packets of 5G-V2X-based NSA in highway scenarios. In this paper, the latency is improved by preventing the physical layer of gNodeB and the user equipment (UE) from sending redundant packets for service in a highway scenario. The proposed approach adopts an adaptive neuro-fuzzy inference system (ANFIS), a powerful modeling technique based on artificial neural networks ,and a fuzzy inference system. The performance of ANFIS is compared with that of the traditional 5G V2X NSA architecture in a simulation study using Voice over Internet Protocol (VoIP) traffic. The delays, throughputs, and packet losses of both architectures are determined in radio link control (RLC) and VoIP applications. The switch-modes, signal-to-interference-noise ratios (SINRs), hybrid automatic repeat request (HARQ) error rate, channel quality indicators (CQIs), served blocks, and transmission-state of gNodeB are computed for the two architectures for device-to-device (D2D), uplink (UL) and downlink (DL) traffic directions. The simulation results show comparable SINRs, CQIs, served blocks ,andswitchmodes in both scenarios, but the presented ANFIS model significantly outperforms the traditional architecture in delay by 66% in D2D, 29% in UL, 25% in DL, and packet loss by 21% in UL in RLC, the HARQ error rate by 9% in D2D, 30% in UL, 95% in DL, transmission-state in gNodeB by 29%, and the delay by 4% for UEs, and frame loss by 90%for UE in VoIP.

AN ENHANCED RESOURCE ALLOCATION SCHEME FOR LONG TERM EVOLUTION ADVANCED NETWORK

The emerging Long-Term Evolution Advanced (LTE-A) is an extended version of Long-Term Evolution (LTE) fourth-generation (4G) wireless broadband technologies that supports high speed transmission of multimedia applications, like video streaming, multimedia online game, Voice over IP (VoIP), browsing over the web etc. The Third Generation Partnership Project (3GPP) does not define a standard packet scheduling strategy in LTE-A. Therefore, it is expected for researchers and network service providers to design efficient schemes. This paper proposed an Enhanced Resource Allocation Scheme (E-RAS) that improves the packet scheduling strategy of LTE-A by addressing issues related to delay and fairness of users with both high and low channel conditions, for both Real Time (RT) and Non-Real Time (NRT) flows. The aim was achieved by using a fair spectrum allocation mechanism that consider the priority of RT flows of cell-centre users, without neglecting the throughput requirements of edge-centre users. Moreover, the proposed algorithm introduces an efficient traffic load balancing mechanism to efficiently regulate traffic load in LTE-A network. To evaluate the proposed scheme, simulation was conducted using the MATLAB programming with LTE-A system level simulator. The result obtained shows that the proposed scheme achieves better delay for RT and NRT traffics by 7% and 11% respectively as against the Resource Allocation Scheme to Optimize the Throughput (RASOT) scheme.

A novel DWDM-based comb source for future optical network

In recent times, there has been enormous growth in internet-based applications such as online entertainment platforms, video conferencing and voice over internet protocol (VoIP) applications which might be the trend in future. In order to support such bandwidth-hungry applications, the speed of the network should be enhanced. Increasing the infrastructure of the network requires heavy investment. Another alternative to increase the capacity of the network is to use dense wavelength division multiplexing (DWDM) which also results in increased costs as the lasers and other equipment needed for generation of wavelengths are required. In this work, an alternative cost effective technique is used to create a DWDM scenario using optical flat comb source (OFCS) by combining amplitude modulator (AM) and single drive mach zehnder modulator (SD-MZM). The spectral lines generated using this procedure can be used in Passive Optical Network (PON) as input sources. In this paper, 51 flat spectral lines are generated and transported over a distance of 100 km which are also capable of supporting a data rate of greater than 10 Gbps in a single spectral line.

