VoIP Research Articles

Voice over internet protocol codec performance in interactive streaming environment

Voice over Internet Protocol (VoIP) is an interactive telecommunication technology that suggests shared, cooperative, and quality transactions in the form of communications over the internet. VoIP normally differs from traditional and conventional circuit-based networks in the form of a low rate because all the communication takes place through the internet. The user has to pay only for an internet connection. Real-time voice data transmission is very difficult when compared with ordinary text data transmission. A main challenge in telecom engineering is maintaining quality. The VoIP transmission faces a lot of inconvenience. It suffers from packet loss, low quality, jitter, delay, and on-time delivery. These parameters influence and degrade performance. The codec plays a major role in VoIP transmission. This paper reports on the fundamental functionalities of VoIP, including coder and decoder operations for quality output, broadcast technology, VoIP connection setups, advantages, applications, and emerging trends in VoIP. The performance of various codecs was analyzed to identify the most suitable codec for voice transmission.

Analiza paketske komutacije u VoIP telefoniji na komandnom mestu jedinica taktičkog nivoa

Introduction/purpose: This paper conducts a comprehensive analysis of a potential implementation of Voice over Internet Protocol (VoIP) systems, focusing on network architecture, VoIP phones, and servers. The study explores potential vulnerabilities and proposes solutions. The paper concludes by advocating for a holistic approach to securing VoIP systems, incorporating supplementary services to ensure the confidentiality, integrity, and availability of voice communications in the digital landscape. Methods: Review of the underlying theory, analysis of the end-user needs and potential solutions, practical viability assesment. Results: The theoretical points discussed were proven in practice, using commercially available resources. Communication was established in an expected manner. Conclusions: Implementing solutions similar to the one presented in the paper would be a relatively inexpensive way to make diverse improvements to the operation of tactical level units, both in peacetime and during war.

Dynamic QoS: Automatically Modifying QoS Queue's Maximum Bandwidth Rate-Limit of Network Devices for Network Improvement

The heterogeneous data traffic of today's network is a huge challenge to existing best-effort network technology, particularly in the context of large Ethernet, which handles hundreds to thousands of users. The existing conventional best-effort network technology is no longer efficient to handle the diversity of traffic types in the network and requires network management equipment such as Quality of Service (QOS). Usually, QOS is implemented on the gateway router. However, for better network performance and management, to guarantee high priority for sensitive traffic like video conferencing, Voice over Internet Protocol (VoIP), and streaming media within an internal network, it is nice to have QoS implemented on each router in the LAN network, starting from the access router to the gateway router. This paper is to demonstrate the effectiveness of the proposed dynamic QoS that has been developed and deployed in the LAN, purposely to provide adequate bandwidth for sensitive traffic when the network utilization is high and congested, by automatically modifying the QoS Queue's Maximum Bandwidth Rate-Limit of the best-effort traffic queue of the related router. The performance of the proposed developed dynamic QoS was evaluated via a comparison study before and after the dynamic QoS was presented in the network simulation environment that was built using Mininet. Results from the testing show that the developed dynamic QoS can improve the network's performance by automatically giving the appropriate bandwidth for sensitive traffic on the fly while needed/on demand.

Dynamic QoS: Automatically Modifying QoS Queue's Maximum Bandwidth Rate-Limit of Network Devices for Network Improvement

The heterogeneous data traffic of today's network is a huge challenge to existing best-effort network technology, particularly in the context of large Ethernet, which handles hundreds to thousands of users. The existing conventional best-effort network technology is no longer efficient to handle the diversity of traffic types in the network and requires network management equipment such as Quality of Service (QOS). Usually, QOS is implemented on the gateway router. However, for better network performance and management, to guarantee high priority for sensitive traffic like video conferencing, Voice over Internet Protocol (VoIP), and streaming media within an internal network, it is nice to have QoS implemented on each router in the LAN network, starting from the access router to the gateway router. This paper is to demonstrate the effectiveness of the proposed dynamic QoS that has been developed and deployed in the LAN, purposely to provide adequate bandwidth for sensitive traffic when the network utilization is high and congested, by automatically modifying the QoS Queue's Maximum Bandwidth Rate-Limit of the best-effort traffic queue of the related router. The performance of the proposed developed dynamic QoS was evaluated via a comparison study before and after the dynamic QoS was presented in the network simulation environment that was built using Mininet. Results from the testing show that the developed dynamic QoS can improve the network's performance by automatically giving the appropriate bandwidth for sensitive traffic on the fly while needed/on demand.

