VoIP Research Articles

ANALYSIS OF VOICE OVER INTERNET PROTOCOL ATTACKS DIRECTED AT IP TELEPHONY

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

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

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

Voice Over Internet Protocol (VoIP) User Location Detection Application

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

Conducting Interviews during the COVID-19 Pandemic and Beyond

The use of telephone and VoIP (Voice over Internet Protocol) interviews has become necessary owing to the legal restrictions and safety measures introduced during the COVID-19 pandemic. This has furthered scholarly dialogue surrounding the choice of interview mode, and the assumption that face-to-face interviews are the "gold standard" [Novick, 2008: 397]. The general public has also become more accustomed to utilising VoIP in their day-to-day lives. In this article, I discuss this change in communication methods and the impact it could have on interview modes. Nevertheless, the researcher's and participant's contexts remain paramount when deciding which interview mode to employ. For this reason, Oltmann's model [Oltmann, 2016] has been extended to include VoIP, providing a more complete framework for researchers to make an informed decision when contemplating interview modes.

Performance Evaluation of VoIP Codecs over Network Coding in Wireless Mesh Networks

Voice over Internet protocol (VoIP) is used for transmitting voice signals in a packet-switched Internet protocol (IP) networks in real time. For transmitting voice over a wireless mesh networks (WMNs), the analog voice signal has to be digitalized, encoded, and packetized. Codecs based on different Quality of Service (QoS) requirements are used. One of the main QoS requirements is that packets are transmitted through the network in real time; one-way transmission time or End-to-End (ETE) packet delay, and packet delay variation or jitter have to be lower than thresholds. ETE delay depends on various parameters; among them is also network delay. Various mechanisms are used to lower the network delay in WMNs. A promising mechanism, for improving the performance of streaming services such as the case also in VoIP, is network coding. In this paper, we evaluate the benefits of using wireless network coding for VoIP in WMNs. Network coding procedure in combination with various VoIP codecs is used to observe the impact on network delay and jitter of the VoIP application. The simulation results show that network coding can decrease network delay and jitter. Moreover, results show that network coding benefits are codec dependent.

Back to basics: fundamental principles of system dynamics and queueing theory

Back to basics: fundamental principles of system dynamics and queueing theory

EU Criminal Law and Electronic Surveillance: The Pegasus System and Legal Challenges It Poses

Abstract The introduction of modern forms of communication, such as encrypted messengers or VoIP telephony, has forced law enforcement agencies to use new technologies to carry out surveillance of people facing criminal proceedings. Rather than relying on the interception of communications during transmission, modern surveillance systems are often based on breaking or bypassing the security features of a user’s mobile device – making it possible to conduct various forms of surveillance that include audio and video recording. One example of such a system is Pegasus – a tool that was initially used to pursue national security objectives but is now increasingly applied in criminal surveillance. The introduction of technical innovations in criminal surveillance must include an examination of their compatibility with legal constraints laid down to protect the individual against the risk of arbitrariness and abuse of power. The effectiveness of surveillance should never be the sole determinant for tasks undertaken by public authorities. The aim of this article is to demonstrate that the implementation of modern surveillance measures such as Pegasus must also include a review of existing legal regulations to ensure that the use of these products’ extended capabilities is under proper control and complies with the rule of law.

LAN Traffic Capture Applications Using the Libtins Library

Capturing traffic and processing its contents is a valuable skill that when put in the right hands makes diagnosing and troubleshooting network issues an approachable task. Apart from aiding in fixing common problems, packet capture can also be used for any application that requires getting a deeper understanding of how things work under the hood. Many tools have been developed in order to allow the user to study the flow of data inside of a network. This paper focuses on documenting the process of creating such tools and showcasing their use in different contexts. This is achieved by leveraging the power of the C++ programming language and of the libtins library in order to create custom extensible sniffing tools, which are then used in VoIP (Voice over IP) and IDS (Intrusion Detection System) applications.

