VoIP Research Articles

Quality Evaluation of VoIP Codecs in Cellular Mobile Networks

Maintaining the Quality-of-Service (QoS) of Voice-over-Internet Protocol (VoIP) is a major problem due to the instability of network conditions. One factor that influences VoIP call quality is the employed codec. In this research, the influence of codec on the QoS of VoIP calls is evaluated under Wi-Fi, 2G, 3G and 4G networks. VoIP calls are established using PCMA, PCMU, GSM and G.722 codecs and QoS parameters including jitter, round trip time and packet loss are measured. QoS measurements are compared against considered networks and codecs. Results illustrate that there exists a codec that better suits each network with considerable advantages over other codecs and minimum drawbacks.

Characteristic analysis of queue theory in Wi-Fi applications using OPNET 14.5 modeler

Wireless Fidelity (Wi-Fi) broadband network technology has created great influence in the evolution of broadband wireless networks that are anticipated to progress regarding broadband speed and coverage. Several Wi-Fi hotspots are available everywhere, making it a medium of internet access that is easier to use compared to a local area network (LAN). However, the internet being the best effort network doesn’t provide the required Quality of Service (QoS) and there is no differentiation of service traffic. The chief aim of the current paper is to study the operation of the three organizing mechanisms: First-In, First-Out (FIFO) method – the standard method of network implementation to process the packets one by one as it arrives, Priority Queuing (PQ) and Weighted Fair Queuing (WFQ) whereas PQ and WFQ classify the types of traffic based on service priority. In addition, WFQ assigns fair weight to each service on multiple traffic classes like video conferencing, Voice over Internet Protocol (VoIP), and File Transfer Protocol (FTP), using Telkom ST3’s Wi-Fi network. This study applies four different scenarios: the first scenario applies the methods without any queuing discipline; the second scenario implements the methods with FIFO; the third scenario carries out the methods with PQ and the last scenario applies the methods with WFQ. The studies have shown that “end-to-end packet delay and packet delay variation for VoIP in the WFQ scenario” is good when compared to other queuing mechanisms with values of 171.717 ms and 0.977 ms, respectively. In the case of videotape conferencing also, the performance is better in the case of WFQ with values of 32.495 ms and 7.207 ms, correspondingly, since the WFQ has a “bandwidth allocation” tailored to the requirements

Development of crypto-code constructs based on LDPC codes

The results of developing post-quantum algorithms of McEliece and Niederreiter crypto-code constructs based on LDPC (Low-Density Parity-Check) codes are presented. With the rapid growth of computing capabilities of mobile technologies and the creation of wireless mesh and sensor networks, Internet of Things technologies, and smart technologies on their basis, information security is becoming an urgent problem. At the same time, there is a need to consider security in two circuits, internal (directly within the network infrastructure) and external (cloud technologies). In such conditions, it is necessary to integrate threats to both the internal and external security circuits. This allows you to take into account not only the hybridity and synergy of modern targeted threats, but also the level of significance (degree of secrecy) of information flows and information circulating in both the internal and external security circuits. The concept of building security based on two circuits is proposed. To ensure the security of wireless mobile channels, it is proposed to use McEliece and Niederreiter crypto-code constructs based on LDPC codes, which allows integration into the credibility technology of IEEE 802.15.4, IEEE 802.16 standards. This approach provides the required level of security services (confidentiality, integrity, authenticity) in a full-scale quantum computer. Practical security technologies based on the proposed crypto-code constructs, online IP telephony and the Smart Home system based on the use of an internal server are considered

The effect of internet telephony and a cochlear implant accessory on mobile phone speech comprehension in cochlear implant users.

