ITC/USA 2011 Conference Proceedings / The Forty-Seventh Annual International Telemetering Conference and Technical Exhibition / October 24-27, 2011 / Bally's Las Vegas, Las Vegas, Nevada / Bidirectional networked radio frequency (RF) communications between the ground and test articles are quickly becoming a normal mode of operation. Not only can devices be remotely controlled, but other networking technologies are emerging into flight test. Voice over IP (VoIP) is ubiquitous in the workplace and in homes, but it presents unique challenges when used to communicate between test articles. This paper presents some issues to be considered and test results to help aid deployment of VoIP systems in network-based test systems such as iNET's Telemetry Network System (TmNS).

Siendo ex-alumnos de la Universidad Ricardo Palma, ahora Bachilleres de la carrera profesional de Ingeniería Electrónica, tenemos la gran responsabilidad de ejercer nuestra profesión de manera adecuada, esto quiere decir, usar el poder del conocimiento adquirido para el bien y el desarrollo. El tener presente esta premisa ayuda a comprender la justificación de un proyecto enfocado a las zonas rurales de nuestro territorio. Cursos como Comunicaciones Rurales muestran realidades ignoradas por muchos en un país centralizado como es el Perú. Parte del desarrollo consiste en romper barreras sociales e ideologías que nos separan a pesar que todos pertenecemos a la misma patria y, de cierta forma, dotar de tecnología a las zonas que lo necesiten intercomunicándonos unso a otros ayudará a la integración. Reconozcamos que el crecimiento y beneficio de un grupo social determinado no traerá nayor trascendencia, los verdaderos cambios positivos significativos para el Perú llegarán con la descentralización, la unión y el apoyo del conjunto, siendo la tecnología de la información vital para este propósito.

In the presented work we focus on both implementation and protocol of the voice over IP application Skype. We analyse several versions of Skype clients and deduce inner workings of the Skype protocol. We present details about the cryptographic primitives used by the Skype clients and their impact on the security of the communication. We emphasize several places of suspicious leaks of the internal states of random generators and deduce rules for the detection of the Skype traffic on the firewall. In the end, we mention a simple enhancement of the current Skype clients that, in practice, can detect an ongoing eavesdropping.

Depuis l'invention du premier téléphone par Alexandre Graham Bell en 1869, la téléphonie n'a cessé d'évoluer : de la commutation de circuit à la commutation par paquet, d'un réseau fixe à un réseau mobile. Plusieurs architectures ont été crées où la voix est combinée aux données et à l'imagerie pour être transportées sur un seul réseau data. La nature de ces réseaux ouverts a un impact sur la voix en terme de sécurité. D'où le besoin imminent de sécuriser la voix tout en assurant une bonne qualité de service à la voix aussi bien dans un réseau fixe que dans un réseau mobile ou IP. Des solutions de sécurisation sont proposées pour les données. Mais des solutions partielles voire incomplètes sont proposées pour la voix. Dans un premier temps, nous définissons les besoins de la sécurisation de la téléphonie et les services de sécurité nécessaires à son déploiement. Ainsi, nous analysons la sécurité offerte par les différents réseaux offrant un service de téléphonie, à savoir la sécurité dans le réseau téléphonique traditionnel (RTC et RNIS), dans les réseaux mobiles (GSM et UMTS), et dans le réseau de données IP avec les deux architectures prépondérantes H.323 et SIP. Ceci va nous permettre de comparer les solutions de sécurité offertes par ces architectures de téléphonie et de pouvoir aborder d'une part, leur avantages et inconvénients et d'autre part, les exigences qu'ils ne peuvent pas satisfaire. Cette analyse nous conduit à un résultat éloquent qui est l'absence d'une solution de sécurité complète qui répond aux exigences de la téléphonie et qui permet d'effectuer un appel sécurisé de bout en bout.

