VoIP Research Articles

Case Study of Voice Disconnection During Calls on the Hikari Denwa IP Telephone Service due to DHCP Conflicts

Case Study of Voice Disconnection During Calls on the Hikari Denwa IP Telephone Service due to DHCP Conflicts

Design of an asterisk-based VoIP system and the implementation of security solution across the VoIP network

Voice over Internet Protocol (VoIP) is a rapidly advancing technology that facilitates the transmission of voice and audio signals over the Internet or an IP-based network in real-time. This technology has seen a significant rise in demand due to its advantages over traditional circuit-switched telephony, including lower call rates, reduced operational costs, easier management, and enhanced call features. However, the growth in VoIP usage has also increased the potential for various security threats and attacks, jeopardizing the privacy, confidentiality, and integrity of transmitted data. This paper presents the design of an Asterisk-based VoIP system and the implementation of a comprehensive security solution across the VoIP network. The study involves an in-depth analysis of VoIP technology, identifying its vulnerabilities and addressing potential threats. A security framework is proposed and implemented to safeguard the VoIP network. The designed system and security solutions are rigorously tested and evaluated to ensure robustness and effectiveness. The findings highlight critical security measures necessary for protecting VoIP infrastructures and provide a framework for future research and development in securing VoIP networks.

QoS-Aware Power-Optimized Path Selection for Data Center Networks (Q-PoPS)

Data centers consume significant amounts of energy, contributing indirectly to environmental pollution through greenhouse gas emissions during electricity generation. According to the Natural Resources Defense Council, information and communication technologies and networks account for roughly 10% of global energy consumption. Reducing power consumption in Data Center Networks (DCNs) is crucial, especially given that many data center components operate at full capacity even under low traffic conditions, resulting in high costs for both service providers and consumers. Current solutions often prioritize power optimization without considering Quality of Service (QoS). Services such as video streaming and Voice over IP (VoIP) are particularly sensitive to loss or delay and require QoS to be maintained below certain thresholds. This paper introduces a novel framework called QoS-Aware Power-Optimized Path Selection (Q-PoPS) for software-defined DCNs. The objective of Q-PoPS is to minimize DCN power consumption while ensuring that an acceptable QoS is provided, meeting the requirements of DCN services. This paper describes the implementation of a prototype for the Q-PoPS framework that leverages the POX Software-Defined Networking (SDN) controller. The performance of the prototype is evaluated using the Mininet emulator. Our findings demonstrate the performance of the proposed Q-PoPS algorithm in three scenarios. Best-case: Enhancing real-time traffic protocol quality without increasing power consumption. midrange-case: Replacing bottleneck links while preserving real-time traffic quality. Worst-case: Identifying new paths that may increase power consumption but maintain real-time traffic quality. This paper underscores the need for a holistic approach to DCN management, optimizing both power consumption and QoS for critical real-time applications. We present the Q-PoPS framework as evidence that such an approach is achievable.

Active Queue Management in L4S with Asynchronous Advantage Actor-Critic: A FreeBSD Networking Stack Perspective

Bufferbloat is one of the leading causes of high data transmission latency and jitter on the Internet, which severely impacts the performance of low-latency interactive applications such as online streaming, cloud-based gaming/applications, Internet of Things (IoT) applications, voice over IP (VoIP), real-time video conferencing, and so forth. There is currently a pressing need for developing Transmission Control Protocol (TCP) congestion control algorithms and bottleneck queue management schemes that can collaboratively control/reduce end-to-end latency, thus ensuring optimal quality of service (QoS) and quality of experience (QoE) for users. This paper introduces a novel solution by experimentally integrate the low latency, low loss, and scalable throughput (L4S) architecture (specified by the IETF in RFC 9330) in FreeBSD framework with the asynchronous advantage actor-critic (A3C) reinforcement learning algorithm. The first phase involves incorporating a modified dual-queue coupled active queue management (AQM) system for L4S into the FreeBSD networking stack, enhancing queue management and mitigating latency and packet loss. The second phase employs A3C to adjust and fine-tune the system performance dynamically. Finally, we evaluate the proposed solution’s effectiveness through comprehensive experiments, comparing it with traditional AQM-based systems. This paper contributes to the advancement of machine learning (ML) for transport protocol research in the field. The experimental implementation and results presented in this paper are made available through our GitHub repositories.

