VoIP Networks Research Articles

Detecting of Flow Timing Known Attacks and Protection in VoIP Networks

Detecting of Flow Timing Known Attacks and Protection in VoIP Networks

Low-Cost, Open-Source, Experimental Setup Communication Platform for Emergencies, Based on SD-WAN Technology

The rapid advancement of communication technologies underscores the urgent need for robust and adaptable emergency communication systems (ECSs), particularly crucial during crises and natural disasters. Although network-based ECSs have been extensively studied, integrating open-source technologies, such as software-defined wide area networks (SD-WAN) with private long-term evolution (LTE) base stations, is a relatively unexplored domain. This study endeavors to fill this gap by introducing an experimental ECS platform that utilizes a hybrid network, incorporating a VoIP network to enhance open-source and on-premises communications in targeted areas. Our hypothesis posits that a hybrid network architecture, combining SD-WAN and private LTE, can substantially improve the reliability and efficiency of ECSs. Our findings, supported by the open-source OMNeT++ simulator, illuminate the enhanced communication reliability of the network. Moreover, the proposed platform, characterized by autonomous wireless 4G/LTE base stations and an Asterisk VoIP server, demonstrates improved quality of service (QoS) and quality of experience (QoE), with minimal data loss. This research not only has immediate practical applications but also bears significant implications for the development of cost-effective, open-source communication networks, optimized for emergencies, critical infrastructure, and remote areas.

Performance analysis: Securing SIP on multi-threaded/multi-core proxy server using public keys on Diffie-Hellman (DH) in single and multi-server queuing scenarios.

The rapid replacement of PSTN with VOIP networks indicates the definitive phase-out of the PBX/PABX with smartphone-based VOIP technology that uses WLAN connectivity for local communication; however, security remains a key issue, regardless of the communication coverage area. Session initiation protocol (SIP) is one of the most widely adopted VOIP connection establishment protocols but requires added security. On the Internet, different security protocols, such as HTTPS (SSL/TLS), IPSec, and S/MIME, are used to protect SIP communication. These protocols require sophisticated infrastructure and some pose a significant overhead that may deteriorate SIP performance. In this article, we propose the following: i) avoid using Internet bandwidth and complex Internet protocols for local communication within an organization, but harness WLAN connectivity, ii) use multi-threaded or multicore computer systems to handle concurrent calls instead of installing hardware-based SIP servers, and iii) run each thread in a separate core. Cryptography is a key tool for securely transmitting confidential data for long- and short-range communication, and the Diffie-Hellman (DH) protocol has consistently been a popular choice for secret key exchanges. Primarily, used for symmetric key sharing, it has been proven effective in generating public/private key pairs, sharing public keys securely over public channels, and subsequently deriving shared secret keys from private/public keys. This key exchange scheme was proposed to safeguard VOIP communication within WLANs, which rely on the SIP for messaging and multimedia communication. For ensuring an efficient implementation of SIP, the system was rigorously analyzed using the M/M/1 and M/M/c queuing models. We analyze the behavior of SIP servers with queuing models with and without end-to-end security and increase users' trust in SIP security by providing a transparent sense of end-to-end security as they create and manage their private and public keys instead of relying on the underlying SIP technology. This research implements instant messaging, voice conversation, and secret key generation over DH while implementing and observing the role of multi-threading in multiqueue systems that serve incoming calls. By increasing the number of threads from one to two, the SIP response time improved from 20.23809 to 0.08070 min at an arrival rate of 4250 calls/day and a service rate of three calls/min. Similarly, by adding one to seven threads, the queue length was reduced by four calls/min. Implementing secure media streaming and reliable AES-based signaling for session confidentiality and integrity introduces a minor 8-ms tradeoff in SIP service performance. However, the advantages of implementing added security outweigh this limitation.

