VoIP System Research Articles

ОБЗОР ПУБЛИКАЦИЙ ЗАРУБЕЖНОГО СЕГМЕНТА ПО БЕЗОПАСНОСТИ VOIP-СЕТЕЙ: ГЕНЕРАЦИЯ DOS-АТАК И ИХ ОБНАРУЖЕНИЕ

The paper is devoted to the problem of protecting VoIP systems from DoS/DDoS attacks, as one of the most relevant for the field of digital telecommunications. A review of a significant number of publications by foreign scientists devoted to methods of creating this type of attack, as well as countering them, was carried out. A systematization of the review results is proposed in the form of a comparative table according to the following 10 criteria: year of publication, stage of the life cycle of the attack, its type, protection method, degree of implementation, its resource intensity, practical applicability, effectiveness of the method, application of machine learning. Fundamental conclusions were drawn for each of the criteria, a brief description of the research was given, as well as ways to continue it.

Онтологічний підхід в системі забезпечення безпеки використання IP-телефонії

In today's information world, the use of VoIP has become an attractive option for user communication. With the downward trend in paying for basic broadband services and the rapid increase in internet speeds, the use of VoIP should only continue to grow in popularity. However, as the use of VoIP increases, so do the potential threats to ordinary users. This article examines the peculiarities of organizing corporate VoIP telephony systems, highlights the main problems in information protection systems in VoIP telephony, and outlines ways to solve them. The development of methods for speech analysis and corresponding processing of natural language, which allows for creating more accurate and effective systems for detecting anomalous traffic and potentially dangerous communications, is especially relevant. With the continuous development of artificial intelligence technologies, the direction of using intelligent means for content analysis in the VoIP system is becoming interesting. A method for detecting anomalies in IP-telephony traffic, based on grouping VoIP messages through context-frequency analysis, is proposed. Additionally, a method for automated filling of the ontology of thematic messages in corporate IP-telephony systems is proposed, based on the formalized presentation of messages using tree-like structures and the description of interaction operations through tuple algebra. Furthermore, a software implementation for converting voice messages into text representations using the SpeechRecognition library for voice-to-text conversion in the Python programming language was created. Experimental studies of the proposed approaches were conducted, and a software subsystem for detecting anomalous messages based on the ontological approach was implemented in the current corporate IP-telephony system.

Implementing a campus VoIP intercom VLAN: A technology review, system requirements and architecture

Voice over the internet can be used with existing Local Area Network infrastructure within an organisation such as DIT to cut down the communication costs within the campus premises. This paper presents how can VoIP intercom be set and the requirements of the system. Voice is very important method of communication and hence is important to make it cheap and easily available. In this work, a VLAN was created in the DIT LAN then an appliance IP-PBX was installed, the phones were connected to a nearest by backbone switch. With gateway of the phones being the IP-PBX and VLAN tagging of the respective connected switch ports, the phones were able to connect to the PBX. The analysis of the system during busy hour shows that the system reliable and suitable to be used if the concurrent calls do not exceed the maximum capacity of the PBX. It is on this basis that this paper tends to critically review this new technology VoIP, x-raying the different types. It further more discusses in detail the VoIP system, VoIP protocols, and a comparison of different VoIP protocols. The compression algorithm used to save network bandwidth in VoIP, advantages of VoIP and problems associated with VoIP implementation were also critically examined. It equally discussed the trend in VoIP security and Quality of Service challenges. It concludes by reiterating the need for a cheap, reliable and affordable means of communication that would not only maximize cost but keep abreast with the global technological change.

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.

Detecting Parallel Covert Data Transmission Channels in Video Conferencing Using Machine Learning

Covert communication channels are a concept in which a policy-breaking method is used in order to covertly transmit data from inside an organization to an external or accessible point. VoIP and Video systems are exposed to such attacks on different layers, such as the underlying real-time transport protocol (RTP) which uses Transmission Control Protocol (TCP) or User Datagram Protocol (UDP) packet streams to punch a hole through Network address translation (NAT). This paper presents different innovative attack methods utilizing covert communication and RTP channels to spread malware or to create a data leak channel between different organizations. The demonstrated attacks are based on a UDP punch hole created using Skype peer-to-peer video conferencing communication. The different attack methods were successfully able to transmit a small text file in an undetectable manner by observing the communication channel, and without causing interruption to the audio/video channels or creating a noticeable disturbance to the quality. While these attacks are hard to detect by the eye, we show that applying classical Machine Learning algorithms to detect these covert channels on statistical features sampled from the communication channel is effective for one type of attack.

