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This discussion paper considers some of the practical and ethical aspects of doing qualitative interviews using synchronous online visual technologies within a shifting research context. It is argued that the immediate access to potential participants and subsequent data collection necessitate adjustment to the ways in which qualitative researchers understand and apply ethics, accountability, and responsibility in their data collection processes. We examine the parallels between interviewing face-to-face and interviewing using technology from a practical and integral perspective. In the online environment researchers require a heightened sensitivity and awareness of their attitudes, knowledge, and skills before, during and after the interview to ensure that the process is safe, rigorous and meaningful for collecting comprehensive qualitative data. To do this, we consider how to plan, conduct and end online interviews using voice over internet protocol.

Voice over internet protocol (VoIP) calls are increasingly transported over computer-based networking due to several factors, such as low call rates. However, point-to-point (P-P) calls, as a division of VoIP, are encountering a capacity utilization issue. The main reason for that is the giant packet header, especially when compared to the runt P-P calls packet payload. Therefore, this research article introduced a method to solve the liability of the giant packet header of the P-P calls. The introduced method is named voice segment compaction (VSC). The VSC method employs the unneeded P-P calls packet header elements to carry the voice packet payload. This, in turn, reduces the size of the voice payload and improves network capacity utilization. The preliminary results demonstrated the importance of the introduced VSC method, while network capacity improved by up to 38.33%.

The growing demand for IP addresses, driven by the proliferation of devices, has depleted the internet protocol (IP) version 6 (IPv6) reserves of some regional internet registries (RIRs). It is imperative to migrate to IPv6, offering an extended addressing space. This transition is no longer a choice but a necessity due to the exhaustion of IP version 4 (IPv4) addresses. The internet engineering task force (IETF) has implemented various transition strategies, such as the use of dual stack, IPv6-in-IPv4 tunnels, and address translation, due to the inconsistency between the two versions of the IP (IPv4 and IPv6). IPv4/IPv6 address translation mechanisms are crucial for the coexistence of networks using both protocols, with scalability playing a central role. Although these mechanisms offer advantages such as optimizing addressing space, their ability to scale effectively must be evaluated, especially in demanding scenarios such as voice over IP (VoIP). This article examines the scalability of two mechanisms, network address translation 64 (NAT64) and stateless IP/internet control message protocol (ICMP) translation (SIIT), in terms of VoIP clients in the graphical network simulator 3 (GNS3) environment. The results indicate that the SIIT mechanism is more performant and scalable than NAT64 in all measured parameters.

The Session Initiation Protocol (SIP) has revolutionized the way of controlling Voice over Internet Protocol (VoIP) based communication sessions over an open channel. The SIP protocol is insecure for being an open text-based protocol inherently. Different solutions have been presented in the last decade to secure the protocol. Recently, Zhang et al. authentication protocol has been proposed with a sound feature that authenticates the users without any password-verifier database using smart card. However, the scheme has a few limitations and can be made more secure and optimized regarding cost of exchanged messages, with a few modifications. Our proposed key-agreement protocol makes a use of two server secrets for robustness and is also capable of authenticating the involved parties in a single round-trip of exchanged messages. The server can now authenticate the user on the request message received, rather than the response received upon sending the challenge message, saving another round-trip of exchanged messages and hence escapes a possible denial of service attack.

In recent years, the application of Mobile Ad-hoc Network (MANET) with Voice over Internet Protocol (VoIP) has been increased.  However, the level of Quality of Service (QoS) for VoIP traffic in MANET, while there is no infrastructure, will reduce when dealing with a large number of calls. In this type of dynamic environment, the developing of a new infrastructure becomes more costly and time-consuming. In this paper, we proposed an efficient method, called the Quality of Service-Nearest Neighbor (QoS-NN), to improve the QoS level for VoIP in order to manage the huge number of calls over MANET network. We utilized the Ad-hoc On-demand Distance Vector (AODV) protocol as the underlying routing protocol to implement our proposed method. We evaluated the proposed QoS-NN method using Network Simulator version 2 (NS2). The performance of the proposed QoS-NN method was compared with Lexicographic order method. The comparison was evaluated in terms of R-factor, end-to-end delay, packet loss ratio, and packet delivery ratio performance metrics. In addition, the proposed method evaluated under different network parameters such as VoIP CODECs, node mobility speed, number of calls and number of nodes. The comparison results indicate that the proposed QoS-NN outperform the Lexicographic order method.