<p>Current Voice over IP (VoIP) services are regarded less secure than the traditional public switched telephone network (PSTN). This is due to the fact that VoIP services are frequently deployed in an relatively open environment so that VoIP infrastructures can be easily accessed by potential attackers. Furthermore, current VoIP services heavily rely on other public Internet infrastructures shared with other applications. Thus, the vulnerabilities of these Internet infrastructures can affect VoIP applications as well. Nevertheless, deployed in a closed environment with independent protocols, PSTN has never faced similar risks.</p><p>The main goal of this licentiate thesis is the discussion of security issues of the Session Initiation Protocol (SIP), which serves as a signalling protocol for VoIP services. This work especially concentrates on the security risks of SIP related to performance. These risks can be exploited by attackers in two ways: either actively or passively. The throughput of a SIP proxy can be actively manipulated by attackers to reduce the availability of services. It is defined as Denial of Service (DoS) attacks. On the other hand, attackers can also profile confidential information of services (e.g., calling history) by passively observing the performance of a SIP proxy. It is defined as a timing attack. In this thesis, we carefully studied four concrete vulnerabilities existing in current SIP services, among which, three of them can lead to DoS attacks and one can be exploited for timing attacks. The results of our experiments demonstrate that these attacks can be launched easily in the real applications.</p><p>Moreover, this thesis discusses different countermeasure solutions for the attacks respectively. The defending solutions have all in common that they are influencing the performance, by either enhancing the performance of the victim during a DoS attack, or abating the performance to obscure the time characteristic for a timing attack. Finally, we carefully evaluated these solutions with theoretical analyses and concrete experiments.</p>

Objective quality evaluation is a key element for the success of the emerging Voice over IP (VoIP) technologies. Although there are extensive economic incentives for the convergence of voice, data, and video networks, packet networks such as the Internet have inherent incompatibilities with the transport of real time services. Under this paradigm, network planners and administrators are interested in ongoing mechanisms to measure and ensure the quality of these real time services. Objective quality assessment algorithms can be broadly divided into a) intrusive (methods that require a reference signal), and b) non intrusive (methods that do not require a known reference signal). The latter group, typically requires knowledge of the network conditions (level of delay, jitter, packet loss, etc.), and that has been a very active area of research in the past decade. The state of the art methods for objective non-intrusive quality assessment provide high correlations with the subjective tests. Although good correlations have been achieved already for objective non-intrusive quality assessment, the current large voice transport networks are in a hybrid state, where the necessary network parameters cannot easily be observed from the packet traffic between nodes. This thesis proposes a new process, the Network Conditions Estimator (NCE), which can serve as bridge element to real-world hybrid networks. Two classifications systems, an artificial neural network and a C4.5 decision tree, were developed using speech from a database collected from experiments under controlled network conditions. The database was composed of a group of four female speakers and three male speakers, who conducted unscripted conversations without knowledge about the details of the experiment. Using mel frequency cepstral coefficients (MFCCs) as the feature-set, an accuracy of about 70% was achieved in detecting the presence of jitter or packet loss on the channel. This resulting classifier can be incorporated as an input to the E-Model, in order to properly estimate the QoS of a network in real time. Additionally, rather than just providing an estimation of subjective quality of service provided, the NCE provides an insight into the cause for low performance.

The cellular networks are evolving to meet the future requirements of data rate,coverage and capacity. The fourth generation mobile communication system, LTEhas been developed to meet these goals. LTE uses multiple antenna features andlarger bandwidths in order to accomplish this task. These features will furtherextend the requirements of data rate, coverage, latency and flexibility. LTE also utilizes the varying quality of the radio channel and the interferencefrom other transmitters by adapting the data rate to the instantaneous channelquality at all the time. This is typically referred to as Link Adaptation. Thelink adaptation fails from time to time due to the varying channel quality as wellas the interference from other transmitters. In order to counteract these failures,retransmission methods are employed. These methods detect the errors on thereceiver side and signals the transmitter for the retransmission of the erroneousdata. The efficiency of link adaptation increases if combined with a properly designedretransmission scheme at the expense of delays due to retransmissions. This master thesis focuses on the study of the retransmission schemes with fasterfeedback, resulting in a reduction in delay. The feedback is generated by makingan early estimate of the decoding outcome and sending it early to the transmitterresulting in faster retransmission. This is important in certain applications wherethe data transmission is intolerant to delays.The thesis work shows by system performance simulations that fast packet retransmission,precisely called Early HARQ Feedback, significantly affects the systemperformance together with the utilization of the link adaptation. The study alsoshows that the link adaptation, in certain scenarios, can be optimized to improvethe system performance. In that respect, it is also possible to increase the numberof retransmissions within the same resource utilization. That optimization is basicallycalled aggressive link adaptation. Consequently, Early HARQ Feedback incombination with aggressive link adaptation provides a large improvement in thedownlink performance of the studied cases.

