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).

ITC/USA 2012 Conference Proceedings / The Forty-Eighth Annual International Telemetering Conference and Technical Exhibition / October 22-25, 2012 / Town and Country Resort & Convention Center, San Diego, California / The expected efficiency of network-based telemetry systems vs. the tried and true PCM-based approaches is a debated topic. This paper chooses to use a lighthearted voice to pull the two sides of the "war" to a table of negotiation based on metrics. Ultimately, focusing on metrics that truly define efficiency is the key to understanding the varying points of view. A table of these metrics along with the "why and when" criteria for their use is presented based on historic mathematical information theory, true flight test data requirements, and lab analysis. With these metrics, the negotiation and reasonable compromises in the war may become clear. In other words, this paper attempts to provide a methodology that can be used by the community to aid in choosing the appropriate (or good enough) technologies for current and future telemetry testing demands.

With the rising popularity of Voice over IP (VoIP) services, understanding the effects of a global network on Quality of Service is critical for the providers of VoIP applications. This thesis builds on a model that analyzes the round trip time, packet delay jitter, and packet loss between endpoints on an Autonomous System (AS) level, extending it by mapping AS pairs onto an Internet topology. This model is used to produce a mean opinion score estimate. The mapping is introduced to reduce the size of the problem in order to improve computation times and improve accuracy of estimates. The results of testing show that estimating mean opinion score from this model is not desirable. It also shows that the path mapping does not affect accuracy, but does improve computation times as the input data grows in volume.

Das Ziel dieser Arbeit ist es, ein Handover-Verfahren für ein Siemens Handset zu entwickeln. Die Entwicklungsumgebung beruht auf den Wireless-LAN Standards 802.11 der IEEE (Institute of Electrical and Electronics Engineers). Dabei liegen die Schwerpunkte auf den Standardisierungen 802.11f und 802.11i, wobei sich eine neue Arbeitsgruppe (IEEE 802.11r) direkt mit dem Thema "Handover" beschäftigen wird. Das Handset soll selbständig die Verwaltung und Einleitung des Handovers übernehmen und lediglich insofern vom Access Point unterstützt werden, dass dieser als Informationssammler dient und somit Entscheidungshilfen geben kann.

This master thesis deals with the design and implementation of an application for analyzing Voice over IP quality using NetFlow. In the beginning, there is summarized basic information about VoIP technology and NetFlow - its principles, the most used protocols, factors that have influence on call quality and call quality rating methods. Later there is presented proposal of application and then described its implementation. The created application was tested on samples, which simulate calls in network with delays and packet-loss. Within testing was made the comparison with commercial application and the results are discussed.

Voice Over IP (VoIP) telephony is becoming widespread, and is often integrated into computer networks. Because of his, it is likely that malicious software will threaten VoIP systems the same way traditional computer systems have been attacked by viruses, worms, and other automated agents. While most users have become familiar with email spam and viruses in email attachments, spam and malicious traffic over telephony currently is a relatively unknown threat. VoIP networks are a challenge to secure against such malware as much of the network intelligence is focused on the edge devices and access environment. A novel security architecture is being developed which improves the security of a large VoIP network with many inexperienced users, such as non-IT office workers or telecommunication service customers. The new architecture establishes interaction between the VoIP backend and the end users, thus providing information about ongoing and unknown attacks to all users. An evaluation of the effectiveness and performance of different implementations of this architecture is done using virtual machines and network simulation software to emulate vulnerable clients and servers through providing apparent attack vectors.

Approved for public release, distribution is unlimited / Converging voice and data networks has the potential to save money and is the main reason Voice over Internet Protocol (VoIP) is quickly becoming mainstream in corporate America. The potential VoIP offers to more efficiently utilize the limited connectivity available to ships at sea makes it an attractive option for the Navy. This thesis investigates the usefulness of VoIP for the communications needs of a unit level ship. This investigation begins with a review of what VoIP is and then examines the ship to shore connectivity for a typical unit level ship. An OMNeT++ model was developed and used to examine the issues that affect implementing VoIP over this type of link and the results are presented. / Lieutenant Commander, United States Navy

Das Ziel dieser Arbeit ist es, ein Handover-Verfahren für ein Siemens Handset zu entwickeln. Die Entwicklungsumgebung beruht auf den Wireless-LAN Standards 802.11 der IEEE (Institute of Electrical and Electronics Engineers). Dabei liegen die Schwerpunkte auf den Standardisierungen 802.11f und 802.11i, wobei sich eine neue Arbeitsgruppe (IEEE 802.11r) direkt mit dem Thema "Handover" beschäftigen wird. Das Handset soll selbständig die Verwaltung und Einleitung des Handovers übernehmen und lediglich insofern vom Access Point unterstützt werden, dass dieser als Informationssammler dient und somit Entscheidungshilfen geben kann.

Θα περιγράψουμε τις βασικές λειτουργίες ενός VoIP δικτύου, τα συστατικά του μέρη, καθώς και τα πρωτόκολλα που είναι υπεύθυνα για την εγκατάσταση, τον έλεγχο και τον τερματισμό μιας VoIP υπηρεσίας-συνομιλίας. Τα πρωτόκολλα αυτά ονομάζονται πρωτόκολλα σηματοδοσίας. Τα πρωτόκολλα σηματοδοσίας για VoIP εφαρμογές και ιδιαίτερα το πρωτόκολλο SIP (Session Initiation Protocol) είναι το βασικό θέμα της παρούσας εργασίας. Συγκεκριμένα, έγινε ανάπτυξη ενός SIP User Agent, το λογισμικό του οποίου θα εκτελείται στο δικτυακό επεξεργαστή IXP425 της Intel, μαζί με τα απαραίτητα πρωτόκολλα για την κωδικοποίηση-αποκωδικοποίηση και μετάδοση δειγμάτων φωνής σε μορφή πακέτων δεδομένων. Το αποτέλεσμα αναμένεται να είναι ένα ολοκληρωμένο προϊόν (VoIP phone) για την πραγματοποίηση VoIP κλήσεων. / -

The rapid growth of the Internet and the proliferation of its new applications pose a serious challenge in network performance management and monitoring. The current Internet has no mechanism for providing feedback on network congestion to the end-systems at the IP layer. For applications and their end hosts, end-to-end measurements may be the only way of measuring network performance. Understanding the packet delay and loss behavior of the Internet is important for proper design of network algorithms such as routing and flow control algorithms, for the dimensioning of buffers and link capacity, and for choosing parameters in simulation and analytic studies. In this thesis, round trip time (RTT), one-way network delay and packet loss in the Internet are measured at different times of the day, using a Voice over IP (VoIP) device. The effect of clock skew on one-way network delay measurements is eliminated by a Linear Programming algorithm, implemented in MATLAB. Distributions of one-way network delay and RTT in the Internet are determined. It is observed that delay distribution has a gamma-like shape with heavy tail. It is tried to model delay distribution with gamma, lognormal and Weibull distributions. It is observed that most of the packet losses in the Internet are single packet losses. The effect of firewall on delay measurements is also observed.