Introduction

Communication is the most important thing in the success of an activity. Communication that occurs in a computer network does not only process data packet transfers between computers, but computer technology is currently able to transfer data in the form of sound frequencies that are processed into data packets and sent between computers on the internet network. The development of internet technology is not only using for getting information, but also that is widely used. One of them is audio call or VoIP (Voice over Internet Protocol), this technology has been used by several messenger service providers, such as Yahoo Messenger, Goggle Talk, Skype, and others. Voice over Internet Protocol is a technology that makes internet media able to conduct voice communication remotely directly. Analog sound signals, like what you hear when communicating on the telephone are converted to digital data and sent over the network in the form of data packets in real time. Asterisk is a softswitch to operate the proxies, with session initiation protocol (SIP) based. The aim is to build an Asterisk-VoIP based server, so that it can be developed in future research as needed. The methodology in outline consists of two lines, namely literature study and PPDIOO (Plan, Prepare, Design, Implement, Operate, Optimize). It has been implemented in the Banten Province Communication and Information Service, which has been previously established by the internet network. So that VoIP here is functioned for maximizing it.

The Internet Protocol (IP) environment poses two relevant sources of distortion to the speech recognition problem: lossy speech coding and packet loss. In this paper, we propose a new front-end for speech recognition over IP networks. Specifically, we suggest extracting the recognition feature vectors directly from the encoded speech (i.e., the bit stream) instead of decoding it and subsequently extracting the feature vectors. This approach offers two significant benefits. First, the recognition system is only affected by the quantization distortion of the spectral envelope. Thus, we are avoiding the influence of other sources of distortion due to the encoding-decoding process. Second, when packet loss occurs, our front-end becomes more effective since it is not constrained to the error handling mechanism of the codec. We have considered the ITU G.723.1 standard codec, which is one of the most preponderant coding algorithms in voice over IP (VoIP) and compared the proposed front-end with the conventional approach in two automatic speech recognition (ASR) tasks, namely, speaker-independent isolated digit recognition and speaker-independent continuous speech recognition. In general, our approach outperforms the conventional procedure, for a variety of simulated packet loss rates. Furthermore, the improvement is higher as network conditions worsen.

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.

Chapter 16 - The IP Multimedia Subsystem: True Converged Communications

Voice over Internet Protocol (VoIP) is one of the most popular technologies nowadays that facilitate the user by providing different features as instant messages, phone calls, video calls, and voicemails. Basic VoIP protocols were designed to be efficient instead of secure. After numerous attacks on these protocols several solutions were proposed to prevent against these threats. In this paper, we focus on the security of Session Initiation Protocol (SIP) that is used to initiate, modify, and terminate the VoIP sessions. The paper presents the design and implementation of secure layered architecture for SIP, which adds a new layer to the standard SIP layer model and entitled as Security layer. The Security layer provides authentication, authorization, adaptable feature, and secure key exchange, based on our newly designed protocol, named as Session Initiation Protocol using Single Sign-On (SIPSSO). In order to implement the secure layered architecture based on SIPSSO, we have developed an Android Secure Call application and extend the open source Asterisk accordingly. After the designing and implementation phases, we have verified the SIPSSO protocol formally by using an automated security verification tool, Scyther. Our analysis results reveal that by adding Security layer, we ensured protection against different SIP attacks such as Eavesdropping, Man In The Middle (MITM) attack, Message Tampering, Replay attack, Session Teardown, and Spam over Internet Telephony (SPIT).

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.

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.

Accurate modeling of VoIP traffic QoS parameters in current and future networks with multifractal and Markov models

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

Internet Protocol (IP) and Voice over Internet Protocol (VoIP) are technologies that are changing the way wired and wireless communication networks are developed. More recently, VoIP applications using the Internet are allowing people to communicate with one another without using traditional telephone networks. As these applications are incorporated into wireless devices, such as cordless phones and cellular phones, new wireless VolP applications as well as traditional IP applications are allowing telecommunication providers to develop services with greater flexibility and lower cost structures than traditional telephone service applications. This paper reviews industry and market information on VoIP, and discusses emerging standards that will influence the development of wireless IP applications. New customer premise equipment (CPE) devices and hand-sets are presented along with the emerging set of applications and services currently available and in development. A discussion of network architectures necessary to provide wireless IP applications including voice over IP is also presented.

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.

In this paper, we analyze the jitter and packet loss behavior of voice over Internet protocol (VoIP) traffic by means of networks measurements and simulations results. As result of these analyses, we provide a detailed characterization and accurate modeling of these Quality of Service (QoS) parameters. Our studies have revealed that VoIP jitter can be modeled by selfsimilar and multifractal models. We present a methodology for simulating packet loss. Besides, we found relationships between Hurst parameter (H) with packet loss rate (PLR).

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.