Pengembangan Media Pembelajaran Berbasis Google Sites pada Mata Pelajaran Teknologi Layanan Jaringan

The Government requires the classroom learning system to be replaced with learning at home through an online system. This research was conducted to develop learning media based on Google sites in the subject of network service technology. This research was conducted to see the feasibility of the developed Google sites-based learning media. The method used in this research is Research and Development with ADDIE development were Analysis, Design, Development, Implementation, and Evaluation. The development stage involved four validators as media experts and materials experts to assess the feasibility of media and materials. The implementation phase was limited to involving 34 students of class XII Computer Network Engineering. Collecting data using validation questionnaires, response questionnaires and interviews. Data analysis techniques are descriptive qualitative and quantitative. The results of this study are learning media products based on Google sites in the course of network service technology. One of the materials discussed is a firewall on a VoIP network. The results of the feasibility test show that the Google sites-based learning media is feasible to use. This is based on the validation of media experts who obtained an average result of 97,3% in the “strongly agree” category, material expert validation obtained an average result of 88,6% in the “strongly agree” and based on the results of the questionnaire students obtained an average result of 92,86% in the “strongly agree”. This indicates that all aspects assessed are in the very good category.

ОПТИМІЗАЦІЯ КОМП’ЮТЕРНОЇ ТЕХНІКИ ПРИ МОДЕРНІЗАЦІЇ ЄДИНОЇ СИСТЕМИ ЦЕНТРАЛІЗОВАНОГО КОНТРОЛЮ І ВИМІРІВ МЕРЕЖІ СКС-7 З ВИКОРИСТАННЯМ РЕСУРСІВ ЦЕНТРУ ОБРОБКИ ДАНИХ

The architecture of the unified system of centralized control and measurements of the SKS-7 network is outdated, the technologies used to organize its work and implement data storage need to be updated and optimized, because the current state of the system causes a decrease in speed and performance, as well as a critical drop in reliability. Considering the factors, there is also a need to modernize the software and hardware of the IP protocol analysis and monitoring system. The system does not provide the full range of functions necessary for a full-fledged analysis of the VoIP network, at the moment. Updating the system architecture became a necessity to improve efficiency and reliability. The work describes the process of modernization of the SKS-7 network control and measurement system using data center resources by optimizing computer equipment and expanding the capabilities of the systems used to monitor the OKS-7 network. The system of analysis and control and measurements of the SKS-7 network is presented and its main functionalities are defined. The main signaling protocols were analyzed and a system of analysis and monitoring of IP protocols was developed. The modernized control and measurement system of the SKS-7 network allows to identify deficiencies when testing elements of telecommunication networks and analyzing their impact on operation, analyzing the information and signal load of SKS-7 and VoIP networks in real time and in previous periods.

VoIPChain: A decentralized identity authentication in Voice over IP using Blockchain

Confidentiality, availability, integrity, authentication, and non-repudiation when together became the top priority for secure communication. However, authentication is the first line of defense in all these security parameters. Unfortunately, several authentication mechanisms in the traditional methods cannot provide secure communication due to various vulnerabilities such as single point of failure and privacy issues. Furthermore, key distribution mechanisms such as Certificate Authority (CA) and Trusted Third Party (TTP) distribute keys over the central architecture fail, thus secure communication cannot occur. As a solution to these drawbacks, this paper proposes a new decentralized blockchain-based identity authentication mechanism for VoIP networks, VoIPChain. The proposed scheme utilizes the main features of the blockchain platform, such as immutability, transparency, and fault tolerance, to provide data privacy and secure communication in VoIP applications. The proposed scheme is evaluated as an actual implementation using the virtual Ethereum platform and Python language. The experimental results show that the proposed scheme is an efficient and cost-effective solution for the call process as a decentralized identity authentication system in VoIP. In addition, end-to-end secure call performance outperforms the existing blockchain-based authentication mechanisms by 30%–70% in terms of average time delay. The proposed scheme is almost ten times faster than the TLS process to authenticate between parties. Moreover, compared to fast but less secure basic methods over the SIP authentication the proposed scheme has an acceptable time delay in a call process.

