145 research outputs found
Micro protocol engineering for unstructured carriers: On the embedding of steganographic control protocols into audio transmissions
Network steganography conceals the transfer of sensitive information within
unobtrusive data in computer networks. So-called micro protocols are
communication protocols placed within the payload of a network steganographic
transfer. They enrich this transfer with features such as reliability, dynamic
overlay routing, or performance optimization --- just to mention a few. We
present different design approaches for the embedding of hidden channels with
micro protocols in digitized audio signals under consideration of different
requirements. On the basis of experimental results, our design approaches are
compared, and introduced into a protocol engineering approach for micro
protocols.Comment: 20 pages, 7 figures, 4 table
A New covert channel over RTP
In this thesis, we designed and implemented a new covert channel over the RTP protocol. The covert channel modifies the timestamp value in the RTP header to send its secret messages. The high frequency of RTP packets allows for a high bitrate covert channel, theoretically up to 350 bps. The broad use of RTP for multimedia applications, including VoIP, provides plentiful opportunities to use this channel. By using the RTP header, many of the challenges present for covert channels using the RTP payload are avoided. Using the reference implementation of this covert channel, bitrates of up to 325 bps were observed. Speed decreases on less reliable networks, though message delivery was flawless with up to 1% RTP packet loss. The channel is very difficult to detect due to expected variations in the timestamp field and the flexible nature of RTP
Quality of Service Controlled Multimedia Transport Protocol
PhDThis research looks at the design of an open transport protocol that supports a range of
services including multimedia over low data-rate networks. Low data-rate multimedia
applications require a system that provides quality of service (QoS) assurance and flexibility.
One promising field is the area of content-based coding. Content-based systems use an array
of protocols to select the optimum set of coding algorithms. A content-based transport
protocol integrates a content-based application to a transmission network.
General transport protocols form a bottleneck in low data-rate multimedia
communicationbsy limiting throughpuot r by not maintainingt iming requirementsT. his work
presents an original model of a transport protocol that eliminates the bottleneck by
introducing a flexible yet efficient algorithm that uses an open approach to flexibility and
holistic architectureto promoteQ oS.T he flexibility andt ransparenccyo mesi n the form of a
fixed syntaxt hat providesa seto f transportp rotocols emanticsT. he mediaQ oSi s maintained
by defining a generic descriptor. Overall, the structure of the protocol is based on a single
adaptablea lgorithm that supportsa pplication independencen, etwork independencea nd
quality of service.
The transportp rotocol was evaluatedth rougha set of assessmentos:f f-line; off-line
for a specific application; and on-line for a specific application. Application contexts used
MPEG-4 test material where the on-line assessmenuts eda modified MPEG-4 pl; yer. The
performanceo f the QoSc ontrolledt ransportp rotocoli s often bettert hano thers chemews hen
appropriateQ oS controlledm anagemenatl gorithmsa re selectedT. his is shownf irst for an
off-line assessmenwt here the performancei s compared between the QoS controlled
multiplexer,a n emulatedM PEG-4F lexMux multiplexers chemea, ndt he targetr equirements.
The performanceis also shownt o be better in a real environmentw hen the QoS controlled
multiplexeri s comparedw ith the real MPEG-4F lexMux scheme
Multimedia in mobile networks: Streaming techniques, optimization and User Experience
1.UMTS overview and User Experience
2.Streaming Service & Streaming Platform
3.Quality of Service
4.Mpeg-4
5.Test Methodology & testing architecture
6.Conclusion
Robust header compression over IEEE 802 networks
Tese de mestrado. Redes e Serviços de Comunicação. Faculdade de Engenharia. Universidade do Porto, INESC Porto. 200
Content-Aware Multimedia Communications
The demands for fast, economic and reliable dissemination of multimedia
information are steadily growing within our society. While people and
economy increasingly rely on communication technologies, engineers still
struggle with their growing complexity.
Complexity in multimedia communication originates from several sources. The
most prominent is the unreliability of packet networks like the Internet.
Recent advances in scheduling and error control mechanisms for streaming
protocols have shown that the quality and robustness of multimedia delivery
can be improved significantly when protocols are aware of the content they
deliver. However, the proposed mechanisms require close cooperation between
transport systems and application layers which increases the overall system
complexity. Current approaches also require expensive metrics and focus on
special encoding formats only. A general and efficient model is missing so
far.
This thesis presents efficient and format-independent solutions to support
cross-layer coordination in system architectures. In particular, the first
contribution of this work is a generic dependency model that enables
transport layers to access content-specific properties of media streams,
such as dependencies between data units and their importance. The second
contribution is the design of a programming model for streaming
communication and its implementation as a middleware architecture. The
programming model hides the complexity of protocol stacks behind simple
programming abstractions, but exposes cross-layer control and monitoring
options to application programmers. For example, our interfaces allow
programmers to choose appropriate failure semantics at design time while
they can refine error protection and visibility of low-level errors at
run-time.