There are two main architectures in VOIP system at present. First is peer-to-peer, it has highly scalable, fault-tolerant and also can lighten the number and reliance of server. But there is no standard protocol between peers with different architecture, cause the clients unable to communication with each other. This problem can be solved by communicating from one P2P network to another. Second is Client-Server, it has mass of research data, lots of actual products, and standard protocol. This architecture is the most perfect one with simple structure, easy to maintain, lower response time than peer-to-peer structure, and has a variety of additional services, for instance Voice Mail, conference call, etc. All the client need is to obey sip standard protocol and it can register to any sip proxy to make a phone call. The disadvantage is no server no use. These two architectures have both good side and bad side, none of them is absolutely perfect. Our thesis is proposed a all new idea about ¡§Hybrid¡¨, this idea combine P2P and Client-Server architecture together to design a flexible soft phone that can be used is normal condition to register to a proxy, or setup a P2P network instantly in our own local area network. Finally our DCHS Mechanism is workable even when the sip proxy is maintaining or failure the client can use this mechanism to call any other user outside the P2P network by sharing other peers¡¦ call history.

Because of the maturation of the VoIP technique, VoIP can not only satisfy the basic requirement of telecommunication but also provide multimedia communication services. As a result, it is very attractive in recent years. Through VoIP, the cost of communication can be saved. It can be very competitive. In addition, VoIP can be combined with PSTN (Public Switched Telephone Network). This helps traditional PSTN users to be able to use traditional telephones to make VoIP calls.Besides, VoIP can also extend other services. It can achieve diversification of services, comfortable using and reducing the cost requirement. Moreover, with the increasing of the VoIP population, the traditional method using single server is unable to afford so much loading. It is possible that the large load makes the service stop anytime. This makes the usability and the reliability decrease. To make the VoIP service work anytime, we implement a method in both client side and server side to achieve the goal of continuous providing of the service. From this implementation, the service of VoIP can be provided anytime. The users, however, have no need to be aware of the different operation style in VoIP.

eBay chief executive, Meg Whitman, at a press conference expressed to the investors that ¡§communications plays a key role in e-commerce and society. This makes Skype become the most suitable cooperator with eBay.¡¨ When integrating with Skype, eBay makes buyers and sellers communicate with each other through VoIP. This removes the biggest obstacle between buyers and sellers and achieves an ¡§unparalleled e-commerce and communications engine.¡¨ ¡§eBay with Skype¡¨ is the best example of 3PCC with e-commerce. 3PCC is a model that allows an entity (which is called controller) to manage and set up a communication between two or more other parties. It has already existed in the PSTN for a long time. Although there are many applications designed for SIP, they are not 3PCC with e-commerce model. Therefore, we attempt to design an application that integrating 3PCC with e-commerce. In this paper, we not only introduce how 3PCC is achieved but also express how REFER (a new method in SIP) can be used for replacing the traditional 3PCC mechanism in chapter 2. Chapter 3 will introduce the S/H development framework, the flows of SIP and the functions or libraries related to the Gateway. In chapter 4, we will first explain the design concept about our systems and then express how we implement the system. These include the website database structures, the Gateway programs and the packet analysis and verification. Finally, we will conclude this paper in Chapter 5. In addition, we will show the system and operation guide in appendix.