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).

A research area that has recently gained great interest is the development of network architectures relating to the tracking of wireless VoIP devices. This is particularly so for architectures based on the popular Session Initiation Protocol (SIP). Previous work, however, in this area does not consider the impact of combined VoIP and tracking on the capacity and call set-up time of the architectures. Previous work also assumes that location information is always available from sources such as GPS, a scenario that rarely is found in practice. The inclusion of multiple positioning systems in tracking architectures has not been hitherto explored. It is the purpose of this thesis to design and test SIP-based architectures that address these key issues. Our first main contribution is the development of a tracking-only SIP based architecture. This architecture is designed for intermittent GPS availability, with wireless network tracking as the back-up positioning technology. Such a combined tracking system is more conducive with deployment in real-world environments. Our second main contribution is the development of SIP based tracking architectures that are specifically aimed at mobile wireless VoIP systems. A key aspect we investigate is the quantification of the capacity constraints imposed on VoIP-tracking architectures. We identify such capacity limits in terms of SIP call setup time and VoIP QoS metrics, and determine these limits through experimental measurement and theoretical analyses. Our third main contribution is the development of a novel SIP based location tracking architecture in which the VoIP application is modified. The key aspect of this architecture is the factor of two increase in capacity that it can accommodate relative to architectures utilizing standard VoIP. An important aspect of all our tracking architectures is the Tracking Server. This server supplies the location information in the event of GPS unavailability. A final contribution of this thesis is the development of novel particle-filter based tracking algorithms that specifically address the GPS intermittency issue. We show how these filters interact with other features of our SIP based architectures in a seamless fashion.

Call admission control (CAC) extends the capabilities of Quality of service (QoS) tools which protect voice traffic from the negative effects of other voice traffic. It does not allow oversubscription of a Voice over Internet Protocol (VoIP) network. To achieve better performance for efficient call admission control, various dynamic routings are being proposed. In the dynamic routing mechanism, the condition of the network is learned by observing the network condition via the probe packets and according to the defined threshold, routes are chosen dynamically. In such schemes, various combination of route selection is used such as two routes are used where one is fixed and other is random or two random routes are chosen and after observation one is chosen if it passes the test. Few schemes use a route history table along with the two random routes. But all have some issues like it selects random routes (not considering the number of hops), does not process memorization before admission threshold test, it calculates all selected paths regardless of the fact that they are selected or not, thereby wasting central processing unit (CPU) time and since these uses two routes so obviously the call admission probability is less. In this thesis work, a new dynamic routing scheme is proposed which considers a routing history table with endpoint admission control increasing the call admission probability, makes call establishment time faster and it saves valuable CPU resources. The proposed scheme considers a combination of three routes with routing history table--one is the direct route and the other two are selected randomly from all available routes and the routing history table is used to memorize the rejected calls. CAC tests like Admission Threshold were performed on the selected routes. Various parameters such as delay, packet loss, jitter, latency etc from the probe packets are used to carry out the tests. Performance of the proposed scheme with respect to other dynamic routing schemes is studied using a mathematical / analytical model. Also, effect of arrival rate probe packets on utilization, busy period, waiting period, acceptance probability of calls, probe packets, and the number of successful calls was also studied. / Thesis (M.S.)--Wichita State University, College of Engineering, Dept. of Electrical and Computer Engineering. / "July 2006."

Workshop Mensch-Computer-Vernetzung Stand und Perspektiven des Einsatzes einer "Voice over IP"-Lösung für die Telefonie am URZ der TU Chemnitz auf Grundlage der VOCAL-Server-Suite

