39 research outputs found

    RTP Payload Format for Uncompressed Video

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    Performance Evaluation of H.264/AVC encoded Video over TETRA Enhanced Data Service (TEDS)

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    Public Safety Systems (PSS) are communication networks oriented towards supporting activities of public safety actors (police, medical, fire-fighters, etc.). TErrestrial Trunked RAdio (TETRA) is a Professional Mobile Radio (PMR) standard designed to meet PSS requirements with specialized voice communication features and reliable, secure communication links. TETRA Release 2 introduces TETRA Enhanced Data Service (TEDS), to support emerging data-intensive applications such as online navigation and tele-medicine by providing higher, scalable data rates. This thesis studies the feasibility of streaming video over a wideband TEDS link using the H.264/AVC codec, a video compression standard that manages to retain high decoded video quality while dramatically reducing streaming bit rate. A bandwidth limiter is used to emulate a link that supports data rates equivalent to those specified in the TEDS standard. Effects of video streaming parameters such as codec rate and play-out buffer size coupled with link-induced delay variation on decoded video quality are investigated. Visual quality is rated using objective quality metrics to quantify results with some measure of reliability. The overall aim is to identify the technical requirements needed to support an acceptable quality of video transmission over TEDS. To this end, we measure decoded video quality in different channel loss conditions, varying video streaming parameters and at different channel bandwidths, plus enhancements such as data traffic prioritisation as defined in the TEDS specification

    Quality of Service Controlled Multimedia Transport Protocol

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    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

    Video Content-Based QoE Prediction for HEVC Encoded Videos Delivered over IP Networks

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    The recently released High Efficiency Video Coding (HEVC) standard, which halves the transmission bandwidth requirement of encoded video for almost the same quality when compared to H.264/AVC, and the availability of increased network bandwidth (e.g. from 2 Mbps for 3G networks to almost 100 Mbps for 4G/LTE) have led to the proliferation of video streaming services. Based on these major innovations, the prevalence and diversity of video application are set to increase over the coming years. However, the popularity and success of current and future video applications will depend on the perceived quality of experience (QoE) of end users. How to measure or predict the QoE of delivered services becomes an important and inevitable task for both service and network providers. Video quality can be measured either subjectively or objectively. Subjective quality measurement is the most reliable method of determining the quality of multimedia applications because of its direct link to users’ experience. However, this approach is time consuming and expensive and hence the need for an objective method that can produce results that are comparable with those of subjective testing. In general, video quality is impacted by impairments caused by the encoder and the transmission network. However, videos encoded and transmitted over an error-prone network have different quality measurements even under the same encoder setting and network quality of service (NQoS). This indicates that, in addition to encoder settings and network impairment, there may be other key parameters that impact video quality. In this project, it is hypothesised that video content type is one of the key parameters that may impact the quality of streamed videos. Based on this assertion, parameters related to video content type are extracted and used to develop a single metric that quantifies the content type of different video sequences. The proposed content type metric is then used together with encoding parameter settings and NQoS to develop content-based video quality models that estimate the quality of different video sequences delivered over IP-based network. This project led to the following main contributions: (1) A new metric for quantifying video content type based on the spatiotemporal features extracted from the encoded bitstream. (2) The development of novel subjective test approach for video streaming services. (3) New content-based video quality prediction models for predicting the QoE of video sequences delivered over IP-based networks. The models have been evaluated using subjective and objective methods

    Entwicklung eines JPEG-Dateianalysators

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    Die Arbeit befasst sich mit der Verbesserung und Erweiterung eines bestehenden Softwareprojektes zum Streaming von JPEG-Bildern per RTP. In diesem Projekt werden JPEG-Bilder aus MJPEG-Dateien von einem Server zum Client übertragen. Um eine fehlerfreie Übertragung zu gewährleisten, soll zuvor die Eignung der Dateien für eine solche geprüft werden. Die entsprechenden Anforderungen finden sich in RFC 2435, welcher die Übertragung von JPEG-Bildern per RTP standardisiert. Zur Automatisierung der Überprüfung wurde diese in einem eigenen Programm implementiert. Weitere Verbesserungen wurden hinsichtlich der Ausführbarkeit des Projektes auf verschiedener Hardware getroffen. So wurden interne Algorithmen verbessert, um auch auf schwächerer Hardware einen flüssigen Ablauf zu ermöglichen. Außerdem wurde die Kompatibilität der RTSP-Implementierung im Projekt mit jener im VLC Media Player hergestellt. Zuletzt wurde das Softwareprojekt hinsichtlich der Verschlüsselung der Übertragung erweitert. Die Grundlage dafür legt die Anlayse von Anforderungen an die Verschlüsselung von Mediendaten. Es wurden zwei verschiedene Verfahren betrachtet und implementiert: Zum einen das weitverbreitete SRTP-Protokoll, zum anderen eine eigene JPEG-Verschlüsselung. Anschließend wurde die Komplexität der Entwicklung eines Verschlüsselungsverfahrens gezeigt, indem das selbst ent wickelte Verfahren durch einen Ersetzungsangriff gebrochen wurde.This work deals with the improvement and extension of an existing software project for streaming JPEG images via RTP. In the project JPEG images are read in from MJPEG files and transmitted from a server to a client. To guarantee a faultless transmission the fitness of the files for transmission should be checked. The corresponding requirements can be found in RFC 2435 in which the transmission of JPEG images via RTP is standardized. An automation of this verification is realized in an own program. Further improvements are done in regard to the executability of the project on different hardware. Internal algorithms are improved to get a smooth execution even on weaker hardware. Additionally the compatibility of the implementation of RTSP in the project with that in the VLC Media Player is established. Finally the software project is extended in terms of the encryption of the transmission. The requirements of media data encryption are analyzed and used as the base for the following considerations. There are two operations which were examined and implemented: On the one hand the SRTP protocoll which is widely used. On the other hand an own JPEG encryption. Following that the complexity of developing an own encryption method is shown by breaking the developed JPEG encryption with a replacement attack

