Voice Over IP Systems Research Articles

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.

Analiza paketske komutacije u VoIP telefoniji na komandnom mestu jedinica taktičkog nivoa

Introduction/purpose: This paper conducts a comprehensive analysis of a potential implementation of Voice over Internet Protocol (VoIP) systems, focusing on network architecture, VoIP phones, and servers. The study explores potential vulnerabilities and proposes solutions. The paper concludes by advocating for a holistic approach to securing VoIP systems, incorporating supplementary services to ensure the confidentiality, integrity, and availability of voice communications in the digital landscape. Methods: Review of the underlying theory, analysis of the end-user needs and potential solutions, practical viability assesment. Results: The theoretical points discussed were proven in practice, using commercially available resources. Communication was established in an expected manner. Conclusions: Implementing solutions similar to the one presented in the paper would be a relatively inexpensive way to make diverse improvements to the operation of tactical level units, both in peacetime and during war.

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

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

Automated Speech Audiometry for Integrated Voice Over Internet Protocol Communication Services.

Problems in speech recognition are often apparent in telecommunication situations. For ecologically valid assessments of such conditions, it is important to quantify the impact of real environments including acoustic conditions at a far-end communication device and all paths of transmission degradation. This study presents an automated matrix sentence test procedure based on automatic speech recognition (ASR) integrated in a Voice over Internet Protocol (VoIP) infrastructure and compares the individual effects of transmission degradations with results from laboratory measurements. Speech recognition thresholds (SRTs) were measured in 16 normal-hearing subjects in four test conditions: (a) a laboratory condition guided by a human experimenter, (b) a laboratory condition with reduced bandwidth and (c) additionally reduced headset quality to simulate typical communication systems, and (d) an automated, ASR-controlled adaptive test procedure over a real VoIP infrastructure. Errors of the ASR system were analyzed to show possible effects on measurement outcome Results: Measured SRTs showed a highly significant correlation (r = .93) between the fully automatic and "laboratory" conditions, with a constant bias of about 1 dB indicating a linear shift of the data without affecting the distribution around the mean. The individual impact of the different system degradations on SRTs could be quantified Conclusions: This study provides a proof of concept for automated ASR-based SRT measurements over VoIP systems for speech audiometric testing in real communication systems, as it produced results comparable to traditional laboratory settings for this group of 16 normal-hearing subjects. This makes VoIP services a promising candidate for speech audiometric testing in real communication systems.

Novel Method of Reducing the Allocated Bandwidth by Voip

Abstract Old telecommunication systems are gradually being replaced by a new system that works over IP networks, which is known as voice over internet protocol (VoIP). VoIP has several merits (e.g., very cheap call rate), which make it increasingly popular in the telecommunication world. However, VoIP faces numerous impediments that decelerate its promotion. One of the top impediments is the wasted bandwidth caused by VoIP systems. Numerous methods have been proposed to handle this impediment, including packet aggregation methods. This paper proposes a novel aggregation method, called packet aggregation and carrier header (PA-CH), to reduce the amount of the large bandwidth caused by VoIP. As the name suggests, PA-CH saves in bandwidth by aggregating the packets in a header and using the redundant fields in the packet header to carry a portion of the packet voice data. The performance of the introduced PA-CH method was investigated based on three main metrics, namely, link capacity, allocated bandwidth reduction, and voice data shortening. Simulation results indicate that the proposed PA-CH method outperforms the comparison methods in three factors. For instance, the proposed method’s allocated bandwidth reduction ratio reaches 51% when the number of calls running concurrently reaches 100. Therefore, the proposed PA-CH method achieves its goal of reducing the wasted bandwidth caused by VoIP.

A Comprehensive Review of Intrusion Detection and Prevention Systems against Single Flood Attacks in SIP-Based Systems

