429 research outputs found

    Frame delay and loss analysis for video transmission over time-correlated 802.11A/G channels

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    Doctor of Philosophy

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    dissertationThe wireless radio channel is typically thought of as a means to move information from transmitter to receiver, but the radio channel can also be used to detect changes in the environment of the radio link. This dissertation is focused on the measurements we can make at the physical layer of wireless networks, and how we can use those measurements to obtain information about the locations of transceivers and people. The first contribution of this work is the development and testing of an open source, 802.11b sounder and receiver, which is capable of decoding packets and using them to estimate the channel impulse response (CIR) of a radio link at a fraction of the cost of traditional channel sounders. This receiver improves on previous implementations by performing optimized matched filtering on the field-programmable gate array (FPGA) of the Universal Software Radio Peripheral (USRP), allowing it to operate at full bandwidth. The second contribution of this work is an extensive experimental evaluation of a technology called location distinction, i.e., the ability to identify changes in radio transceiver position, via CIR measurements. Previous location distinction work has focused on single-input single-output (SISO) radio links. We extend this work to the context of multiple-input multiple-output (MIMO) radio links, and study system design trade-offs which affect the performance of MIMO location distinction. The third contribution of this work introduces the "exploiting radio windows" (ERW) attack, in which an attacker outside of a building surreptitiously uses the transmissions of an otherwise secure wireless network inside of the building to infer location information about people inside the building. This is possible because of the relative transparency of external walls to radio transmissions. The final contribution of this dissertation is a feasibility study for building a rapidly deployable radio tomographic (RTI) imaging system for special operations forces (SOF). We show that it is possible to obtain valuable tracking information using as few as 10 radios over a single floor of a typical suburban home, even without precise radio location measurements

    Packet Loss in Terrestrial Wireless and Hybrid Networks

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    The presence of both a geostationary satellite link and a terrestrial local wireless link on the same path of a given network connection is becoming increasingly common, thanks to the popularity of the IEEE 802.11 protocol. The most common situation where a hybrid network comes into play is having a Wi-Fi link at the network edge and the satellite link somewhere in the network core. Example of scenarios where this can happen are ships or airplanes where Internet connection on board is provided through a Wi-Fi access point and a satellite link with a geostationary satellite; a small office located in remote or isolated area without cabled Internet access; a rescue team using a mobile ad hoc Wi-Fi network connected to the Internet or to a command centre through a mobile gateway using a satellite link. The serialisation of terrestrial and satellite wireless links is problematic from the point of view of a number of applications, be they based on video streaming, interactive audio or TCP. The reason is the combination of high latency, caused by the geostationary satellite link, and frequent, correlated packet losses caused by the local wireless terrestrial link. In fact, GEO satellites are placed in equatorial orbit at 36,000 km altitude, which takes the radio signal about 250 ms to travel up and down. Satellite systems exhibit low packet loss most of the time, with typical project constraints of 10−8 bit error rate 99% of the time, which translates into a packet error rate of 10−4, except for a few days a year. Wi-Fi links, on the other hand, have quite different characteristics. While the delay introduced by the MAC level is in the order of the milliseconds, and is consequently too small to affect most applications, its packet loss characteristics are generally far from negligible. In fact, multipath fading, interference and collisions affect most environments, causing correlated packet losses: this means that often more than one packet at a time is lost for a single fading even

    Quality aspects of Internet telephony

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    Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

    Roaming Real-Time Applications - Mobility Services in IPv6 Networks

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    Emerging mobility standards within the next generation Internet Protocol, IPv6, promise to continuously operate devices roaming between IP networks. Associated with the paradigm of ubiquitous computing and communication, network technology is on the spot to deliver voice and videoconferencing as a standard internet solution. However, current roaming procedures are too slow, to remain seamless for real-time applications. Multicast mobility still waits for a convincing design. This paper investigates the temporal behaviour of mobile IPv6 with dedicated focus on topological impacts. Extending the hierarchical mobile IPv6 approach we suggest protocol improvements for a continuous handover, which may serve bidirectional multicast communication, as well. Along this line a multicast mobility concept is introduced as a service for clients and sources, as they are of dedicated importance in multipoint conferencing applications. The mechanisms introduced do not rely on assumptions of any specific multicast routing protocol in use.Comment: 15 pages, 5 figure

