Wireless VoIP Research Articles

EVALUATION OF VOIP TRAFFICS OVER TIKRIT UNIVERSITY NETWORKS

In recent years, VoIP is a rapidly evolving technology that could possibly revolutionize the telecommunication industry. When implemented on wireless data networks, VoIP could prove to be instrumental in the convergence of existing fixed and cellular telephony networks with the fast growing wired and wireless data networks. Whereas, the growth of the VoIP applications with additional mobile devices in the institution of higher education and increasing number of users led to slow VoIP services. In addition, the throughput and latency (or delay, jitter, and jabber) are two of the most important issues that need to be resolved before the commercial deployment of wireless VoIP. The trace composes of two parts: firstly, collecting of data, secondly, performing the traffics analysis. Therefore, we have analyzed the performance offered by SCTP, TCP and UDP over VoIP at Tikrit University, Iraq. In this paper, we can see that the kind of performance given by SCTP is best then UDP, but the latency is decreased it performance significantly compared to UDP and the SCTP throughput is looks in a better manner. The measurements were verified by simulating the VoIP traffics using Network Simulator 2

Voice Over Internet Protocol Based Communication Design (VoIP) With 3CXSystemPhone On Android Smartphone

TCP\IP protocol can be connected to various computer data networks in the world. This protocol increasingly exists and is needed so that many parties develop it to vote through this protocol. Voice Over Internet Protocol technology is the answer to that desire. This technology is able to convert analog voice (human voice) into data packets then through public internet data networks and private intranet data packets are passed, so that communication can occur. With VoIP communication costs can be reduced so that it can reduce investment costs and conversations (cost saving) or even up to 100% free. VoIP implementation can be done by designing a wireless VoIP network (cable) using 3CXSystemPhone software as a PBX. In this scientific work the software used is 3CXSystemPhone 11.0, where SIP is a VoIP server which is a freeware software, in its application only requires one PC server and several PC clients (2 for example) that are connected to each other

End-to-End Backlog and Delay Bound Analysis for Multi-Hop Vehicular Ad Hoc Networks

Vehicular ad hoc network (VANET) is able to facilitate data exchange among vehicles and provides diverse data services. Intuitively, end-to-end backlog and delay bounds are considered significant metrics to evaluate the quality of service in VANETs. In order to analyze how the multi-hop transmission impacts the delay performance, we model the multi-hop service process into a virtualized single service in a min-plus convolution form. To obtain multi-hop end-to-end backlog and delay bound, we consider the stochastic network traffic characteristics and the highly dynamic channel environment under the static priority, first in first out, and earliest deadline first scheduling policies by applying the martingale theory. The IEEE 802.11p enhanced distributed channel access mechanism is also adopted to analyze the access performance in the MAC sub-layer. With three kinds of real wireless data traces, i.e., VoIP, gaming, and UDP, we verify our algorithm by considering the double Nakagami- $m$ fading channel model among vehicles. From the simulation results, we can see that the supermartingale end-to-end backlog and delay bound are remarkably tight to the real simulation results when compared with the existing standard bounds. The effect of the number of vehicles on the highway on the end-to-end backlog and delay performance is also investigated.

Performance Analysis of Data Transmission over Wireless VOIP using Wavelets

Objectives: To analyze the performance of wireless VOIP system using wavelets. Method: In this paper a Wavelet based Data (Image, Audio and Video) Transmission System for wireless VOIP is proposed. The performance of each wavelet in this proposed system is analyzed on the basis of SNR, PSNR, RMSE, PRSE and Compression ratio. Packet Loss and Delay is also evaluated to analyze transmission capabilities. All these parameters are evaluated for the proposed system without using wavelets and with using different wavelets. Findings: The simulation results shows that the proposed system gives enhanced performance by using different wavelets when compared by without using wavelets. Finally, the Daubechies family has shown better results when compared with its counterpart’s. Applications: This wavelet based proposed system can used for high speed 4G / 5G IP based systems.