Enabling Ultra Reliable Low Latency Communications in rural areas using UAV swarms

Latency is a critical aspect for a broad spectrum of applications that relies on the internet, such as, voice over IP (VoIP) or teleconferencing, and the lack of ultra-fast and highly reliable communications is prominent in rural areas even in mature economies. Our proposal focuses on optimizing the deployment of microservice-oriented architectures (MSA) in computing and routing enabled unmanned aerial vehicles (UAVs). For that matter, an information system which gathers all the information of the flying ad hoc network (FANET) is developed. From there, we propose multiple approaches, based on integer linear programming (ILP) and heuristics, to tackle the minimization of end-to-end latency by deploying multiple instances of microservices in the UAVs that are close to the users that make use of them. Extensive experiments based on network emulation prove the performance of our ILP formulation of the problem and address the optimality gap between the ILP-based approach and the heuristics ones, which are highly scalable and usable in real-time for large-scale scenarios.

Studying the Impact of Different TCP DoS Attacks on the Parameters of VoIP Streams

In today’s digital world, no one and nothing is safe from potential cyberattacks. There is also no 100% protection from such attacks. Therefore, it is advisable to carry out various studies related to the effects of the different cyberattacks on the performance of the specific devices under attack. In this work, a study was carried out to determine how individual TCP DoS attacks affect the parameters of VoIP (Voice over IP) voice and video streams. For the purpose of this work, a model of a simple IP network has been created using the GNS3 IP network-modeling platform. The VoIP platform used was Asterisk Free PBX. Tools from Kali Linux were used to implement the individual TCP DoS attacks; IP-network-monitoring tools and round-trip-delay-measurement tools were also used. The proposed study is applicable to multiple VoIP platforms wherein voice and video traffic are passed/processed by the VoIP server. From the obtained results, it was found that Asterisk Free PBX is very well secured against TCP DoS attacks, which do not affect the platform performance or the parameters of the voice and video streams. The values of the observed parameters, such as jitter, packet loss, round-trip delay, etc., are very far from the maximum allowable values. We also observed a low load on the CPU and RAM of the system during the whole study.

VoIP Security Threats

Nowadays, communication is based on a wide variety of solutions, especially using the real-time ones. People use a lot of online tools that offer them audio-video solutions, and the most important thing is that anyone has access to them, and there are easy to use. VoIP (Voice over IP) is one of the most used solutions today, with which everyone can make calls over the internet. Like any other service that is exposed to the internet, it presents risks for the data violation and administrators or any people that use this service, needs to secure the system.

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.

Supporting VoIP communication in IEEE 802.11ax networks: A new admission control and scheduling resource allocation scheme

The current IEEE 802.11ax standard enhances Wi-Fi networks with a series of new features, such as multi-user (MU) transmission, an Orthogonal Frequency Division Multiple Access (OFDMA) scheme and the ability to multiplex traffic from different access categories (ACs). These features can be utilized to enhance the QoS support for VoIP traffic and optimize the usage of IEEE 802.11ax network resources. This work proposes a new scheme to multiplex VoIP calls in IEEE 802.11ax MU frames and a new scheduler and resource allocation algorithm specifically designed for VoIP data traffic. Our scheduler allocates VoIP packets requiring longer transmission times into the same frame(s), minimizing the channel air-time assigned to VoIP transmission. In addition, unused radio resources in the MU frame are leveraged to transmit best-effort packets along with VoIP packets. For completeness, we also define a call admission control (CAC) algorithm that anticipates channel saturation conditions to ensure VoIP users can maintain a guaranteed level of QoS. Based on simulation results, our scheme is more efficient in reducing channel utilization than other schedulers such as multi-user round-robin (RR) (implemented by ns-3) or single-user FIFO. For example, for 30 VoIP stations using the G.711 codec under mixed channel conditions, our scheme reduces by 30% the air-time required to transmit VoIP packets. When coupled with the ability to also send best-effort packets along with VoIP packets, this translates into a higher throughput (i.e. 10 Mbit/s vs 4 Mbit/s) and more simultaneous VoIP users with guaranteed QoS (up to 46 VoIP users vs 26 and 28 users for the multi-user RR and single-user FIFO scheduling algorithms, respectively).

Hybrid Non-Intrusive QoE Assessment of VoIP Calls Based on an Ensemble Learning Model

Hybrid Non-Intrusive QoE Assessment of VoIP Calls Based on an Ensemble Learning Model

A mean opinion score prediction model for VoIP calls offloading handover from LTE to WiFi

A mean opinion score prediction model for VoIP calls offloading handover from LTE to WiFi

Sketching of interactive VoIP traffic with multivariate statistical learning-based classification