PurposeIn individuals with severe hearing loss, mobile phone communication is limited despite treatment with a cochlear implant (CI). The goal of this study is to identify the best communication practice for CI users by comparing speech comprehension of conventional mobile phone (GSM) calls, Voice over Internet Protocol (VoIP) calls, and the application of a wireless phone clip (WPC) accessory. MethodsThis study included 13 individuals (mean age 47.1 ± 17.3 years) with at least one CI. Frequency response and objective voice quality were tested for each device, transmission mode and the WPC. We measured speech comprehension using a smartphone for a GSM call with and without WPC as well as VoIP-calls with and without WPC at different levels of white background noise.ResultsFrequency responses of the WPC were limited (< 4 kHz); however, speech comprehension in a noisy environment was significantly improved compared to GSM. Speech comprehension was improved by 9–27% utilizing VoIP or WPC compared to GSM. WPC was superior in noisy environments (80 dB SPL broadband noise) compared to GSM. At lower background noise levels (50, 60, 70 dB SPL broadband noise), VoIP resulted in improved speech comprehension with and without WPC. Speech comprehension scores did not correlate with objective voice quality measurements. ConclusionSpeech comprehension was best with VoIP alone; however, accessories such as a WPC provide additional improvement in the presence of background noise. Mobile phone calls utilizing VoIP technology, with or without a WPC accessory, result in superior speech comprehension compared to GSM.

Enhanced Network Security for Improved Trustworthiness of VoIP Applications via Cuckoo Search and Machine Learning

Objectives: To offer network design for secure Voice over Internet Protocol (VoIP) services with improved Quality of Services (QoS) parameters. Methods: The network area is created with required number of nodes. INTRA-SR process is employed for tracing of the route from the source node and INTER-SR is involved to reach a destination node. Cuckoo Search (CS) based optimization followed for the broadcasted voice packets. The Machine learning classifiers as SVM and ANN applied to decrease the instances of loss of voice packets. Simulation work is performed by using MATLAB 2018 and results obtained plotted in graphs using MS-Excel. Findings: The proposed design evaluated by incorporating CS algorithm to minimize the packet drops. SVM and ANN hybrid used to locate secure routing path. The QoS for throughput, latency and jitter are observed. The results exhibited higher average throughput of 98.8% irrespective of the attack instances. Lower average latency and jitter of 1.9s and 2.51ms are also exhibited by the proposed work. Similarly, latency work employing Ant Colony Optimization (ACO) with multiplex and multicasting (MM) is 2.66s that get increased to 3.16s due to attack. Novelty/Applications: The proposed algorithm significantly enhanced by deployment of highly protected network design with QoS and security. Application of ANN and SVM shown the improvement in performance for VoIP services. In addition to the regression analysis validates the results by applying other optimization algorithms. Keywords: Voice over Internet Protocol; Cuckoo Search; Support Vector Machine; Artificial Neural Network; Machine Learning

The Effect of Decreased Voice Quality in a VoIP Network

Voice over Internet Protocol (VOIP) is a telephone system telecommunication technology that uses an IP (Internet Protocol) network. VoIP offers numerous advantages for customers and communication, including cost savings, phone and service portability, mobility, and integration with other applications. Because voice is sensitive to delay and jitter, bandwidth must be guaranteed while it is being transported. There are some issues with Quality of Service (QoS), the most significant of which is latency in VOIP. The purpose of this paper is to investigate the effect of delay on speech quality in Voice Over Internet Protocol (VOIP). One source of delay is the buffer that already exists in the original multiplexing'-multicast node. There will be no buffering to store packets in the proposed multiplexing'-multicast algorithm, so any packet received will be sent immediately unless the packet arrives after a specific time or in a different order, where OPNET IT Guru Academic Edition simulation was used to analyze network performance.

