VoIP Technology Research Articles

Performance analysis: Securing SIP on multi-threaded/multi-core proxy server using public keys on Diffie-Hellman (DH) in single and multi-server queuing scenarios.

The rapid replacement of PSTN with VOIP networks indicates the definitive phase-out of the PBX/PABX with smartphone-based VOIP technology that uses WLAN connectivity for local communication; however, security remains a key issue, regardless of the communication coverage area. Session initiation protocol (SIP) is one of the most widely adopted VOIP connection establishment protocols but requires added security. On the Internet, different security protocols, such as HTTPS (SSL/TLS), IPSec, and S/MIME, are used to protect SIP communication. These protocols require sophisticated infrastructure and some pose a significant overhead that may deteriorate SIP performance. In this article, we propose the following: i) avoid using Internet bandwidth and complex Internet protocols for local communication within an organization, but harness WLAN connectivity, ii) use multi-threaded or multicore computer systems to handle concurrent calls instead of installing hardware-based SIP servers, and iii) run each thread in a separate core. Cryptography is a key tool for securely transmitting confidential data for long- and short-range communication, and the Diffie-Hellman (DH) protocol has consistently been a popular choice for secret key exchanges. Primarily, used for symmetric key sharing, it has been proven effective in generating public/private key pairs, sharing public keys securely over public channels, and subsequently deriving shared secret keys from private/public keys. This key exchange scheme was proposed to safeguard VOIP communication within WLANs, which rely on the SIP for messaging and multimedia communication. For ensuring an efficient implementation of SIP, the system was rigorously analyzed using the M/M/1 and M/M/c queuing models. We analyze the behavior of SIP servers with queuing models with and without end-to-end security and increase users' trust in SIP security by providing a transparent sense of end-to-end security as they create and manage their private and public keys instead of relying on the underlying SIP technology. This research implements instant messaging, voice conversation, and secret key generation over DH while implementing and observing the role of multi-threading in multiqueue systems that serve incoming calls. By increasing the number of threads from one to two, the SIP response time improved from 20.23809 to 0.08070 min at an arrival rate of 4250 calls/day and a service rate of three calls/min. Similarly, by adding one to seven threads, the queue length was reduced by four calls/min. Implementing secure media streaming and reliable AES-based signaling for session confidentiality and integrity introduces a minor 8-ms tradeoff in SIP service performance. However, the advantages of implementing added security outweigh this limitation.

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.

Utilizing Voip Packet Header’s Fields to Save the Bandwidth

Abstract Voice over IP (VoIP) is widely utilized by organizations, schools, colleges, and so on. Nevertheless, VoIP numerous challenges that hinder its spread. One of the significant challenges is the poor exploit of the VoIP technology network bandwidth (BW), caused by the huge preamble of the VoIP packet. This paper suggests a novel methodology to manage this huge preamble overhead challenge. The proposed methodology is named runt payload VoIP packet (RPV). The core principle of the RPV methodology is to reemploy and exploit the VoIP packet preamble’s data (fields) that are superfluous by VoIP technology, especially for unicast IP voice calls. Generally, those fields will be used to convey the VoIP packet payload. Consequently, diminish or zero the length of the payload and, therefore, spare the BW. The results of the investigation into the suggested RPV methodology indicated significant enhancement in the BW exploitation of VoIP technology. For instance, the saved BW in the examined environment with the LPC codec came to up to 25.9%.

Conducting Grounded Theory Research in the Early Days of the Coronavirus Pandemic: Process Interruptions, Barriers, and Innovative Approaches to Study Design

<i>The COVID-19 pandemic changed the ways by which qualitative research is designed, participants are recruited and engaged, and results are shared regardless of research disciplines or epistemological perspectives. As a result of on-going requirements for social distancing and remote working or learning, researchers have had to redefine effective research methods by reimaging recruitment and data collection, investing in VoIP technology, and developing innovative ways to build trust across digital platforms. This manuscript explores how one Grounded Theory research study evolved in response to the emergence of COVID-19 and presents recommendations for innovative approaches to study design, data collection, and continuous participant interaction. We also briefly discuss ethical considerations that have emerged over the past two years.</i>

Design a Real-time Communication System using 3CX Software-based Private Branch Exchange Phone System on Raspberry Pi Device

With the advancement of technology, the telephone network has become the primary mode of communication worldwide, and private businesses have increased their reliance on telephone communication. Many organizations choose to establish their own service in order to manage internal calls. Voice over Internet Protocol (VoIP) is one of the emerging technologies that may provide low-cost service with high-quality and availability. VOIP technology enables the transfer of multimedia data such as audio and video. While some VoIP services require a computer or a dedicated VoIP phone, others allow you to make VoIP calls using your landline phone via a special adaptor. Rather than using a traditional private branch exchange (PBX), we used a Raspberry pi, which is a set of credit card-sized single-board computers, as a server for handling voice and video call communications over a wired or wireless LAN network while monitoring the entire system. Wireshark is a software application that is used to capture packets in a network and present information about certain packets in as much detail as possible.