Security Enhancement of SIP Protocol in VoIP Communication

In Voice over IP (VoIP) systems, calls are started, maintained, and ended using the Session Initiation Protocol (SIP). One of the SIP flaws is the lack of enough data to validate users, and its contents can be changed to fake the Caller Identification (CID). Fake CID can be used by hackers to trick receivers and obtain valuable materials. Existing solutions have a number of faults, including the use of more network resources, the use of insufficient data to identify fake CID, and high call setup delay for caller and callee during the validation process. This work presents an algorithm to enhance SIP protocol security to mitigate the flaws of the previous solutions that are used to address CID spoofing. By using the Media Access Control (MAC) address, the algorithm can validate the CID and warn the callee for the fake CID. The algorithm was developed and tested by using Mininet, Python based open source OpenFlow (POX) controller, SIPp simulator, Linphone softphone, and freePBX. All faked CID were detected and receivers were alerted on the linphone screen. Session setup delay obtained are within the 15.1 ms to 27.3 ms, which are within the acceptable Quality of Service (QoS) ranges.

Tel-MPLS: a new method for maximizing the utilization of IP telephony over MPLS networks

Currently, the multiprotocol label switching (MPLS) standard is extremely prevalent. By exploiting the features provided by MPLS technology, a range of services, including IP telephony, have enhanced their overall performance. However, due to the size of the packet header, the IP telephony service consumes a significant portion of the MPLS network's available bandwidth. For instance, in IP telephony over MPLS networks, the packet header might account for as much as 80% of lost time and bandwidth. Designers working on IP telephony are making substantial efforts to address this issue. This study contributes to current efforts by proposing a novel approach called Tel-MPLS, which involves IP telephony over MPLS. TelMPLS approach uses the superfluous fields in the IP telephony packet's header to retain the packet data, therefore lowering or zeroing the IP telephony packet's payload. Tel-MPLS is an approach that significantly reduces the bandwidth of IP telephony MPLS networks. According to the findings, the Tel-MPLS approach is capable of reducing the amount of bandwidth that is lost by 12% when using the G.729 codec.

Traffic prediction in SDN for explainable QoS using deep learning approach

The radical increase of multimedia applications such as voice over Internet protocol (VOIP), image processing, and video-based applications require better quality of service (QoS). Therefore, traffic Predicting and explaining the prediction models is essential. However, elephant flows from those applications still needs to be improved to satisfy Internet users. Elephant flows lead to network congestion, resulting in packet loss, delay and inadequate QoS delivery. Recently, deep learning models become a good alternative for real-time traffic management. This research aims to design a traffic predicting model that can identify elephant flows to prevent network congestion in advance. Thus, we are motivated to develop elephant flow prediction models and explain those models explicitly for network administrators’ use in the SDN network. H2O, Deep Autoencoder, and autoML predicting algorithms, including XGBoost, GBM and GDF, were employed to develop the proposed model. The performance of Elephant flow prediction models scored 99.97%, 99.99%, and 100% in validation accuracy of under construction error of 0.0003952, 0.001697, and 0.00000408 using XGBoost, GBM, and GDF algorithms respectively. The models were also explicitly explained using Explainable Artificial Intelligence. Accordingly, packet size and byte size attributes need much attention to detect elephant flows.

