277 research outputs found

    RTP Payload Format for Uncompressed Video

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    Experiences with high definition interactive video conferencing

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    We review the design and implementation of UltraGrid, a new high definition video conferencing system, and present some experimental results. UltraGrid was the first system to support gigabit rate high definition interactive video conferencing on commodity systems and networks, and we present measurements to illustrate behavior of production networks subject to such real time traffic. We illustrate the benefits of hybrid IP/provisioned optical networks over best effort IP networks for this class of traffic, and motivate the development of congestion control algorithms for interactive conferencing on best effort IP networks

    Streaming DICOM Real-Time Video and Metadata Flows Outside The Operating Room

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    International audienceWith the current advancement in the medical world, surgeons are faced with the challenge of handling many sources of medical information in more and more complex and technological Operating Rooms (ORs). Obviously, in the next generation ones, there will be an increasing number of video flows during the surgery (e.g. endoscopes, cameras, ultrasounds, etc.), which can be also displayed all over the OR in order to facilitate the task for the surgeon and to avoid any adverse events or problems related to inadequate communication in the OR. Additionally, other information needs to be shared, pre/post/during an operation, such as the history of the digital images related to the patient in the PACS and the metadata coming from medical sensors. Moreover, these medical videos captured from the OR can be either displayed on a large screen in the OR in order to provide the surgeon with more visibility, in this case via DICOM-RTV, or streamed outside the OR via a P2P solution. The latter one can serve various purposes such as for teaching medical student in real-time or for remote-expertise with a remote senior surgeons. Hence, this paper addresses the challenges of streaming DICOM-RTV video and metadata flows live from the operating room, typically during an ongoing surgery, in real-time to the outside world. A Proof of Concept is also presented in order to demonstrate the feasibility of our solution

    Robust header compression over IEEE 802 networks

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    Tese de mestrado. Redes e Serviços de Comunicação. Faculdade de Engenharia. Universidade do Porto, INESC Porto. 200

    Unidirectional and bidirectional optimistic modes IP header compression for real-time video streaming

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    Communication over Internet Protocol (IP) networks, has become crucial component of day everyday activities. They are utilized over the Internet to support a wide range of services. The flexibility of this kind of transmission relies on the IP User Datagram Protocol (UDP), IP/UDP/Real-time Transport Protocol (RTP) and IP/Transmission Control Protocol (TCP). Unfortunately, the weight of encapsulated protocol headers affects the transmission efficiency. This research aims at improving a technique that reduce the packets header size by compression. Performance analysis of the enhanced efficient techniques in both unidirectional and bidirectional optimistic modes applied to real-time video streaming traffic for UDP/IP and HTTP/TCP flows over free error channel has been conducted. The finding shows that the header compression ratio in each case is good and better than the previous studies. The technique achieved a reduction up to 90% for RTP/UDP/IP, 89% for UDP /IP and 77.5 to 86.5 % for TCP/IP profile. This research contribution is restricted to compression gain and saving for 0x0000, 0x0001, 0x0002 and 0x0006 profiles in the unidirectional and bidirectional optimistic mode

    Database of audio records

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    Diplomka a prakticky castDiplome with partical part

    Performance characterization of robust header compression (ROHC) over satellite based unidirectional link (UDL).

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    Tesis ini menilai penggunaan Pemampatan Teguh Kepala Paket (RObust Header Compression (ROHC)) untuk trafik Pengkapsulan Ringan Sehala (Unidirectional Lightweight Encapsulation (ULE)) dari segi prestasi rangkaian serta implementasi praktikal dan reka bentuk sistem pemampat and penyahmampat ROHC. This thesis evaluates RObust Header Compression (ROHC) for Unidirectional Lightweight Encapsulation (ULE) in terms of network performance as well as the practical implementation and the design of ROHC compressor and ROHC decompressor system

    Evaluating the Performance of the AES70/AES 67-based Network Architectures for Audio Streaming