Bridging Knowledge Gaps through Embedded Librarianship Practices in Nigerian Universities

This study investigates the role of embedded librarianship in the production and diffusion of knowledge in Nigerian universities. A quantitative approach was adopted using a cross-sectional survey design with a sample of 187 library staff drawn from a population of 370 library staff from three university libraries. The findings reveal that the types of embedded librarianship practices adopted in the universities include faculty embedded librarianship, research and bibliographic services embedded librarianship, and information literacy embedded librarianship. Knowledge production and dissemination were enhanced by the practice of embedded librarianship through fostering collaboration between embedded faculties and the library, improving students’ research and information searching skills, and participating in teaching information literacy skills. The findings also revealed that phones, emails, web forums/blogs/wikis, instant messaging services, Voice over Internet Protocol (VoIP), social networking sites and face-to-face discussions were the means of communication and interaction between the embedded librarians and their clients in the three universities. The implications of this study will serve as a springboard for adopting and strengthening the practice of embedded librarianship among Nigerian academic libraries since the concept is new and emerging globally, thereby bridging the knowledge gaps that exist between the haves and have nots in terms of knowledge in the Nigerian higher education sub-sector.

Fraud in a digital society in the context of social change

The article analyzes the problems of fraud in the modern digital society in the context of social changes caused by both the COVID-19 pandemic and the current geopolitical situation. The authors study in detail the types of financial fraud in this area. They pay special attention to the types of digital fraud: telephone fraud and IP-telephony, SMS fraud, phishing, sniffering (message interception), skimming (theft of card data), which are associated with the emergence of cryptocurrencies. The article presents the data of the authors’ sociological research conducted by the method of an online questionnaire survey (n = 765). The study showed that only 44 % of respondents are sufficiently informed about the types of fraud to protect themselves from modern internet scammers. An analysis of data on certain types of digital fraud showed that 20 % of respondents lost money due to telephone or Internet fraud, 16 % of respondents personally encountered IP telephony, and 40 % have no idea about such a type of fraud as sniffering and existing anti-virus protection against it. The research also showed the vulnerability, first of all, of the male half of the population over 40 years of age in the field of internet fraud. Therefore, public authorities, together with the media, business and other actors of Russian society, need to strengthen the policy in the field of improving the level of financial and digital literacy of the population.

Real Time Cloud based Fault Detection and Alert for Antenna Array Using CNN

The rise of smart antennas made it imperative to use adaptive beam forming, but single-element failure in array antenna disturbs the radiation pattern. Hence, to achieve adaptive beam forming, detecting and locating faults in an antenna array is a must for correction in the radiation pattern. The presented work in this paper provides real-time monitoring and alert for fault occurrence in antenna arrays. Fault detection is performed by an optimized Convolutional Neural Network (CNN), and the Thingspeak web service is utilized to store fault detection information on the cloud to get the visualization of faults. Voice over Internet Protocol (VoIP) Application programming interface (API) provides an alert in real-time once the fault is detected. The filter kernel parameters, number of convolutional layers, and dense layers of the fully connected neural network are tuned to minimize the mean square error (MSE). 16 element (4 × 4) planar microstrip array antenna is simulated on Ansys HFSS, and by introducing discontinuity, i.e. one type of defect in the corporate feed network, a training and testing set of 374 samples considering single element fault are used to train the CNN Model with a 0.3 validation split. Further performance of the same model by changing the last layer of a fully connected layer is evaluated to check the effectiveness of model when two elements are faulty.

Enhanced Multiple Speakers’ Separation and Identification for VOIP Applications Using Deep Learning

Institutions have been adopting work/study-from-home programs since the pandemic began. They primarily utilise Voice over Internet Protocol (VoIP) software to perform online meetings. This research introduces a new method to enhance VoIP calls experience using deep learning. In this paper, integration between two existing techniques, Speaker Separation and Speaker Identification (SSI), is performed using deep learning methods with effective results as introduced by state-of-the-art research. This integration is applied to VoIP system application. The voice signal is introduced to the speaker separation and identification system to be separated; then, the “main speaker voice” is identified and verified rather than any other human or non-human voices around the main speaker. Then, only this main speaker voice is sent over IP to continue the call process. Currently, the online call system depends on noise cancellation and call quality enhancement. However, this does not address multiple human voices over the call. Filters used in the call process only remove the noise and the interference (de-noising speech) from the speech signal. The presented system is tested with up to four mixed human voices. This system separates only the main speaker voice and processes it prior to the transmission over VoIP call. This paper illustrates the algorithm technologies integration using DNN, and voice signal processing advantages and challenges, in addition to the importance of computing power for real-time applications.