Classifying VoIP (Voice over Internet Protocol) traffic is vital for optimizing network performance and Quality of Service (QoS). This study introduces the Multivariate Statistical-Based Classification (MVSC) system, designed to classify network traffic with high accuracy and efficiency. As traditional methods struggle in the diverse and complex landscape of today’s network traffic, which includes voice, video, gaming, and data, the MVSC algorithm rises to the challenge. It employs Statistical Dissemination and leverages various statistical features such as Packet Size, Inter-Arrival Statistics, Packet and Data rates, Flow Length, and Five-tuple information to create nuanced profiles of network traffic packets. These packets are then grouped into distinct clusters based on their statistical attributes through Application Flow Cluster Grouping. A unique aspect of the MVSC system is its approach to representing each application flow as points in a two-dimensional space, where distances to predefined application profiles are calculated. The nearest profile then determines the type of VoIP traffic. Experimental results using university traffic data (KU-IDS) underscore the system’s high accuracy, consistently around 98-99%. These findings affirm the system’s suitability for real-time deployment. In summary, the MVSC system offers a robust and efficient solution for VoIP traffic classification, significantly boosting network performance and QoS, and proving to be an invaluable asset in contemporary network management.

Unidirectional switch model for multiservice traffic with polymodal distribution

The modern stage of development of telecommunication networks is characterized by active improvement of routing algorithms in packet-switched networks taking into account the provision of the required quality of service. The need to transfer large amounts of information is due to the need to ensure the normal functioning of modern multimedia applications, in particular such as IP telephony, video and radio broadcasting, interactive distance learning, which determines the multiservice nature of traffic, which is heterogeneous, and its statistical properties in most cases are of a complex polymodal nature. Taking into account the fact that traditional Poisson models providing FIFO servicing of network traffic for solving modern telecommunication problems lead to overly optimistic results in terms of delays, there is a need to develop a new mathematical model of a unidirectional switch providing FIFO servicing of multiservice traffic with polymodal distribution taking into account the provision of the required quality of subscriber service and an algorithm for estimating its main parameters. Development of a mathematical model of a unidirectional switch FIFO-service of multiservice traffic with polymodal distribution taking into account the provision of the required quality of subscriber service and an algorithm for estimating its main parameters. A new mathematical model of a unidirectional switch implementing FIFO-service of multiservice traffic with polymodal distribution has been developed, allowing one to estimate the probabilities that packets arriving at the switch input will not be processed due to overflow of its buffer memory and exceeding the maximum packet processing time. The model takes into account the delays introduced by the channel and channel equipment during the distribution and processing of packets from the source to the switch in question. Recommendations for modeling traffic processing to ensure the specified accuracy of the results are presented.

Comparison of Usage between Voice Switching Communication System (Vscs) Direct Speech (Ds) Vsat (Very Small Aperture Terminal) System and Voip (Voice Over Internet Protocol)

One of the ground-to-ground DS coordination communications that need to be adapted to use on an IP basis, hereinafter referred to as Voice over Internet Protocol (VoIP). However, to be fully used and also as a substitute for VSAT, VoIP needs to be analyzed and also compared reliability and Cost Benefit Analysis (CBA) with VSAT. This is needed so that it is known that the adjustment to the improvement and development of technology and the future plan is ready to be implemented and also has a positive impact on aviation safety, operations and finances. The purpose of this study was to determine the comparison between the use of DS VSAT system with VoIP. The method used in this study is a mixed-method, with the category of descriptive analysis and the final results/conclusions of the study using the calculation of the percentage of data collecting (reliability test) and the ratio of cost benefit analysis (literature/archive studies). The results of the reliability test for 1 (one) year in 2021 obtained results for VSAT 99.53%-100% and for VoIP 99.5%-100%, from these results concluded in 1 (one) year the reliability between DS VSAT system and VoIP is the same ( Note: in December 2021 on the DS VSAT system there was a failure so that the percentage generated for 1 (one) month was only 99.95%, while the DS VoIP system in June to December never failed (failures that occurred in February - May could be due to the transition period for new equipment) While the results of Cost Benefit Analysis (CBA) between the two are 55.85%, where the use of DS VoIP systems is more financially efficient (saving) than the use of DS VSAT systems.

Implementation of a Contact Center with Interactive Voice Response Using Cisco Unified Communications

A popular voice-based communication service used to receive calls automatically is Interactive Voice Response (IVR). Generally, IVR is widely used by institutions to provide services to the public in consulting, offering, or providing information about a particular matter. One of the uses of IVR, among others, is the Contact Center. By using Voice over Internet Protocol (VoIP) technology and using the SIP Trunk service from the service provider, institutions can take advantage of the existing network infrastructure to save on telephone usage. VoIP technology also allows it to be integrated with Private Automatic Branch Exchange (PABX) using Cisco Unified Communications Manager (CUCM). While the IVR system uses Cisco Unified Contact Center Express (UCCX). In this study, the implemented contact center serves to receive calls from external parties. The telephone number uses a call hunting system that will distribute incoming calls to officers on duty at PT. Multipolar Technology Tbk.