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.

Real-time steganalysis for streaming media based on multi-channel convolutional sliding windows

In recent years, covert communication technologies based on Voice over Internet Protocol (VoIP) have received more and more attention, which meanwhile poses a significant threat to the security of cyberspace. In this paper, we are chiefly concerned with improving the accuracy and efficiency of detection of covert communications, and we propose a real-time VoIP steganalysis model to tackle these issues. Multi-channel convolutional sliding windows (CSW) are developed to analyze the correlations between a given frame and its neighboring frames in a VoIP signal. Within each sliding window, we employ two feature extraction channels to extract correlation features from the input signal. Each channel is constructed of multiple convolutional layers having a large number of convolution kernels. The extracted features are then fed to a forward fully connected network for feature fusion. By analyzing the statistical distribution of these features, the discriminator will determine whether the input speech signal contains covert information or not. We designed several experiments to test the proposed model’s detection performance under various conditions, including different embedding rates, different speech lengths, etc. Experimental results show that the proposed model can efficiently and accurately detect steganographic voice streams, especially in the case of low embedding rates. In addition, further experiments demonstrate that the proposed model can attain nearly real-time detection of VoIP speech signals and achieve state-of-the-art performance.

Low-cost wireless mesh communications based on openWRT and voice over internet protocol

Technology makes it easier for us to communicate over a distance. However, there are still many remote areas that find it difficult to communicate. This is due to the fact that communication infrastructure in some areas is expensive to build while the profit will be low. This paper proposes to combine voice over internet protocol (VoIP) over mesh network implemented on openWRT router. The routers are performing mesh functions. We set up a VoIP server on a router and enabled session initiation protocol (SIP) clients on other routers. Therefore, we only need routers as a means of communication. The experiment showed very good results, in the line-of-sight (LOS) condition, they are limited to reception distances up to 145 meters while in the non-line-of-sight (NLOS) condition, they are limited to reception distances up to 55 meters.

An Efficient Method to Enhance IP Telephony Performance in IPV6 Networks

Abstract IP telephony have played an essential role during the COVID 19 pandemic lockdown. One of the issues that lower the service level of the IP telephony solutions is the inefficient bandwidth exploitation. This paper proposes a Smallerize/Zeroize (SmlZr) method to enhance bandwidth exploitation. The SmlZr method is explicitly designed for the P2P IP telephony calls over IPv6 networks. The essence concept of the proposed method is to use the unnecessary fields in the header to keep the voice media of the packet. Doing so leads to smallerize or zeroize the packet payload and, thus, enhance the bandwidth exploitation. The SmlZr method has outperformed the RTP method for all the comparison parameters. For instance, the SmlZr method shrinks the bandwidth by 25% compared to the RTP protocol. Bandwidth saving is helpful for P2P IP telephony calls because it alleviates the traffic load. Thus, improve the call capacity boosts the call clarity.

Implementasi dan Optimalisasi Voice Order Internet Protocol (VOIP) Menggunakan NDLC Untuk Komunikasi Antar Lantai

VoIP as an internet service is commonly called ip telephone. Internet infrastructure is needed in order to use or provide VoIP services. Some offices use local communication to communicate within the office environment itself, but cannot connect to other offices because it is only a local phone. This research aims to implement the Voice Over Internet Protocol (VoIP) by utilizing wireles local area network (WLAN) facilities by building a VoIP server. The development model used in this research is a user interface that is built alone to be applied on Android phones. This user interface is as a face-to-face tool for fellow VoIP users. The study was conducted at the West Bogor sub-district office with 23 respondents as a sample of application trial subjects. The results of the analysis showed 78.66% of the application eligibility level, showing the application or system made was considered good.