Design and implementation of a VoIP PBX integrated Vietnamese virtual assistant: a case study

ABSTRACT As digitization is integrated into daily life, media are increasingly transferred over the Internet. Voice-over-Internet Protocol (VoIP), the most popular media transfer technology, is attracting many researchers and investments. The application of Artificial Intelligence (AI) technology into the Private Branch Exchange (PBX) has played a pivotal role in enhancing the customer experience and is able to unite employees in any company. One technology application used to optimize customer experience in a call centre is the use of an automatic PBX integrated with a Virtual Assistant (VA), which interacts directly with the PBX through voice and in multiple languages without any keystrokes. The Interactive Voice Response (IVR) module forwards the customer’s call to an operator or supports automatic processing. This solution can help businesses to handle thousands of calls per day with optimal performance, thus creating a customer care campaign that quickly reaches many users. A PBX integrated with Vietnamese Virtual Assistants (VVA) on an AI technology platform will also help businesses to cut down on operator costs with automated calls. Through comparison with a traditional PBX, this article analyzes, evaluates and optimizes an automatic PBX system with integrated VVA, thereby offering efficient solutions for interest companies.

The voice as a material clue: a new forensic Algerian Corpus.

Dialects have received bigger interest in recent years as they are increasingly used on the web and social media. Because Algerian Arabic dialects suffer from a lack of appropriate speech corpora for speech recognition, a rich dialect corpus is needed to approach Algerian Accent recognition. The latter remains a key feature in the field of Forensic Voice Comparison (FVC) systems. This paper presents a new large-scale forensic Algerian speech corpus called Sawt El-Djazaïr. An important criterion in dealing with forensic corpora is the presence of session variability. For this purpose, we collected celebrity recordings in various regions of Algeria, from different social networks, in various scenarios, and at different times. In addition, we also recorded 87 participants using cellular calls and voice over IP (VoIP) applications including Viber, WhatsApp, and Google Meet. The corpus of approximately 50hours covers various speech topics and is spoken in twelve Algerian sub-dialects. The design guidelines of the proposed corpus are described along with the grouping of dialects across different geographical locations. Sawt El-Djazaïr is available to the research community upon request.

911 Swatting, VoIP, and Doxxing

When it comes to 911 swatting and doxxing, Voice over Internet Protocol (VoIP) takes this digital mischief to a new level.

Conducting qualitative interviews via VoIP technologies: reflections on rapport, technology, digital exclusion, and ethics

ABSTRACT Qualitative research has been strongly affected by the COVID-19 pandemic, highlighting the possibilities that Voice over Internet Protocol (VoIP) technologies such as Skype, WhatsApp, and Zoom offer to qualitative scholars. Based on the experience of using such technologies to collect qualitative data for our PhD studies, we present how we dealt with the challenges of this interview mode. Precisely, we discuss problems related to rapport, technology, digital exclusion, and ethics frequently pointed out in the methodological literature on online interviews. Thereby we put forward strategies and techniques that helped us to 1) build a rapport, 2) manage technical difficulties, 3) reflect on risks of digital exclusion, and 4) comply with the ethical standards of our institution. In doing so, we draw on our qualitative data to support the arguments. The aim of this paper is, thus, to deepen the methodological debate on online interviews in social sciences.

