VoIP Applications Research Articles

An Analysis of VoIP Application in Fixed and Mobile WiMAX Networks

WiMAX is a wireless broadband technology, which promises maximum coverage area and high data rates. WiMAX provides last mile connectivity. This network defines two working models such as fixed WiMAX and mobile WiMAX. The main aim of this paper is to compare and also analyze the performance of the VoIP application over fixed and mobile WiMAX, with respect to various codecs such as G.711, G.723.1ar5.3, G.726ar24, G.728ar16 and G.729. We have considered few QoS parameters such as average end-to-end delay, throughput, average jitter, average one-way delay and average MOS. From the obtained result, codec G.711 and G.726ar24 performs better when compared to other codecs in both fixed and mobile WiMAX network.

Scalable classification of QoS for real-time interactive applications from IP traffic measurements

Measurement of network Quality of Service (QoS) has attracted considerable research effort over the last two decades. The recent trend towards Internet Service Providers (ISP's) offering application-specific QoS is creating possibilities for more sophisticated QoS metrics to be offered by ISP's in service level agreements. This in turn could be used for the purposes of improved network optimization and billing according to application specific QoS guarantees. We report a scalable near real-time approach using passively logged IP traffic data for classification of application latency and packet loss across a range of real-time interactive applications. We run six experiments involving Minecraft, Quake 3 Urban Terror, VLC video streaming and the commercial Wirofon VOIP application. We use a mixture of laboratory and real-world deployments, with run times ranging from hours to days, and observe a combination of real and simulated ISP latency and packet loss events. Our binary classification (i.e. classes 'OK' or 'lag') 10-fold cross validation F1 scores are between 0.80 and 0.93 depending on the application type. Our multi-class classification (i.e. classes representing discrete packet loss or latency ranges) 10-fold cross validation F1 scores for Minecraft are 0.89 for latency and 0.90 for packet loss. With new business models between ISP's and application developers being actively considered this work represents a significant contribution to the debate by providing scientific evidence relating to a novel approach to scalable QoS measurement

3AKEP: Triple-authenticated key exchange protocol for peer-to-peer VoIP applications

Peer-to-peer architectures have become very popular in the last years for a variety of services and applications they support, such as collaborative computing, streaming and VoIP applications. The security of the protocols involved in such operations is, however, a fundamental prerequisite for a widespread diffusion of such a technology. In this paper, we focus on the establishment of a security association in a distributed scenario and we propose a new key exchange protocol authenticated through three different methods: i) the verification of a signature, based on the identifier of the remote peer, ii) the use of retained secrets from previously established sessions with the same peer, iii) the exchange of a Short Authentication String through a proper “trusted means”. We also provide a possible implementation for peer-to-peer VoIP applications for setting up secure multimedia communications through the standard SIP protocol. Our proposal does not require pre-shared secrets, trusted third parties, nor a Public Key Infrastructure. In addition, we investigate different ways of distributing cryptographic peer identities in a sort of P2P web-of-trust. The proposed protocols have been also implemented and integrated into an open source SIP User Agent, for functional validation.

Media Access Delay and Throughput Analysis of Voice Codec with Silence Suppression on Wireless Ad Hoc Network

The rapid growth in internet has led to the emergence of different wireless technologies and simultaneously there has been a tough challenge to provide QoS over such wireless technologies.The demand for various real time multimedia application especially voice and video applications to be available on the internet has also increased. Multimedia applications need a lot more bandwidth and have different QoS requirements than the applications that were used in early years of the internet. The main reason for the poor quality of voice over IP is because we are transmitting the real time applications over the IP network which is mainly meant to transfer only data and not for voice transmission. The purpose of this paper is to check the performance of VOIP application under different CODEC's such as G.711, G.729 and G.723.1 over wireless network with and without silence suppression. Another issue addressed in this paper is the effect of increasing nodes on voice transmission for the different codecs. Parameters like media access delay and throughput are taken into consideration. Simulation is performed using OPNET modeler simulator.

SEAMLESS VOIP SERVICE USING CENTRALIZED CODEC ADJUSTMENT: A FRAMEWORK IN WIRELESS NETWORK

The evolution of wireless technology leads VoIP application to be a good preferred choice among mobile users. Customers can having a VoIP service while being mobile regardless of location, terminal, and it is independent of network access. It is believed to be an alternative to the PSTN and its service is expected to be as good as PSTN. In essence, QoS plays a crucial part in order to maintain the seamless service during the VoIP session. As wireless networks are prone to channel error and VoIP application is known to be a delay-sensitive application, network traffic congestion could easily affect the voice quality of VoIP. Therefore, the codec selection within the VoIP application must be improved so that the codec can operate at various bit rates in order to cater for different traffic conditions. In this paper, we explain how centralize codec switching can help user to communicate seamlessly. We use WiFi network traffic as an example scenario.

