Analyzing the use of VoIP technology in collaborative modeling

Voice over Internet protocol (VoIP) is the transmission of voice over networks using the internet protocol. IP networks have become increasingly popular in the past few years, due to the exponential growth of the public internet leading the way in to the IP world. Long distance calls, especially international subscriber dialing (ISD), can be made significantly less expensive when supported by an IP network rather than by the PSTN. Any call made is supported by VoIP technology and it involves the transmission of many individual packets over an IP network. Thus the cost of VoIP calls in part depends on the number and size of packets that must be transmitted. So voice (source) compression technology is used to reduce the amount of bandwidth required in order to reduce cost and to reduce the delay impact from network. But, compression techniques increase the network impairments also. When packets are transmitted through network they are affected by impairments, like packet drop and end-to-end delay. The main agenda of VoIP service providers is to provide good quality of service (QoS). The objective of this research work done is to minimize, the error introduced in channel due to the above mentioned network impairments and hence improve the quality of sound, and also analyze the voice quality with and without use of channel coding scheme. There are two measuring techniques, subjective and objective. In subjective method we measure the mean opinion score (MOS) and use matching algorithms in objective methods to measure the difference between decoded outputs of source coding system and joint source-channel coding system. In this work, the objective method is used and it is observed that the performance of the system improves significantly

Voice over Internet protocol (VoIP) is becoming increasingly popular due to its significant advantages regarding cost and support of enhanced multimedia services. Despite of the substantial advantages of using the Internet as the transmission medium for voice calls, we can foresee many undesirable uses, especially in terms of spam. VoIP technology may provide new means for the transmission of bulk unsolicited messages and calls (spam over Internet telephony - SPIT), mainly due to the reduced cost compared to traditional telephony. Therefore, mechanisms are essential to address the SPIT problem, while maintaining the advantages of VoIP technology. In this paper we conduct a comprehensive threat analysis that addresses the ascendant VoIP protocol, the session initiation protocol, in terms of its vulnerability regarding SPIT deliverance. Our analysis introduces the requirements that the mechanisms dealing with the SPIT phenomenon should take into account, in order to be feasible and efficient.

Voice over IP (VoIP) is a developing technology and a key factor in both the emerging cyberspace engineering and also an accomplishment to set up its position in the telecom industry. VoIP technology is based on internet technology; where data packets switching system is used rather than circuit switching. Whereas, an analog signal is changed over into the digital signals in the full-duplex transmission. VoIP technology is replaced with the conventional public switched telephone network (PSTN) system due to the high flexibility and low-cost. The purpose of this research work is to deliberate experimental and computational performance in the view of quality of service (QoS) parameters of VoIP over local area network (LAN) network. The VoIP systems implementation is based on two different operating system framework (Linux and Windows), whereas, Linux-based and Windows-based private branch exchanges (PBXs), such as Asterisk (Linux-open source) and Axon (window-close source) are configured, installed and verified. QoS factors (such as packet loss, delay, jitter, etc.) are observed over the Asterisk and Axon PBXs in a LAN domain with the assistance of Paessler switch activity grapher (PRTG) monitoring tool. The validations of results are looked at for QoS parameters crosswise over both PBXs with data load (i.e., file transfer and HTTP traffic) during VoIP calls. The productivity and execution of Axon and Asterisk have been equated and analyzed over experimental based outcomes.

Cable telecommunication technology allows high-speed bi-directional transfer of Internet protocol (IP) traffic, between the CATV headend system and customer locations, over all-coaxial or hybrid fiber-coax (HFC) cable network. With the rapid progress of VoIP (voice over IP) technology, IP telephony service via cable networks has become a preferred strategy for cable data service providers. This paper presents an implementation of VoIP cable modem, which is named DCM-100. Design issues, hardware architecture, software architecture, and performance testing results are described.