Novel Method of Improving Quality of Service for Voice over Internet Protocol Traffic in Mobile Ad Hoc Networks

In recent years, the application of Mobile Ad-hoc Network (MANET) with Voice over Internet Protocol (VoIP) has been increased. However, the level of Quality of Service (QoS) for VoIP traffic in MANET, while there is no infrastructure, will reduce when dealing with a large number of calls. In this type of dynamic environment, the developing of a new infrastructure becomes more costly and time-consuming. In this paper, we proposed an efficient method, called the Quality of Service-Nearest Neighbor (QoS-NN), to improve the QoS level for VoIP in order to manage the huge number of calls over MANET network. We utilized the Ad-hoc On-demand Distance Vector (AODV) protocol as the underlying routing protocol to implement our proposed method. We evaluated the proposed QoS-NN method using Network Simulator version 2 (NS2). The performance of the proposed QoS-NN method was compared with Lexicographic order method. The comparison was evaluated in terms of R-factor, end-to-end delay, packet loss ratio, and packet delivery ratio performance metrics. In addition, the proposed method evaluated under different network parameters such as VoIP CODECs, node mobility speed, number of calls and number of nodes. The comparison results indicate that the proposed QoS-NN outperform the Lexicographic order method.

Method for ensuring information security of a VOIP telephony network with a forecast of an intruder's intrusion strategy

Introduction: the development of technologies in the field of information and telecommunications, as well as the unification of networks, and in particular the construction of distributed VoIP telephony networks, allow us to formulate the problem that the known methods of managing the protection of VoIP networks are not effective enough in modern conditions, since they take into account only one side of the information confrontation. Purpose: To develop a method for ensuring the information security of a VoIP telephony network, which allows to increase the probability of VoIP network security by reducing the time required for analyzing the actions of the violator, analyzing and processing risks under the influence of the violator. Results: Based on the proposed structure of an information security management system integrated into a VoIP network, a method for ensuring the information security of a VoIP telephony network under the influence of an intruder has been developed by introducing decision-making support processes in the VoIP network information security management system using intelligent intrusion detection tools distributed across segments. This method allows you to build a graph of events of the intruder's actions, on the basis of which mathematical modeling of MiTM and SPIT attacks on the VoIP telephony network is carried out. As a result of the simulation, the dependence of the successful impact on the internal and external characteristics of attacks is obtained, which is the main one of the developed software, which allows to obtain the values of the probability of security of the VoIP network from the parameters of the intruder's impact for further selection of adequate measures for managing the information security of the VoIP telephony network. The method includes the processes of analyzing the digital stream and determining the parameters of protocols and profiles of intruder attacks. Practical relevance: The developed method provides an opportunity to study issues aimed at the security of the VoIP-telephony network, which is affected by violators.

EVALUATING QOS IN NETWORKS WITH ACTIVE QUEUE MANAGEMENT ALGORITHMS

Packets queuing and scheduling in network routers is a key point of overall network performance. Many applications, especially applications require Quality of Services (QoS), need techniques to pass their packets throughout routers and control and/or avoid congestions in highly congested routes. Therefore, many Active Queue Management (AQM) algorithms have been developed to avoid or control congestion in routers and provide fairness among traffic flows. This paper provides an extensive evaluation performance analysis of three well-known AQM algorithms including RED, REM and traditional Drop-Tail with QoS application requirements. The evaluation performance is conducted by employing network simulator version 2 (NS2). The network performance is measured with Voice over Internet Protocol (VoIP) traffic and three performance metrics including, throughput, latency, and PSD (Probability of Sequential Drop). The analysis shows no AQM algorithm achieve all the VoIP QoS requirements, A new AQM is needed to fulfil QoS requirements and manage queue to handle unresponsive flows.