MiniatureVQNet: A Light-Weight Deep Neural Network for Non-Intrusive Evaluation of VoIP Speech Quality

In IP audio systems, audio quality is degraded by environmental noise, poor network quality, and encoding–decoding algorithms. Therefore, there is a need for a continuous automatic quality evaluation of the transmitted audio. Speech quality monitoring in VoIP systems enables autonomous system adaptation. Furthermore, there are diverse IP audio transmitters and receivers, from high-performance computers and mobile phones to low-memory and low-computing-capacity embedded systems. This paper proposes MiniatureVQNet, a single-ended speech quality evaluation method for VoIP audio applications based on a lightweight deep neural network (DNN) model. The proposed model can predict the audio quality independent of the source of degradation, whether noise or network, and is light enough to run in embedded systems. Two variations of the proposed MiniatureVQNet model were evaluated: a MiniatureVQNet model trained on a dataset that contains environmental noise only, referred to as MiniatureVQNet–Noise, and a second model trained on both noise and network distortions, referred to as MiniatureVQNet–Noise–Network. The proposed MiniatureVQNet model outperforms the traditional P.563 method in terms of accuracy on all tested network conditions and environmental noise parameters. The mean squared error (MSE) of the models compared to the PESQ score for ITU-T P.563, MiniatureVQNet-Noise, and MiniatureVQNet–Noise–Network was 2.19, 0.34, and 0.21, respectively. The performance of both the MiniatureVQNet–Noise–Network and MiniatureVQNet-Noise model depends on the noise type for an SNR greater than 0 dB and less than 10 dB. In addition, training on a noise–network-distorted speech dataset improves the model prediction accuracy in all VoIP environment distortions compared to training the model on a noise-only dataset.

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.

Perspectives on Pandemic: Pedagogies, Prejudice and a Paradigm Shift

"Cities have the capability of providing something for everybody, only because, and only when, they are created by everybody." (Jacobs, 1961) Urban planning, particularly for historic cities, requires extensive community involvement, given a large number of public and private stakeholders. Unilateral top-down decisions of the early 20th century made it clear that community involvement was essential to planning. Understandably, this has been difficult with the ongoing global pandemic. Lockdowns, confinement, quarantine, limited travel, and outright bans on gatherings have made public participation in urban decisions nearly impossible – yet planning for the future has not ceased. Of course, technology is aiding communication, including multiple channels of video conferencing, instant messaging, VoIP or internet phone systems, relay chat, social media, and even email. But these means of communication are no substitute for in-person face-to- face interaction and have raised new and challenging questions – How can community involvement be accommodated? And what other relationships or forms of communication are impacted? Successful planning also requires forming pedagogical relationships, another problem created by the pandemic. Most often, pedagogy refers to an academic setting or teacher-student relationship. But pedagogy also relates to other relationships necessary for auspicious planning, including the planner-stakeholder and professional- client associations. Frequently, architects, urban planners, and conservators must explain the problems, technical details, results of surveys, and other studies to clients, decision-makers, and stakeholders. Often people are involved in making decisions and are not well versed with urban planning, architecture, or the conservation of historic cities. Another relationship that is has been impacted by the pandemic is peer-to-peer exchange, essential for complex urban planning. This paper will explore various issues and challenges of this new mode of work inter-pandemic. First, there will be an examination of three types of pedagogical relationships: planner-stakeholders, professional-client, and between professionals or peer-to-peer. Second, three examples are included to illustrate the drawbacks and benefits of this new mode of work, specifically related to planning and conservation projects in historic cities. These examples are all drawn from projects executed by various firms during the pandemic and led by the author. Finally, the presentation discusses the disadvantages and advantages of new forms of pedagogy, including removing prejudices. Pre-pandemic, there were many objections to holding virtual meetings or relying extensively on digital communications from stakeholders, clients, and team partners. The pandemic has effectively removed this prejudice against online encounters and subsequently pedagogical relationships, thus accelerating the paradigm shift related to virtual communications Keywords: Pandemic, Urban Planning, Heritage, Conservation

ANALISIS KEAMANAN VOICE OVER INTERNET PROTOCOL (VOIP) OVER VIRTUAL PRIVATE NETWORK (VPN)