Information technology is developing rapidly, especially in the field of computer science, telecommunications and information systems. In the early days of telecommunications, voice networks could not be combined with data networks. However, more recently it became possible to integrate data, audio and video services onto a single network, known as a multimedia network. The capabilities of the Internet are also rapidly increasing, and now, in addition to data packets (email, web browsing) the Internet transfers audio and video packets. Voice over Internet Protocol (VoIP) is a multimedia service. This technology can be used to transmit audio, video, and data packets. Consequently, it is becoming the most preferred communications technology. VoIP will eventually replace the use of traditional telephony – Public Switched Telephone Networks (PSTN), although the switchover process is challenging. A chief concern it that, while PSTN transmits audio packets over a closed network, VoIP sends audio packets using an open network – the Internet. In PSTN, eavesdroppers must have direct access to physical network to obtain information from communications. Whereas with VoIP, eavesdroppers can monitor data packets from they are connected to the Internet. Hence, security and privacy are important considerations when switching to VoIP systems. One solution is anonymous systems, which can be used to implement security and privacy protocols. However, this typically reduces the Quality of Service (QoS) of the VoIP. This empirical research study examines VoIP performance in an anonymous network – The Onion Routing (Tor). Two scenarios are implemented using the real Tor network to investigate three QoS metrics for VoIP: latency, jitter and packet loss. As recommended, latency in VoIP should be less than 400 ms, jitter should be less than 50 ms and packet loss should not be more than 5%. In addition, the research calculates the probability of attackers in the two scenarios implemented and evaluate Tor network forecasting. Experiments were conducted in reference to two scenarios. The first scenario is VoIP calls routed through a Tor network with three Tor relays (default Tor). And the second scenario is VoIP calls routed through a Tor network with two Tor relays. Experiments were performed in three periods; December 2012, July 2013, and October 2013. During each experimental time period, a hundred calls were captured for each scenario. Experimental results show that the QoS of VoIP over the two Tor relays was better than the VoIP over the three Tor relays. However, a probability of attackers calculation found that the VoIP with three Tor relays returned better anonymity than that with two Tor relays. The best results were returned over the experimental period in July 2013, when the acceptable calls using two Tor relays reached 64 calls at 5% packet loss. Meanwhile, the worst experimental results were returned in October 2013, when acceptable calls numbered 11 using three Tor relays at 1% packet loss. At the end of 2013, the actual data for relay numbers and bandwidth approached forecast result with time series analysis. Meanwhile, the numbers of Tor users differed from the Tor users forecast. In August 2013, Tor users increased dramatically; this is attributed to the fact that BotNet attack was using the Tor network to attack their target leading to a fivefold increase.

A long-held desire on the part of many concerned technologists and social scientists has been the creation of a system that would allow individuals in the remotest regions of the globe to have access to communications technologies. Today, Voice over Internet Protocol (VoIP) allows individuals in regions formerly restricted by economic, geographic or cultural reasons to communicate on a broader scale at a reasonable financial cost, through convergent systems and wireless means, for purposes defined by themselves. This analysis explains the core components of VoIP and how it facilitates communication, considers the social and cultural impact of VoIP on a global scale and examines whether we are entering a truly “global village” through greater use of VoIP technologies.

The lecture will present a test of the VoIP (Voice over Internet Protocol) network at the University of Applied Sciences (FHTW) Berlin based on Cisco VoIP phones model 7970G as an example in test of security policy of VoIP networks to find out the security gaps and to support improving security of the FHTW's VoIP campus network. Configuration options of the VoIP phone used at the University of Applied Sciences Berlin will be presented. The finding, installing and configuring products for testing the security of VoIP network and VoIP phones have to be dicussed and the security test of Cisco VoIP phones presented offer ways for make more secure configuration and steps to improve the overall security of VoIP networks. The presented results are backed by the long term ERASMUS TN co-operation between the University of Applied Sciences Berlin and the University of Rousse.

Systems employing the Internet and Voice over Internet Protocol (VoIP) are alternative to the legacy landline telephony system. The services offered by these systems allow users to communicate with their family members, friends, banks and business partners whenever they are online. Particularly, the VoIP service is popular with the Internet users because charges are bound into usually fixed access costs making the price of the long distance calls themselves appear economical and even free. However, VoIP services have several disadvantages. One of the disadvantages of VoIP is, it is exposed to Internet security vulnerabilities, threats and attacks. The attacks come in many forms. One of the methods to attack the VoIP service, is by sending a Spam over Internet Telephony (SPIT). SPIT is normally executed by malicious parties initiating automated, unsolicited, and unwanted communications that used VoIP or video conferencing services, just like an email SPAM. Solutions to mitigate SPIT are still lacking. To defense a VoIP system from SPIT, we have come out with the idea of Automated Intrusion Detection and Prevention System over SPIT (AIDPoS) that can detect and prevent SPIT from external parties to enter the VoIP network. AIDPoS is a combination system of Intrusion Detection System (IDS) and Intrusion Prevention System (IPS) on Voice over IP (VoIP) server. This is an experimental research to find a solution to combat SPIT if it happens in the future. Our result shows that an automated Intrusion detection and Prevention system over SPIT attack is feasible. Our work is on an open source VoIP network. Other vulnerabilities and threats are important, but beyond our research scope and is not covered in this paper.