A Study of Data Augmentation for ASR Robustness in Low Bit Rate Contact Center Recordings Including Packet Losses

Client conversations in contact centers are nowadays routinely recorded for a number of reasons—in many cases, just because it is required by current legislation. However, even if not required, conversations between customers and agents can be a valuable source of information about clients or future clients, call center agents, markets trends, etc. Analyzing these recordings provides an excellent opportunity to be aware about the business and its possibilities. The current state of the art in Automatic Speech Recognition (ASR) allows this information to be effectively extracted and used. However, conversations are usually stored in highly compressed ways to save space and typically contain packet losses that produce short interruptions in the speech signal due to the common use of Voice-over-IP (VoIP) in these systems. These effects, and especially the last one, have a negative impact on ASR performance. This article presents an extensive study on the importance of these effects on modern ASR systems and the effectiveness of using several techniques of data augmentation to increase their robustness. In addition, ITU-T G.711, a well-known Packet Loss Concealment (PLC) method is applied in combination with data augmentation techniques to analyze ASR performance improvement on signals affected by packet losses.

Изучение влияния стратегии маршрутизации вызовов на эффективность работы колл-центра медицинской организации, оказывающей первичную медико-санитарную помощь

The article provides an estimation of the efficiency of a call-centre work when using two different strategies of targeting incoming calls. The authors clearly demonstrate a reduction in the waiting time for communication with the operator and the frequency of call cancellations, an increase in the number of processed requests and an increase in the level of service when using the strategy of interactive navigation of incoming calls, which generally creates favorable conditions for increasing the availability of medical care to the population. The purpose of the study is to study the effect of routing incoming calls on the efficiency of a medical call center in order to improve its operation. Materials and methods. The comparison of the effectiveness of the polyclinic call center was carried out according to the indicators according to GOST R 55540-2013 under the conditions of using the scheme of random and interactive navigation of incoming calls. The target values of quality indicators are established empirically, taking into account the accepted industrial standards. The analysis of statistical information was carried out using special software “Asternic call center stats”. A total of 72 statistical indicators were studied in both cases within the same seasonal period. Results. It was found that the organization of the scheme of interactive guidance of incoming calls helps to increase the productivity of a medical call center by optimizing the processing of calls without increasing the number of operators. The average waiting time for a patient to communicate with an operator has decreased by 2 times, the frequency of call cancellations has decreased by 3 times, the processing time for patient calls has decreased by 15.6%, the level of service has increased by 1.6 times. The practical significance of the proposed scheme for navigating incoming calls is due not only to the effective increase in the functionality of the medical call center, but also to the ease of scaling due to the unified IP telephony architecture in medical call centers. Conclusions. Thus, as a result of the study, a call routing strategy has been identified, which makes it possible to improve the performance indicators of the medical CC, to service more calls faster without attracting additional personnel, and to create conditions for increasing the availability of medical care to the population.

Real-Time Implementation and Testing of VoIP Vocoders with Asterisk PBX Using Wireshark Packet Analyzer

In the present-day digital world, the transmission of information like voice, text, video, etc. demands more bandwidth for fast and accurate transmission. Usually, 90% of information transmitted is the human voice, and various speech processing techniques are adapted for optimizing the bandwidth requirement and enhancing the performance of multiple modules like vocoders in mobile networks. To test the performance analysis of various vocoders, we used a Wireshark simulator with an Asterisk, and it is a software implementation of a private branch exchange (PBX). This work presents the performance evolution of existing voice over internet protocol (VoIP) vocoders using Wireshark with PBX Asterisk’s, The PBX Asterisk’s to analyze and capture the packets or sniffers. The Wireshark tools can diagnose the errors, the traffic flow with a graphical view and generate the statistics in the network. It is a software tool to analyze the traffic data with an intuitive and simple graphical interface in software tools are used to study. The extensive simulations are performed, and the percentage of data saved was calculated and compared with the various vocoders.