Study of performance indicators and prospects for the development of IP telephony systems

The purpose of the work is to study performance indicators and prospects for the development of IP telephony systems. IP-telephony is a special field of telephone communication that integrates methods and means of digital signal processing, transmission of speech, data, video information and multimedia, management of computing resources based on high technologies. IP telephony systems are in demand and widely used, especially in the corporate sector. In this work, a meaningful study of the principles of functioning of modern IP telephony systems is carried out. It was shown that it is very important to continuously monitor the effectiveness of the functioning of these systems. A systematization of indicators that affect the integrated assessment of the effectiveness of IP telephony systems was carried out. At the same time, the quality of the user experience comes to the fore, which is currently of primary importance for a full assessment of the effectiveness of any modern telecommunications system, the main goal of which is to ensure user satisfaction. In addition, the article also analyzes the prospects for the development of IP telephony systems. In particular, market growth forecasts are demonstrated and directions for further improvements of IP telephony systems are outlined.

MVF: A Novel Technique to Reduce the Voip Packet Payload Length

Abstract The adoption of the Voice over Internet Protocol (VoIP) system is growing due to several factors, including its meagre rate and the numerous contours that can be joined with VoIP systems. However, the wasteful utilisation of the computer network is an inevitable problem that limits the rapid growth of VoIP systems. The essential explanation behind this wasteful utilisation of the computer network bandwidth (BW) is the considerable preamble length of the VoIP packet. In this study, we invent a technique that addresses the considerable preamble length of the VoIP packet. The designed technique is known as the manikin voice frame (MVF). The primary idea of the MVF technique is to utilise the VoIP packet preamble tuples that are not essential to the voice calls, particularly client-to-client calls (voice calls between only two users). Specifically, these tuples will be utilised for reserving the data of the VoIP packet. In certain instances, this will make the VoIP packet data manikin or even make it empty. The performance assessment of the introduced MVF technique demonstrated that the utilisation of the computer network BW has enhanced by 33%. Along these lines, the MVF technique indicates potential progress in resolving the inefficient usage of the computer network BW.

An innovative approach for enhancing capacity utilization in point-to-point voice over internet protocol calls

Voice over internet protocol (VoIP) calls are increasingly transported over computer-based networking due to several factors, such as low call rates. However, point-to-point (P-P) calls, as a division of VoIP, are encountering a capacity utilization issue. The main reason for that is the giant packet header, especially when compared to the runt P-P calls packet payload. Therefore, this research article introduced a method to solve the liability of the giant packet header of the P-P calls. The introduced method is named voice segment compaction (VSC). The VSC method employs the unneeded P-P calls packet header elements to carry the voice packet payload. This, in turn, reduces the size of the voice payload and improves network capacity utilization. The preliminary results demonstrated the importance of the introduced VSC method, while network capacity improved by up to 38.33%.

Overcoming challenges in deep inspect of vpn and proxy by deep learning

The rapid growth of proxying and VPN techniques has presented formidable challenges for internet operators in classifying network traffic. These methods not only hinder effective traffic policing and resource restriction but also introduce complexities by allowing VPNs to masquerade under application protocols. Furthermore, the increasing popularity of applications like Voice over IP (VoIP) and peer-to-peer (P2P) technologies further exacerbates the difficulty in controlling and classifying such traffic. Conventional techniques like Server Name Indication (SNI) analysis are witnessing diminishing effectiveness over time. Network providers now face the critical task of acquiring detailed knowledge about the specific applications and protocols utilised by their customers, enabling them to accurately allocate resources and ensure robust traffic management in this challenging landscape.

Voice over internet protocol codec performance in interactive streaming environment

Voice over Internet Protocol (VoIP) is an interactive telecommunication technology that suggests shared, cooperative, and quality transactions in the form of communications over the internet. VoIP normally differs from traditional and conventional circuit-based networks in the form of a low rate because all the communication takes place through the internet. The user has to pay only for an internet connection. Real-time voice data transmission is very difficult when compared with ordinary text data transmission. A main challenge in telecom engineering is maintaining quality. The VoIP transmission faces a lot of inconvenience. It suffers from packet loss, low quality, jitter, delay, and on-time delivery. These parameters influence and degrade performance. The codec plays a major role in VoIP transmission. This paper reports on the fundamental functionalities of VoIP, including coder and decoder operations for quality output, broadcast technology, VoIP connection setups, advantages, applications, and emerging trends in VoIP. The performance of various codecs was analyzed to identify the most suitable codec for voice transmission.