METHODOLOGY FOR PLANNING THE LOCATION OF SWITCHING NODES OF IP-TELEPHONY NETWORKS

The relevance of the topic is due to the high pace of development and implementation of new systems and processes in information structures and the lack of a methodology for determining the effectiveness of using new infocommunication technologies in them. Currently, there are no unified scientifically based methods for assessing the feasibility and effectiveness of using IP telephony technology. This is due to insufficient experience in the operation of telecommunication systems based on IP telephony technology and their significant difference from traditional telephone systems. It is indicated that for optimal planning of the location of switching nodes of IP-telephony networks, a methodology has been developed, consisting of the development of a mathematical basis for constructing the projected IP-telephony network, algorithmic implementation and estimation of the complexity of the implementation of this algorithm, which can be used in the design of real IP-telephony networks. The algorithmic implementation of the developed technique uses three arrays from among the nodes of the network, i.e. the first array contains labels with two values, the second array contains the distances - the current shortest distances from the specified vertex to the corresponding vertex, and the third array contains the vertex numbers. In turn, the algorithmic implementation depends on how the source vertex is found, how the set of unmarked vertices is stored, and how labels are updated. Keywords: Telecommunications network, IP-telephony technology, packet switching of flows, planning of the location of switching nodes, algorithmic implementation of the planning methodology, subscriber network, parameters of information systems, network parameters.

Comparative performance analysis of different queuing techniques for campus area network

<span>A quality network infrastructure is a user demand for the increment of traffic nowadays. Due to data traffic congestion, a network’s service quality cannot fulfill the user demand, especially for streaming-type traffic. For this, an appropriate queuing method can be a convenient solution for network congestion at the time of picking traffic load. In this paper, a simulation was conducted using the OPNET application to examine the feasibility of three types of queuing methods, namely first in first out (FIFO), priority queuing (PQ), and weighted fair queuing (WFQ). During the simulation, several important parameters such as delay, throughput, and response time are considered as performance metrics and compared among four types of applications as file transfer protocol (FTP), hypertext transfer protocol (HTTP), voice over internet protocol (VoIP), and video conferencing. According to the result, it is observed that WFQ provides comparatively improved results than PQ and FIFO.</span>

Perancangan Infrastruktur VoIP Menggunakan Trixbox Open Source dengan Lapisan Keamanan VPN Antar Klien

PT Batam Bintan Telekomunikasi has several employees scattered across various rooms, and effective communication among them is crucial. Previously, the company relied on a Private Automatic Branch Exchange (PABX) phone system, which proved suboptimal as each room had only one PABX line while accommodating eight employees. Consequently, there was a need to optimize the communication system for PT Batam Bintan Telecommunication employees. The communication system was enhanced by implementing Voice over Internet Protocol (VoIP) as a replacement for the PABX system. VoIP was deployed using a Trixbox server, SIP phones, Zoiper, and MicroSIP applications for making calls on client computers, with VPN (Virtual Private Network) ensuring security between clients. Each employee was provided with a VoIP account registered on the Trixbox server, complete with a telephone number and password. Based on the research findings, the implementation of VoIP on computers within PT Batam Bintan Telekomunikasi's environment, connected through the Trixbox server, significantly facilitated communication among employees, ultimately supporting their performance. This VoIP solution is expected to replace the previous communication system utilized at PT Batam Bintan Telekomunikasi, thus leveraging internet resources to enhance communication efficiency.

The Development of a Secure Internet Protocol (IP) Network Based on Asterisk Private Branch Exchange (PBX)

The research problem described in this article is related to the security of an IP network that is set up between two cities using hosting. The network is used for transmitting telephone traffic between servers located in Germany and the Netherlands. The concern is that with the increasing adoption of IP telephony worldwide, the network might be vulnerable to hacking and unauthorized access, posing a threat to the privacy and security of the transmitted information. This article proposes a solution to address the security concerns of the IP network. After conducting an experiment and establishing a connection between the two servers using the WireShark sniffer, a dump of real traffic between the servers was obtained. Upon analysis, a vulnerability in the network was identified, which could potentially be exploited by malicious actors. To enhance the security of the network, this article suggests the implementation of the Transport Layer Security (TLS) protocol. TLS is a cryptographic protocol that provides secure communication over a computer network, ensuring data confidentiality and integrity during transmission. Integrating TLS into the network infrastructure, will protect the telephone traffic and prevent unauthorized access and eavesdropping.

