Voice Over IP Services Research Articles

Towards a Smart Environment: Optimization of WLAN Technologies to Enable Concurrent Smart Services

In this research paper, the spatial distributions of five different services-Voice over Internet Protocol (VoIP), Video Conferencing (VC), Hypertext Transfer Protocol (HTTP), and Electronic Mail-are investigated using three different approaches: circular, random, and uniform approaches. The amount of each service varies from one to another. In certain distinct settings, which are collectively referred to as mixed applications, a variety of services are activated and configured at predetermined percentages. These services run simultaneously. Furthermore, this paper has established a new algorithm to assess both the real-time and best-effort services of the various IEEE 802.11 technologies, describing the best networking architecture as either a Basic Service Set (BSS), an Extended Service Set (ESS), or an Independent Basic Service Set (IBSS). Due to this fact, the purpose of our research is to provide the user or client with an analysis that suggests a suitable technology and network configuration without wasting resources on unnecessary technologies or requiring a complete re-setup. In this context, this paper presents a network prioritization framework for enabling smart environments to determine an appropriate WLAN standard or a combination of standards that best supports a specific set of smart network applications in a specified environment. A network QoS modeling technique for smart services has been derived for assessing best-effort HTTP and FTP, and the real-time performance of VoIP and VC services enabled via IEEE 802.11 protocols in order to discover more optimal network architecture. A number of IEEE 802.11 technologies have been ranked by using the proposed network optimization technique with separate case studies for the circular, random, and uniform geographical distributions of smart services. The performance of the proposed framework is validated using a realistic smart environment simulation setting, considering both real-time and best-effort services as case studies with a range of metrics related to smart environments.

CO-CAC: A new approach to Call Admission Control for VoIP in 5G/WiFi UAV-based relay networks

Voice over IP (VoIP) requires a Call Admission Control (CAC) mechanism in WiFi networks to preserve VoIP packet flows from excessive network delay or packet loss. Ideally, this mechanism should be integrated with the operational scenario, guarantee the quality of service of active calls, and maximize the number of concurrent calls. This paper presents a novel CAC scheme for VoIP in the context of a WiFi access network deployed with Unmanned Aerial Vehicles (UAVs) that relay to a backhaul 5G network. Our system, named Codec-Optimization CAC (CO-CAC), is integrated into each drone. It intercepts VoIP call control messages and decides on the admission of every new call based on a prediction of the WiFi network’s congestion level and the minimum quality of service desired for VoIP calls. To maximize the number of concurrent calls, CO-CAC proactively optimizes the codec settings of active calls by exchanging signaling with VoIP users.We have simulated CO-CAC in a 50m × 50m scenario with four UAVs providing VoIP service to up to 200 ground users with IEEE 802.11ac WiFi terminals. Our results show that without CAC, the number of calls that did not meet a minimum quality level during the simulation was 10% and 90%, for 50 and 200 users, respectively. However, when CO-CAC was in place, all calls achieved minimum quality for up to 90 users without rejecting any call. For 200 users, only 25% of call attempts were rejected by the admission control scheme. These results were narrowly worse when the ground users moved randomly in the scenario.

Automated Speech Audiometry for Integrated Voice Over Internet Protocol Communication Services.

Problems in speech recognition are often apparent in telecommunication situations. For ecologically valid assessments of such conditions, it is important to quantify the impact of real environments including acoustic conditions at a far-end communication device and all paths of transmission degradation. This study presents an automated matrix sentence test procedure based on automatic speech recognition (ASR) integrated in a Voice over Internet Protocol (VoIP) infrastructure and compares the individual effects of transmission degradations with results from laboratory measurements. Speech recognition thresholds (SRTs) were measured in 16 normal-hearing subjects in four test conditions: (a) a laboratory condition guided by a human experimenter, (b) a laboratory condition with reduced bandwidth and (c) additionally reduced headset quality to simulate typical communication systems, and (d) an automated, ASR-controlled adaptive test procedure over a real VoIP infrastructure. Errors of the ASR system were analyzed to show possible effects on measurement outcome Results: Measured SRTs showed a highly significant correlation (r = .93) between the fully automatic and "laboratory" conditions, with a constant bias of about 1 dB indicating a linear shift of the data without affecting the distribution around the mean. The individual impact of the different system degradations on SRTs could be quantified Conclusions: This study provides a proof of concept for automated ASR-based SRT measurements over VoIP systems for speech audiometric testing in real communication systems, as it produced results comparable to traditional laboratory settings for this group of 16 normal-hearing subjects. This makes VoIP services a promising candidate for speech audiometric testing in real communication systems.

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.

