VoIP Services Research Articles

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

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

Enhanced Secure Communication Protocol with Pipelined Advanced Encryption for Mobile Networks

In today's increasingly connected world, the demand for secure mobile communications is paramount. The security protocols are specifically tailored for mobile communication systems, ensuring the confidentiality and integrity of sensitive data transmitted over wireless networks. It can be seamlessly integrated into various mobile applications such as messaging platforms, VoIP services, and mobile banking apps, providing end-to-end encryption for user privacy. Current encryption protocols used in mobile communications often face challenges related to performance and security. Traditional encryption methods may not adequately address the evolving threats posed by sophisticated attackers. Moreover, the computational overhead associated with encryption and decryption processes can impact the overall efficiency of mobile communication systems. This work presents a novel approach to enhancing the security of mobile communications through the design of a Pipelined Advanced Encryption Based Cryptography Protocol (PAEBCP). The proposed protocol aims to address the vulnerabilities present in existing encryption methods by leveraging a pipelined architecture for efficient encryption and decryption processes. The proposed PAEBCP protocol introduces a pipelined architecture that optimizes the encryption and decryption processes for mobile communication systems. By dividing the encryption process into multiple stages and parallelizing key operations, the protocol enhances both security and performance.

Impact Analysis of Insufficient Destination Resources on VoIP Service Quality: Case Study

Impact Analysis of Insufficient Destination Resources on VoIP Service Quality: Case Study

QUALITY OF EXPERIENCE FOR VOICE OVER INTERNET PROTOCOL (VoIP)

Today, the service of Voice over Internet Protocol or (VoIP) is one of the most used service around the globe in many fields, especially with video conferencing applications. Different applications and also many communications companies around the world use the VoIP service. Communication stakeholders aim to keep old customers happy and attract new customers. There was a need for some kind of service quality measures. Metrics are used to quantify the quality of transmission link are known Quality of Service (QoS). Whereas, user satisfaction level is identified by using Quality of Experience (QoE). This paper conducted a simulation model for VoIP service over heterogeneous, using OPNET modeler. This work aims to compare the user satisfaction level of the VoIP service by using different coding schemes in the application level of the User Equipment (UE). The measured QoS parameters over the heterogeneous transmission link were IP packet delay, IP Packet jitter, IP packet loss, MAC layer delay, PHY layer throughput in the network. Additionally, in this simulation model the use of E-Model to assess QoE level using Mean Opinion Score (MOS) metric in predicting the VoIP call quality. The results using the vocoder G.729A during the (30) sec of the voice call simulation time were as follows: Delay 0.57 mSec, Throughput of 290 Kbits/sec, Jitter of 0.02 uSec, MOS value of 3.76. The results using the vocoder G.711 during the same aforementioned time were as follows: Delay 0.25 mSec, Throughput of 70 Kbits/sec, Jitter of 0 Sec), MOS value of 3.08.

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.

Predictive Modeling of Delay in an LTE Network by Optimizing the Number of Predictors Using Dimensionality Reduction Techniques

Delay in data transmission is one of the key performance indicators (KPIs) of a network. The planning and design value of delay in network management is of crucial importance for the optimal allocation of network resources and their performance focuses. To create optimal solutions, predictive models, which are currently most often based on machine learning (ML), are used. This paper aims to investigate the training, testing and selection of the best predictive delay model for a VoIP service in a Long Term Evolution (LTE) network using three ML techniques: Multilayer Perceptron (MLP), Support Vector Machines (SVM) and k-Nearest Neighbors (k-NN). The space of model input variables is optimized by dimensionality reduction techniques: RReliefF algorithm, Backward selection via the recursive feature elimination algorithm and the Pareto 80/20 rule. A three-segment road in the geo-space between the cities of Banja Luka (BL) and Doboj (Db) in the Republic of Srpska (RS), Bosnia and Herzegovina (BiH), covered by the cellular network (LTE) of the M:tel BL operator was chosen for the case study. The results show that the k-NN model has been selected as the best solution in all three optimization approaches. For the RReliefF optimization algorithm, the best model has six inputs and the minimum relative error (RE) RE = 0.109. For the Backward selection via the recursive feature elimination algorithm, the best model has four inputs and RE = 0.041. Finally, for the Pareto 80/20 rule, the best model has 11 inputs and RE = 0.049. The comparative analysis of the results concludes that, according to observed criteria for the selection of the final model, the best solution is an approach to optimizing the number of predictors based on the Backward selection via the recursive feature elimination algorithm.

