A Frame-Based Selective Encryption Method for Real Time Video Transmission on VoIP

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

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.

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 number of active users using Voice over IP (VoIP) services has an increasing tendency. With an expanding number of users, there is also a rapid increase in the number of hackers interested in attacking the VoIP communication system. This paper aims at detecting malicious SIP traffic and also deals with the security of the VoIP architecture issue. It is not a trivial matter to secure the VoIP system because exploiting the vulnerabilities of IP based telecommunication systems have increased. It is crucial to develop a tool that would be able to detect these attacks, analyse collected data, monitor attackers progress and to prepare an effective way of how to defend against VoIP attackers. That was the primary motivation why we have decided to develop our honeypot solution which can detect attacks on VoIP infrastructure, and it is adapted to the new security threats and which is designed according to the needs of the telecommunications market. Our VoIP honeypot is implemented purely in JAVA programming language and is capable of capturing and processing various types of attacks. The whole project is based on a Linux distribution, ready for the easiest deployment because it is prepared as a virtual machine image.

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

In 2006, the University of Alaska Anchorage (UAA) upgraded the telephone system at its main campus in Anchorage from a traditional private branch exchange (PBX) architecture to a Voice over Internet Protocol (VoIP) system. This case describes the organizational decisions that led to the change; the scope and the process of upgrading; and the current status of the new VoIP system. The actual migration to VoIP was completed less than a year after the start of the project. The transition process went smoothly. User satisfaction with the performance of the VoIP system is very high. Based on extensive interviews with decision makers and the technical personnel involved, this case also describes financial considerations (including “creative” ways to stretch a limited budget), outsourcing considerations, training related issues, as well as lessons learned.

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.

The usage of voice over IP (VoIP) traffic in IEEE 802.11 wireless networks is expected to increase in the near future due to widely deployed 802.11 wireless networks and VoIP services on fixed lines. However, the quality of service (QoS) of VoIP traffic in wireless networks is still unsatisfactory. In this thesis, I identify several sources for the QoS problems of VoIP traffic in IEEE 802.11 wireless networks and propose solutions for these problems. The QoS problems discussed can be divided into three categories, namely, user mobility, VoIP capacity, and call admission control. User mobility causes network disruptions during handoffs. In order to reduce the handoff time between Access Points (APs), I propose a new handoff algorithm, Selective Scanning and Caching, which finds available APs by scanning a minimum number of channels and furthermore allows clients to perform handoffs without scanning, by caching AP information. I also describe a new architecture for the client and server side for seamless IP layer handoffs, which are caused when mobile clients change the subnet due to layer 2 handoffs. I also present two methods to improve VoIP capacity for 802.11 networks, Adaptive Priority Control (APC) and Dynamic Point Coordination Function (DPCF). APC is a new packet scheduling algorithm at the AP and improves the capacity by balancing the uplink and downlink delay of VoIP traffic, and DPCF uses a polling based protocol and minimizes the bandwidth wasted from unnecessary polling, using a dynamic polling list. Additionally, I estimated the capacity for VoIP traffic in IEEE 802.11 wireless networks via theoretical analysis, simulations, and experiments in a wireless test-bed and show how to avoid mistakes in the measurements and comparisons. Finally, to protect the QoS for existing VoIP calls while maximizing the channel utilization, I propose a novel admission control algorithm called QP-CAT (Queue size Prediction using Computation of Additional Transmission), which accurately predicts the impact of new voice calls by virtually transmitting virtual new VoIP traffic.

The use of Voice over IP (VoIP) applications involves a number of security threats and usability issues, leading to possible breaches of security and privacy. With the adoption of future peer-to-peer communication systems, the challenges grow even more as we rely on untrusted peers to access the service. We are developing a peer-to-peer VoIP system which features techniques for improving the security and privacy of users in future networks. However, as the threats are seldom well understood, presenting them in a usable manner is problematic. Implemented on a mobile device, the small user interface provides additional challenges for the end user. Via interviews, a questionnaire and usability testing, we seek to improve both the usability of managing and understanding the additional security, as well as the overall user experience of the emerging application.

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.

VoIP(Voice over IP), so called Internet phone, provides several unique advantages, for instance, inexpensive long distance call service than the existing wire telephone network and various multimedia services from Internet network added to voice call service. This has been widely discussed and has been a research topic throughout the developed countries in the recent years. The paper is intended to describe and analyze the feasibility of the results of the VoIP system designed if implemented in a campus network like Dhaka University telephone network. Thus the convergence strategy of an IP PABX system with the existing wire telephone network has been discussed. While taking this into consideration, upgrading method of the existing system using modern network appliances has been described and at the same time new network topology connecting the present IP network with the telephone network has been designed.

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

Voice over IP (VoIP) is nowadays a common component of multimedia converged networks. In this paper, Quality of Service (QoS) analysis of VoIP traffic is presented in a multimedia network having other types of network traffic also deployed i.e. video, database and web. In such traffic occupied network, VoIP is configured with different codecs and frame counts per packet to observe the effect of these variations on the QoS of VoIP in terms of queuing delay, jitter, packet end-to-end delay and packet delay variation (end-to-end jitter) metrics. The significance of this paper is the emphasis on QoS evaluations in a VoIP deployment while observing the effects of variations in voice codecs and packet lengths. Many previous research papers focus on the perceived speech quality and packet losses; whereas queuing delays and end-to-end latency are usually not considered when analysis is done for different codecs and packet lengths at a time. We consider three main voice codecs (G.711, G.729 and G.723.1) and three frame counts per packet values (10, 25 and 50 voice frames per packet) under each codec. The simulation results show that G.723.1 voice encoding experiences higher queuing delays and queue delay variations (jitter) for higher numbers of frames per packet. However, the least packet end-to-end delay is observed in voice packets encoded with G.729 codec for all configured values of frames per packet.

Voice over Internet Protocol (VoIP) is no longer only feasible in wired networks, but also applicable through wireless networks as prevalence of wireless networks grows with the technology maturity and rising demand. While making a VoIP call over WiFi wireless networks is applicable, this is not a service with guaranteed quality. With limited bandwidth, an overloaded AP may cause wireless users to experience delayed packets, high packet loss rate, and even low throughput. IEEE 802.11 wireless Local Area Network (WLAN) was originally designed only for data traffic transmission, and therefore deficiency in real-time traffic transmission, such as VoIP applications, is inevitable. Not being able to timely transmit real-time packets not only fails to meet the demand of users, but also increases unnecessary loading in the wireless network. Industries and academies have made lots of efforts on providing QoS for real-time voice traffic which usually coexists with other classes of traffic. In this thesis, a call admission control mechanism is proposed which not only timely monitors the wireless networks, but also dynamically adjusts the number of admissible clients to wireless networks. New VoIP users not only know the network condition before entering it, but also accurately predict transmission quality through sending probing packets. With the mechanism, existing VoIP users in the wireless networks have their service protected, and potential VoIP users can expect good quality of service once admitted.