Peer-to-Peer File Sharing WebApp

Peer-to-peer (P2P) networking has emerged as a promising technology that enables distributed systems to operate in a decentralized manner. P2P networks are based on a model where each node in the network can act as both a client and a server, thereby enabling data and resource sharing without relying on centralized servers. The P2P model has gained considerable attention in recent years due to its potential to provide a scalable, fault-tolerant, and resilient architecture for various applications such as file sharing, content distribution, and social networks.In recent years, researchers have also proposed hybrid architectures that combine the benefits of both structured and unstructured P2P networks. For example, the Distributed Hash Table (DHT) is a popular hybrid architecture that provides efficient lookup and search algorithms while maintaining the flexibility and adaptability of the unstructured network.To demonstrate the feasibility of P2P systems, several prototypes have been developed, such as the BitTorrent file-sharing protocol and the Skype voice-over-IP (VoIP) service. These prototypes have demonstrated the potential of P2P systems for large-scale applications and have paved the way for the development of new P2P-based systems.

Packet Loss Compensation for VoIP through Bone‐Conducted Speech Using Modified Linear Prediction

In this paper, we compare air‐conducted (AC) speech with bone‐conducted (BC) speech for the purpose of utilizing them in packet loss concealment (PLC) for speech quality of voice over internet protocol (VoIP). Instead of the autocorrelation method of linear prediction (LP), which was utilized in the conventional PLC techniques, we employ the modified covariance (MC) method. The MC method provides accurate LP estimation from short input data samples and avoids the numerical problem the autocorrelation method suffers from. The lost frame is compensated from both forward and backward directions in which linear gain and weighting are applied. When BC speech is used as the input speech data for PLC in the case where the speech sender is in noisy environments, BC speech behaves more accurately than the corresponding AC speech, resulting in an excellent performance of the LP‐based PLC technique. This unveils a useful use of BC speech in speech information systems. Experiments show that in severe noise environments of AC speech‐to‐noise ratio being less than 10 dB, BC speech is superior to AC speech for PLC. It is also shown that the transmitted BC speech is more accurately reconstructed for PLC than transmitted AC speech is done. © 2023 The Authors. IEEJ Transactions on Electrical and Electronic Engineering published by Institute of Electrical Engineer of Japan and Wiley Periodicals LLC.

Performance Evaluation of VoIP Analysis and Simulation

The use of technology has impacted on communication in so many ways. The advent of voice over internet protocols (VoIP) has made the world a global village where one can reach out to any part of the universe. But a challenge exists as to how to make communication and data transmission faster, the volume of traffic, bandwidth and latency in networks, that has to be transmitted between the sender and the receiver. The overall customer experience can be improved by the use of technology, which also makes it simpler to collect client information. Data packets are addressed and routed by the Internet Protocol (IP). This research aimed at deploying jitter, throughput, network traffic delay and bandwidth (JiTTraB) as a performance metrics to measure voice over internet protocols (VoIP) to measure the Quality of Service (QoS) of networks. This method prioritizes network traffic going through a router and providing acceptable service to most users in a quest to address VoIP concerns. In comparison to an existing university campus network (UCN), JiTTraB has a higher throughput, more data transmission at a given time with a minimal delay of about 0.001 seconds. However, UCN queues data with a high rate of delay before transmission. Data seem to crash considerably on the UCN due to network overload above its throughput margin and growth capacity. Thus, the proposed network design based on JiTTraB outperforms UCN in terms of transmission rate, throughput, bandwidth, delay and jitter. In addition, the JiTTraB communication network enhances QoS by performing better in close-quarters conversations than in far-quarters conversations. With an expansion reserve of 30%, the JiTTraB can handle up to 350 conversations securely while maintaining VoIP QoS standards without degrading the functionality of other network services or applications.