Introducing a Novel Data Over Voice Technique for Secure Voice Communication

The current increasing need for privacy-preserving voice communications is leading to new ideas for securing voice transmission. This paper refers to a relatively new concept of sending encrypted data or speech as pseudo-speech in the audio domain over existing voice communication infrastructures, like 3G cellular network and Voice over IP (VoIP). The distinctive characteristic of such a communication system is that it relies on the robust transmission of binary information in the form of audio signal. This work presents a novel Data over Voice (DoV) technique based on codebooks of short harmonic waveforms. The technique provides a sufficiently fast and reliable data rate over cellular networks and many VoIP applications. The new method relies on general principles of Linear Predictive Coding for voice compression (LPC voice coding) and is more versatile compared to solutions trained on exact channel models. The technique gives by design a high control over the desired rate of transmission and provides robustness to channel distortion. In addition, an efficient codebook design approach inspired by quaternary error correcting codes is proposed. The usability of the proposed DoV technique for secure voice communication over cellular networks and VoIP has been successfully validated by empirical experiments. The paper details the system parameters, putting a special emphasis on system's security and technical challenges.

Investigation of Techniques for VoIP Frame Aggregation Over A-MPDU 802.11n

The widespread and desirable features of IP and IEEE 802.11 networks have made these technologies a suitable medium for carrying voice over IP (VoIP). However, a bandwidth (BW) exploitation obstacle emerges when 802.11 networks are used to carry VoIP traffic. This BW exploitation obstacle is caused by the large 80-byte preamble size of the VoIP packet and a waiting time of 765 μs for each layer 2 VoIP frame. As a solution, IEEE 802.11n was consequently designed with a built-in layer 2 frame aggregation feature, but the adverse impact on the VoIP performance still needed to be addressed. Subsequent VoIP research has proposed numerous enhancements to the built-in IEEE 802.11n aggregation techniques. In this study, we analyse and propose enhancements to the BW exploitation and VoIP performance over IEEE 802.11n. We also recommend new techniques to enhance VoIP performance via IEEE 802.11n frame aggregation implementations. Furthermore, we contribute guidelines for creating a robust, suitable, and reliable aggregation technique for VoIP over the 802.11n standard.

Prenos Q-signalizacije postupkom tunelovanja u automatskoj telefonskoj mreži integrisanih servisa Vojske Srbije

Introduction/purpose: To specify the practical application of ECMA355 and ECMA-336 Standards for Q-SIG tunneling and the implementation of mapping functions via the existing IP (Internet Protocol) network of the Serbian Armed Forces (Intranet SAF), in the Private Automatic Telephone Network SAF (PATN SAF), as the main part of the Private telecommunication-information networks of integrated services SAF (PISN SAF). Methods: Description of the implemented solution and analysis of the software parameters of the established transmission SIP route, with the display of the results obtained in the fight with jitter and echo in the network. Results: With such a solution, it was achieved that participants from the peripheral parts of the PISN SAF, which operate on the principle of transmission and circuit switching by TDM (Time Division Multiplexing), can connect with each other via the newly established central IP network SAF (Core network) which operates on the principle of transmission and switching packets with the SIP (Session Initiation Protocol), without losing the functionality of QSIG from the framework of the digital telecommunication network of integrated services ISDN (Integrated Services Digital Network). Conclusion: The article deals with the modern IP PINX (Private Integrated Services Network Exchange) manufactured by Mitel, type MX-ONE Service Node 6.0, which is implemented at the transit level PATN SAF and which successfully implements the process of tunneling Q-SIG through the IP network and the necessary functions for mapping the transmission of tunneled QSIG messages and mapping voice (and other audio) information to VoIP (Voice over IP) communication media streams through that network. Also, the basic elements for its software preparation during the introduction of a new SIP route, with a capacity of 30 IP trunks in a transmission beam realized with 100 Mb/s-T Ethernet, are given, and the fight with the present jitter and echo in the network is described. Finally, the paper presents the experience-based values of the parameters for reducing the influence of jitter and suppressing echo.

