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VoIP University is a solution to enable the development of communication projects based on IP protocol. This paper presents a low-cost deployment called VoIP UEMA, the first project of VoIP University solution. The technical feasibility of this project was checked as proof of concept. The concept was proven showing it is possible to create user extensions with some information such as name, ID, job role, course, center, and campus getting information of all registered users in the academic and administration system server to make voice and video calls through Android application and browser web phone. A mean opinion score test that measures subjective voice quality for a call was done with 35 people calling each other using two VoIP UEMA softphones, VoIP UEMA application, and VoIP UEMA web phone. The result achieved a weighted average of 3.5, indicating good-quality calls. The low-cost aspect was justified by a financial analysis showing the annual phone bill expenses dropped, representing savings of over 97 %. Keywords: VoIP University. VoIP UEMA. SIP. Asterisk. Web Service. A Universidade VoIP e uma solucao que permite o desenvolvimento de projetos de comunicacao baseados no protocolo IP. Este artigo apresenta uma implantacao de baixo custo chamada VoIP UEMA, o primeiro projeto da solucao Universidade VoIP. A viabilidade tecnica desse projeto foi verificada como prova de conceito. O conceito foi comprovado, mostrando que e possivel criar extensoes de usuario com algumas informacoes, como nome, ID, cargo, curso, centro e campus, obtendo informacoes de todos os usuarios registrados no servidor do sistema academico e administrativo para fazer chamadas de voz e video por meio do aplicativo Android e do navegador do telefone da Web. Um teste de pontuacao media de opiniao que mede a qualidade subjetiva da voz em uma chamada foi feito com 35 pessoas fazendo chamadas entre si usando dois softphones VoIP UEMA, o aplicativo VoIP UEMA e o telefone web VoIP UEMA. O resultado alcancou uma media ponderada de 3,5, indicando chamadas de boa qualidade. O aspecto de baixo custo foi justificado por uma analise financeira que mostrou que as despesas anuais com a conta telefonica cairam, representando uma economia de mais de 97%. Palavras-chave: Universidade VoIP. VoIP UEMA. SIP. Asterisk. Servico da Web. VoIP University es una solucion que permite el desarrollo de proyectos de comunicaciones basados en el protocolo IP. Este articulo presenta una implementacion de bajo costo denominada VoIP UEMA, el primer proyecto de solucion VoIP Universitaria. La viabilidad tecnica de este proyecto se comprobo como prueba de concepto. El concepto fue probado mostrando que es posible crear extensiones de usuario con cierta informacion como nombre, ID, puesto de trabajo, curso, centro y campus obteniendo informacion de todos los usuarios registrados en el servidor del sistema academico y administrativo para hacer llamadas de voz y video a traves de la aplicacion Android y el navegador del telefono web. Se realizo una prueba de puntuacion de opinion media que mide la calidad de voz subjetiva para una llamada con 35 personas que se llamaban entre si utilizando dos softphones VoIP UEMA, una aplicacion VoIP UEMA y un telefono web VoIP UEMA. El resultado alcanzo un promedio ponderado de 3,5, lo que indica llamadas de buena calidad. El aspecto de bajo costo fue justificado por un analisis financiero que mostro que los gastos anuales en la factura telefonica cayeron, lo que representa un ahorro de mas del 97%. Palabras clave: Universidad VoIP. VoIP UEMA. SORBO. Asterisco. Servicio Web.

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.

Voice over Internet Protocol (VoIP) has become increasingly popular among individuals and business organisations, with millions of users communicating using VoIP applications (apps) on their smart mobile devices. Since Android is one of the most popular mobile platforms, this research focuses on Android devices. In this paper we survey the research that examines the security and privacy of mVoIP published in English from January 2009 to January 2014. We also examine the ten most popular free mVoIP apps for Android devices, and analyse the communications to determine whether the voice and text communications using these mVoIP apps are encrypted. The results indicate that most of the apps encrypt text communications, but voice communications may not have been encrypted in Fring, ICQ, Tango, Viber, Vonage, WeChat and Yahoo. The findings described in this paper contribute to an in-depth understanding of the potential privacy risks inherent in the communications using these apps, a previously understudied app category. Six potential research topics are also outlined.

Voice over Internet Protocol (VoIP) an emerging new technologies is based on realtime communication of data by sending and receiving digital signals using Internet Protocols (IP). VoIP has become popular in business organization and scientific communities. Voice coder (Codec) named as coder and decoder, play an important role in packet switch network for transfer of audio and video data. A signalling protocol known as Session Initiation Protocol (SIP) is used for data communication between nodes in VoIP network. In this paper we survey on security related issues in VoIP communication system and covered several security attacks. We proceed with survey on all the problems that have big effects on VoIP security and challenges such as data manipulation, substitution attack, power backups, jitter, latency and delay. Finally, we conclude the paper with further research on security of VoIP communication system.

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.