African Rigorous Innovative Stroke Epidemiological Surveillance: Protocol for a Community-Based Mobile-Health Study

Despite projections of stroke being a leading cause of mortality in Africa, reliable estimates of stroke burden on the continent using rigorous methods are not available. We aimed to implement a mobile-Health community-based interactive Stroke Information and Surveillance System to sustainably measure stroke burden and improve stroke health literacy and outcomes in urban and rural sites in Nigeria. African Rigorous Innovative Stroke Epidemiological Surveillance (ARISES) is an observational cohort study, which will be conducted in urban (Ibadan North LGA Ward 3) and rural (Ibarapa Central LGA) sites with a combined base population of over 80,000 people. The study will use a consultative approach to establish an mHealth-based Stroke Information and Surveillance System comprising a Stroke Alert System and a Stroke Finding System. These systems will enable the community to report stroke events and the research team/health workers find stroke cases using phone calls, short message service, and Voice Over Internet Protocols (VoIP). We will update community household data and geo-locate all households. Over the 5 years of the study, the system will collect information on stroke events and educate the community about this disease. Reported stroke cases will be clinically adjudicated at home and in prespecified health facilities. Baseline and endline community surveys will be conducted to assess stroke occurrence and other important study variables. The proportion of strokes alerted and found will be determined over the study period. Focus group discussions and key informant interviews will be conducted to understand community stroke literacy and perspectives. The study will also assess any impact of these efforts on time from stroke onset to referral, community uptake of orthodox health services for stroke patients. ARISES is anticipated to establish proof of concept about using mHealth for stroke surveillance in Africa. The potential impact of the SISS on improving patient outcomes will also be determined.

A Survey On Wireless Communications : 6g And 7g

One of the fastest growing sectors is wireless technology, which is evolving in all areas of mobile and wireless communications. Wireless technology has increased greatly in the last decade. 7.5 Generation (G) represents the history of wireless technology today. With 6G and 7G, data transmission rates will be higher over Future Generation wireless technology. With new technologies emerging in all fields of mobile and wireless communications, wireless technology continues to emerge as one of the hottest sectors with a high rate of development. Currently, 5G mobile communications systems are just getting started. Our current infrastructure supports a number of technologies, including voice over IP (VoIP), broadband data access over wireless, and more. This paper discusses several generations of wireless technologies from 0G to 7G. Wireless technology is important and beneficial to society. In this paper, we compare all of the generations and explain how each generation uses technology in its execution, application, and usage.

An optimized link state routing protocol for real-time application over Vehicular Ad-hoc Network

Voice over Internet Protocol (VoIP) provides good services through Vehicular Ad-hoc Networks (VANET) platform. These networks encounter various challenges to support voice calls with acceptable Quality of Service (QoS). The Optimized Link State Routing Protocol (OLSR) is used to investigate the performance of VoIP applications in the VANET network. The network was tested before and after running the OLSR algorithm, the testing focused on Quality of Service (QoS) parameters like end-to-end delay, delay variation (jitter), and probability of packet loss between two moving hops through multi-hop Ad-hoc networks in different scenarios, using the ITU G.711 VoIP codec. After measured the end-to-end delay, jitter, and probability of packet loss for two nodes the algorithm decreased delay by 18.72%, while decreased jitter by about 20.42% and decreased packet loss by about 128.6%. However, The OLSR has shown initial good performance for four hops, and then added more hops the delay exceeded 400 ms which is not acceptable according to ITU-T recommendations. This was achieved by implementing a testbed to extract the desired results as opposed to using a simulation.

Performance assessment of VoIP service over different handover mechanisms in UMTS networks

Many researchers have discussed various topics in universal mobile telecommunication system (UMTS) networks: the process of switching from one cell to another for the subscriber and the impact of the quality of the connection during the transition process, quality of services (QoS), the quality of the uplink and downlink carrier line, the various types of code for the voice transmitted through the Internet, especially the research that discussed voice over internet protocol (VoIP) technology as voice travels from cell to cell in mobile networks, depending on the type of delivery. In this paper, a proposed scenario of a UMTS network was implemented to evaluate the multicellular VoIP movement; the proposed UMTS network was simulated using the OPNET 14.5 simulator. The calculation and analysis of the different parameters of the user while moving from one cell to another with different movement speeds considered, the best mean opinion score (MOS) value (3.19) registered for the scenario (soft handover) comparing with another type of handover (3.00).