Call Me Maybe: Using Dynamic Protocol Switching to Mitigate Denial-of-Service Attacks on VoIP Systems

Voice over IP is quickly becoming the industry standard voice communication service. While using an IP-based method of communication has many advantages, it also comes with a new set of challenges; voice networks are now accessible to a multitude of internet-based attackers from anywhere in the world. One of the most prevalent threats to a VoIP network are Denial-of-Service attacks, which consume network bandwidth to congest or disable the communication service. This paper looks at the current state of research into the mitigation of these attacks against VoIP networks, to see if the mechanisms in place are enough. A new framework is proposed titled the “Call Me Maybe” framework, combining elements of latency monitoring with dynamic protocol switching to mitigate DoS attacks against VoIP systems. Research conducted around routing VoIP over TCP rather than UDP is integrated into the proposed design, along with a latency monitoring mechanism to detect when the service is under attack. Data gathered from a Cisco Packet Tracer simulation was used to evaluate the effectiveness of the solution. The gathered results have shown that there is a statistically significant improvement in the response times of voice traffic when using the “Call Me Maybe” framework in a network experiencing a DoS attack. The research and findings therefore aim to provide a contribution to the enhancement of the security of VoIP and future IP-based voice communication systems.

Analysis Of Voice Quality on Aircraft Telephone Through Internet Telephony Gateway in Voice Over Internet Protocol

Along with the development of the era, analog telephones have been replaced by IP Phones whose communication range can be wider and more flexible. Therefore, analog telephones are almost forgotten because of the limited communication range. With the addition of ITG (Internet Telephony Gateway) then analog telephones can communicate in VoIP networks. This research will discuss the analysis of voice quality in the communication process between telephone sets via ITG in a VoIP network based on the shape of the input and output, the amplitude response to frequency changes and the delay. Data retrieval is done by using a transmission test set as a substitute for human voice and knowing the amplitude value generated every time there is a change in frequency and an oscilloscope as a tool to determine the shape of the input and output waves as well as the delay. The results of this study indicate that delay generated by communication between analog telephones is <1 ms. When viewed from the form of the information signal generated from communication between telephone sets, that is, the shape of the input and output information signals are both sines, so it can be said that the information signal has not decreased in quality. The amplitude response resulting from frequency changes in communication between analog telephones is, the greater the value of the communication frequency.

Analisis Keamanan Komunikasi VoIP Server Portable Dilengkapi OpenVPN Menggunakan Linux Asterisk

Communication is a very important thing, VoIP is one of the supports that is often used to facilitate communication. VoIP is connected to the internet network so that communication can occur, but VoIP is very vulnerable to security problems in communication. It is very easy for eavesdropping to occur by parties who want to know ongoing information for certain purposes, therefore VoIP security must exist, openVPN is one that is often used. With services that are private / private so that the communication network is protected. In this security research, testing is carried out on a VoIP network with a linux asterisk server using a VPN and not using a VPN. With different results on tests of delay, packet loss, jitter. In the test made a phone call that was registered on the voip server so that the call could be made. The call test was carried out several times and for 6 days in order to see if the results produced were as expected because the internet network could also be disrupted by weather condition. The first phone call that was made without using a VPN the results that were obtained were not very good, in contrast to the second phone call that used a VPN the results produced by the analysis were better.

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.

Softwarization and virtualization of VoIP networks

Softwarization and virtualization of VoIP networks

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.