EVALUATION OF VOIP TRAFFICS OVER TIKRIT UNIVERSITY NETWORKS

In recent years, VoIP is a rapidly evolving technology that could possibly revolutionize the telecommunication industry. When implemented on wireless data networks, VoIP could prove to be instrumental in the convergence of existing fixed and cellular telephony networks with the fast growing wired and wireless data networks. Whereas, the growth of the VoIP applications with additional mobile devices in the institution of higher education and increasing number of users led to slow VoIP services. In addition, the throughput and latency (or delay, jitter, and jabber) are two of the most important issues that need to be resolved before the commercial deployment of wireless VoIP. The trace composes of two parts: firstly, collecting of data, secondly, performing the traffics analysis. Therefore, we have analyzed the performance offered by SCTP, TCP and UDP over VoIP at Tikrit University, Iraq. In this paper, we can see that the kind of performance given by SCTP is best then UDP, but the latency is decreased it performance significantly compared to UDP and the SCTP throughput is looks in a better manner. The measurements were verified by simulating the VoIP traffics using Network Simulator 2

Performance Evaluation of VoIP Service for Emergency Disaster Responses over WiMAX-WLAN Integrated Network

As a very large archipelago country, Indonesia has a high intensity of the natural disaster. In many cases, the telecommunication infrastructure collapse during the disaster. Therefore, a prominent and robust emergency telecommunication network is required. In this paper, an implementation of VoIP services over the WiMAX-WLAN integrated network is proposed. The integrated network is designed to provide an alternative of the emergency network during the disaster response. The VoIP service is selected for the study case because this service can be used as a complementary of the telephony service. In this paper, the WiMAX network is designed based on the IEEE Standard 802.16-2004 with the frequency band of 3.5 GHz while the WLAN network is designed based on the IEEE Standard 802.11n. To support many scenarios of network topology, the proposed integrated network is designed in P2P and PMP topologies. Within these network topologies, the network performance is evaluated in terms of delay, jitter, throughput and packet loss, as well as MOS value. The network performance is evaluated over various background traffic. From the experimental evaluation, it shows that the network becomes worse when the network load that is represented by the background traffic exceeding 3 Mbps. In general, there is no significant performance different between P2P and PMP network topology.

NR Coverage Enhancements for PUSCH

Coverage performance is very important for operators when commercializing NR networks due to its direct impact on service quality as well as CAPEX and OPEX. Compared to LTE, NR systems may operate at higher frequency bands, resulting in larger path loss and worse coverage performance. In this article, potential coverage issues for NR networks are discussed. Through link budget analysis, the gaps between baseline and target performance for typical NR physical channels are described; then the bottleneck channels are identified, including PUSCH, PUCCH, and PRACH. This article mainly focuses on PUSCH coverage enhancement, and potential solutions are discussed in a comprehensive manner. On top of the potential solutions, four joint schemes, which take feasibility as well as specification impact into consideration, are proposed for VoIP and eMBB services to further improve the coverage performance.