Based on some examples we show how our middleware simplifies the
integration of stream-based communication into large-scale application
architectures. An important result of this work is that despite cross-layer
cooperation, neither application nor transport protocol designers
experience an increase in complexity. Application programmers can even
reuse existing streaming protocols which effectively increases system
robustness.Der Bedarf unsere Gesellschaft nach kostengünstiger und
zuverlässiger
Kommunikation wächst stetig. Während wir uns selbst immer mehr von modernen
Kommunikationstechnologien abhängig machen, müssen die Ingenieure dieser
Technologien sowohl den Bedarf nach schneller Einführung neuer Produkte
befriedigen als auch die wachsende Komplexität der Systeme beherrschen.
Gerade die Übertragung multimedialer Inhalte wie Video und Audiodaten ist
nicht trivial. Einer der prominentesten Gründe dafür ist die
Unzuverlässigkeit heutiger Netzwerke, wie z.B.~dem Internet. Paketverluste
und schwankende Laufzeiten können die Darstellungsqualität massiv
beeinträchtigen. Wie jüngste Entwicklungen im Bereich der
Streaming-Protokolle zeigen, sind jedoch Qualität und Robustheit der
Übertragung effizient kontrollierbar, wenn Streamingprotokolle
Informationen über den Inhalt der transportierten Daten ausnutzen.
Existierende Ansätze, die den Inhalt von Multimediadatenströmen
beschreiben, sind allerdings meist auf einzelne Kompressionsverfahren
spezialisiert und verwenden berechnungsintensive Metriken. Das reduziert
ihren praktischen Nutzen deutlich. Außerdem erfordert der
Informationsaustausch eine enge Kooperation zwischen Applikationen und
Transportschichten. Da allerdings die Schnittstellen aktueller
Systemarchitekturen nicht darauf vorbereitet sind, müssen entweder die
Schnittstellen erweitert oder alternative Architekturkonzepte geschaffen
werden. Die Gefahr beider Varianten ist jedoch, dass sich die Komplexität
eines Systems dadurch weiter erhöhen kann.
Das zentrale Ziel dieser Dissertation ist es deshalb,
schichtenübergreifende Koordination bei gleichzeitiger Reduzierung der
Komplexität zu erreichen. Hier leistet die Arbeit zwei Beträge zum
aktuellen Stand der Forschung. Erstens definiert sie ein universelles
Modell zur Beschreibung von Inhaltsattributen, wie Wichtigkeiten und
Abhängigkeitsbeziehungen innerhalb eines Datenstroms. Transportschichten
können dieses Wissen zur effizienten Fehlerkontrolle verwenden. Zweitens
beschreibt die Arbeit das Noja Programmiermodell für multimediale
Middleware. Noja definiert Abstraktionen zur Übertragung und Kontrolle
multimedialer Ströme, die die Koordination von Streamingprotokollen mit
Applikationen ermöglichen. Zum Beispiel können Programmierer geeignete
Fehlersemantiken und Kommunikationstopologien auswählen und den konkreten
Fehlerschutz dann zur Laufzeit verfeinern und kontrolliere
Analysing the characteristics of VoIP traffic
In this study, the characteristics of VoIP traffic in a deployed Cisco VoIP phone system and a SIP based soft phone system are analysed. Traffic was captured in a soft phone system, through which elementary understanding about a VoIP system was obtained and experimental setup was validated. An advanced experiment was performed in a deployed Cisco VoIP system in the department of Computer Science at the University of Saskatchewan. Three months of traffic trace was collected beginning October 2006, recording address and protocol information for every packet sent and received on the Cisco VoIP network. The trace was analysed to find out the features of Cisco VoIP system and the findings were presented.This work appears to be one of the first real deployment studies of VoIP that does not rely on artificial traffic. The experimental data provided in this study is useful for design and modeling of such systems, from which more useful predictive models can be generated. The analysis method used in this research can be used for developing synthetic workload models. A clear understanding of usage patterns in a real VoIP network is important for network deployment and potential network activities such as integration, optimizations or expansion. The major factors affecting VoIP quality such as delay, jitter and loss were also measured and simulated in this study, which will be helpful in an advanced VoIP quality study. A traffic generator was developed to generate various simulated VoIP traffic. The data used to provide the traffic model parameters was chosen from peak traffic periods in the captured data from University of Saskatchewan deployment. By utilizing the Traffic Trace function in ns2, the simulated VoIP traffic was fed into ns2, and delay, jitter and packet loss were calculated for different scenarios. Two simulation experiments were performed. The first experiment simulated the traffic of multiple calls running on a backbone link. The second experiment simulated a real network environment with different traffic load patterns. It is significant for network expansion and integration
Měření Triple play služeb v hybridní síti
The master's thesis deals with a project regarding the implementation, design and the quality of IPTV, VoIP and Data services within the Triple Play services. In heterostructural networks made up of GEPON and xDSL technologies. Different lengths of the optical and metallic paths were used for the measurements.