Wireless Mesh Networks (WMNs) have gained prominence in modern communication technology due to their flexibility and ease of deployment, which are advantageous in scenarios like disaster management and rescue operations. However, existing methods for enhancing the performance of WMNs, such as increasing the number of gateways, are costly, introduce interference, and complicate deployment. Moreover, current routing protocols often suffer from suboptimal packet delivery due to inadequate traffic flow management and packet loss. This research addresses these gaps by proposing a novel optimization model that integrates Artificial Bee Colony (ABC) and Particle Swarm Optimization (PSO) techniques to enhance packet delivery ratio in WMNs using Voice over Internet Protocol (VoIP). Unlike traditional approaches that overlook efficient traffic management, our proposed model focuses on optimizing packet transmission by selecting efficient routes and minimizing packet loss. The novelty of this solution lies in its hybrid use of ABC and PSO for dynamic node and route selection, which significantly improves network performance, reduces control overhead, and minimizes packet loss. Experimental results demonstrate that the proposed model outperforms existing protocols, making it a promising approach for enhancing network reliability and efficiency in WMNs.

Voice over Internet Protocol (VoIP) is widely used by companies, schools, universities, and other institutions. However, VoIP faces many issues that slow down its propagation. An important issue is poor utilization of the VoIP service network bandwidth, which results from the large header of the VoIP packet. The objective of this study is to handle this poor utilization of the network bandwidth. Therefore, this study proposes a novel method to address this large header overhead problem. The proposed method is called zero size payload (ZSP), which aims to reemploy and use the header information (fields) of the VoIP packet that is dispensable to the VoIP service, particularly the unicast IP voice calls. In general, these fields are used to carry the VoIP packet payload. Therefore, the size of the payload is reduced to save bandwidth. The performance estimation results of the proposed ZSP method showed a considerable improvement in the bandwidth utilization of the VoIP service. For example, the saved bandwidth in the tested scenario with the G.723.1, G.729, and LPC codecs reached 32%, 28%, and 26% respectively.

Individuals in clinical training programs concerned with critical medical care must learn to manage clinical cases effectively as a member of a team. However, practice on live patients is often unpredictable and frequently repetitive. The widely substituted alternative for real patients—high-fidelity, manikin-based simulators (human patient simulator)—are expensive and require trainees to be in the same place at the same time, whereas online computer-based simulations, or virtual worlds, allow simultaneous participation from different locations. Here we present three virtual world studies for team training and assessment in acute-care medicine: (1) training emergency department (ED) teams to manage individual trauma cases; (2) prehospital and in-hospital disaster preparedness training; (3) training ED and hospital staff to manage mass casualties after chemical, biological, radiological, nuclear, or explosive incidents. The research team created realistic virtual victims of trauma (6 cases), nerve toxin exposure (10 cases), and blast trauma (10 cases); the latter two groups were supported by rules-based, pathophysiologic models of asphyxia and hypovolemia. Evaluation of these virtual world simulation exercises shows that trainees find them to be adequately realistic to “suspend disbelief,” and they quickly learn to use Internet voice communication and user interface to navigate their online character/avatar to work effectively in a critical care team. Our findings demonstrate that these virtual ED environments fulfill their promise of providing repeated practice opportunities in dispersed locations with uncommon, life-threatening trauma cases in a safe, reproducible, flexible setting.

As digitization is integrated into daily life, media are increasingly transferred over the Internet. Voice-over-Internet Protocol (VoIP), the most popular media transfer technology, is attracting many researchers and investments. The application of Artificial Intelligence (AI) technology into the Private Branch Exchange (PBX) has played a pivotal role in enhancing the customer experience and is able to unite employees in any company. One technology application used to optimize customer experience in a call centre is the use of an automatic PBX integrated with a Virtual Assistant (VA), which interacts directly with the PBX through voice and in multiple languages without any keystrokes. The Interactive Voice Response (IVR) module forwards the customer's call to an operator or supports automatic processing. This solution can help businesses to handle thousands of calls per day with optimal performance, thus creating a customer care campaign that quickly reaches many users. A PBX integrated with Vietnamese Virtual Assistants (VVA) on an AI technology platform will also help businesses to cut down on operator costs with automated calls. Through comparison with a traditional PBX, this article analyzes, evaluates and optimizes an automatic PBX system with integrated VVA, thereby offering efficient solutions for interest companies.

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.