Analisis Keamanan Komunikasi VoIP Server Portable Dilengkapi OpenVPN Menggunakan Linux Asterisk

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

Node Density Performance Analysis on IEEE 802.11ah Standard for VoIP Service

The growing of internet and IoT application makes large number of devices will be connected each other to exchange data communication. IEEE made standard of WLAN technology to support the needs. But from the existing standard technologies, there are still deficiencies in delivering the data package for communication. IEEE 802.11ah was made as a new standard of Wi-Fi technology that supports wireless networking scenarios and solving the problem of capacity, coverage and power consumption in IoT era. The wireless access network is very attractive for best-effort data package transmission, but VoIP service on the wireless network is very critical due to its real-time communication characteristic. This research addresses the performance analysis of 802.11 ah standard against VoIP services. To determine the influence of node density in 802.11ah standard, the simulation is done by changing the number of station that access the channel via RAW mechanism. The simulation is performed using network simulator NS3 version 3.23. Based on the result, it can be concluded that the evaluation of IEEE 802.11ah standard in the node density scenario accompanied by the changing of RAW station and RAW group number, the use of MCS 5 with 2 MHz bandwidth resulting better network performance than MCS 7 with 1 MHz bandwidth. Using MCS 5 obtained delay value of 0.241761 s on average, the throughput value of 0,5648 Mbps on average, and the PDR value of 19.84 % on average.

Efficient Detection of Spam Over Internet Telephony by Machine Learning Algorithms

Recent trends show a growing interest in VoIP services and indicate that guaranteeing security in VoIP services and preventing hacker communities from attacking telecommunication solutions is a challenging task. Spam over Internet Telephony (SPIT) is a type of attack which is a significant detriment to the user’s experience. A number of techniques have been produced to detect SPIT calls. We reviewed these techniques and have proposed a new approach for quick, efficient and highly accurate detection of SPIT calls using neural networks and novel call parameters. The performance of this system was compared to other state-of-art machine learning algorithms on a real-world dataset, which has been published online and is publicly available. The results of the study demonstrated that new parameters may help improve the effectiveness and accuracy of applied machine learning algorithms. The study explored the entire process of designing a SPIT detection algorithm, including data collection and processing, defining suitable parameters, and final evaluation of machine learning models.

Chebyshev chaotic map-based efficient authentication scheme for secure access of VoIP services through SIP

Chebyshev chaotic map-based efficient authentication scheme for secure access of VoIP services through SIP

Chebyshev chaotic map-based efficient authentication scheme for secure access of VoIP services through SIP

Chebyshev chaotic map-based efficient authentication scheme for secure access of VoIP services through SIP

HIGH QUALITY LOW BITRATE VOICE CODEC FOR TRANSMISSION OVER ADVANCED LTE

In mobile communication systems bit-rate reductions while maintaining an acceptable voice quality are necessary to achieve efficiency in channel bandwidth utilization and users satisfaction. As Long-Term Evolution(LTE) converging towards all-IP solutions and supporting VOIP service, the voice signals are converted into coded digital bit-stream and sent over the network. This paper proposes the implementation of codebook excited linear prediction (CELP) voice codec algorithm based on two source-rates of low 9.6Kbps and medium 16Kbps for achieving a perceptible level of voice quality, while efficiently using available bandwidth during the transmission over advanced LTE. The architecture of proposed CELP codec model is implemented to decompose the voice signal into a set of parameters that characterize each particular frame at the encoder part, these parameters are quantized and encoded for transmission to the decoder. The investigation showed that the configuration of the link and the applied CELP codec mode mainly influence on the obtained voice capacity and quality. The quantifying also shows that the voice quality can be traded for the enhanced capacity, since the low rate codec will produce lower voice quality than higher rate codec. Also, this paper is achieved, during theconfiguration of the system with higher channel quality indicator (CQI) index, increasing in the capacity gain to a saturated value of about 500 and 1000 users per cell over 5MHz bandwidth for transmit diversity (TD) and Open-Loop Spatial Multiplexing (OLSM) respectively and up to 1000 and 2000 users per cell over 10MHz channel bandwidth for TD and OLSM respectively.

