An overview of some techniques to exploit VoIP over WLAN

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

After analyzing SRTP (Secure Real-time Transport Protocol) and SIP (session initiation protocole), we identified the need to secure VoIP (Voice over IP) network. Our study assesses the risk of attacks on the VoIP network. Additionally, it offers solutions to the existing VoIP network vulnerabilities while issuing Instant Messages. Using cryptography, we present a general proposal for secure VoIP instant messages and SIP protocol.

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.

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.

Visual and Vocal communication can be transferred through Circuit switched Network or Packet Switched Network. Public Switched Telephone Network (PSTN) is not an affordable option therefore over existing packet switched network, Voice over Internet Protocol (VoIP) has become a preferable alternative due to its reduced cost. However, despite its reduced cost it has so many challenges which affect its successful deployment. This is because; the quality of VoIP is mainly affected by jitter, delay, packet loss and some other parameters. This research was carried out to evaluate voice quality in VoIP experimentally under different scenarios using OPNET network simulator. A VoIP network was simulated using Riverbed modeler academic edition 17.5 and the behavior and quality of VoIP was studied and analyzed under different scenarios. The results of the analysis and the performance evaluation are presented in this paper. This work can guide researchers and designers to design a network for VoIP services and its deployment. It can also guide the operators to choose speech compression technique for better voice quality.

Voice over Internet Protocol (VoIP) is a popular communication service nowadays. VoIP reduces the cost of call transmission by passing voice and video packets through the available bandwidth for data packets through Internet protocol. The quality of the VoIP signal is degraded due to the various network impairments. The proposed scheme, interpolated finite impulse response, is implemented as post-processor after decoding the signal in VoIP system. The performance of the proposed scheme is evaluated for various network conditions. The results of the proposed scheme are measured with the objective measurement methods for signal quality evaluation. The performance of the proposed system is compared with the existing techniques for quality improvement in VoIP system. The results show much improvement in speech quality with the proposed scheme in comparison to other similar schemes.

The rapid emergence of voice over Internet Protocol (VoIP) networks and their enticing appeal to a broad base of customers are driven by economic incentives, improved productivity, and creation of new services. Thus, VoIP networks provide an attractive value proposition to service providers, as they enable additional revenue opportunities through new services coupled with reduced network operational costs. An effective operations support system (OSS) is needed to manage a VoIP network by performing a full range of network management tasks (fault, configuration, accounting, performance, and security [FCAPS]). However, its primary goal is to ensure that customers do not perceive any difference in the transition from the public switched telephone network (PSTN) to converged services architecture in terms of toll-quality voice and 99.999% network availability. An OSS for managing VoIP networks requires greater focus on service-oriented operations and management. In this paper, we discuss the challenges that are unique to managing VoIP networks. Specifically, we propose an OSS for VoIP networks with emphasis on provisioning, monitoring, capacity planning, and service creation.

Recently, with the expansion and advent of the wireless Internet such as WiFi, mobile WiMax, and mobile peer-to-peer (P2P) voice over IP (VoIP) communication through P2P overlay network in all-IP area has been deeply investigated as a research area. When the connected peer terminal leaves hot spot of all-IP area, it will be lost the communication session. In this paper, we propose seamless voice connection over P2P mobile VoIP and conventional wireless network. In this case, the concept of vertical handoff is not the handoff of IP layer but that of session layer to create new VoIP session in latency before the previous session ends. We have two kind of wireless communication network; one is all-IP and the other is CDMA cellular and assumes that the mobile node has the functionality to communicate with each other network. We certain the dynamically adjusted trigger method is better than the fixed trigger in the success of new call setup processing and the decrease of unnecessary signaling in cellular network. We think that this work is able to achieve seamless and low-cost communication between P2P VoIP and cellular network.

IP Telephony, Internet Telephony, Digital Phone or often also called VoIP (Voice Over Internet Protocol) is a technology that allows long-distance voice conversations with the Internet. The increasing number of VoIP users and other IP-based multimedia streaming services naturally raises security issues. Many users are likely to lose their privacy in communication. To overcome this security problem a security system must be implemented. Implementing a security system will use VPN Gateway using SSL and TLS encryption on the VoIP server. The VPN Gateway method is used to build a private network so that only certain users can use the private network. The TLS method is used to secure a user signaling session to the server. From the test results obtained that the VoIP server that uses VPN Gateway and TLS on the server can overcome the attacks e.g., eavesdropping, attacking authentication, teardown session, and denial of service.

Dynamic voice spammers detection using Hidden Markov Model for Voice over Internet Protocol network

An energy efficient authenticated key agreement protocol for SIP-based green VoIP networks

Playout delay adaptation algorithms are often used in real time voice communication over packet-switched networks to counteract the effects of network jitter at the receiver. Whilst the conventional algorithms developed for silence-suppressed speech transmission focused on preserving the relative temporal structure of speech frames/packets within a talkspurt (intertalkspurt adaptation), more recently developed algorithms strive to achieve better quality by allowing for playout delay adaptation within a talkspurt (intratalkspurt adaptation). The adaptation algorithms, both intertalkspurt and intratalkspurt based, rely on short term estimations of the characteristics of network delay that would be experienced by up-coming voice packets. The use of novel neural networks and fuzzy systems as estimators of network delay characteristics are presented in this paper. Their performance is analyzed in comparison with a number of traditional techniques for both inter and intratalkspurt adaptation paradigms. The design of a novel fuzzy trend analyzer system (FTAS) for network delay trend analysis and its usage in intratalkspurt playout delay adaptation are presented in greater detail. The performance of the proposed mechanism is analyzed based on measured Internet delays. Index Terms-Fuzzy delay trend analysis, intertalkspurt, intratalkspurt, multilayer perceptrons (MLPs), network delay estimation, playout buffering, playout delay adaptation, time delay neural networks (TDNNs), voice over Internet protocol (VoIP).

Voice over Internet Protocol (VoIP) networks have recently emerged as a promising telecommunication medium for transmitting voice signal. One of the essential aspects that interests researchers is how to estimate the quality of transmitted voice over VoIP for several purposes such as design and technical issues. Two methodologies are used to evaluate the voice, which are subjective and objective methods. In this study, the authors propose a non‐intrusive machine learning‐based (NIML) objective method to estimate the quality of voice. In particular, they build a training set of parameters – from the network and the voice itself – along with the quality of voices as labels. The voice quality is estimated using the perceptual evaluation of speech quality (PESQ) method as an intrusive algorithm. Then, the authors use a set of classifiers to build models for estimating the quality of the transmitted voice from the training set. The experimental results show that the classifier models have a valuable performance where Random Forest model has superior results compared to other models of precision 94.1%, recall 94.2%, and receiver operating characteristic area 99.2% as evaluation metrics.

Design and realization of ad-hoc VoIP with embedded p-SIP server