Implementing OLSR and Wireless VoIP as Low-Cost Infrastructure Telephony for Rural Area

Communication is one of essential elements in this life, including the people who live at rural areas. However, telecommunications in rural area especially in Indonesia have many constraints like expensive infrastructure, difficult terrain and electrical problem. This led to a lot of areas that uncovered by telecommunication access. In this paper, we proposed a design and architecture of mobile wireless communication based on modified openWRT as a solution for low-cost communication and telephony infrastructure in rural areas. The design combines VoIP over mesh ad-hoc network as a potential solution for cheap calls and use modified embedded device to save electrical resources as a node to connect each other with multi hop transmissions. With this approach, we get many advantages such as high mobility, easy to use, scalable, low-cost implementation, and low power consumption. This paper presents the measurement result of performance to provide access telecommunication in several conditions. In addition, this paper also shows benchmarking results and lifetime of the node that we use in this prototype. The results show that our node can provide telecommunications access in terms of making calls with good quality using wireless LAN. To sum this up, The OpenVoice node is expected to contribute bridging digital divide in rural areas.

Analysis and Simulation of Wired VoIP Networks compared to Wireless VoIP

Analysis and Simulation of Wired VoIP Networks compared to Wireless VoIP

WAVELETS based wireless VOIP and its future scenario

Those who use VOIP, know that a good quality service cannot be ensured over the internet as the internet is not well suited to render real time services. Besides these users do not pay much for the VOIP calls as compared to the circuit switched phone calls. So they do not bother much about the quality of services. However, it is expected that the quality should be sound enough to pay for their time as well as money. In this paper, the recently published literature has been reviewed along with the introduction to wavelets.

Implementation and Analysis of Wireless VoIP Networks with Routerboard Management at State Polytechnic of Malang

In this article, the implementation of wirelessVoIP at State Polytechnic of Malang uses microtic management with the aim of knowing the QoS (Quality of Services) performance between systems without microtic management and systems using microtic management. In the implementation of this system, two tests were carried out, namely QoS testing when without proxy management and when using proxy management. From the two tests, the performance will be compared by doing data compilation using the VQ manager software. The QoS parameters to be taken are delay, jitter, packet loss and throughput. From the measurement it is known that when the VoIP server serves <= 3 calls simultaneously, the MOS value between the managed system (MOS = 3.7) and the system without management (MOS = 3.7) is almost the same because the value of delay and packet loss in the system without management and systems with management did not differ much, namely 107 ms and 83 ms, and the packet loss value was the same, namely 5%. And when serving> 3 simultaneous calls, there is a difference of 0.18 from the MOS value between the managed system (MOS = 3.48) and a system without management (MOS = 3.3) with a value of delay and packet loss for systems without management and systems with management, namely 527 ms and 340 ms, and the packet loss values ??were the same, namely 8% and 7.2%.

A New Wireless VoIP Signaling Device Supporting SIP and H.323 Protocols

Nowadays, VoIP is a technology with a great demand and wireless networks are increasingly deployed. Each of these has its own technology constraints. For VoIP, it is very important to take into consideration the need to provide a high quality service according to well-defined standard transmission (jitter, end-to-end delay, MOS, and packet loss). However, wireless networks (IEEE 802.11) are based on radio which undergoes a number of technical constraints to achieve theoretical transmission rates; among these constraints the number of users of the networks, the distance between the client and the access, and the amount of data transmitted point are included. In this term, a study is made by simulating wireless network in OPNET Modeler with a fairly large number of VoIPs (15 users) whose signaling is handled via a new node that was created specifically to manage the signaling tasks under SIP and H.323 in order to minimize the number of nodes in the network and avoid the congestion. In this paper, two scenarios are compared; the first contains a number of VoIP users with SIP and H.323 signaling handled by the new created device; the second scenario is similar to the first except that the distance between the stations is remarkably lower.