The first part of the thesis is theoretically analyzed the development and trend of optical and metallic networks.
The second part deals with the measurement of typical optical and metallic parameters on the constructed experimental network, where its integrity was tested.
Another part of the thesis is the evaluation of Triple play results, regarding the test where the network was variously tasked/burdened with data traffic and evaluated according to defined standards.
The last part is concerned with the Optiwave Software simulation environment.Diplomová práce se zabývá návrhem, realizací a kvalitou služeb IPTV, VoIP a Data v rámci Triple play služeb v heterostrukturní sítí tvořené GEPON a xDSL technologiemi. Pro měření byli využity různé délky optické a metalické trasy.
První části diplomové práce je teoreticky rozebrán vývoj a trend optických a metalických sítí.
Druhá část se zaměřuje na měření typických optických a metalických parametrů na vybudované experimentální síti, kde byla následně testována její integrita.
Dalším bodem práce je vyhodnocení výsledků Triple play, kde síť je různě zatěžována datovým provozem a následně vyhodnocována podle definovaných norem.
Závěr práce je věnovaný simulačnímu prostředí Optiwave.440 - Katedra telekomunikační technikyvýborn
Improving the robustness of CELP-like speech decoders using late-arrival packets information : application to G.729 standard in VoIP
L'utilisation de la voix sur Internet est une nouvelle tendance dans Ie secteur des télécommunications et de la réseautique. La paquetisation des données et de la voix est réalisée en utilisant Ie protocole Internet (IP). Plusieurs codecs existent pour convertir la voix codée en paquets. La voix codée est paquetisée et transmise sur Internet. À la réception, certains paquets sont soit perdus, endommages ou arrivent en retard. Ceci est cause par des contraintes telles que Ie délai («jitter»), la congestion et les erreurs de réseau. Ces contraintes dégradent la qualité de la voix. Puisque la transmission de la voix est en temps réel, Ie récepteur ne peut pas demander la retransmission de paquets perdus ou endommages car ceci va causer plus de délai. Au lieu de cela, des méthodes de récupération des paquets perdus (« concealment ») s'appliquent soit à l'émetteur soit au récepteur pour remplacer les paquets perdus ou endommages. Ce projet vise à implémenter une méthode innovatrice pour améliorer Ie temps de convergence suite a la perte de paquets au récepteur d'une application de Voix sur IP. La méthode a déjà été intégrée dans un codeur large-bande (AMR-WB) et a significativement amélioré la qualité de la voix en présence de <<jitter » dans Ie temps d'arrivée des trames au décodeur. Dans ce projet, la même méthode sera intégrée dans un codeur a bande étroite (ITU-T G.729) qui est largement utilise dans les applications de voix sur IP. Le codeur ITU-T G.729 défini des standards pour coder et décoder la voix a 8 kb/s en utilisant 1'algorithme CS-CELP (Conjugate Stmcture Algebraic Code-Excited Linear Prediction).Abstract: Voice over Internet applications is the new trend in telecommunications and networking industry today. Packetizing data/voice is done using the Internet protocol (IP). Various codecs exist to convert the raw voice data into packets. The coded and packetized speech is transmitted over the Internet. At the receiving end some packets are either lost, damaged or arrive late. This is due to constraints such as network delay (fitter), network congestion and network errors. These constraints degrade the quality of speech. Since voice transmission is in real-time, the receiver can not request the retransmission of lost or damaged packets as this will cause more delay. Instead, concealment methods are applied either at the transmitter side (coder-based) or at the receiver side (decoder-based) to replace these lost or late-arrival packets. This work attempts to implement a novel method for improving the recovery time of concealed speech The method has already been integrated in a wideband speech coder (AMR-WB) and significantly improved the quality of speech in the presence of jitter in the arrival time of speech frames at the decoder. In this work, the same method will be integrated in a narrowband speech coder (ITU-T G.729) that is widely used in VoIP applications. The ITUT G.729 coder defines the standards for coding and decoding speech at 8 kb/s using Conjugate Structure Algebraic Code-Excited Linear Prediction (CS-CELP) Algorithm
- …