S-Kademlia: A trust and reputation method to mitigate a Sybil attack in Kademlia

Peer-to-peer architectures have become very popular in the last years for a variety of services and applications such as collaborative computing, streaming and VoIP applications. The security and integrity of the overlay involved in such networks is a fundamental prerequisite for deploying such a technology. Withstanding multiple false identities in the overlay, also known as a Sybil attack, is one of the main challenges in securing structured peer-to-peer networks. Poisoning routing tables through these identities may make the routing and storage and retrieval processes extremely difficult and time consuming. In this paper we investigate possible countermeasures and propose a novel method for making the routing and the storage and retrieval of resources in a Kademlia network more secure through the use of a combined trust-based algorithm exploiting reputation techniques. Our solution provides a balanced mixing of standard Kademlia algorithms and trust-based algorithms showing promising results in thwarting a Sybil attack in a Kademlia network, in comparison with similar methods as well.

Fast and Fair Handling of Multimedia CAPTCHA Flows

Multimedia CAPTCHAs play a crucial role in bot filtering policies for VoIP applications. Network flows carrying CAPTCHA content have a very small lifespan and require high transmission quality, making for special handling at transport layer. The present study introduces an analysis-derived, rapidly converging control mechanism for CAPTCHA flows. Instead of relying on generic heuristics, the novel scheme enables the flows to estimate their deviation from the equilibrium and adapt to it in a single step. Simulations demonstrate the high efficiency and convergence speed of the scheme, highlighting its unique fitness for the appointed task.

Performance Analysis of WSN based Technologies: Zigbee and Wimax

Wireless Sensor Network is composed of hundreds or thousands of sensor nodes scattered in a geographical area and one or multiple sinks collecting information or transmitting it through wireless channels. Zigbee and WiMAX are the most advancing communication technologies of Wireless Sensor Network. WiMAX is a standard IEEE 802.16 based technology which supports internet access and VOIP application whereas Zigbee is the only standard IEEE 802.15.4 based technology designed to address the unique needs of low cost, low power WSNs which provide high data rate at complexity and costs. In this paper, a comparative analysis of Zigbee and WiMAX technologies is made in a 100 node network scenario, by incorporating different routing protocols and studying the affect on the QoS parameters such as throughput and PDR(Packet Delivery Ratio).

Queuing Disciplines for VoIP and High Resolution Video Application Services Using OPNET Tool

Queuing Disciplines for VoIP and High Resolution Video Application Services Using OPNET Tool

WiFi Network Virtualization to Control the Connectivity of a Target Service

This paper proposes a WiFi network virtualization technique to control the connectivity of a target service. The packet-level delay violation ratio can be reduced even in a congested situation by provisioning dedicated base station (BS) resources (a set of dedicated BSs) to the target service and allowing only the corresponding terminals to associate with the BSs. The proposed technique is novel in that BSs are specially configured to use the same MAC address, and thus all the decisions on BS selection and handover are separated from those BSs and terminals and are put together into a centralized controller, while consistent layer-2 data paths in a backhaul network are also cooperatively configured. Simulation results show that the proposed technique can control the delay violation ratio of a target VoIP service and keep the ratio extremely low and comparable to that under IEEE 802.11e. A proof-of-concept prototype including two multi-channel virtualization-capable WiFi BSs and a BS switch is developed using off-the-shelf WiFi modules and a commercial OpenFlow switch. Experimental results show that the terminals can make a handover to a dedicated BS in less than 65 ms without any packet drop and association break, and confirm that the effect of the managed handover is limited even in a VoIP application.

Security and privacy of smartphone messaging applications

Purpose – This paper aims to give an overview on a number of selected applications in comparison to a previous evaluation conducted two years ago, as well as performing an analysis on several new applications. Mobile messaging and VoIP applications for smartphones have seen a massive surge in popularity, which has also sparked the interest in research related to their security and privacy protection, leading to in-depth analyses of specific applications or vulnerabilities. Design/methodology/approach – The evaluation methods mostly focus on known vulnerabilities in connection with authentication and validation mechanisms but also describe some newly identified attack vectors. Findings – The results show a positive trend for new applications, which are mostly being developed with security and privacy features, whereas some of the older applications have shown little progress or have even introduced new vulnerabilities. In addition, this paper shows privacy implications of smartphone messaging that are not ...