A Link-Layer Virtual Networking Solution for Cloud-Native Network Function Virtualisation Ecosystems: L2S-M

Microservices have become promising candidates for the deployment of network and vertical functions in the fifth generation of mobile networks. However, microservice platforms like Kubernetes use a flat networking approach towards the connectivity of virtualised workloads, which prevents the deployment of network functions on isolated network segments (for example, the components of an IP Telephony system or a content distribution network). This paper presents L2S-M, a solution that enables the connectivity of Kubernetes microservices over isolated link-layer virtual networks, regardless of the compute nodes where workloads are actually deployed. L2S-M uses software-defined networking (SDN) to fulfil this purpose. Furthermore, the L2S-M design is flexible to support the connectivity of Kubernetes workloads across different Kubernetes clusters. We validate the functional behaviour of our solution in a moderately complex Smart Campus scenario, where L2S-M is used to deploy a content distribution network, showing its potential for the deployment of network services in distributed and heterogeneous environments.

Involved Security Solution in Voice over IP Networks

In this paper, the work involves setting up a secure VoIP solution. At first, we presented some attacks scenarios affecting confidentiality, authenticity and integrity in the VoIP network. In the second part, we focused on VOIP security themes, so our work has focused on areas where enhancing VoIP security is essential. The convergence towards everything of IP is now a reality. The efforts made by infrastructure providers, whether fixed, mobile, and cabled or wireless. This is also evident in operators who now only design the services they offer in the context of everything IP. But with the increasing use of Voice over IP (VoIP) technology, services and protocols are becoming increasingly vulnerable. In this context work several organizations that promote the research and development of network security, warn users of the attacks risks that may affect the communications confidentiality. This work proposed to explore the security vulnerabilities associated with VoIP networks and to present some solutions to avoid and minimize these threats.

NISQE: Non-Intrusive Speech Quality Evaluator Based on Natural Statistics of Mean Subtracted Contrast Normalized Coefficients of Spectrogram

With the evolution in technology, communication based on the voice has gained importance in applications such as online conferencing, online meetings, voice-over internet protocol (VoIP), etc. Limiting factors such as environmental noise, encoding and decoding of the speech signal, and limitations of technology may degrade the quality of the speech signal. Therefore, there is a requirement for continuous quality assessment of the speech signal. Speech quality assessment (SQA) enables the system to automatically tune network parameters to improve speech quality. Furthermore, there are many speech transmitters and receivers that are used for voice processing including mobile devices and high-performance computers that can benefit from SQA. SQA plays a significant role in the evaluation of speech-processing systems. Non-intrusive speech quality assessment (NI-SQA) is a challenging task due to the unavailability of pristine speech signals in real-world scenarios. The success of NI-SQA techniques highly relies on the features used to assess speech quality. Various NI-SQA methods are available that extract features from speech signals in different domains, but they do not take into account the natural structure of the speech signals for assessment of speech quality. This work proposes a method for NI-SQA based on the natural structure of the speech signals that are approximated using the natural spectrogram statistical (NSS) properties derived from the speech signal spectrogram. The pristine version of the speech signal follows a structured natural pattern that is disrupted when distortion is introduced in the speech signal. The deviation of NSS properties between the pristine and distorted speech signals is utilized to predict speech quality. The proposed methodology shows better performance in comparison to state-of-the-art NI-SQA methods on the Centre for Speech Technology Voice Cloning Toolkit corpus (VCTK-Corpus) with a Spearman’s rank-ordered correlation constant (SRC) of 0.902, Pearson correlation constant (PCC) of 0.960, and root mean squared error (RMSE) of 0.206. Conversely, on the NOIZEUS-960 database, the proposed methodology shows an SRC of 0.958, PCC of 0.960, and RMSE of 0.114.