Voice over Internet Protocol (VoIP) is a recent voice communication technology and due to its variety of calling capabilities, the system is expected to fuel the market value even further in the next five years. However, there are serious concerns since VoIP systems are frequently been attacked. According to recent security alliance reports, malicious activities have increased largely during the current pandemic against VoIP and other vulnerable networks. This hence implies that existing models are not sufficiently reliable since most of them do not have a hundred percent detection rate. In this paper, a review of our most recent Intrusion Detection & Prevention Systems (IDPS) developed is proposed together with a comparative analysis. The final work consisted of ten models which addressed flood intentional attacks to mitigate VoIP attacks. The methodological approaches of the studies included the quantitative and scientific paradigms, for which several instruments (comparative analysis and experiments) were used. Six prevention models were developed using three sorting methods combined with either a modified galloping algorithm or an extended quadratic algorithm. The seventh IDPS was designed by improving an existing genetic algorithm (e-GAP) and the eighth model is a novel deep learning method known as the Closest Adjacent Neighbour (CAN). Finally, for a better comparative analysis of AI-based algorithms, a Deep Analysis of the Intruder Tracing (DAIT) model using a bottom-up approach was developed to address the issues of processing time, effectiveness, and efficiency which were challenges when addressing very large datasets of incoming messages. This novel method prevented intruders to access a system without authorization and avoided any anomaly filtering at the firewall with a minimum processing time. Results revealed that the DAIT and the e-GAP models are very efficient and gave better results when benchmarking with models. These two models obtained an F-score of 98.83%, a detection rate of 100%, a false rate of 0%, an accuracy of 98.7%, and finally a processing time per message of 0.092 ms and 0.094 ms respectively. When comparing with previous models in the literature from which it is specified that detection rates obtained are 95.5% and false-positive alarm of around 1.8%, except for one recent machine learning-based model having a detection rate of 100% and a processing time of 0.53 ms, the DAIT and the e-GAP models give better results.

VoIP Security: A Performance and Cost-benefit Analysis

Voice over IP (VoIP) has become the standard technology for telephony and has replaced the old Public Switched Telephone Network (PSTN). This research focuses on the security aspect of VoIP systems. Unsecured VoIP systems are vulnerable to malicious attacks. However, the overhead of the security techniques hampers the performance of VoIP systems. This research analyzes how a VoIP system performs with different security techniques. The performance of the VoIP system is analyzed on different types of data networks such as IPv4, IPv6, and IPv4/IPv6 mixed networks, and in scenarios such as with and without network traffic. Additionally, the research includes a cost-benefit analysis of the security techniques, to determine their cost effectiveness. Based on the performance analysis and cost-benefit analysis, this research proposes three security techniques that can be applied to VoIP systems deployed on IPv4, IPv6, and IPv4/IPv6 networks.

A COMPARATIVE ANALYSIS OF PROTOCOLS FOR VOIP SERVICES

Although the widespread of broadband network has boosted the deployment of voice over IP (VoIP) systems, most part of the telephone service is still provided on the traditional PSTN lines. It can be expected that the transition from traditional telephone to VoIP systems proceed, and that the network traffic due to the phone calls will become significant. Hence, it is still important to select the best protocols, both for traffic forwarding and for encoding to ensure the required quality. In this paper we present the results of a series of experiments to evaluate the protocols involved in VoIP service. On a laboratory testbed we reproduce a common situation of two interconnected branches and compare the performance achievable by selecting different network protocols like SIP and IAX combined with current voice codecs like G.711 and G.729.

Evaluating QoE in VoIP networks with QoS mapping and machine learning algorithms

The quality of experience (QoE) of the end-users is a critical criterion of measurement in VoIP (Voice over Internet Protocol) systems for technical and commercial purposes. We investigate how quality of service (QoS) influences QoE and assesses the QoE in VoIP communication. Our contributions are three-fold. First, the impacts of QoS on QoE are comprehensively analyzed by experimental means and an association test method, instead of independently studying each parameter. Second, an algorithm is proposed to integrate the effects of QoS parameters with spatial or temporal characteristics on QoE. Third, we apply machine learning regression algorithms with QoS impairments, noise and echo impairments to nonintrusive voice quality prediction in different network environments. The results from numerous experiments show that fairly accurate prediction can be obtained from these models. Our work will achieve a more accurate evaluation of the QoE in VoIP by using QoS parameters, clarify the influence of IP network environments, noise and echo impairments on the quality and reliability of VoIP traffic, and provide QoS parameter requirements for the VoIP application that runs at the desired QoE level.

VoIP QoS Analysis over Asterisk and Axon Servers in LAN Environment

Voice over IP (VoIP) is a developing technology and a key factor in both the emerging cyberspace engineering and also an accomplishment to set up its position in the telecom industry. VoIP technology is based on internet technology; where data packets switching system is used rather than circuit switching. Whereas, an analog signal is changed over into the digital signals in the full-duplex transmission. VoIP technology is replaced with the conventional public switched telephone network (PSTN) system due to the high flexibility and low-cost. The purpose of this research work is to deliberate experimental and computational performance in the view of quality of service (QoS) parameters of VoIP over local area network (LAN) network. The VoIP systems implementation is based on two different operating system framework (Linux and Windows), whereas, Linux-based and Windows-based private branch exchanges (PBXs), such as Asterisk (Linux-open source) and Axon (window-close source) are configured, installed and verified. QoS factors (such as packet loss, delay, jitter, etc.) are observed over the Asterisk and Axon PBXs in a LAN domain with the assistance of Paessler switch activity grapher (PRTG) monitoring tool. The validations of results are looked at for QoS parameters crosswise over both PBXs with data load (i.e., file transfer and HTTP traffic) during VoIP calls. The productivity and execution of Axon and Asterisk have been equated and analyzed over experimental based outcomes.