A research area that has recently gained great interest is the development of network architectures relating to the tracking of wireless VoIP devices. This is particularly so for architectures based on the popular Session Initiation Protocol (SIP). Previous work, however, in this area does not consider the impact of combined VoIP and tracking on the capacity and call set-up time of the architectures. Previous work also assumes that location information is always available from sources such as GPS, a scenario that rarely is found in practice. The inclusion of multiple positioning systems in tracking architectures has not been hitherto explored. It is the purpose of this thesis to design and test SIP-based architectures that address these key issues. Our first main contribution is the development of a tracking-only SIP based architecture. This architecture is designed for intermittent GPS availability, with wireless network tracking as the back-up positioning technology. Such a combined tracking system is more conducive with deployment in real-world environments. Our second main contribution is the development of SIP based tracking architectures that are specifically aimed at mobile wireless VoIP systems. A key aspect we investigate is the quantification of the capacity constraints imposed on VoIP-tracking architectures. We identify such capacity limits in terms of SIP call setup time and VoIP QoS metrics, and determine these limits through experimental measurement and theoretical analyses. Our third main contribution is the development of a novel SIP based location tracking architecture in which the VoIP application is modified. The key aspect of this architecture is the factor of two increase in capacity that it can accommodate relative to architectures utilizing standard VoIP. An important aspect of all our tracking architectures is the Tracking Server. This server supplies the location information in the event of GPS unavailability. A final contribution of this thesis is the development of novel particle-filter based tracking algorithms that specifically address the GPS intermittency issue. We show how these filters interact with other features of our SIP based architectures in a seamless fashion.

Title page, table of contents and abstract only. The complete thesis in print form is available from the University of Adelaide Library. / Current best-effort packet-switched Internet is not a perfect environment for real-time applications such as transmitting voice-over the network (Voice Over Internet Protocol or VOIP). Due to the unlimited concurrent access to the Internet by users, the packet loss problem cannot be avoided. Therefore, the VOIP based applications encompass problems such as "voice quality degradation caused by lost packets". The effects of lost packets are fundamental issues in real-time voice transmission over the current unreliable Internet. The dropped packets have a negative impact on voice quality and concealing their effects at the receiver does not deal with all of the drop consequences. It has been observed that in a very lossy network, the receiver cannot cope with all the effects of lost packets and thereby the voice will have poor quality. At this point the Active Networks, a relatively new concept in networking, which allows users to execute a program on the packets in active nodes, can help VOIP regenerate the lost packets, and improve the quality of the received voice. Therefore, VOIP needs special voice-packing methods. Based on the measured packet loss rates, many new methods are introduced that can pack voice packets in such a way that the lost packets can be regenerated both within the network and at the receiver. The proposed voice-packing methods could help regenerate lost packets in the active nodes within the network to improve the perceptual quality of the received sound. The packing methods include schemes for packing samples from low and medium compressed sample-based codecs (PCM, ADPCM) and also include schemes for packing samples from high compressed frame-based codecs (G.729). Using these packing schemes, the received voice has good quality even under very high loss rates. Simulating a very lossy network using NS-2 and testing the regenerated voice quality by an audience showed that significant voice quality improvement is achievable by employing these packing schemes. / http://proxy.library.adelaide.edu.au/login?url= http://library.adelaide.edu.au/cgi-bin/Pwebrecon.cgi?BBID=1147315 / Thesis (Ph.D.) -- University of Adelaide, School of Electrical and Electronic Engineering, 2004