Enhanced Network Security for Improved Trustworthiness of VoIP Applications via Cuckoo Search and Machine Learning

Objectives: To offer network design for secure Voice over Internet Protocol (VoIP) services with improved Quality of Services (QoS) parameters. Methods: The network area is created with required number of nodes. INTRA-SR process is employed for tracing of the route from the source node and INTER-SR is involved to reach a destination node. Cuckoo Search (CS) based optimization followed for the broadcasted voice packets. The Machine learning classifiers as SVM and ANN applied to decrease the instances of loss of voice packets. Simulation work is performed by using MATLAB 2018 and results obtained plotted in graphs using MS-Excel. Findings: The proposed design evaluated by incorporating CS algorithm to minimize the packet drops. SVM and ANN hybrid used to locate secure routing path. The QoS for throughput, latency and jitter are observed. The results exhibited higher average throughput of 98.8% irrespective of the attack instances. Lower average latency and jitter of 1.9s and 2.51ms are also exhibited by the proposed work. Similarly, latency work employing Ant Colony Optimization (ACO) with multiplex and multicasting (MM) is 2.66s that get increased to 3.16s due to attack. Novelty/Applications: The proposed algorithm significantly enhanced by deployment of highly protected network design with QoS and security. Application of ANN and SVM shown the improvement in performance for VoIP services. In addition to the regression analysis validates the results by applying other optimization algorithms. Keywords: Voice over Internet Protocol; Cuckoo Search; Support Vector Machine; Artificial Neural Network; Machine Learning

Back to basics: fundamental principles of system dynamics and queueing theory

Back to basics: fundamental principles of system dynamics and queueing theory

Implementasi dan Optimalisasi Voice Order Internet Protocol (VOIP) Menggunakan NDLC Untuk Komunikasi Antar Lantai

VoIP as an internet service is commonly called ip telephone. Internet infrastructure is needed in order to use or provide VoIP services. Some offices use local communication to communicate within the office environment itself, but cannot connect to other offices because it is only a local phone. This research aims to implement the Voice Over Internet Protocol (VoIP) by utilizing wireles local area network (WLAN) facilities by building a VoIP server. The development model used in this research is a user interface that is built alone to be applied on Android phones. This user interface is as a face-to-face tool for fellow VoIP users. The study was conducted at the West Bogor sub-district office with 23 respondents as a sample of application trial subjects. The results of the analysis showed 78.66% of the application eligibility level, showing the application or system made was considered good.

Performance assessment of VoIP service over different handover mechanisms in UMTS networks

Many researchers have discussed various topics in universal mobile telecommunication system (UMTS) networks: the process of switching from one cell to another for the subscriber and the impact of the quality of the connection during the transition process, quality of services (QoS), the quality of the uplink and downlink carrier line, the various types of code for the voice transmitted through the Internet, especially the research that discussed voice over internet protocol (VoIP) technology as voice travels from cell to cell in mobile networks, depending on the type of delivery. In this paper, a proposed scenario of a UMTS network was implemented to evaluate the multicellular VoIP movement; the proposed UMTS network was simulated using the OPNET 14.5 simulator. The calculation and analysis of the different parameters of the user while moving from one cell to another with different movement speeds considered, the best mean opinion score (MOS) value (3.19) registered for the scenario (soft handover) comparing with another type of handover (3.00).

Zeroize: A New Method to Improve the Utilization of 5G Networks When Running VoIP over IPv6

5G technology is spreading extremely quickly. Many services, including Voice Over Internet Protocol (VoIP), have utilized the features of 5G technology to improve their performance. VoIP service is gradually ruling the telecommunication sector due to its various advantages (e.g., free calls). However, VoIP service wastes a substantial share of the VoIP 5G network’s bandwidth due to its lengthy packet header. For instance, the share of the packet header from bandwidth and channel time reaches 85.7% of VoIP 5G networks when using the IPv6 protocol. VoIP designers are exerting considerable efforts to solve this issue. This paper contributes to these efforts by designing a new technique named Zeroize (zero sizes). The core of the Zeroize technique is based on utilizing the unnecessary fields of the IPv6 protocol header to keep the packet payload (voice data), thereby reducing or “zeroizing” the payload of the VoIP packet. The Zeroize technique substantially reduces the expanded bandwidth of VoIP 5G networks, which is reflected in the wasted channel time. The results show that the Zeroize technique reduces the wasted bandwidth by 20% with the G.723.1 codec. Therefore, this technique successfully reduces the bandwidth and channel time of VoIP 5G networks when using the IPv6 protocol.