Detecting SPIT Attacks in VoIP Networks Using Convolutional Autoencoders: A Deep Learning Approach

Voice over Internet Protocol (VoIP) is a technology that enables voice communication to be transmitted over the Internet, transforming communication in both personal and business contexts by offering several benefits such as cost savings and integration with other communication systems. However, VoIP attacks are a growing concern for organizations that rely on this technology for communication. Spam over Internet Telephony (SPIT) is a type of VoIP attack that involves unwanted calls or messages, which can be both annoying and pose security risks to users. Detecting SPIT can be challenging since it is often delivered from anonymous VoIP accounts or spoofed phone numbers. This paper suggests an anomaly detection model that utilizes a deep convolutional autoencoder to identify SPIT attacks. The model is trained on a dataset of normal traffic and then encodes new traffic into a lower-dimensional latent representation. If the network traffic varies significantly from the encoded normal traffic, the model flags it as anomalous. Additionally, the model was tested on two datasets and achieved F1 scores of 99.32% and 99.56%. Furthermore, the proposed model was compared to several traditional anomaly detection approaches and it outperformed them on both datasets.

Real Impact of the Blockchain in Securing a ToIP Network

Telephony over IP (ToIP) is a cost-saving communication technology based on voice over IP (VoIP) that enables enterprises to reduce communication fees. However, ToIP faces many security threats due to its IP-based nature. This work aims to improve ToIP security using cryptography and blockchain technology. The authors propose a secure approach to user registration, authentication, communication session establishment, and communication data storage. The proposed solution leverages blockchain technology to ensure the integrity, confidentiality, and availability of communication data. By implementing this solution, the researchers aim to enhance the security of ToIP networks and protect them from cyber threats. This approach provides a secure and reliable way to support ToIP services while preserving confidentiality and privacy.

Возможности перераспределения функций городской поликлиники при открытии подразделения – Дистанционный консультативный центр

Goal. To evaluate the interim results of a pilot project on the organization of an independent unit of a city clinic that provides remote support to citizens – the Distant Consultation Center of the Government Autonomous Organization Healthcare “City Clinic No. 8” (hereinafter referred to as the DCC). M e t h o d s o f w o r k . To assess the effectiveness of the pilot project, a number of indicators of the work of the medical organization were analyzed before the start of the project and 12 months after the start of its implementation. The analysis was carried out in two way of work of the Distant Consultation Center: 1. the process of providing medical care to patients with acute respiratory viral infections / Influenza / COVID‑19; 2. the process of providing medicines to the Subsidized category of citizens. The volumes of work that can be transferred to the Distant Consultation Center from the department of primary care physicians care and from the emergency medical care are analyzed. The source of the data was the medical information system of the medical organization “URMO TO” (modules Registry, Control of realization, doctor’s ARM, Disability sheets, Additional processing). The period of work of the medical organization from 01.01.2022 to 31.12.2022 is analyzed. Statistical processing of the results was carried out using the Statistica 6.0 application software package. Distance consultation was carried out via IP telephony, as well as using specialized Telemedicine software. R e s u l t s o f t h e w o r k . Before the organization of the DCC, the median indicator of the actual workload per hour of the covid team was 2,5 [2,3–2,7] patients, p<0,07. While the doctor of DCC carried out an average of 56,9 [44–68] remote consultations, which is an average of 8,8 patients per hour, p< 0,07. The median value of patients for prescriptions for Subsidized medicines before the start of the project was 4 [3–6] patients per day (p<0,07), on average 22.2% time of the initial consultation falls on patients requiring a Subsidized medicine. The median value of applications for the purpose of providing Subsidized categories of citizens with medicines when discharged by a medical assistant at the DCC was 1 [0–3] (p<0,08). Thus, this section began to occupy 0,06% in the structure of reasons for the initial appointment to the primary care physician. C o n c l u s i o n s . The organization of the work of the DCC allowed us to develop approaches, approve and adapt them to the processes of filling out the schedule of primary care doctors in order to increase accessibility to the primary consultations of the primary care physician’s department. Scaling up the organizational model and processes of the Distance consultation center of the Municipal Polyclinic No. 8 can serve as an effective tool in achieving the goals announced within the framework of the pilot project of the Ministry of Health of the Russian Federation “Incident 38” launched in 2022. T h e s c o p e o f t h e r e s u l t s . The presented experience shows the possibility of managing patients with respiratory viral diseases in outpatient polyclinic conditions on an ongoing basis, as well as the telemedicine format of care can be extended and adapted to non-infectious nosologies.