Towards a Smart Environment: Optimization of WLAN Technologies to Enable Concurrent Smart Services

In this research paper, the spatial distributions of five different services-Voice over Internet Protocol (VoIP), Video Conferencing (VC), Hypertext Transfer Protocol (HTTP), and Electronic Mail-are investigated using three different approaches: circular, random, and uniform approaches. The amount of each service varies from one to another. In certain distinct settings, which are collectively referred to as mixed applications, a variety of services are activated and configured at predetermined percentages. These services run simultaneously. Furthermore, this paper has established a new algorithm to assess both the real-time and best-effort services of the various IEEE 802.11 technologies, describing the best networking architecture as either a Basic Service Set (BSS), an Extended Service Set (ESS), or an Independent Basic Service Set (IBSS). Due to this fact, the purpose of our research is to provide the user or client with an analysis that suggests a suitable technology and network configuration without wasting resources on unnecessary technologies or requiring a complete re-setup. In this context, this paper presents a network prioritization framework for enabling smart environments to determine an appropriate WLAN standard or a combination of standards that best supports a specific set of smart network applications in a specified environment. A network QoS modeling technique for smart services has been derived for assessing best-effort HTTP and FTP, and the real-time performance of VoIP and VC services enabled via IEEE 802.11 protocols in order to discover more optimal network architecture. A number of IEEE 802.11 technologies have been ranked by using the proposed network optimization technique with separate case studies for the circular, random, and uniform geographical distributions of smart services. The performance of the proposed framework is validated using a realistic smart environment simulation setting, considering both real-time and best-effort services as case studies with a range of metrics related to smart environments.

Utilizing Voip Packet Header’s Fields to Save the Bandwidth

Abstract Voice over IP (VoIP) is widely utilized by organizations, schools, colleges, and so on. Nevertheless, VoIP numerous challenges that hinder its spread. One of the significant challenges is the poor exploit of the VoIP technology network bandwidth (BW), caused by the huge preamble of the VoIP packet. This paper suggests a novel methodology to manage this huge preamble overhead challenge. The proposed methodology is named runt payload VoIP packet (RPV). The core principle of the RPV methodology is to reemploy and exploit the VoIP packet preamble’s data (fields) that are superfluous by VoIP technology, especially for unicast IP voice calls. Generally, those fields will be used to convey the VoIP packet payload. Consequently, diminish or zero the length of the payload and, therefore, spare the BW. The results of the investigation into the suggested RPV methodology indicated significant enhancement in the BW exploitation of VoIP technology. For instance, the saved BW in the examined environment with the LPC codec came to up to 25.9%.

A Novel Approach for Efficient Mitigation against the SIP-Based DRDoS Attack

Voice over Internet Protocol (VoIP) and its underlying Session Initiation Protocol (SIP) are widely deployed technologies since they provide an efficient and fast means of both voice and data communication over a single network. However, in spite of their advantages, they also have their security threats due to the inherent vulnerabilities in the underlying Internet Protocol (IP) that can potentially be exploited by hackers. This study introduces a novel defense mechanism to effectively combat advanced attacks that exploit vulnerabilities identified in some less-known features of SIP. The SIP-DRDoS (SIP-based distributed reflection denial of service) attack, which can survive the existing security systems, is an advanced attack that can be performed on an SIP network through the multiplication of legitimate traffic. In this study, we propose a novel defense mechanism that consists of statistics, inspection, and action modules to mitigate the SIP-DRDoS attack. We implement the SIP-DRDoS attack by utilizing our SIP-based audit and attack software in our VoIP/SIP security lab environment that simulates an enterprise-grade SIP network. We then utilize our SIP-based defense tool to realize our novel defense mechanism against the SIP-DRDoS attack. Our experimental results prove that our defense approach can do a deep packet analysis for SIP traffic, detect SIP flood attacks, and mitigate them by dropping attack packets. While the SIP-DRDoS attack with around 1 Gbps of traffic dramatically escalates the CPU (central processing unit) usage of the SIP server by up to 74%, our defense mechanism effectively reduces it down to 17% within 6 min after the attack is initiated. Our approach represents a significant advancement over the existing defense mechanisms and demonstrates the potential to effectively protect VoIP systems against SIP-based DRDoS attacks.