Optimum availability and security for IMS-based VOIP networks using weight product method

Optimum availability and security for IMS-based VOIP networks using weight product method

Performance of VoIP Using Routing Open Shortest Path First with Multi-Protocol Label Switching

Voice over internet protocol is a communication technology in the world of computer network that can be used for sending voice or video and data transmission over Internet Protocol in real time. VoIP network can be implemented with Asterisk applications as a server to a Private Automatic Branch eXchange applied in a Graphical Network Simulator 3. In this study, VoIP communication using routing OSPF within MPLS will be calculated, the QoS value collected based on impact performance under normal condition and not normal condition (link failure) different varian bandwidth in the network. The results from the simulation show that in normal condition and not normal condition there is average delay value with routing OSPF 4 ms and routing OSPF with MPLS 5 ms, the value of jitter max which same of 6 ms using varian bandwidth from 256 kbps and 512 kbps. All of the QoS parameters, such as delay, jitter and packet loss will be compare to standard ITU-T G.114. This research can be extended with addition of another measurement or another protocol.

Factors Affecting the Accessibility of Voice Telephony for People with Hearing Loss: Audio Encoding, Network Impairments, Video and Environmental Noise

This paper describes four studies with a total of 114 individuals with hearing loss and 12 hearing controls that investigate the impact of audio quality parameters on voice telecommunications. These studies were first informed by a survey of 439 individuals with hearing loss on their voice telecommunications experiences. While voice telephony was very important, with high usage of wireless mobile phones, respondents reported relatively low satisfaction with their hearing devices’ performance for telephone listening, noting that improved telephone audio quality was a significant need. The studies cover three categories of audio quality parameters: (1) narrowband (NB) versus wideband (WB) audio; (2) encoding audio at varying bit rates, from typical rates used in today's mobile networks to the highest quality supported by these audio codecs; and (3) absence of packet loss to worst-case packet loss in both mobile and VoIP networks. Additionally, NB versus WB audio was tested in auditory-only and audiovisual presentation modes and in quiet and noisy environments. With WB audio in a quiet environment, individuals with hearing loss exhibited better speech recognition, expended less perceived mental effort, and rated speech quality higher than with NB audio. WB audio provided a greater benefit when listening alone than when the visual channel also was available. The noisy environment significantly degraded performance for both presentation modes, but particularly for listening alone. Bit rate affected speech recognition for NB audio, and speech quality ratings for both NB and WB audio. Packet loss affected all of speech recognition, mental effort, and speech quality ratings. WB versus NB audio also affected hearing individuals, especially under packet loss. These results are discussed in terms of the practical steps they suggest for the implementation of telecommunications systems and related technical standards and policy considerations to improve the accessibility of voice telephony for people with hearing loss.

Implementation of VOIP (Voice Over Internet Protocol) at Bhayangkara Hospital Bengkulu

In communication technology, voice communication is something that will be a very important part, because currently voice communication is considered the most practical communication. Bhayangkara Bengkulu Hospital is one of the Government Hospitals engaged in the health sector in Bengkulu Province. So far in communicating, Bhayangkara Bengkulu Hospital is still simple and communication media is still limited. Communication media between officers still uses HT and HP, because only a few rooms use telephones. The purpose of this research is to build a VoIP network at Bhayangkara Bengkulu Hospital so that it can facilitate communication at Bhayangkara Bengkulu Hospital. VoIP implementation at Bhayangkara Bengkulu Hospital is made using Mikrotik RB951 and Trixbox which is placed on a computer that will be used as a server. The Zoiper application is installed on each computer in each treatment room and other agencies. The functionality of this VoIP network makes it easier and can help the Bhayangkara Bengkulu Hospital in communicating. This system has been tested and the results are satisfactory as desired, although there are still obstacles in the practice of using it

Pengembangan Server VOIP Menggunakan FREEPBX Dan Asterisk Berbasis Raspberry PI

Raspberry Pi is a single board computer that the author uses as a voip server based on asterisk. To use this voip network, each device must have a static IP as well as be connected to a stable network. The advantage of using raspberry pi is that it no longer costs money to make voice calls for both package and pulse costs. The purpose and benefits of this research to know how raspi works, know how to install voip server using freepbx and asterisk, know how to install voip client on smartphones and laptops dang know the topology of raspi-based voip network, and know how to test stages of searching for throughput, loss package, delay, and jitter using wireshark