Wireless Last Mile Study in Rural Areas

Internet access has become a primary human need. During the COVID 19 pandemics, almost all activities such as studying and working online. However, several rural areas still did not have internet access due to weak cellular signals. Internet access in rural areas can have various alternatives, such as VSAT (satellite). It needs an inexpensive and precise way to complete the internet access coverage because many customers were concentrated in areas far from the VSAT terminal. WLAN 802.11n was an alternative way to expand the range of internet coverage with low cost and easy implementation. Our case study uses the 5GHz frequency with a sectoral antenna as the multi-hop point-to-point backhaul network's frequency to avoid the risk of channel interference on the backhaul. On the other hand, 2.4GHz frequency with an omnidirectional antenna serves smartphone customers. On the test results, the network latency between multihop access point, has a maximum value of 19,894 ms. This result means that the latency obtained can be categorized as preferred VoIP services based on ITUT G.1010 in the local network. Then for the customer on the fourth hop, 3968 meters from the VSAT terminal, the UDP data rate of 1.04Mbps was stable, and the TCP data rate decreased to 1.26Mbps. This paper emphasizes the use of multihop 5Ghz WLAN 802.11n as a backhaul to expand internet access coverage from VSAT in rural areas, where the concentration of customers was far apart and there were many buffalo horn barriers in traditional homes and buildings.

A Two Stage Queuing System with delay time

This paper discusses about a two stage queueing system with shared server and the delay time between two end to end responses. This system is examined as almost the process of quasi-birth and death and the necessary and sufficient conditions for system stability are discussed. The ErntPS algorithm for VoIP Services has recently been investigated for average delay in queueing and it has been proven that ErtPS can allow more users than UGS. However, we need the distribution of probabilities of queue delays in order to include information necessary for QoS rather than the average queue delays. Some fascinating performance measures are achieved by matrix geometric method. The server wishes to determine the optimal point of differentiation and its optimal buffer size. Finally we find the delay time differentiation and the specific probabilistic system descriptors.

Pengembangan Server VOIP Menggunakan FREEPBX Dan Asterisk Berbasis Raspberry PI

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

Improved algorithm for enhance robustness of IPTV based on GEPON

Abstract Ten Gigabit-capable ethernet passive optical network (GEPON) can quickly improve network bandwidth, meet multilevel business needs, and quickly improve the comprehensive competitiveness of the network. It is well known that the 10G EPON access network technology is used to carry IPTV services, VoIP services and high-speed Internet access, known as “triple play”, and IPTV services have been particularly popular in recent years. As the demand for IPTV service increases exponentially, a large amount of data will be generated, occupying a large access bandwidth and causing an impact on other services. Therefore, reasonable scheduling of IPTV service queue is an urgent problem to be solved under multiservice environment at present. Therefore, the design of PON bearing IPTV robustness algorithm to meet user experience is the focus of this paper. This paper proposes a Forward Error Correction with Weight Round Robin algorithm (FWRR), which adopts RS error correction code and Weighted Round Robin algorithm based on IP layer. When IPTV packet is lost at optical line terminal (OLT) end, downlink optical network unit (ONU) can recover the lost data through received packets. This mechanism can effectively solve the problem of packet loss caused by the blocking of OLT end, and realize the scheme of PON carrying IPTV multicast service. Simulations results in OPNET show that compared with weighted round robin (WRR) algorithm and round robin (RR) algorithm, FWRR mechanism can reduce packet loss, delay and jitter, in addition, improve the robustness of IPTV service in GEPON.

Advanced Handover Techniques in 5G LTE-A Networks

In High-speed and seamless access is the goal of next-generation mobile communication systems, especially for VoIP and multimedia services. The supported mobile speeds are expected to reach 500 km/h in the Fifth Generation Networks (5G), lead to the handover will take place continuously as the mobile user moves through the cells. Therefore, efficient radio resource management, including the handover, load balancing technologies is vital problems. In this work, we perform a comprehensive survey of advanced handover techniques in the next-generation mobile communication system (5G) as well as requirements and features for the LTE-Advanced system. In the LTE-Advanced system, Fractional Soft Handover (FSHO) with 5 CCs improves latency and better handover quantity when not using CA. Currently, the proposed handover techniques have many advantages but have not yet resolved important issues in the handover process. Therefore, new handover techniques are required to support a fast and seamless handover process in the LTE-Advanced system. As a result, an advanced transfer technique proposed by combining FSHO, SSHO techniques can reduce latency, enhance efficiency and reliability, especially in the border areas between the cells.