In order to provide the same or better service quality in the Internet than traditional circuit-switched telephone network, a number of issues have to be solved which have hampered it in the Internet. Therefore, in this paper the VoIP (Voice over Internet Protocol) network design problems are modeled as nonlinear non convex combinatorial integer mathematical programming problems. The optimization problems have great practice value for the network service providers. This problem is a constrained multicast QoS routing and is proved to be a NP complete problem. The total link capacity augmented cost should be minimized when we design a new VoIP network or when the original network could not serve all of the traffic demands. It is also known as a complicated problem. Introduction Originally, the Internet was designed to provide best-effort services for the data generated by computers. The timeliness of data communication is generally delay tolerant. Quality-of-Service (QoS) constraints are not as important as routing flexibility and connectivity. Hence, using IP to transport voice data is contradictory to the basic requirement of the voice service: a timely delivery of voice samples. Although IP was not initially designed to provide services for real-time traffic, recent technical progress has made IP have the capabilities to provide real-time services in near future. In order to provide the same or better service quality in the Internet than traditional circuit-switched telephone network, we must deal with a number of issues that have hampered it in the Internet. Voice service requirements could be discussed from two perspectives: (1) application requirements such as end-to-end delay, jitter, packet loss and overdue probability; (2) user’s perspective such as reliability, availability, and supplementary services [1][2]. Telephony service providers must guarantee the quality of service they provide e.g. maximum one way delay does not exceed 150ms. Internet telephony requires a range of protocols, ranging from those needed for transporting real-time data across to the network e.g. Real-time Transport protocol (RTP) [3], to Quality-of-Service aware routing (QoS routing), signaling protocol, resource reservation, internetworking between IP networks and PSTN, QoS-aware network management and billing protocols. ITU-T defined H.323 to provide multimedia communication in packet networks. From the perspective of network service providers, they want to optimize the network performance such as minimizing the total bandwidth consumption, maximizing throughput or total revenue [4][5] subject to user and application constraints. In this paper, we want to develop the mathematical model for VoIP network. We minimize the total bandwidth consumption under users’ QoS requirements, the network topology and the network capacity. Performance Optimization Model It is generally accepted that Internet telephony and traditional circuit-switched telephony will coexist for quite some time. The VoIP architecture must deal with interworking between IP networks and PSTN, so we need gateways between the two worlds. There are four possible models of VoIP [6]. They are PC-to-PC, Gateway-to-Gateway, PC-to-Gateway, and Gateway-to-PC models. The architecture of VoIP is shown in Fig.1. The first model of VoIP is PC-to-PC architecture, which based 2nd International Conference on Electrical, Computer Engineering and Electronics (ICECEE 2015) © 2015. The authors Published by Atlantis Press 1144 on the assumption that two or more users have access to multimedia computers that are connected to the Internet. Fig.1 VoIP Architecture The VoIP system is modeled as a graph, where the hosts, VoIP gateways, and switches are represented by nodes and communication link sets are represented by links. Let N = {1, 2,..., n} be the set of nodes and L be the set of links in the graph (network). Let G be the set of all user groups. An user group g is a voice communication session requesting for transmission in the network. An user group g may be an unicast from the source to a destination or a multicast from the source to multiple destinations. For each user group g, the traffic is transmitted exactly over one tree. Dg represents the set of destinations of user group g. g λ is the required bandwidth of the voice transmission for each user group g. Below is a verbal description of the VoIP system design problem we considered. Given : Network topology Capacities of network links Equivalent bandwidth of each multicast/unicast group Time threshold and tolerable overdue probability for each multicast/unicast group To determine : The minimum overall bandwidth consumption Routing tree for each multicast/unicast group Overdue probability of each multicast/unicast group Objective : To minimize the total bandwidth consumption Subject to : End-to-end QoS (overdue probability) constraint Tree constraint Multi-commodity flow constraint Capacity constraint Integrality constraint Hop constraint Table 1 and 2 are the notations we use in this paper . Internet VoIP Gateway VoIPGatewa Central office switch or local hub Central office switch or local hub Multimedia PC IP Router Multimedia PC IP Router Gatekeepe r

Recent voice over Internet protocol (VoIP) technology is attracting more and more attention and interest in the industry. VoIP applications such as IP telephony systems involve sending voice transmissions as data packets over private or public IP networks as well as reassembled and decoded on the receiving side. Security concerns are always the major barrier that prevents many businesses from employing VoIP technologies. In this paper, we discuss various VoIP security threats and possible approaches to tackle the threats in VoIP applications