    Design of Media Access Control Schemes for Performance Enhancement of Future Generation Wireless Systems

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    Wireless Local Area Networks (WLANs) now provide connectivity to many businesses, homes and educational institutions. The wireless channel itself is plagued with numerous problems, such as it does not natively allow sharing of the wireless resource. WLAN devices utilize a complex medium access control (MAC) mechanism to allow multiple users to share the wireless resource. The distributed coordination function (DCF) is the most commonly used multiple access scheme in WLANs and a member of the 802.11 standard [1]. In this thesis, two major roles of MAC protocols are examined: maximizing network throughput and service differentiation. Firstly, a novel MAC scheme is proposed that makes use of Multiple-Input, Multiple-Output (MIMO) antenna technology to improve overall network throughput. The proposed MIMO-A ware MAC (MA-MAC) scheme utilizes the beamforming feature available in MIMO systems to allow two simultaneous transmissions of the wireless channel overlapped in time. This results in increased aggregate network throughput. This proposed scheme is shown to offer better throughput and delay performance versus existing MAC schemes proposed for simultaneous transmission. In addition, this MAC scheme is able to achieve this performance in a manner compatible with the existing standard. The latter part of this thesis proposes a new Time Division Multiple Access (TDMA) based scheme for providing video, voice and data services (also known as the Triple-Play services) in a point-to-multipoint network. By dynamically allocating transmission slots, the proposed Television TDMA (TV-TDMA) scheme is shown to better meet delay requirements for video and voice traffic, and is able to achieve higher overall saturation throughput for best-effort traffic than existing Quality of Service enabled protocols

    Assessing Readiness of IP Networks to Support Desktop Videoconferencing Using OPNET

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    OPNET is a powerful network design and simulation tool that has gained popularity in industry and academia. However, there exists no known simulation approach on how to deploy a popular real-time network service such as videoconferencing. This paper demonstrates how OPNET can be leveraged to assess the readiness of existing IP networks to support desktop videoconference. To date, OPNET does not have built-in features to support videoconferencing or its deployment. The paper offers remarkable details on how to model and configure OPNET for such a purpose. The paper considers two types of video traffic (viz. fixed and empirical video packet sizes). Empirical video packet sizes are collected from well-known Internet traffic traces. The paper presents in-depth analysis and interpretation of simulation results and shows how to draw proper engineering conclusions

    Assessing Readiness of IP Networks to Support Desktop Videoconferencing Using OPNET

    Get PDF
    OPNET is a powerful network design and simulation tool that has gained popularity in industry and academia. However, there exists no known simulation approach on how to deploy a popular real-time network service such as videoconferencing. This paper demonstrates how OPNET can be leveraged to assess the readiness of existing IP networks to support desktop videoconference. To date, OPNET does not have built-in features to support videoconferencing or its deployment. The paper offers remarkable details on how to model and configure OPNET for such a purpose. The paper considers two types of video traffic (viz. fixed and empirical video packet sizes). Empirical video packet sizes are collected from well-known Internet traffic traces. The paper presents in-depth analysis and interpretation of simulation results and shows how to draw proper engineering conclusions

    Assessing Readiness of IP Networks to Support Desktop Videoconferencing Using OPNET

    Get PDF
    OPNET is a powerful network design and simulation tool that has gained popularity in industry and academia. However, there exists no known simulation approach on how to deploy a popular real-time network service such as videoconferencing. This paper demonstrates how OPNET can be leveraged to assess the readiness of existing IP networks to support desktop videoconference. To date, OPNET does not have built-in features to support videoconferencing or its deployment. The paper offers remarkable details on how to model and configure OPNET for such a purpose. The paper considers two types of video traffic (viz. fixed and empirical video packet sizes). Empirical video packet sizes are collected from well-known Internet traffic traces. The paper presents in-depth analysis and interpretation of simulation results and shows how to draw proper engineering conclusions
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