Improving VoIP Transmission for IEEE 802.11n 5GHz MANET

In catastrophic scenarios, rescue teams need to communicate with each other without any fixed infrastructure while moving, such as search and rescue, firefighter, police, and tactical networks i.e. MANET. In these scenarios Mobile Ad hoc Networks (MANETs) can supports immediately Voice Over Internet Protocol (VoIP) service to be used, with fast deployment, low cost, and flexibility during movement. Recently the new mobile devices support the new frequency of IEEE 802.11n 5GHz that gives better performance requirements to MANET, with enhanced Quality of Service (QoS) and bandwidth (B.W). This paper will calculate and improve VoIP service in MANET that operates under IEEE 802.11n 5GHz frequency, with comparing two main routing protocols AODV and OLSR, and analyzing some QoS requirement such as end-to-end delay, throughput, and retransmission attempts using Network Simulator (NS2).

Enhanced Routing Algorithm Based on Reinforcement Machine Learning-A Case of VoIP Service.

The routing algorithm is one of the main factors that directly impact on network performance. However, conventional routing algorithms do not consider the network data history, for instances, overloaded paths or equipment faults. It is expected that routing algorithms based on machine learning present advantages using that network data. Nevertheless, in a routing algorithm based on reinforcement learning (RL) technique, additional control message headers could be required. In this context, this research presents an enhanced routing protocol based on RL, named e-RLRP, in which the overhead is reduced. Specifically, a dynamic adjustment in the Hello message interval is implemented to compensate the overhead generated by the use of RL. Different network scenarios with variable number of nodes, routes, traffic flows and degree of mobility are implemented, in which network parameters, such as packet loss, delay, throughput and overhead are obtained. Additionally, a Voice-over-IP (VoIP) communication scenario is implemented, in which the E-model algorithm is used to predict the communication quality. For performance comparison, the OLSR, BATMAN and RLRP protocols are used. Experimental results show that the e-RLRP reduces network overhead compared to RLRP, and overcomes in most cases all of these protocols, considering both network parameters and VoIP quality.

Down to Zero Size of VoIP Packet Payload

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.

Fast Detection of Distributed Denial of Service Attacks in VoIP Networks Using Convolutional Neural Networks

Voice over Internet Protocol (VoIP) is a recent technology used to transfer media and voice over Internet Protocol (IP). Many organizations moved to VoIP services instead of the traditional telephone systems because of its low cost and variety of introduced services. The Session Initiation Protocol (SIP) is the most used protocol for signaling functions in VoIP networks. It has simple implantation but suffers from less protection against attacks. The Distributed Denial of Service (DDoS) attack is a dangerous attack that preventing legitimate users from using VoIP services and draining their resources. In this paper, we proposed an approach that utilizes deep learning to detect DDoS attacks. The proposed approach uses token embedding to improve the extracted features of SIP messages. Then, Convolutional Neural Network (CNN) was used to detect DDoS attacks with different intensities. Furthermore, a real VoIP dataset that contains different scenarios of attacks was used to evaluate the proposed approach. Our experiments find that the CNN model achieved a high F1 score (99-100%) as another deep learning approach that utilizes Recurrent Neural Network (RNN) but with less detection time. Also, it outperforms another system that depends on classical machine learning in case of low-rate DDoS attacks.

A COMPARATIVE ANALYSIS OF PROTOCOLS FOR VOIP SERVICES

Although the widespread of broadband network has boosted the deployment of voice over IP (VoIP) systems, most part of the telephone service is still provided on the traditional PSTN lines. It can be expected that the transition from traditional telephone to VoIP systems proceed, and that the network traffic due to the phone calls will become significant. Hence, it is still important to select the best protocols, both for traffic forwarding and for encoding to ensure the required quality. In this paper we present the results of a series of experiments to evaluate the protocols involved in VoIP service. On a laboratory testbed we reproduce a common situation of two interconnected branches and compare the performance achievable by selecting different network protocols like SIP and IAX combined with current voice codecs like G.711 and G.729.

A Comparison of Voice-over-IP Capacity and Performance for Different Scheduling Techniques

To ensure Quality of Service (QoS), different scheduling techniques play a very vital role for Voice over IP (VoIP) traffic. The traffic for VoIP service is constant bit traffic and has low latency, which implies that small VoIP packets are imparted in the typical duration of time. Also, for QoS related to packet delay and packet loss, the Voice over IP packet needs to be anticipated periodically. Dynamic scheduling is used for packet scheduling in the Evolved UTRAN (E-UTRAN) type of radio systems. In present work, we will compare different type of scheduling techniques such as dynamic scheduling, persistent scheduling, and semi-persistent (with SID packets) scheduling used for VoIP packets in Long Term Evolution (LTE). We try to find out the number of users supported for different Control Channels (CC) with and without restrictions. The results show that semi-persistent scheduling is the most efficient scheduling as it helps in better-optimized resource utilization while dynamic scheduling has a large amount of signaling involved which reduces its efficiency. Persistent scheduling technique fails to utilize the resource as compared to semi-persistent scheduling. VoIP capacity related to the number of users is calculated for Semi-persistent Scheduling without any condition on control channels and as a result, we observed some prominent data.