Managing Quality of Experience for Wireless VOIP Using Noncooperative Games

We model the user's quality of experience (QoE) in a wireless voice over IP (VoIP) service as a function of the amount of effort the user has to put to continue her conversation. We assume that users would quit or terminate an ongoing call if they have to put more efforts than they could tolerate. Not knowing the tolerance threshold of each individual user, the service provider faces a decision dilemma of whether to fix the network problem immediately whenever he detects a user effort in the VoIP system, or ignore it with the hope that the user may still continue the call anyway. In this paper, we formulate the provider's dilemma as a non-cooperative game between the provider and the VoIP user experiencing a deteriorating QoE. We demonstrate that providers implementing the equilibrium solutions can expect to not only increase their revenues, but also reduce the number of cases when users quit out of frustration thus minimizing potential churning. We also discuss conditions under which a sophisticated user may or may not benefit from faking unwarranted efforts with a goal of receiving a better service from the provider. Finally, we conduct a subjective experiment of VoIP over WiFi, which verifies the key model assumption that perceptual quality is negatively correlated to the amount of effort the user has to put to continue the call.

유무선 VoIP 취약점 분석과 해킹공격 및 보안 연구

Recently VoIP has provided voice(both wired and wireless from IP-based) as well as the transmission of multimedia information. VoIP used All-IP type, Gateway type, mVoIP etc. Wired and wireless VoIP has security vulnerabilities that VoIP call control signals, illegal eavesdropping, service misuse attacks, denial of service attack, as well as wireless vulnerabilities etc. from WiFi Zone. Therefore, the analysis of security vulnerabilities in wired and wireless VoIP and hacking incidents on security measures for research and study is needed. In this paper, VoIP (All-IP type, and for Gateway type) for system and network scanning, and, IP Phone to get the information and analysis of the vulnerability. All-IP type and Gateway type discovered about the vulnerability of VoIP hacking attacks (Denial of Service attacks, VoIP spam attacks) is carried out. And that is a real VoIP system installed and operated in the field of security measures through research and analysis is proposed.

NCTU‐VT: a freeware for wireless VoIP performance measurement

AbstractVoice over IP (VoIP) is a promising low‐cost voice communication over the wireless IP network. To provide satisfactory VoIP services, the Quality of Service (QoS) of the wireless network should be guaranteed. This paper proposes a VoIP performance measurement freeware called NCTU VoIP Testing Tool (NCTU‐VT). We compare NCTU‐VT with two commercial tools SmartVoIPQoS and IxChariot in terms of packet loss, latency, and Mean Opinion Score (MOS) of the VoIP sessions in Wi‐Fi network. Our study indicates that these three tools can accurately measure VoIP performance in Wi‐Fi environment. Copyright © 2010 John Wiley & Sons, Ltd.

Binary CDMA 기반의 무선 VoIP 시스템의 구현 및 성능평가

Binary CDMA는 음성 영상 등의 멀티미디어 데이터를 초고속으로 우수한 전송품질로 통신할 수 있는 우리나라가 개발한 새로운 표준 무선통신 기술이다. 본 논문에서는 전화선의 설치나 매설 작업 없이 원하는 장소에 자유롭게 설치할 수 있는 Binary CDMA 기반의 무선 VoIP 시스템 개발을 위한 개발방법 및 구현에 관해 연구하였다. Binary-CDMA VoIP 시스템의 구조를 설계하고, 호제어를 위한 메세지와 자료구조의 정의한 후, 사용자간 통화를 위한 착발신 단말 간의 호제어 메세지 흐름을 설계하고 구현하였다. Binary CDMA is a new standard technology for wireless communication developed by our country that makes high speed communications and good quality of services. In this paper we researched the development methods and implementation of a wireless VoIP System based on the Binary-CDMA technology that makes it freely installed in any place without phone cables and laying works. We designed the structure of the Binary-CDMA Wireless VoIP System and defined messages and data structures for call control. And then we designed and implemented call control message flow between VoIP terminals for user's communication.

Binary CDMA 기반의 무선 VoIP 시스템의 개발방법 및 설계

Binary CDMA는 음성 영상 등의 멀티미디어 데이터를 초고속으로 우수한 전송품질로 통신할 수 있는 우리 나라가 개발한 새로운 표준 무선통신 기술이다. 본 논문에서는 전화선의 설치나 매설 작업 없이 원하는 장소에 자유롭게 설치할 수 있는 Binary CDMA 기반의 무선 VoIP 시스템 개발을 위한 개발방법 및 설계에 관해 연구하였다. Binary CDMA is a new standard technology for wireless communication developed by Korea that makes high speed communications and good quality of services. In this paper we researched on the development methods and design of a wireless VoIP system based on the Binary-CDMA technology that makes it freely installed in any place without phone cables and laying works.