VOIP technology in grassroots politics: Transforming political culture and practice?

This article investigates how the adoption and use of digital technologies shape political culture and practice in grassroots political groups, particularly focusing on how VOIP technologies enable and/or constrain groups to work across physical space and form political relationships among participants. While this article is grounded in a case study of one broad-based coalition in Sydney, Australia, the findings expand our understanding of how digital technology shapes political culture and practice in grassroots spaces by (a) analysing an organisation both before and after the adoption of VOIP technology and (b) focusing on a case study where the organisation attempted to maintain rather than transform their political culture and practice with the adoption of new digital organising methods. The article argues that the instrumental benefits of digital technologies come at a cost: VOIP technologies may constrain the formation of deep relationships and flatten distinctive political practices within grassroots political organisations.

Multi-service threats: Attacking and protecting network printers and VoIP phones alike

Printing over a network and calling over VoIP technology are routine at present. This article investigates to what extent these services can be attacked using freeware in the real world if they are not configured securely. In finding out that attacks of high impact, termed the Printjack and Phonejack families, could be mounted at least from insiders, the article also observes that secure configurations do not appear to be widely adopted. Users with the necessary skills may put existing security measures in place with printers, but would need novel measures, which the article prototypes, with phones in order for a pair of peers to call each other securely and without trusting anyone else, including sysadmins.

Perancangan Jaringan Voice Over Internet Protokol (Voip) Menggunakan Platform Asterisknow

Communication is an important stage to support the running of a work program, without good communication it can cause misunderstandings between the two parties, the importance of the role of communication means several companies are growing rapidly in order to innovate in the manufacture of equipment in the telecommunications sector, which is commonly used in long-distance communication. far is the use of telephone services, both individual telephones and company-based telephones, for some companies that use high-enough telephone activity, this will result in soaring telephone bills that must be paid, thus to reduce the level of communication with high costs, we can use VoIP technology ( Voice Over Internet Protocol) which can be used by utilizing the TCP / IP protocol where the protocol without us realizing it is a protocol used for exchanging data, thus the protocol can also be accessed. used as voice voice data exchange using Open Source Software in the Asterisk application which is used as a substitute for PBX (Private Branch Exchange) devices, where in the application it is used to make the TCP / IP protocol as an intermediary for data exchange to communicate or Voice. The advantage of this application is that it is able to serve a maximum of 1000 online accounts on one server and 240 concurrent calls. This is of course suitable for companies who want to implement this technology based on open source applications.

High Pass Filter Application to Reduce Voice Communication Delays on IP Phones

Abstract: VoIP technology is equipped with several functions, namely, the signalling function, meaning the VoIP is in charge of receiving the network from the caller, after which the conversation delivered. This technology is capable to pass voice traffic in the form of packets over an IP network. Packets of sound undergo a long process or delay to get to the destination it can damage the voice’s quality being heard. It happened because there is continuous delay in the communication between IP phone set on VoIP technology that causes echo in the receiver’s voice. Echo can occur during communication with IP phone set with average delay capacity above 5ms. The delay on the delivery and reception as in the results of communication between telephones can be adjusted for sound frequency of 300 Hz to 1000 Hz as a cut off by adding High Pass Filter (HPF) application. HPF filter application is able to stabilize the amplitude of about -21 dBm from the set of transmission test on the receiver when there is weakening at low frequencies, but when the frequency is raised to 1500-2250 Hz the amplitude strengthens to -12,2 dBm or increases the value of about 8.8 dB. Meanwhile, for the lower frequency such as 300 – 950 Hz, the filter would not pass it since the frequency is designed to be cut-off at 1000 Hz. The value of delay is narrower to 0.08ms by HPF application at the frequency of sound upper limit received by any 1000-3400 Hz telephone set. Keywords: VoIP technology, echo cancelling, HPF filter, delay, voice quality

A Survey on Voice over Internet Protocol (VoIP) Reliability Research

VoIP technology deals with the real-time data communication for voice transfer in the form of digital packets through internet communication and facilitates public to make use of internet for video and phone calls. The voice data packets are transferred from source to the destination and vice versa. VoIP require high speed internet connection for data transfer on data network. It is necessary to arrive the data packet from its source to the destination with high level of reliability. It is very important to analyse link failure, packet loss, delay and jitter during the data communication. Components involved in data communication should be reliable. This paper provides a survey for step wise development and use of reliability techniques for gaining high quality of voice in VoIP network.