ITTP-PG: A Novel Grouping Technique to Enhance VoIP Service Bandwidth Utilization

Recently, the field of telecommunications started to migrate to Voice over Internet Protocol (VoIP) service. VoIP service applications produce packets with short payload sizes to reduce packetization delay. That is, increasing the preamble size and expends the network link bandwidth. Packet grouping is a technique to enhance the employment of network link bandwidth. Numerous grouping techniques are suggested to enhance link bandwidth employment when using RTP/UDP protocols. Unlike previous research, this article suggests a packet grouping technique that works over the Internet Telephony Transport Protocol (ITTP), not RTP/UDP. This technique is called ITTP Packet Grouping (ITTP-PG). The ITTP-PG technique groups VoIP packets, which exist in the same route, in a single ITTP/IP preamble instead of an ITTP/IP preamble to each packet. Consequently, preamble size is diminished and network link bandwidth is saved. ITTP-PG also adds 3-byte runt-preamble to each packet to distinguish the grouped packets. The suggested ITTP-PG technique is simulated and compared with the conventional ITTP protocol (without grouping) using three elements, namely, the number of concurrent VoIP calls, preamble overhead, and bandwidth usage. Based on all these elements, the ITTP-PG technique outperforms the conventional ITTP protocol. For example, the result shows that bandwidth usage improved by up to 45.9% in the tested cases.

Effectiveness of SIP Server Under SIP Flooding Attack During VoIP Calls

Voice over Internet Protocol (VoIP) service is increasingly one of the most widespread applications since few decades. Data traffic of VoIP service faces various challenges and threats. The Session Initiation Protocol (SIP) flooding attack is one of the popular threats to this service. Flooding attack is the most severe attack as it is easy to generate and capture the networks and nodes that further lead to Denial of Service attack. Therefore, the performance analysis of SIP servers requires more focus. In this paper, effectiveness of SIP server under SIP flooding attack is evaluated using various message scenarios by setting up an experimental test bed. The stress testing of SIP server is carried out with CPU and memory utilization check. The quality of VoIP calls is measured in term of three quality metrics jitter, delay and packet loss. It is observed that with the increase of simultaneous calls, quality of VoIP calls degraded.

Performance Investigation of VoIP Over Mobile WiMAX Networks through OPNET Simulation

Worldwide Interoperability for Microwave Access (WiMAX) is regarded as a promising technology that can provide wireless communication because of its advantages which include, high-speed data rates, high coverage and low cost of development and maintenance. WiMAX also supports the performance of Voice over Internet Protocol (VoIP), which is expected to replace conventional circuit switched voice services. VoIP requires to accurately design of QoS configurations over WiMAX networks. This paper focuses on studying and analyzing the performance of VoIP over WiMAX mobile networks. WiMAX network and VoIP technology such as mobility, WiMAX service classes, number of nodes and VoIP codecs are studied and analyzed. WiMAX network is simulated in a different manner using the simulation program known as OPNET Modeler. Simulation results established that the service layer Unsolicited Grand Service (UGS) is more appropriate for VoIP service because it has the best standard and performance. It was also observed that the least delay and highest value of customer satisfaction rate of services is demonstrated by the G.723.1 best coding. It also maintains the minimum consumption of capacity.

Quality of Service of VoIP over WiMAX

In this paper, the Voice over Internet Protocol (VoIP) was analyzed, and the performance of the quality of service (QoS) parameters obtained through the OPNET simulation software was evaluated. The case study started with the analysis of the network structure .It examined the optimal planning parameters and considered to optimize the performance of the Worldwide Interoperability for Microwave Access network to increase the VoIP service performance. The simulation covered three scenarios with different configurations; each scenario was elevated in term of QoS. The simulations studied various cases including variation in cell radius, different number of users, and different number of cells. Results showed that increasing the cell radius does not increase the quality of the network, and increasing the number of users inside one cell also degrades the quality of the network. By contrast, distributing users into different cells increasesthe network performance but enhances the delay time and increases the throughput.

On the Cryptographic Features of a VoIP Service

Security issues of typical Voice over Internet Protocol (VoIP) applications are studied in this paper; in particular, the open source Linphone application is being used as a case study. An experimental analysis indicates that protecting signalling data with the TLS protocol, which unfortunately is not always the default option, is needed to alleviate several security concerns. Moreover, towards improving security, it is shown that a VoIP application may operate over a virtual private network without significantly degrading the overall performance. The conclusions of this study provide useful insights to the usage of any VoIP application.