Qos Enhancement in Wireless VoIP Networks Using Interactive Multiple Model Based Kalman Filter

In this paper, a QoS enhancement scheme is proposed using an Interacting Multiple Model (IMM) for wireless VoIP (WVoIP) networks. The quality of speech varies in WVoIP due to variations in channel parameters. Kalman filter can be used to enhance speech quality by predicting the channel variations. IMM consists of two models namely static and dynamic, representing the wireless VoIP mobility states and both filters work in parallel. The IMM filter estimate is computed as a weighted sum of individual filter estimates. As the computational complexity is more in IMM, a systolic array based parallel architecture is proposed to implement IMM for real time applications.

Improvement in Performance of the VoIP over WLAN

Wireless VoIP, typically over 802.11 WLAN, is becoming increasingly popular, but even further elevates the challenge of delay and loss reduction. Degradation of speech quality caused by packet delay and loss of voice traffic is still one of the critical technical barriers of the VoIP system. Furthermore, apart from these limitations WLAN will need to support a large number of concurrent VoIP communication since VoIP is spreading rapidly. These motivations led me to do the analysis of VoIP capacity in IEEE 802.11e WLAN and to investigate the methodology to enhance this capacity while maintaining an overall good quality of service.

Improving wireless VoIP quality by using adaptive packet coding

The transfer of real-time VoIP data over wireless and mobile networks is subject to packet losses that yield unacceptable quality. This work proposes that the quality can vary significantly depending on the packet coding method used. An adaptive piggyback packet coding that can provide the VoIP user acceptable quality over the wide range of packet loss conditions observed in wireless and mobile networks is found.

A Cross Layer Perceptual Speech Quality Based Wireless VoIP Service

This paper proposes a cross layer wireless VoIP service which integrates an Adaptive QoS Playout (AQP) algorithm, E-model, Stream Control Transmission Protocol (SCTP), IEEE 802.21 Media Independent Handover (MIH) middleware and two user motion detection services. The proposed AQP algorithm integrates the effect of playout control and lost packet retransmission based on the E-model. Besides, by using the partial reliable transmission service from SCTP and the handoff notification from MIH services in a cross layer manner, AQP can reduce the lateness loss rate and improve speech quality under high frame error rates. In the simulations, the performance of AQP is compared with a fixed playout algorithm and four adaptive playout strategies. The simulation results show that the lateness loss rate of AQP is 2% lower than that of existing playout algorithms and the R-factor is 16% higher than the compared algorithms when a network has 50ms wired propagation delay and 2.5% frame error rate.

Advances in wireless VoIP [Guest Editorial

The six papers in this special issue cover a wide spectrum of topics related to VoIP communications, ranging from QoS mechanisms to the design and performance evaluation of different VoIP components with the purpose of informing both industry and academia about this important research area.

Efficient silence suppression and call admission control through contention-free medium access for VoIP in WiFi networks

In view of the rapidly growing trend of migrating customers from traditional wired phones to mobile phones and then to VoIP services in the recent past, there is a tremendous demand for wireless technologies to support VoIP, specially on WiFi technologies which have already matured commercially. This has put forth great research challenges in the area of wireless VoIP. In this article we have addressed two core issues, efficient silence suppression and call admission control, in QoS provisioning for VoIP services in WiFi networks. In this connection we present a QoS-aware wireless MAC protocol called hybrid contention-free access (H-CFA) and a VoIP call admission control technique called the traffic stream admission control (TS-AC) algorithm. The H-CFA protocol is based on a novel idea that combines two contention-free wireless medium access approaches, round-robin polling and TDMA-like time slot assignment, and provides substantial multiplexing capacity gain through silence suppression of voice calls. The TS-AC algorithm ensures efficient admission control for consistent delay bound guarantees and further maximizes the capacity through exploiting the voice characteristic so that it can tolerate some level of non-consecutive packet loss. We expose the benefits of our schemes through numerical results obtained from simulations.