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    O consumo de mutimédia via streaming tornou-se um aspecto central da vida moderna. Tal papel acarreta enormes desafios para garantir geral satisfação com a qualidade dos serviços, o que naturalmente tende a acelerar o seu desenvolvimento, isto é, tende a traduzir-se em sucessivos aperfeiçoamentos. O utilizador tem tremenda facilidade em avaliar a qualidade de um destes serviços: não pode apresentar latência e deve apresentar a maior fidelidade possível. São precisamente estes parâmetros que o presente projeto pretende avaliar na performance do protocolo AES70 e do standard AES67, constituindo assim o seu principal objetivo. Esta avaliação está contextualizada dentro daquilo que são as opções de transmissão de áudio sobre uma rede local Ethernet, principalmente quando aplicada a micro-controladores embutidos. Para tal, o trabalho realizado para esta dissertação prendeu-se inicialmente por dominar o controlador disponibilizado para testagem, no caso, uma placa STM com um micro-controlador ARM Cortex-M7. Durante este processo foi possível analisar sobretudo a idiossincrasia do controlador, descobrindo-se lacunas que põem em causa não só o processo de avaliação dos referidos protocols, mas antes ainda, a sua implementação no sistema embutido. Contudo, foi possível implementar uma aplicação de áudio streaming sobre Ethernet, a qual se espera ser útil para trabalho futuro. Este é, na verdade, o factor que esta dissertação mais acaba por desempenhar, uma sucinta explicação para entender todas as vertentes do controlador disponibilizado, nomeadamente as mais relevantes a aplicações de áudio e Ethernet, deixando-o perparado para cenários de teste e posterior comparação com abordagens semelhantes.The consumption of multimedia services via streaming has become a core aspect of modern life. Such a role brings enormous challenges to ensure general satisfaction with the quality of those services, which naturally tends to accelerate their development, i.e., it reflects in a tendency for successive improvements. Their quality, for an user, is measured quite elementally: the service must not present any discernible latency and must deliver the highest possible fidelity. These parameters are precisely the ones this project intends to evaluate in the performance of the AES70 protocol and the AES67 standard, thus constituting the project's main objective. This evaluation is contextualized within the available options for audio transmission over an Ethernet local area network, especially when applied to embedded microcontrollers. As such, the work that comprises this dissertation was initially concerned with mastering the testing environment, i.e., the controller provider for that role, in this case, an STM board with an ARM Cortex-M7 microcontroller. During this process, the analysis of the controller's idiosyncrasies was the main object of focus, contributing to the uncovering of important shortcomings which further challenged not only the evaluation process of these protocols, but particularly, their implementation in the embedded system. However, an implementation of an audio streaming over Ethernet application was accomplished and it is expected to be useful for future work. This is, in fact, the most valuable factor of this dissertation: it incorporates a precise explanation for understanding all key aspects of the STM controller, namely those most relevant to audio and Ethernet applications, rendering it ready for test scenarios and, subsequently, for the comparison of results with similar audio streaming approaches

    Fusion and Perspective Correction of Multiple Networked Video Sensors

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    A network of adaptive processing elements has been developed that transforms and fuses video captured from multiple sensors. Unlike systems that rely on end-systems to process data, this system distributes the computation throughout the network in order to reduce overall network bandwidth. The network architecture is scalable because it uses a hierarchy of processing engines to perform signal processing. Nodes within the network can be dynamically reprogrammed in order to compose video from multiple sources, digitally transform camera perspectives, and adapt the video format to meet the needs of specific applications. A prototype has been developed using reconfigurable hardware that collects and processes real-time, streaming video of an urban environment. Multiple video cameras gather data from different perspectives and fuse that data into a unified, top-down view. The hardware exploits both the spatial and temporal parallelism of the video streams and the regular processing when applying the transforms. Recon-figurable hardware allows for the functions at nodes to be reprogrammed for dynamic changes in topology. Hardware-based video processors also consume less power than high frequency software-based solutions. Performance and scalability are compared to a distributed software-based implementation. The reconfigurable hardware design is coded in VHDL and prototyped using Washington University’s Field Programmable Port Extender (FPX) platform. The transform engine circuit utilizes approximately 34 percent of the resources of a Xilinx Virtex 2000E FPGA, and can be clocked at frequencies up to 48 MHz. The com-position engine circuit utilizes approximately 39 percent of the resources of a Xilinx Virtex 2000E FPGA, and can be clocked at frequencies up to 45 MHz
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