This research was conducted on the basis of the influence of security systems that play a role in encrypting data on VoIP communication systems, with the security technology of PPTP VPN computer networks with the system passing data in a virtual private ip or as a tunnel for secure data transmission media. The results obtained from security analysis for the implementation of security methods on PPTP VPNs, then the data to help developers in terms of building a secure VoIP communication system. Basiclly ZRTP uses the Diffie-Hellman key exchange as a key exchange of communication between clients, which is the key for communication between clients using hashes from Diffie-Hellman and is done peer-to-peer through the VOIP RTP package, while the Point-to-Point Tunneling Protocol (PPTP) is a network protocol that allows the secure transmission of data from the remote client to the server by creating a virtual private network (VPN) through a network of data. TCP/IP or UDP is dedicated to encryption and creates RTP tunnel transport on VoIP communication systems. VoIP communication system research is conducted using 2 security methods, namely VoIP VPN PPTP, VoIP ZRTP, with the aim to find out the results of VoIP communication testing using PPTP and ZRTP VPN security methods if an attack occurs during VoIP communication

VoIP Speech Encryption System Using Stream Cipher with Chaotic Key Generator

Recently, with the development multimedia technologies and wireless telecommunication, Voice over Internet Protocol, becomes widely used in communication between connecting people, VoIP allows people that are connected to the local network or the Internet to make voice calls using digital connection instead of based on the analog traditional telephone network. The technologies of Internet doesn’t give any security mechanism and there is no way to guarntee that the voice streams will be transmitted over Internet or network have not been intercepted in between. In this paper, VoIP is developed using stream cipher algorithm and the chaotic cryptography for key generator. It is based on the chaotic maps for generating a one-time random key used to encrypt each voice data in the RTP packet. Chaotic maps have been used successfully for encryption bulky data such as voice, image, and video, chaotic cryptography has good properties such as long periodicity, pseudo-randomness, and sensitivity to initial conditions and change in system parameters. A VoIP system was successfully implemented based on the on ITU-T G.729 for voice codec, as a multimedia encoding of Real-time Transport Protocol payload data, then, apply a proposed method to generate three-mixed logistic chaotic maps [1] and then analysis the encryption/ decryption quality measures for speech signal based this method. The experimental work demonstrates that the proposed scheme can provide confidentiality to voice data with voice over IP performance quality, minimum lost in transmitted packet, minimum average delay, and minimum jitter.

Data in Transit

Abstract Peter Cox, CEO and Founder of UM Labs, explores VoIP and messaging systems and explains why organisations need to ensure high levels of security around data-in-transit, warning that failure to do so may lead to GDPR compliance issues.

Design and Simulation of VoIP System for Campus usage A Case Study at PTU

Design and Simulation of VoIP System for Campus usage A Case Study at PTU

Survey on Detection Techniques for Denial-of-Service Attacks in SIP-Based Voice Over IP Networks

VoIP provides real-time voice communications over packet-switched networks. VoIP is widely used as it offers more flexibility when compared to the traditional Public Switched Telephone Network (PSTN). Session Initiation Protocol (SIP) is a protocol used for initiating a session between the caller and the callee in the VoIP system. SIP is a text-oriented protocol and is implemented in a susceptible environment. These features make SIP protocol vulnerable to various security risks such as Denial of Service, Registration hijacking and Eavesdropping. Detection of these attacks plays a vital role. The Survey focuses on different techniques used for detection of DoS attacks and differentiating between the proposed mechanisms for identifying abnormalities and then notifying the end-users about the possible intrusions.

Secure and efficient ECC based SIP authentication scheme for VoIP communications in internet of things

Since last decade, VoIP is transitioning from being a voice scheme to the most powerful unified communications engine. All VoIP systems uses the Session Initiation Scheme that defines the procedures and messages used to set up a phone call – or any other kind of communication. IoT is another paradigm-shifting idea that is going to change VoIP communications, since it offers a seamless way to connect all of the devices, applications and platforms. Embedding VoIP in IoT solutions provides a competitive advantage over the traditional telephony system of being interconnected to Internet of Things. With the IoT, value proposition of VoIP broadens so with IoT, however, VoIP is vulnerable to all of the intrinsic security problems in IP. In this paper, a new biometrics based authentication scheme using ECC has been proposed. The formal and informal security analysis of the scheme proves the security strength of the scheme. Simulation of the scheme using AVISPA also proves the scheme is secure against potential threats. Comparison of the proposed scheme in terms of computation cost and security features with other related schemes proves the superiority of the scheme.