The rapid advance and popularization of VoIP (Voice over IP) has also brought security issues. VoIP-based secure voice communication has two sides: first, for legitimate users, the secret voice can be embedded in the carrier and transmitted safely in the channel to prevent privacy leakage and ensure data security; second, for illegal users, the use of VoIP Voice communication hides and transmits illegal information, leading to security incidents. Therefore, in recent years, steganography and steganography analysis based on VoIP have gradually become research hotspots in the field of information security. Steganography and steganalysis based on VoIP can be divided into two categories, depending on where the secret information is embedded: steganography and steganalysis based on voice payload or protocol. The former mainly regards voice payload as the carrier, and steganography or steganalysis is performed with respect to the payload. It can be subdivided into steganography and steganalysis based on FBC (fixed codebook), LPC (linear prediction coefficient), and ACB (adaptive codebook). The latter uses various protocols as the carrier and performs steganography or steganalysis with respect to some fields of the protocol header and the timing of the voice packet. It can be divided into steganography and steganalysis based on the network layer, the transport layer, and the application layer. Recent research results of steganography and steganalysis based on protocol and voice payload are classified in this paper, and the paper also summarizes their characteristics, advantages, and disadvantages. The development direction of future research is analyzed. Therefore, this research can provide good help and guidance for researchers in related fields.

Voice over Internet Protocol (VoIP) is gradually dominating the telecommunication sector because it is free or inexpensive. However, one of the key problems of VoIP growth is inefficient bandwidth utilization. Several methods have been proposed to improve the VoIP bandwidth utilization, and they include VoIP packet aggregation and header compression. In this study, we investigate a new dimension to improve VoIP bandwidth utilization, that is, VoIP packet payload compression. The main idea of the proposed method, which is called the voice frame pruning (VFP) method, is to prune the leading/trailing zeros/ones of the VoIP packet payload on the basis of a certain mechanism. The VoIP packet payload is pruned at the sender side’s wide area network (WAN) gateway and restored to its original form at the receiver side’s WAN gateway. The implementation results of the proposed VFP method indicate good bandwidth savings based on the VoIP codec used. For example, the bandwidth savings of the LPC, G.723.1, and G.729 improved by up to approximately 5%, 8%, and 3.5%, respectively, thereby improving the VoIP bandwidth utilization and the capacity of VoIP calls.

The International Telecommunication Union (ITU) predicts a substantial and swift increase in global mobile data traffic. The predictions suggest that this growth will vary from 390 EB (exabytes) to 5,016 EB (exabytes) from 2024 to 2030, accordingly. This work presents a new maximum capacity model (MCM) to improve the dynamic resource allocation, robust encryption, and Quality of Service (QoS) in 5G networks which helps to meet the growing need for high-bandwidth applications such as Voice over Internet Protocol (VoIP) and video streaming. Our proposed MCM model enhances data transmission by employing dynamic resource allocation, prioritised traffic management, and robust end-to-end encryption techniques, thereby guaranteeing efficient and safe data delivery. The encryption procedure is applied to the header cypher, while the output parameters of the payload are altered. This indicates that only the sender and recipient will possess exclusive knowledge of the final outcome. In result, the comparative analyses clearly show that the MCM model outperforms over conventional models in terms of QoS packet planner, QoS packet scheduler, standard packet selection, traffic management, maximum data rate, and bandwidth utilisation.

The inefficient use of the IP network bandwidth is a fundamental issue that restricts the exponential spreading of Voice over IP (VoIP). The primary reason for this is the big header size of the VoIP packet. In this paper, we propose a method, called Short Voice Frame (SVF), that addresses the big header size of the VoIP packet. The main idea of the SVF method is to make effective use of the VoIP packet header fields that are unneeded to the VoIP technology. In particular, these fields will be used for temporarily buffering the voice frame (VoIP packet payload) data. This will make the VoIP packet payload short or even zero in some cases. The performance evaluation of the proposed SVF method showed that the use of the IP network bandwidth has improved by up to 28.3% when using the G.723.1 codec.

The use of Voice over Internet Protocol (VoIP) innovation is rising due to its various merits. Nevertheless, the ineffective use of bandwidth is a key dilemma that restricts the fast-rising use of VoIP innovation. The main factor behind this ineffective use of the bandwidth is the sizable VoIP packet preamble. This research creates a technique to address this dilemma of packet preamble. The created technique is known as payload elimination (PldE). The fundamental concept of the PldE technique is to exploit the information (elements) of the VoIP packet preamble that is superfluous for point-to-point calls. In general, these elements are utilized to transport the payload of VoIP packets. Consequently, the payload size of VoIP packet will be lowered or removed, preserving the available bandwidth. The performance test of the PldE technique indicated an improvement of up to 41.6% in the exploitation of IP network bandwidth. So, the PldE technique is showing signs that it could help solve the problem of the IP network's inefficient use of bandwidth.