Comparison of Voice over Internet Protocol (VoIP) Performances in Various Network Topologies

VoIP is a digital communication technology that is currently developing because VoIP can be implemented on several network topologies, such as bus, star, and ring. Each of these topologies has advantages and disadvantages. So, a study is required to find out in which topology can VoIP be implemented optimally. In this research, VoIP is implemented in several topologies and furthermore the performance measurements are carried out for each topology. VQ manager is installed in order to measure the VoIP performance. For the server, we used Elastix and for the node implementation network topologies, we use several access points. From the results of the research, the performance of VoIP implemented in the star topology produces QoS that is better than other topologies with a delay value of 185 ms, 18 ms jitter, and 1% packet loss. This happens because in the star topology, all packets are distributed centrally. It is expected that the results of this research can be used as a reference in the application of VoIP technology in several types of topologies.

Design and Implementation of a Collaborative Communication Network for Video Conference Services

Under the current global conditions, strong emphasis is put on saving resources and reducing costs, migrating to collaborative unified communications solutions and developing open-source Web applications. The secure video-conferencing system will mainly use VoIP and WebRTC technologies to make multimedia calls, allowing end-users mobile and remote access (MRA) and therefore to facilitate communication between different networks, as well as the accessibility and the interoperability with any type of technology and device. This article will analyze the scalable implementation of the real-time communications network, providing redundancy and load balancing, therefore ensuring the high availability of the entire collaborative communications system. The call control and the dial plan are done through the Unified Call Manager software and WebRTC videoconferencing external access functionality is provided by configuring Traversal Using Relays around NAT (TURN) service across the Expressway Series server pair (Core & Edge). The ability to manage and monitor virtual meetings remains the responsibility of the open-source management tools.

МОДЕЛЮВАННЯ ТА АНАЛІЗ РОБОТИ ВНУТРІШНЬОЇ ТЕЛЕФОННОЇ МЕРЕЖІ НА БАЗІ IP-СТАНЦІЇ З ВИКОРИСТАННЯМ ПРОТОКОЛУ SIP

The development of Internet technologies is characterized by wide possibilities for introducing various telecommunication services, in particular IP telephony, which provides users with a new technology for multimedia traffic transmittion using Internet networks. Special data transfer protocols (SIP and H.323) are being actively developed to apply this technology, which enables the organization of private and corporate data networks. The paper analyzes the use of these protocols based on the Panasonic KX-NS500 IP station, which has made it possible to give recommendations on improving the operation of internal telephone network for the transmission of voice traffic for various types of codecs. The article discusses the problem of improving the efficiency of IP based voice communications using SIP and H.323 protocols. A communication model, that uses three analog phones and three IP phones serviced by an IP station, is proposed. The telephone line uses SIP protocol. The OPNET Modeler software environment is used to investigate the performance of the proposed network model. The optimal type of voice codec for SIP protocol and the possibilities of using the protocol in the organization of internal telephone network using the IP station are determined. The telecommunication network segment using SIP protocol has been simulated and the characteristics of two communication codecs most commonly used in VoIP technology have been analyzed. Based on the simulation, the assessment of the quality of the speech signal by the MOS rating, the range of delay of the speech signal and the choice of a codec to play a voice signal for the proposed network model have been made.

Traffic classification for connectionless services with incremental learning

The technological advancement in VoIP technology and P2P streaming led to the development of novel applications. Most of these applications use UDP traffic. The availability of UDP services for applications such as streaming, trivial file transfer, are denied to legitimate users due to malicious traffic, intentionally created by abnormal requesting behaviour of the botnets. Categorizing the traffic is required to discriminate the malicious traffic that occur due to attacks from normal traffic for better real time resource allocation. For this purpose, this paper proposes a two level hybrid classification model based on incremental learning to detect high and low rate attacks that deny the legitimate access to connectionless services. The simulation results show that the proposed incremental learning strategy improves the classification accuracy of the proposed hybrid classifier compared to existing traditional learning methods.