An optimal rejuvenation scheme for improving VoIP service reliability in the existence of resource exhaustion and security breaches

The escalating number of end-users of VoIP and the challenges faced in furnishing first-class quality of service (QoS) are the major driving forces for research in the area of VoIP. When the resources allotted to every end-user get exhausted, it leads to a decline in the QoS. Further, as VoIP is an application operating on the internet, it is prone to security breaches. This paper addresses the issues of resource exhaustion and security breaches in a VoIP network. To help avert performance degradation in a VoIP system, software rejuvenation is also performed. The performance of a VoIP system in the existence of resource exhaustion and security breaches is modelled by means of a continuous time Markov chain (CTMC) to study the system reliability. From the CTMC the mean time to failure (MTTF) is obtained. Furthermore, an optimal rejuvenation scheme is proposed for achieving the best attainable value of MTTF.

Pengembangan Sistem VOIP Menggunakan Server Issabel Versi 4.0 dan Tunnel EOIP pada OMNI Hospital Alam Sutera

The VoIP communication system at OMNI Hospital Alam Sutera uses the Elastix 2.5 server with the Centos 5.11 operating system. Elastix 2.5 by the developer has been declared End of Life. The server security system is a serious concern considering that VoIP servers can be accessed from the internet. Iptables and fail2ban applications are applications that are used to limit and counteract those who try to attack the VoIP server. One application that can be used as an open source VoIP server is the Issabel Application version 4.0. The migration process from Elastix 2.5 application to Issabel 4.0 by backing up all configurations in the Elastix 2.5 application through a web browser including the configuration of endpoints, fax, e-mail, asterisk. After the backup file is downloaded then upload the backup file to the Issabel 4.0 application then run the migration process. Adding a backup path as a failover connection is needed because the VoIP communication protocol between the OMNI Hospitals Group still uses one path so that when there is a problem in the connection path, the communication protocol will stop. The tunnel EoIP is a protocol used as a backup path between the OMNI Hospitals Group site.

Teletraffic Analysis for VoIP Services in WLAN Systems with Handoff Capabilities

In this work, a TDMA-based system is studied for VoIP services with and without mobility. The system is considered to be dedicated for voice-only services such as a WLAN environment under the PCF mode or a cellular system. In such environments, mobiles usually have mobility. This introduces the need to study the system when a handoff procedure is enabled. The teletraffic analysis considers call arrivals, call completions, and ON/OFF activity processes for individual VoIP sessions. The VoIP system is simulated in order to verify the accuracy of the analytical results. Since many papers published in the literature consider a fixed number of VoIP users in the system, this analysis can be useful to estimate the system's performance when the system is in statistical equilibrium. As an additional feature of this work, a simple fluid model to calculate the number of active and inactive users in a VoIP system with and without handoff capabilities is proposed and developed.

Survey of Research on IP-DECT and VOIP Systems Safety and a Novel Counter-Measure Approach

Survey of Research on IP-DECT and VOIP Systems Safety and a Novel Counter-Measure Approach

Encrypted Fingerprint into VoIP Systems using Cryptographic Key Generated by Minutiae Points

The transmission of the encryption voice over IP is challenging. The voice is recorded, eavesdropping, change and theft voice, etc. The voice over IP is encrypted by using Advance Encryption Standard (AES) Algorithm. AES key is generated from Minutiae Points in fingerprint. By other way, we talk about biometric-cryptosystem, which is hybrid between one of the cryptosystems and biometric systems, such as fingerprint using for authentication as well as to generate cryptographic key to encrypt voice over IP by applying AES. In this paper, we define a new term which is Fingerprint Distribution Problem (FDP) based on Key Distribution Problem. Also, we suggest a solution for this problem by encrypted fingerprint before sending between users by using one of the public key cryptosystems which is RSA Algorithm.

An Optimal Rejuvenation Strategy for Increasing Service Reliability of a VOIP System with Multiple Components

In this day and age Voice over Internet Protocol (VoIP) has turned out to be a revolutionary technology in the field of telecommunication. Because of the escalating number of users of VoIP, it is essential to intend for a reliable VoIP system providing good quality of service. In this paper a stochastic model based on continuous time Markov chain is developed to analyze the reliability of a server based VoIP system comprising of significant components only. The concurrent breakdown of more than one component is taken into modeling concern. Redundancy at the component level is used with the purpose to boost the system reliability of VoIP. As a defensive course of action, software rejuvenation is implemented to avert or suspend software failures. And an optimal software rejuvenation strategy is proposed which leads to increased system reliability. Numerical results are presented for the quantitative examination of the suggested model.