Zeroize: A New Method to Improve the Utilization of 5G Networks When Running VoIP over IPv6

5G technology is spreading extremely quickly. Many services, including Voice Over Internet Protocol (VoIP), have utilized the features of 5G technology to improve their performance. VoIP service is gradually ruling the telecommunication sector due to its various advantages (e.g., free calls). However, VoIP service wastes a substantial share of the VoIP 5G network’s bandwidth due to its lengthy packet header. For instance, the share of the packet header from bandwidth and channel time reaches 85.7% of VoIP 5G networks when using the IPv6 protocol. VoIP designers are exerting considerable efforts to solve this issue. This paper contributes to these efforts by designing a new technique named Zeroize (zero sizes). The core of the Zeroize technique is based on utilizing the unnecessary fields of the IPv6 protocol header to keep the packet payload (voice data), thereby reducing or “zeroizing” the payload of the VoIP packet. The Zeroize technique substantially reduces the expanded bandwidth of VoIP 5G networks, which is reflected in the wasted channel time. The results show that the Zeroize technique reduces the wasted bandwidth by 20% with the G.723.1 codec. Therefore, this technique successfully reduces the bandwidth and channel time of VoIP 5G networks when using the IPv6 protocol.

Enhancement of WiMAX networks using OPNET modeler platform

Worldwide interoperability microwave access (WiMAX) is an 802.16 wireless standard that delivers high speed, provides a data rate of 100 Mbps and a coverage area of 50 km. Voice over internet protocol (VoIP) is flexible and offers low-cost telephony for clients over IP. However, there are still many challenges that must be addressed to provide a stable and good quality voice connection over the internet. The performance of various parameters such as multipath channel model and bandwidth over the Star trajectoryWiMAX network were evaluated under a scenario consisting of four cells. Each cell contains one mobile and one base station. Network performance metrics such as throughput and MOS were used to evaluate the best performance of VoIP codecs. Performance was analyzed via OPNET program14.5. The result use of multipath channel model (disable) was better than using the model (ITU pedestrian A). The value of the throughput at 15 dB was approximately 1600 packet/sec, and at -1 dB was its value 1300 packet/se. According to data, the Multipath channel model of the disable type the value of the MOS was better than the ITU Pedestrian A type.

Using the Discord Application to Facilitate EFL Vocabulary Acquisition

Vocabulary knowledge is a prerequisite for mastering any foreign language. It takes a lot of time and effort to develop English as a foreign language (EFL) vocabulary skills. Efficient ways to facilitate this process should be studied and analyzed. The present study aimed to investigate the role of the Discord application in teaching and learning EFL vocabulary. Discord is a free messenger with support for IP telephony and video conferencing, as well as the possibility to create public and private chats for exchanging text and voice messages. The study relied on pre-test and post-test design to get the necessary data of one experimental and one control group, 80 university students altogether. The results of the pre-test showed no significant difference between the two groups. The experimental group was taught with the help of the Discord application, while the control group was dealing with the traditional educational process, that is, 180 min every week in a classroom with a textbook and no electronic educational resources. Results of the t-test indicated that the experimental group significantly outperformed the control group on a researcher-made vocabulary post-test. A post-treatment speaking test was administered—the participants took part in speaking interviews where they were asked to give a short speech on one of the topics studied during the experimental training. The representatives of the experimental group managed to use more vocabulary items per person correctly (pronunciation, lexical combinability, etc.). The findings revealed that the suggested way of using the Discord application may positively influence the acquisition of EFL vocabulary and its further application in speaking. Hence, it is recommended that the Discord application be used as a tool in the EFL vocabulary teaching and learning process to reinforce in-class tasks and activities.