LTE (Long Term Evolution), a 4G mobile communication standard has a support for higher data rates and lower latency. Since LTE is completely an Internet Protocol (IP) based technology, Network operators of LTE can handle voice via a “fall-back” technique by using previously existed circuit switched networks. To avoid this, Voice over IP (VoIP) services need to be used to deliver the voice over IP networks, such as Internet. Voice Codec is one of the critical components in VoIP used to convert analog signals to compressed digital packets and vice-versa that can be sent over the internet. But there are different kinds of codecs exist that optimize the voice stream based on bandwidth and application requirement. Unfortunately, there is no standard codec that is acceptable for all circumstances, some codecs provides better quality, some other uses lesser bandwidth. All these features will effect on the ultimate selection of codec by an organization. This paper focuses on the quality of service (QOS) analysis of VoIP traffic over LTE by configuring different voice codecs such as G.711, G.722.2, G.723.1 and G.729. To analyze the performance of these codecs ns-3 simulator is being used since its core has a support for LTE models. The comparative study of above selected VoIP codecs over LTE network in terms of Packet Loss, End to End delay, Jitter, and Throughput in normal and error prone scenarios has been done. From the analyzed results and the performance evaluated, the selection of codec will become easy to provide better VoIP services for customer satisfaction.

One of the most common ways, where anyone can communicate with each other very easily, and instantly, is, of course, the voice exchange. Public Switched Telephone Networks (PSTN), and Voice over Internet Protocol (VoIP) are common ways of transferring voice communications by using telephone network. These two ways have their own strengths and weaknesses. The two major positive points of the PSTN are reliability, and stability. One of the drawbacks of the PSTN is that it has complex connectivity. The VoIP strengths are cheaper in costs compared to PSTN, less usage of cable, user-friendly, simple connectivity, and variety of services. On the other hand, the biggest challenge of VoIP is frequent disconnects of voice communication. In this paper, a survey is being performed on a selected group of people to understand the quality differences between PSTN, and VoIP. This paper shows that the Quality of Service (QoS) of today's VoIP is not good enough but users still continue to use it. However, this survey is also trying to find ways on how to improve existing QoS of VoIP. Plus, it points out what type of users prefers to utilize by using PSTN or VoIP.

Emerging markets are increasingly competitive under the heading of convergence and substitution, including device convergence, fixed-mobile convergence, fixed-mobile substitution, service convergence, VoIP (Voice over Internet Protocol) telephony substituting for circuit switched voice telephony, bundled offers of mobile, fixed, broadband and TV and, finally, truly unified communications and digital media services that are delivered irrespective of anytime, anywhere, and anyhow. With the increasing use of wireless networking, convergence technology combined with user mobility is expected to introduce a significant change in how and when people communicate, using a variety of new applications on ubiquitous wireless devices. This chapter aims to discuss VoIP over enterprise fixed mobile converged networks (FMCN) and points out some open issues. We first review VoIP basics including transport and signalling protocols, speech coding, and VoIP quality assessment model, followed by introducing enterprise FMCN architecture and service characteristics. Then we review the performance of VoIP over enterprise wireless networks and discuss voice over enterprise unified communications. Finally, several future directions including VoIP over femtocell, location/presence-enriched enterprise VoIP, and admission control in enterprise wireless network are briefly listed.

The continuous expand of network communications has reached a great level in last years. Internet protocol (IP) has transformed the delivery of content and services over networks. The perspective of lower costs communications had determined a very fast development of Voice over Internet Protocol (VoIP) technologies. Along with voice, VoIP communications channels can transport video streams also. This type of VoIP, known as videoconferencing communications, demands even a higher level of quality of services (QoS) assurance for network resources. The paper is centered on the essential role of basic network bandwidth control within VoIP communications, using as investigation subject IP video calls or videoconferencing communication.

Voice over Internet protocol (VoIP) has become a popular alternative to traditional public-switched telephone network (PSTN) networks that provides advantages of low cost and flexible advanced ldquodigitalrdquo features. The flexibility of the VoIP system and the convergence of voice and data networks brings with it additional security risks. These are in addition to the common security concerns faced by the underlying IP data network facilities that a VoIP system relies on. The result being that the VoIP network further complicates the security assurance mission faced by enterprises employing this technology. It is time to document various security issues that a VoIP infrastructure may face and analyze the challenges and solutions that may guide future research and development efforts. In this paper, we examine and investigate the concerns and requirements of VoIP security. After a thorough review of security issues and defense mechanisms, we focus on attacks and countermeasures unique to VoIP systems that are essential for current and future VoIP implantations. Then, we analyze two popular industry best practices for securing VoIP networks and conclude this paper with further discussion on future research directions. This paper aims to direct future research efforts and to offer helpful guidelines for practitioners.