Robust application identification methods for P2P and VoIP traffic classification in backbone networks

Application identification plays an essential role in network management such as intrusion detection and security monitoring. But the continuous growth of bandwidth and massive amount of packets pose serious challenges for efficacious and accurate application identification. In this paper, we develop a new method to reduce the number of packets being processed while achieving the goal of accurate P2P and VoIP application identification. Firstly, we employ the Bi-flow model to aggregate traffic packets into Bi-flow, which can capture the exchange behavior characteristics between different terminals. Then we employ the signature of Packet Size Distribution (PSD) to capture flow dynamics, which is defined as the payload length distribution probability of the packets in one Bi-flow. Secondly, we collect PSD of several different P2P and VoIP applications and the analysis results show that PSD of different applications are different with each other, which can be used as features to perform traffic identification. We also find the PSD characteristics of one Bi-flow can be captured by its first few packets, which demonstrate our methods can identify the Bi-flow quickly after its establishment. We employ the Renyi cross entropy to perform identification by calculating the similarity between PSD of the Bi-flow being identified and that of specific application. If the similarity is higher than a selected threshold, the Bi-flow being identified is classified to the specific application. Finally, as the PSD is a type of probability feature which is not sensitive to packet lose, we integrate the Poisson sampling method into our framework to process the massive data in backbone networks. Experimental results using the artificial and actual traces collected from monitoring platform in the Northwest Center of CERNET (China Education and Research Network) verify the accuracy and robustness of our method.

Competing One-Way Essential Complements: The Forgotten Side of Net Neutrality

We analyze the incentives of internet service providers (ISPs) to break net neutrality by excluding internet applications competing with their own products, a typical example being the exclusion of VoIP applications by telecom companies offering internet and voice services. Exclusion is not a concern when the ISP is a monopoly because it can extract the additional surplus created by the application through price rebalancing. When ISPs compete, it could lead to a fragmented internet where only one firm offers the application. We show that, both in monopoly and duopoly, prohibiting the exclusion of the app and surcharges for its use –a strong form of net neutrality– is not welfare improving.

Metodología para Seleccionar Aplicaciones de la Tecnología de Comunicaciones Voz sobre la IP (VoIP) para Pequeñas y Medianas Empresas (Pymes): El caso de una Agencia de Viajes

The aim of this paper is to propose a methodology for selecting VoIP applications to be implemented in small and medium size enterprises. The methodology was applied to a representative SME such as a travel agency. To achieve this the following phases were developed: Identification of offered services, identification of providers in the geographic area, study VoIP application, study communication processes of SMEs and development of the matrix for the selection of applications and providers. The results of each phase provide data necessary for the implementation of the following step. To validate the usefulness and applicability of the proposed methodology, tests were conducted to analyze the quality of VoIP applications. Finally on the basis of the results obtained, it was found that the proposed methodology is useful to facilitate the selection of VoIP applications, according to the convenience of each SME and of their communication processes and economic conditions.

Energy-Efficient Security for Voice over IP

The fast spread of handheld smart devices contributed to the development of VoIP softphones running over such devices. Most security mechanisms were mainly designed for desktop PCs and hence did not take into consideration the power constraints of handheld devices. This fact highly motivated the development of new security mechanisms that try to minimize the energy consumption without compromising the security of the exchanged data. In this paper, we propose an energy-efficient security algorithm for VoIP applications running on mobile devices (SecVoIP). The algorithm resolves several weaknesses available in current algorithms while maintaining an appropriate security level. Several experiments were conducted and the results showed significant improvement in processing time, CPU cycles, and consumed energy as compared to SRTP, one of the most widely used security protocols for VoIP. Moreover, we present the results of extensive experimental work that demonstrates that known plaintext attacks against audio streams are not feasible.