Enhancing Bandwidth Utilization of IP Telephony Over IPv6 Networks

The demand for the telecommunication services, such as IP telephony, has increased dramatically during the COVID-19 pandemic lockdown. IP telephony should be enhanced to provide the expected quality. One of the issues that should be investigated in IP telephony is bandwidth utilization. IP telephony produces very small speech samples attached to a large packet header. The header of the IP telephony consumes a considerable share of the bandwidth allotted to the IP telephony. This wastes the network's bandwidth and influences the IP telephony quality. This paper proposes a mechanism (called Smallerize) that reduces the bandwidth consumed by both the speech sample and the header. This is achieved by assembling numerous IP telephony packets in one header and use the header's fields to carry the speech sample. Several metrics have been used to measure the achievement Smallerize mechanism. The number of calls has been increased by 245.1% compared to the typical mechanism. The bandwidth saving has also reached 68% with the G.28 codec. Therefore, Smallerize is a possible mechanism to enhance bandwidth utilization of the IP telephony.

Coverless Data Hiding in VoIP based on DNA Steganography with Authentication

Data hiding in Voice over Internet Protocol (VoIP) using coverless approach improves the undetectability by preserving the cover bits from modification. This paper focuses on hiding the secret message in VoIP streams using Deoxyribonucleic Acid (DNA) steganography approach. DNA steganography is known for its low cracking probability. The embedding process is done in two steps. The first step converts the VoIP sample, secret message and a user generated key (for Authentication) into m-RNA pattern during transcription and the second step converts the m-RNA to form a triplet during translation process to create a protein array, where the secret message is embedded. The secret message is extracted from the protein array by applying reverse translation and Transcription. The proposed approach improves the undetectability by leaving the cover bits unmodified with PESQ values 84% comparatively greater than the state of art techniques.

STFF-SM: Steganalysis Model Based on Spatial and Temporal Feature Fusion for Speech Streams

The real-time detection of speech steganography in Voice-over-Internet-Protocol (VoIP) scenarios remains an open problem, as it requires steganalysis methods to perform for low-intensity embeddings and short-sample inputs, as well as provide rapid detection results. To address these challenges, this paper presents a novel steganalysis model based on spatial and temporal feature fusion (STFF-SM). Differing from the existing methods, we take both the integer and fractional pitch delays as input, and design subframe-stitch module to organically integrate subframe-wise integer delays and frame-wise fractional pitch delays. Further, we design a spatial fusion module based on pre-activation residual convolution to extract the pitch spatial features and gradually increase their dimensions to discover finer steganographic distortions to enhance the detection effect, where a Group-Squeeze-Weighting block is introduced to alleviate the information loss in the process of increasing the feature dimension. In addition, we design a temporal fusion module to extract pitch temporal features using the stacked LSTM, where a Gated Feed-Forward Network is introduced to learn the interaction between different feature maps while suppressing the features that are not useful for detection. We evaluated the performance of STFF-SM through comprehensive experiments and comparisons with the state-of-the-art solutions. The experimental results demonstrate that STFF-SM can well meet the needs of real-time detection of speech steganography in VoIP streams, and outperforms the existing methods in detection performance, especially with low embedding strengths and short window sizes.

Session Initiation Protocol Proxy in a Role of a Quality of Service Control Application in Software-Defined Networks

This article deals with quality of service (QoS) in internet protocol (IP) telephony by applying software-defined networking (SDN) tools. The authors develop a new design that deterministically classifies real-time protocol (RTP) streams based on data found in session initiation protocol (SIP) using SIP proxy as a mediator, and the concept making this possible is called SDN. Compared to traditional networks, SDN allows us to approach network configuration differently. SDN networks are programmable through software applications running on top of the SDN controller. One of the technologies that might benefit from this concept is IP telephony, which often needs an additional priority management configuration to ensure consistent quality of its real-time media exchange. Typically, a session protocol for real-time communications is SIP, and as such, its infrastructure may be used to classify the traffic in question and take advantage of the centralized approach of SDN networks to distribute the class information across the switching devices. Different approaches and possible applications are discussed in the conclusion. The contribution of this paper lies in the proposal of SDN-based QoS mechanisms. The entire design of the concept was implemented and validated in a laboratory environment. The results clearly demonstrate the efficiency of the proposed approach.