A Survey on Voice over Internet Protocol (VoIP) Reliability Research

VoIP technology deals with the real-time data communication for voice transfer in the form of digital packets through internet communication and facilitates public to make use of internet for video and phone calls. The voice data packets are transferred from source to the destination and vice versa. VoIP require high speed internet connection for data transfer on data network. It is necessary to arrive the data packet from its source to the destination with high level of reliability. It is very important to analyse link failure, packet loss, delay and jitter during the data communication. Components involved in data communication should be reliable. This paper provides a survey for step wise development and use of reliability techniques for gaining high quality of voice in VoIP network.

Digital divides across consumers of internet services in Spain using panel data 2007–2019. Narrowing or not?

Digital gaps have the potential to exacerbate the inequalities that exist in society. The main objective of this paper is to study the gaps that occur in the use of internet services by households in Spain during the period 2007–2019 and to become useful in the design of policies addressed to narrow specific digital divides. The data is a panel obtained from the ICT-H Survey on Households of the National Statistics Institute. This paper defines the gaps as the differences in the use of internet services across individual consumers. A selected group of twelve digital services are considered: e-commerce, e-selling, e-tourism, e-learning, e-health, e-banking, e-government, VoIP, chat, email, cloud services, and social networks. The second level digital gaps are analyzed in each service according to six relevant socio-demographic characteristics: gender, age, education, digital skills, population size, and income. A set of graphs show the evolution of the gaps. Gaps are narrowing in most dimensions and specific characteristics, but not in others such as age, education, and digital skills. The gaps reveal the evolution of digitization and in some cases of digital exclusion for specific groups. Specific knowledge about digital gaps is useful for policymakers, since closing the digital divide is an explicit policy goal in this country, as well as in other parts of Europe. Then, a dynamic panel data model was proposed and estimated using Arellano and Bond techniques. A dynamic/network effect was found, as well as other socio-demographic determinants. Finally, the paper contains conclusions, policy recommendations and an agenda for future research. The policy recommendations consist of digital education programs targeted at the most exposed groups such as the elderly, the less well-educated and people with lower digital skills.

USE OF THE STATISTICAL ANALYSIS METHODS TO DETECT VOIP NETWORK TRAFFIC ANOMALIES

The purpose of the study is to automate detection of signs of technical malfunction and deterioration in the quality of services in the VoIP network. Deterioration of quality leads to a decrease in the volume of services provided, and consequently a profit, a decrea se in customer loyalty and loss of market share. There are three levels of quality degradation in VoIP networks: lack of access at the data network layer, inability to establish a voice connection and lack of access at the level of application services, and degradation of service quality. The analysis of absolute and relative statistical quality indicators of IP-telephony in accordance with the requirements of the international ITU-T standards has been performed, and main indicators and formulas for their calculation are listed. The values of statistical quality indicators are periodically calculated for external channels, subscriber groups and tariff directions. The primary data is CDR (call data records). To detect anomalous changes in the values of the quality indicators it is proposed to use the exponential smoothing method – the Holt-Winters additive model. The deviation of the current value of the quality indicators from the forecasted one is calculated. The confidence range is calculated using the Brutlag method. If the deviation goes beyond the confidence range, the change in the value of the indicator is considered anomalous and the value of the anomaly coefficient of the indicator is set. The period is characterized by a vector of anomaly coefficients of all quality indicators. To classify the period as anomalous, the value of the module of the anomaly coefficients vector of the given period is used. The features of the method application are also considered, in particular the choice of the seasonal period and the calculation of anomaly coefficients during periods of minimum load. The proposed method allows to diagnose an abnormal change in the values of VoIP service quality indicators in automated mode. The prototype of an automated system for the quality of VoIP service monitoring was developed on the base of the method described.