VoIP Accessibility: A Usability Study of Voice over Internet Protocol (VoIP) Systems and a Survey of VoIP Users with Vision Loss

Introduction Accessibility of Voice over Internet Protocol (VoIP) systems was tested with a hands-on usability study and an online survey of VoIP users who are visually impaired. The survey examined the importance of common VoIP features, and both methods assessed difficulty in using those features. Methods The usability test included four paid participants who are blind and four who have low vision. Four different tasks using four different VoIP systems (two Windows-based, two iOS-based) were presented in random order. The online survey included participants with prior VoIP experience: 50 individuals who were blind and 22 who have low vision. Results Usability test participants found that receiving an incoming call was the easiest task and transferring a call was the most difficult. Those with previous iOS experience had a large advantage with the two iOS systems over those with little experience. For the online survey, most respondents (81%) had used VoIP at home for personal use and 49% had used it in their workplace. The caller ID feature was most important to participants, yet only slightly more than half found the feature easy to use; this feature was the most discrepant between importance and ease of use (p < .01). Call management was the most difficult feature for respondents. Generally, those with low vision found features to be more accessible than did those who are blind. Discussion Almost, all usability study participants reported that they could use all four systems in a real-world setting. Although some of the features were readily usable, many of the VoIP features that respondents to the online survey considered to be important were difficult for them to use. This finding indicates a serious gap between what is presently offered and what is needed for true accessibility by those with vision loss, indicating the necessity of adding accessible features to all VoIP systems. Implications for practitioners Rehabilitation staff members and teachers of young adults with vision loss should become familiar with current VoIP systems and introduce them to their clients and students in order to increase their future prospects for employment.

Study SIP Protocol on Asterisk Phone System and Offer Solutions to Its Security

Undoubted every organization’s heart is it’s phone system. The old phone systems couldn’t perform any method to make phone center and voice transmission intelligent on network and they had determinate abilities. Meantime Voice over Internet Protocol (VoIP) introduced itself to the world and performed a lot of abilities for clients like voice transmission on network. Many companies Investment on voip systems and implemented their methods on software and hardware packages. But between them a different production had designed and Performanced by Mark Spenser from Digium Company in 1992 which named Asterisk. Asterisk’s increasing popularity’s reason was its open code programs and its flexibility. VoIP systems such as Asterisk use voice transmission protocols to transfer voice over network. One of the voice transmission protocols is Session Initiation Protocol (SIP), which is one of the Asterisk’s voice transmission protocols. The first and the most important point in voice transmission over network is security. Security can be divided to two parts as inscrutability of invaders to network and coding transmitted voices over network to prevention of illegal listening. In this project at first we tried to introduce Asterisk phone system’s structure and Session Initiation Protocol (SIP) and then scrutiny method of Invader’s dominance to this protocol and Performance modern methods to prevent hacker’s dominance and also coding voice packages to Obscure them in transmission way.

VoIP Technology: Investigation of QoS and Security Issues

Voice over IP (VoIP) is the technology allowing voice traffic transmission as data packets over a private or a public IP network. VoIP allows significant benefits for customers and communication services providers. The main are cost savings, rich media service, phone and service portability and mobility, and the integration with other applications. Nevertheless, the deployment of the VoIP technology encounters many challenges such as architecture complexity, interoperability problems, QoS concerns, and security issues. Due to the inability of the IP networking technology to support the stringent QoS constraints of voice traffic, and the incapability of traditional security mechanisms to adequately protect VoIP systems from recent intelligent attacks, QoS and security issues are considered as the most serious challenges for successful deployment of the VoIP technology. The aim of this paper is to carry out a deep analysis of the security issues and QoS concerns of the VoIP technology. Firstly, we present a brief overview about the VoIP technology. Then, we discuss the QoS problems encountering the deployment of the VoIP technology. The presented discussion mainly address the QoS issues related to the use of the IP networking technology, the QoS concerns related to voice clarity, and the QoS mechanisms proposed to support voice traffic QoS constraints. After that, we investigate the security issues of the VoIP technology. The presented investigation mainly address the vulnerabilities and security attacks of VoIP systems, as well as the countermeasures that should be considered to help the deployment of secured VoIP systems.