Cloud based VoIP Application in Aircraft Data Networks

There has been a high increase in air traffic for the past ten years and to attract more customers many in-flight facilities are introduced by the carriers. These include Video-OnDemand, use of Personal Digital Assistants (PDA) etc. To provide these facilities the VoIP technology is used more often. But maintaining VoIP infrastructure along with the Aircraft Data Network equipment’s will be more cumbersome for the carriers, along with the added infrastructure cost. Thus an improvised architecture has been proposed which helps carriers provide these facilities to the customers in a cost effective and reliable manner. Most of the aircraft communications to the ground rely on satellite for providing these services to the customers. The proposed method uses satellite communication along with cloud based VoIP to provide an efficient quality of service to the passengers. The ground stations used for satellite communication would be connected to a high speed Storage Area Network, to which the cloud network on which the VoIP application is running is also connected. So that whenever a passenger on the flight sends a message/data to the VoIP application first it would be send to the ground stations via the satellite which is then accessed through the SAN server, thus in turn providing faster, reliable and efficient communication to the customers.

Design and analysis of channel reservation scheme in Cognitive Radio Networks

The primary aim of a Cognitive Radio Network (CRN) is to provide higher spectrum utilization through an opportunistic usage of the licensed spectrum bands by the unlicensed Secondary Users (SUs). However, these SUs face unpredictable interruptions in their transmissions due to sudden arrival of the licensed Primary Users (PUs), resulting in either handoff or dropping instances. The channel reservation policy for the PUs is one such solution that reduces the probability of their interference, though at the cost of lower system throughput. This paper provides a complete study of the channel reservation scheme in a CRN. Initially, the system model is defined along with an architectural framework to address the practical constraints involved in implementing the channel reservation strategy. A novel design approach is then adopted to develop the mathematical models, followed by an evaluation of the network parameters to obtain a tradeoff between the decrease in interference and low spectrum utilization when some channels are reserved in the CRN. The optimal number of channels to be reserved is also determined. In addition, as a potential application of the channel reservation policy, an analytical framework is designed for the VoIP applications in the CRN, that highlights significant improvement in the VoIP call quality. Finally, an extensive analysis in the real-life like simulation models validates the inferences drawn from the mathematical models, recording over 50% improvement in the interference-free transmission of the SUs with the channel reservation scheme in the CRN.

Analysis the Performance of MANET Protocol under Black Hole Attack for E-Mail Application

A Mobile Ad hoc Network (MANET) is a collection of wireless mobile nodes having ability to communicate with each other without any fixed network infrastructure. Due to the unavailability of controlling entity, routing and network management are done cooperatively by respective nodes. MANET is an autonomous system where each node act as an end system as well as a router to forward packets for other nodes., intermediate nodes are used to transmitting the packets from the source to the destination node, through different routing protocols i.e. AODV, DSR, GRP, OLSR etc. In this Paper the simulation analysis of the performance of AODV, DSR and TORA is compared for VOIP application under black hole attack. The performance is compared in terms of data dropped, network load, traffic sent and traffic received. The result shows that the overall performance of TORA is better than DSR and AODV.

Audio Codecs in VoIPv6: A Performance Analysis

Audio communications in IP based networks have been revolutionized by the introduction of VoIP applications. High cost-efficiency has made VoIP to be the communication means in today’s world; and this trend is anticipated to be continued on an ongoing basis. The performance of VoIP significantly depends on the efficiency of the audio codecs used in any communication scenario which make the study on the performance issues of audio codecs in VoIP applications worth investigating. IPv6 is the new version of IP, which will gradually replace the current IPv4 as the transition from IPv4 to IPv6 is already in place. This demands the scrutiny of the audio codecs being used in IPv4 to be tested for their compatibility in IPv6 in terms of desired performance. This paper presents the study on the performance of selected audio codecs that are widely used in VoIPv4. G.711, G.729A and G.723.1 codecs were chosen for the study in VoIPv6 based scenarios presented in this paper. The selected audio codecs were applied in IPv6 based voice communication network scenarios to determine their performance efficiency by observing various QoS parameters. The study was done by means of simulation using OPNET.

Study on the QoE for VoIP Networks

In recent years, people are recognizing that the quality in VoIP application should be evaluated according to the QoE (Quality of Experience). The main goal of this paper is to analyze the different factors on the impact of voice quality for VoIP networks. Our contributions are thus threefold: First, we establish a new VoIP simulation platform. The network simulation software is WANem, the voice communication protocol is implemented by Open Phone. This simulation system is more ‘real’ than other researcher’s system. Secondly, we analyze the factors that affect the voice quality of VoIP networks. Thirdly, we use the VoIP networks simulation platform to test the network performance impact on the quality of voice service. Through the experiment result, we can conclude that in order to get the better voice quality, we use the iLBC codec when the VoIP network is exist packet loss. And use the AMR 12.2 kb/s when the VoIP network is exist time delay