    Wireless Communication Options for a Mobile Ultrasound System

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    A mobile ultrasound system has been developed, which makes ultrasound examinations possible in harsh environments without reliable power sources, such as ambulances, helicopters, war zones, and disaster sites. The goal of this project was to analyze three different wireless communication technologies that could be integrated into the ultrasound system for possible utilization in remote data applications where medical information may be transmitted from the mobile unit to some centralized base station, such as an emergency room or field hospital. By incorporating wireless telecommunication technology into the design, on site medical personnel can be assisted in diagnostic decisions by remote medical experts. The wireless options that have been tested include the IEEE 802.11g standard, mobile broadband cards on a 3G cellular network, and a mobile satellite terminal. Each technology was tested in two phases. In the first phase, a client/server application was developed to measure and record general information about the quality of each link. Four different types of tests were developed to measure channel properties such as data rate, latency, inter-arrival jitter, and packet loss using various signal strengths, packet sizes, network protocols, and traffic loads. In the second phase of testing, the H.264 Scalable Video Codec (SVC) was used to transmit real-time ultrasound video streams over each of the wireless links to observe the image quality as well as the diagnostic value of the received video stream. The information gathered during both testing phases revealed the abilities and limitations of the different wireless technologies. The results from the performance testing will be valuable in the future for those trying to develop network applications for telemedicine procedures over these wireless telecommunication options. Additionally, the testing demonstrated that the system is currently capable of using H.264 SVC compression to transmit VGA quality ultrasound video at 30 frames per second (fps) over 802.11g while QVGA resolution at frame rates between 10 and 15 fps is possible over 3G and satellite networks

    Wireless Sensor Networks in Industrial Automation

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    Treatment-Based Classi?cation in Residential Wireless Access Points

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    IEEE 802.11 wireless access points (APs) act as the central communication hub inside homes, connecting all networked devices to the Internet. Home users run a variety of network applications with diverse Quality-of-Service requirements (QoS) through their APs. However, wireless APs are often the bottleneck in residential networks as broadband connection speeds keep increasing. Because of the lack of QoS support and complicated configuration procedures in most off-the-shelf APs, users can experience QoS degradation with their wireless networks, especially when multiple applications are running concurrently. This dissertation presents CATNAP, Classification And Treatment iN an AP , to provide better QoS support for various applications over residential wireless networks, especially timely delivery for real-time applications and high throughput for download-based applications. CATNAP consists of three major components: supporting functions, classifiers, and treatment modules. The supporting functions collect necessary flow level statistics and feed it into the CATNAP classifiers. Then, the CATNAP classifiers categorize flows along three-dimensions: response-based/non-response-based, interactive/non-interactive, and greedy/non-greedy. Each CATNAP traffic category can be directly mapped to one of the following treatments: push/delay, limited advertised window size/drop, and reserve bandwidth. Based on the classification results, the CATNAP treatment module automatically applies the treatment policy to provide better QoS support. CATNAP is implemented with the NS network simulator, and evaluated against DropTail and Strict Priority Queue (SPQ) under various network and traffic conditions. In most simulation cases, CATNAP provides better QoS supports than DropTail: it lowers queuing delay for multimedia applications such as VoIP, games and video, fairly treats FTP flows with various round trip times, and is even functional when misbehaving UDP traffic is present. Unlike current QoS methods, CATNAP is a plug-and-play solution, automatically classifying and treating flows without any user configuration, or any modification to end hosts or applications
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