Complexity reduction in Karhunen-Loeve transform based speech coder for voice transmission

For gain-shape split vector quantization, an incremental M (IM) search algorithm is proposed. In the conventional M search, every possible combination of subvectors is evaluated. However, in the proposed IM search, an intermediate subvector is constructed by merging adjacent subvectors in an incremental and recursive way. Based on the IM search, the high-dimensional Karhunen-Loeve Transform (KLT) based speech coder is designed in practice. IM search requires 1800 times lower complexity than M search in designing a high-dimensional KLT-CVQ. Based on the proposed KLT based speech coder, a novel voice over IP (VoIP) system is designed, which produces superior voice quality and rate-distortion performance to conventional VoIP systems.

Design and implementation of a reconfigurable test-bed for real-time security measurements in VoIP systems

Voice over Internet Protocol (VoIP) systems have to face a great number of security threats, which are basically connected to the underlying Internet Protocol (IP) infrastructure. Along with relevant economic losses and malfunction for the VoIP system itself, external attacks exploiting a security hole could also bring to a major security failure for the interconnected Public-Switched Telephone Network (PSTN), usually regarded as a reliable and secure domain. This is the reason why before connecting a private VoIP network to the carrier infrastructure, a great number of tests have to be carried out, to verify the security robustness of the VoIP network to common attacks.Security tools that are commonly used to address potential threats in VoIP networks are: the insertion of firewalls, the separation of data and voice networks and the adoption of access control lists. The introduction of such elements in the network can lead to unpredictable modifications on the performance of the network and, consequently, of the VoIP service. The paper presents the design and implementation of a reconfigurable test-bed for real-time measurements on VoIP systems, which provides the telecommunication engineer with means to plan the necessary changes in order to reach a higher level of security. In its basic version, the test-bed permits to take into account the right balance between security tools and the required Quality of Service (QoS)/Quality of Experience (QoE) for real time operations. An interesting feature of the enhanced version is the possibility of evaluating QoS and QoE in the presence of firewalls and different security configurations, and assessing the effect of the joint variation of QoS parameters on QoE results.

Reducing Call Setup Latency in Mobile VoIP Systems

In mobile voice over IP (VoIP) systems, user mobility leads to the increased call setup latency to find out the current location of user, which has a negative impact on users call quality. In this letter, we propose a prefetching technique for user location to reduce the call setup latency. We develop an analytical model and investigate the impacts of mobility parameters and thresholds.

Security Test Environment for VoIP Research

Voice over IP (VoIP) is in wide use today, replacing phone lines in many scenarios. However, often, security isn’t considered well enough, even though many security attacks are already known. More research on VoIP security is needed to enhance the level of security of VoIP systems and to show the implications of failing to take appropriate security measures. This paper presents a short introduction in testing VoIP components, proposes an architecture and implementation of a robust, flexible and efficient VoIP test environment for security related tests. Experiences using the implemented environment for different VoIP security tests are shown to demonstrate the suitability of the proposed test environment for research and teaching purposes.

Bootstrap Node-Centered Proximity Identifier Selection for Structured Overlays in Private Networks

Distributed hash tables (DHTs) enable companies or campuses to use voice over IP (VoIP) systems in private networks at low cost. However, DHTs incur latency stretch caused by a topology mismatch between the application layer and the physical layer. Reducing this latency stretch is critical when implementing real-time applications that use DHTs, such as VoIP systems. This letter proposes a new method for reducing the latency stretch by creating node IDs, which form structured overlays that consider node proximity. Simulation results show that the proposed method is superior to Chord in terms of the distance between nodes, look-up path length, network burden, and look-up latency.

Analysis of information leakage from encrypted Skype conversations

Voice over IP (VoIP) has experienced a tremendous growth over the last few years and is now widely used among the population and for business purposes. The security of such VoIP systems is often assumed, creating a false sense of privacy. This paper investigates in detail the leakage of information from Skype, a widely used and protected VoIP application. Experiments have shown that isolated phonemes can be classified and given sentences identified. By using the dynamic time warping (DTW) algorithm, frequently used in speech processing, an accuracy of 60% can be reached. The results can be further improved by choosing specific training data and reach an accuracy of 83% under specific conditions. The initial results being speaker dependent, an approach involving the Kalman filter is proposed to extract the kernel of all training signals.

Comparative Study of M/Er/1 and M/M/1 Queuing Delay Models of the two IP-PBXs

Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies to delivery of voice communications over IP networks such as the Internet or other packet-switched networks. VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. This paper analyzes performance of SIP model (Session Initiation Protocol) based on IPPBX (IP-Private Branch exchange) which is a software application which provides call routing services by parsing and forwarding all the incoming SIP packets in various remote IP telephony network. Then, the M/Er/1 and M/M/1 performance models of the IP-PBX was simulated and some of the key performance benchmarks such as average response time to process the SIP calls and mean number of jobs in the system was reviewed.