Freedom is human’s natural instinct, which was limited by Ethernet and Fixed Telephony Era. With the emergence of new technologies like wireless fidelity (WiFi) and voice over IP (VoIP) humans once again have freedom of movement; which at the very same time provides enough reasons to change the market dynamics of communication industry. The buzz of Voice over WiFi (VoWiFi) in recent years indicates that VoWiFi is shaping up as the next big challenge to traditional telephony, not only due to cost, but also due to range of services and amount of freedom it can offer. However, at the very same time these technologies have evolved to threaten the well-established telephony markets. Enterprise solutions for VoWiFi require enhanced security mechanism and seamless handovers. To address security related issues Wi-Fi Alliance in conjunction with IEEE introduced an enhanced and interoperable security scheme called WiFi Protected Access (WPA). Real time services are sensitive to latency, hence requiring bounded delay time throughout an ongoing session. Handovers in WiFi networks can take fairly long time which real time services cannot tolerate. The problem is further elevated when WiFi networks are secured by using WPA Enterprise. In this thesis we will examine the complete handoff process in WiFi networks. The impact of handovers on VoIP traffic will also be observed. Following the detailed analysis some suggestions will be presented concerning how to reduce this handoff latency. / Friheten som ligger i människans natur begränsades av Ethernet och den fasta telefonin. Med uppkomsten av nya teknologier så som Wireless Fidelity (WiFi) och Voice over IP (VoIP) återfår människan den en gång förlorade friheten. Samtidigt kommer telekommunikationsindustrin att kunna ändras till sin struktur genom WiFi och VoIP. Integreringen av Voice over IP och WiFi , även mer känd som Voice over WiFi, (VoWiFi) har under senare år indikerat att det är en potentiell utmanare till traditionell telefoni inte bara ur ett kostnadsperspektiv utan också för att denna teknologi medför ökade möjligheter när det gäller nya tjänster. Dock återstår en del arbete för VoWiFi för att kunna rubba den fasta telefonin. Företagslösningar av denna teknologi kräver att säkerhetsaspekterna ses över dessutom måste seamless handover fungera på ett tillfredställande sätt. För att se över säkerhetsaspekterna har Wi-Fi Alliance i samarbete med IEEE introducerat säkehetsmekanismen WiFi Protected Access (WPA). Realtidstjänster är känsliga mot fördröjningar. Handover i ett WiFi nätverk kan ta relativt lång tid vilket är oacceptabelt för realtidstjänster. Problemet blir än mer påtagligt när WiFi-nätet är säkrat med hjälp av WPA. I denna exjobbsrapport kommer handoff processen för WiFi nätverk att behandlas. Effekten av handover för VoIP trafik kommer också att beskrivas. Resultat och analyser kommer att föreslås för hur man kan reducera handoff-fördröjningar.

Approved for public release, distribution is unlimited / In this thesis, Voice over Internet Protocol (VoIP) technology will be explored and a recommendation of the operational benefit of VoIP will be provided. A network model will be used to demonstrate improvement of voice End-to-End delay by implementing quality of service (QoS) controls. An overview of VoIP requirements will be covered and recommended standards will be reviewed. A clear definition of a Battle Group will be presented and an overview of current analog RF voice technology will be explained. A comparison of RF voice technology and VoIP will modeled using OPNET Modeler 9.0. / Lieutenant, United States Navy

Thesis (M. Tech. Engineering: Electrical--Vaal University of Technology. / This dissertation discusses the design and implementation of a voice over internet protocol system for the multi-user enterprise. It is limited to small to medium enterprises of which the Vaal University of Technology is an example. Voice communications over existing Internet protocol networks are governed by standards, and to develop such a system it is necessary to have a thorough understanding of these standards. Two such standards namely the International Telecommunications Unions H.323 and the Internet Engineering Task Force's SIP were evaluated and compared to each other in terms of their complexity, extensibility and scalability as well as the services they offer. Based on these criteria it was decided to implement a SIP system. A SIP network consists of application software that act as clients and servers, as well as hardware components such as a proxy and redirect and registrar or location servers that allow users of this network to call each other on the data network. Gateways enable users of the network to call regular public switched telephone network numbers. A test network was set up in the laboratory that contained all the hardware and software components. This was done to understand the installation and configuration options of the different software components and to determine the suitability and interoperability of the software components. This network was then migrated to the network of the Vaal University of Technology which allowed selected users to test and use it. Bandwidth use is a major point of contention, and calculations and measurements showed that the codec being used during the voice call is the determining factor. This SIP system is being used on a daily basis and the users report excellent audio quality between soft phones and soft phones, soft phones and normal telephones and even cellular phones.

Tese de mestrado. Redes e Serviços de Comunicação. Faculdade de Engenharia. Universidade do Porto. 2006