MRTP – A Novel Real-time Protocol for Patient-Doctor Online Meetings and Monitoring

This paper presents the design of a novel Voice over IP (VoIP) protocol that carries real-time voice and medical data in the same stream while preserving good quality of service (QoS) for online medical care. The work deliberates the reasons for the choice of the Session Initiation Protocol (SIP). The proposed medical real-time protocol (MRTP) is designed to be compatible with the existing Real Time Protocol (RTP) and has advanced features for medical data transmission. The MRTP leverages the real-time performance due to less overhead i.e., less delays. A major challenge is that some medical data is life-critical i.e., losses may cause harm. Hence, reliability and satisfactory QoS are crucial. A major result is the decrease in average-delay by around 74% compared to concurrently utilizing RTP for the audio part and TCP-connections (sockets) for the same medical data.

Optimal Hybrid Precoding Based QoE for Partially Structured Massive MIMO System

Precoding is a beamforming technique that supports multi-stream transmission in which the RF chain plays a significant role as a digital precoding at the receiver for wireless communication. The traditional precoding contains only digital signal processing and each antenna connects to each RF chain, which provides high transmission efficiency but high cost and hardware complexity. Hybrid precoding is one of the most popular massive multiple input multiple output (MIMO) techniques that can save costs and avoid using complex hardware. At present, network services are currently in focus with a wide range of traffic volumes. In terms of the Quality of Service (QoS), it is critical that service providers pay a lot of attention to this parameter and its relationship to Quality of Experience (QoE) which is the measurement of the overall level of user satisfaction. Therefore, this paper proposes hybrid precoding of a partially structured system to improve transmission efficiency and allocate resources to provide network services to users for increasing the user satisfaction under power constraints that optimize the quality of baseband precoding and radio frequency (RF) precoding by minimizing alternating algorithms. We focus on the web browsing, video, and Voice over IP (VOIP) services. Also, a Mean Opinion Score (MOS) is employed to measure the level of user satisfaction. The results show that the partially structured system provides a good user satisfaction with the network's services. The partially structured system provides high energy efficiency up to 85%. Considering web service, the partially structured system for 10 users provides MOS at 3.21 which is higher than 1.75 of fully structured system.

ANALYSIS OF VOICE OVER INTERNET PROTOCOL ATTACKS DIRECTED AT IP TELEPHONY

The organizational and technical problems of ensuring the protection of the communication channel via IP telephony against information leakage were analyzed. Possible attacks of this communication channel were considered and protection options based on available methods are proposed.

Saving the Bandwidth of IPv6 Networks Using the Fields of the Packet Header

IPv6 protocol is the future of IP networks due to its large IP address capacity. One of the key consequences of this large capacity is that the IP protocol header has enlarged from 20-byte in IPv4 to 40-byte in IPv6. This will consume a considerable share of the bandwidth (BW) for VoIP applications, which produce small packets in tens of bytes only. To handle this issue, we have introduced an efficient technique to use the superfluous fields in the VoIP packet header, including the IPv6 header, to hold the voice data of the packet. This introduced technique is called the Eliminator technique because it shortens or eliminates the packet payload. The Eliminator technique has been analyzed against the typical technique of transferring the VoIP data. The BW utilization of the two techniques has been compared with different codecs namely G.726, G.728, and G.723.1 codecs. The analysis showed that the BW wasting has alleviated by 16.4%, 19.1%, and 25% with these codecs respectively. This BW utilization improvement will indirectly boost the quality of the VoIP calls, especially in the wireless environments. Accordingly, the Eliminator technique is a viable solution for alleviating the wasted BW when running VoIP in IPv6 networks.

Voice Over Internet Protocol (VoIP) User Location Detection Application

This study aimed to build a Voice Over Internet Protocol (VoIP) User Location Detection Application and address information security for the public if they receive attention from unknown or undesired people. This application displays a map or location of people who communicate with others via mobile phones. The Application Development Methodology used the Iteration Method appeared in the database. Application testing used the black-box testing method, and the test results ran according to the expected function. The final result of this research was a telephone application that functioned by displaying a map on the phone screen of the call’s receiver. Therefore, without being asked, when the recipient picks up the phone, they connect to the map -in case something unwanted happens or when the callers merely want to exchange information with each other.