Analysis of protocols and techniques for transmission of voice over internet protocol

The purpose of the article is to present the process of the evolution of telecommunication networks across the years and are developing at a very rapid pace, with a tendency towards convergence of services. Voice over IP is the preferred method for more and more telecommunications operators, replacing standard telephony and networking. As a result, it can be noted that VoIP technology allows much more information to be transmitted over the network to serve and improve communication needs than traditional telephony. The authors' contribution is the analysis of voice over IP protocols, which clarifies the concepts and rules in this type of communication and presents the functionalities and components of these communications.

Integrating telecollaborative language learning into Higher Education: a study on teletandem practice

Telecollaborative learning can be implemented at higher education institutions as an independent practice or as a blended setting integrated into classroom activities. Considering telecollaborative practices in European universities, some researchers proposed categories and levels of classroom and institutional integration. Teletandem, a model of telecollaboration implemented at UNESP (Universidade Estadual Paulista), involves two speakers of different languages who use the text, voice, and webcam image resources of VOIP technology to help each other learn their native language (or language of proficiency). This paper aims at describing and discussing the different teletandem learning set-ups implemented at the Brazilian university in relation to the levels of integration described by Brammerts (2002; 2003) and Lewis & O’Dowd (2016). The discussion is based on the data collected by means of a questionnaire answered by teletandem practitioners (teachers and researchers) at UNESP. Results revealed low levels of integration both into classroom and into institutional practices.

Modelo Sip Seguro para una Comunicación extremo a extremo sobre IPV6

Modelo Sip Seguro para una Comunicación extremo a extremo sobre IPV6 Sip Security Model for End to End Communication on IPv6 Ross M. Benites, José L Quiroz, Raúl Villafani INICTEL-UNI, Lima 41 DOI: https://doi.org/10.33017/RevECIPeru2011.0024/ RESUMEN Las implementaciones VoIP hoy en día se han incrementado considerablemente, sin embargo no entodos los escenarios se tiene en cuenta los mecanismos de seguridad adecuados. Este último punto es muy importante a considerar el día de hoy , sobre todo por el agotamiento de las direcciones IPV4 y el despliegue hacia IPV6 de muchos de los servicios, donde aparecerán nuevas amenazas a la seguridad que trataran de opacar el gran auge de la tecnología VoIP. Si bien IPv6 fue desarrollado para solucionar muchas vulnerabilidades en seguridad que actualmente se ven presentes en IPv4, el hecho es que no logra alcanzar aún estas metas según pruebas realizadas. El protocolo SIP, el actor principal de la tecnología VoIP , requiere de la implementación de mecanismos de seguridad . Los escenarios actuales requieren terminales de usuario de alto rendimiento y soporte para adaptarse a mecanismos de seguridad heterogéneos o asumir relaciones de confianza. Sin embargo debemos tener en cuenta que hay varias combinaciones de soluciones de seguridad que son proporcionados por usuarios finales y los servidores. En este trabajo se expone un modelo de seguridad aplicado a un escenario experimental VoIP sobre el Internet de Próxima Generación (IPv6), que utiliza la seguridad salto a salto y extremo a extremo. El escenario propuesto se encuentra sobre una red local con direcciones IPv6. Utiliza dos servidores Asterisk implementados bajo las mismas características que cumplen la función de SIP Proxy, y se encuentran conectados mediante un enlace troncal SIP – TLS. Se utilizan además dos terminales de usuario (teléfonos IP) provenientes de una marca comercial conocida, registrados cada uno mediante el protocolo SIP-TLS a cada servidor Asterisk. En trabajos anteriores, se han realizado varios estudios sobre el rendimiento del uso de VoIP sobre IPv4 e IPv6 comparando los resultados [1], evaluación de mecanismos de seguridad para mantener la autenticación de usuario, confidencialidad e integridad de la señalización y media de los mensajes VoIP sobre las redes IPv4 [2] y [3]. Este trabajo se esboza en un marco de seguridad, donde se presenta un escenario basado en una red VoIP en IPV6 utilizando TLS y SRTP. TLS es utilizado para la seguridad en el establecimiento de la sesión con mecanismos de autenticación salto a salto y SRTP (Secure Real Time Protocol) para la seguridad del establecimiento del stream de media. Nos enfocaremos en analizar y evaluar la seguridad de los mensajes en este escenario sobre el protocolo de transporte seguro (TLS). Descriptores: sip, tls ,ipv6, srtp. ABSTRACT VoIP deployments today have increased considerably, but not all the scenarios consider appropriate security mechanisms. This last point is very important to consider today, especially the depletion of IPv4 addresses and the deployment of many services IPv6, where will new security threats to try to overshadow the great technology boom VoIP. Although IPv6 was developed to solve many security vulnerabilities are currently present in IPv4, the fact is that still fails to achieve these goals by testing. The SIP protocol, the main actor of VoIP technology requires the implementation of security mechanisms. The current scenarios require high-end user performance and support to adapt to heterogeneous security mechanisms or assume trust relationships. But keep in mind that there are various combinations of security solutions that are provided by end users and servers. This paper presents a security model applied to an experimental scenario VoIP over Next Generation Internet (IPv6), which uses hop by hop security and end to end. The proposed scenario is on a local network with IPv6 addresses. Use two Asterisk servers implemented under the same characteristics that act as SIP Proxy, and are connected via SIP trunk - TLS. They also use two user terminals (IP phones) from a known trade mark registered by each SIP-TLS protocol for each server Asterisk. In previous work, there have been several studies on the performance of VoIP using IPv6 and IPv4 and comparing the results [1], evaluation of security mechanisms to support user authentication, confidentiality and integrity of the signaling and media messages VoIP over IPv4 networks [2] and [3]. This paper outlines a framework of security, which presents a scenario based on a VoIP network in IPv6 using TLS and SRTP. TLS is used for the security session establishment authentication mechanisms hop by hop and SRTP (Secure Real Time Protocol) for the safety of the establishment of media stream. We will focus on analyzing and evaluating the security of the messages in this scenario the secure transport protocol (TLS) . Keywords: sip, tls ,ipv6, srtp.

A Proposal for A High Availability Architecture for VoIP Telephone Systems based on Open Source Software

The inherent needs of organizations to improve and amplify their technological platform entail large expenses with the goal to enhance their performance. Hence, they have to contemplate mechanisms of optimization and the improvement of their operational infrastructure. In this direction arises the need to guarantee the correct operation and non-degradation of the services provided by the platform during the periods with a significant load of work. This type of scenario is perfectly applicable to the field of VoIP technologies, where users generate elevated loads of work on critical points of the infrastructure, during the process of interaction with their peers. In this research work, we propose a solution for high availability, with the goal of maintaining the continuity of the operation of communication environments based on the SIP protocol in high load. We validate our proposal through numerous experiments. Also, we compare our solution with other classical VoIP scenarios and show the advantages of a high availability and fault tolerance architecture for organizations.

Performance evaluation of VoIP technology in an extended service set, in concordance with IEEE 802.11g

In this paper, we evaluate the performance in function of the metrics associated to Quality of Service [QoS] and Quality of user Experience [QoE] in an experimental way in the VoIP service for G.711 and G.729 códecs. This was performed over an extended service set based on Wi-Fi technology in concordance with IEEE 802.11g standard using embedded systems. QoS related metrics are obtained by using the intrusive traffic injection technique. In addition, we assessed the QoE using the MOSc [Mean Opinion Score conversational] analysis. The best results were obtained for G.729, reaching up to 25 simultaneous injections with optimal delay, jitter and packet loss values according to the ITU-T recommendation for VoIP. However, the G.711 codec presented a better throughput. On the other hand, QoE evaluation indicates a slight superiority of G.729 in the MOSc appreciation. Finally, we conclude that packet loss and delay are the most influential metrics in VoIP service degradation.

Analisis Perbandingan Kinerja Protokol Transport Layer Pada Lalu Lintas Data VIOP dan IPTV

AbstractOne of the use of VOIP and IPTV technology that is increasingly used as a communication application on the internet are video call and video streaming, so when the data required a reliable and fast protocol performance in the process of delivery. The problem of the research is analyzing the performance of layer transport protocol in VOIP and IPTV data traffic. The objective of the research is to know the advantages and disadvantages of VOIP and IPTV data traffic and it is useful to contribute to the development of transport layer protocols in VOIP and IPTV data traffic in the future. This research is conducted independently without related to place and time. This reasearch uses SoftPerfect network Protocol Analyzer software. SoftPerfect network Protocol Analyzer is used to compare the performance of transport layer protocols of TCP, UDP, SCTV based on six parameters using different LAN and Wireless cable, test time, and sharing time. The method used in this research is based on the research approach. The results of this study indicate that SCTV protocol is a protocol that has good performance in VOIP and IPTV data traffic with TCP and UDP protocol.Keywords : Analysis, Performance, TCP, UDP, SCTP, VOIP, IPTV