Desain dan Implementasi Live Streaming Televisi Menggunakan Adaptive H264encoding

Teknologi Informasi yang paling luas penyebarannya adalah Televisi, dengan kemajuan teknologi sarana penyiaran Televisi tidak terbatas lagi ke TV broadcast menggunakan teknologi radio di gelombang khusus seperti saat ini, penyiaran TV telah menyebar ke sarana yang lain termasuk internet. Banyak teknologi yang bisa digunakan di internet, tetapi kandidat yang paling kuat adalah video streaming. Untuk aplikasi real-time atau live seperti kampanye atau siaran pengumuman pemerintah dll, teknologi video streaming yang digunakan adalah teknologi video streaming khusus yang disebut dengan live streaming.Teknologi Live Streaming hampir sama dengan video streaming, hanya saja data yang digunakan langsung bersumber dari televisi atau kamera yang bersifat real time. Live Streaming memerlukan proses live encoding dan minimum buffering, sedangkan di sisi lain diharapkan delay seminimal mungkin. Masalah selanjutnya adalah keterbatasan bandwidth. Jaringan komputer yang digunakan untuk melewatkan berbagai aplikasi akan digunakan juga sebagai media streaming yang membutuhkan bitrate cukup tinggi. Proses ini akan menyebabkan beban jaringan bertambah sehingga service yang ada tidak dapat berjalan dengan baik (terganggu). Pada penelitian ini difokuskan pada proses live streaming H264 dengan metode transmisi multicast dengan ditambahkan sebuah program adaptive streaming. Codec H264 dipilih karena performansinya yang cukup baik pada level bitrate yang lebih rendah. Sistem multicast digunakan untuk mengatasi masalah keterbatasan bandwidth yang digunakan dalam streaming. Adaptive streaming digunakan untuk menyesuaikan bitrate dengan kondisi trafik pada jaringan. Didapatkan nilai PSNR 36,58 dB untuk bitrate 500kbps dan 31,42 dB untuk bitrate 200kbps yang masih berada diatas threshold ITU 20dB dengan MOS 3,4 untuk 50 responden, sistem adaptive menyebabkan berkurangnya paket loss dari 1,53% menjadi 0,46%, bandwitdh stream unucast 1698kbps untuk multicast 558kbps

DESAIN DAN IMPLEMENTASI LIVE STREAMING TELEVISI MENGGUNAKAN ADAPTIVE H264ENCODING

Teknologi Informasi yang paling luas penyebarannya adalah Televisi, dengan kemajuan teknologi sarana penyiaran Televisi tidak terbatas lagi ke TV broadcast menggunakan teknologi radio di gelombang khusus seperti saat ini, penyiaran TV telah menyebar ke sarana yang lain termasuk internet. Banyak teknologi yang bisa digunakan di internet, tetapi kandidat yang paling kuat adalah video streaming. Untuk aplikasi real-time atau live seperti kampanye atau siaran pengumuman pemerintah dll, teknologi video streaming yang digunakan adalah teknologi video streaming khusus yang disebut dengan live streaming.Teknologi Live Streaming hampir sama dengan video streaming, hanya saja data yang digunakan langsung bersumber dari televisi atau kamera yang bersifat real time. Live Streaming memerlukan proses live encoding dan minimum buffering, sedangkan di sisi lain diharapkan delay seminimal mungkin. Masalah selanjutnya adalah keterbatasan bandwidth. Jaringan komputer yang digunakan untuk melewatkan berbagai aplikasi akan digunakan juga sebagai media streaming yang membutuhkan bitrate cukup tinggi. Proses ini akan menyebabkan beban jaringan bertambah sehingga service yang ada tidak dapat berjalan dengan baik (terganggu). Pada penelitian ini difokuskan pada proses live streaming H264 dengan metode transmisi multicast dengan ditambahkan sebuah program adaptive streaming. Codec H264 dipilih karena performansinya yang cukup baik pada level bitrate yang lebih rendah. Sistem multicast digunakan untuk mengatasi masalah keterbatasan bandwidth yang digunakan dalam streaming. Adaptive streaming digunakan untuk menyesuaikan bitrate dengan kondisi trafik pada jaringan. Didapatkan nilai PSNR 36,58 dB untuk bitrate 500kbps dan 31,42 dB untuk bitrate 200kbps yang masih berada diatas threshold ITU 20dB dengan MOS 3,4 untuk 50 responden, sistem adaptive menyebabkan berkurangnya paket loss dari 1,53% menjadi 0,46%, bandwitdh stream unucast 1698kbps untuk multicast 558kbps

DESAIN DAN IMPLEMENTASI LIVE STREAMING TELEVISI MENGGUNAKAN ADAPTIVE H264ENCODING

Teknologi Informasi yang paling luas penyebarannya adalah Televisi, dengan kemajuan teknologi sarana penyiaran Televisi tidak terbatas lagi ke TV broadcast menggunakan teknologi radio di gelombang khusus seperti saat ini, penyiaran TV telah menyebar ke sarana yang lain termasuk internet. Banyak teknologi yang bisa digunakan di internet, tetapi kandidat yang paling kuat adalah video streaming. Untuk aplikasi real-time atau live seperti kampanye atau siaran pengumuman pemerintah dll, teknologi video streaming yang digunakan adalah teknologi video streaming khusus yang disebut dengan live streaming. Teknologi Live Streaming hampir sama dengan video streaming, hanya saja data yang digunakan langsung bersumber dari televisi atau kamera yang bersifat real time. Live Streaming memerlukan proses live encoding dan minimum buffering, sedangkan di sisi lain diharapkan delay seminimal mungkin. Masalah selanjutnya adalah keterbatasan bandwidth. Jaringan komputer yang digunakan untuk melewatkan berbagai aplikasi akan digunakan juga sebagai media streaming yang membutuhkan bitrate cukup tinggi. Proses ini akan menyebabkan beban jaringan bertambah sehingga service yang ada tidak dapat berjalan dengan baik (terganggu). Pada penelitian ini difokuskan pada proses live streaming H264 dengan metode transmisi multicast dengan ditambahkan sebuah program adaptive streaming. Codec H264 dipilih karena performansinya yang cukup baik pada level bitrate yang lebih rendah. Sistem multicast digunakan untuk mengatasi masalah keterbatasan bandwidth yang digunakan dalam streaming. Adaptive streaming digunakan untuk menyesuaikan bitrate dengan kondisi trafik pada jaringan. Didapatkan nilai PSNR 36,58 dB untuk bitrate 500kbps dan 31,42 dB untuk bitrate 200kbps yang masih berada diatas threshold ITU 20dB dengan MOS 3,4 untuk 50 responden, sistem adaptive menyebabkan berkurangnya paket loss dari 1,53% menjadi 0,46%, bandwitdh stream unucast 1698kbps untuk multicast 558kbps

Smart PIN: performance and cost-oriented context-aware personal information network

The next generation of networks will involve interconnection of heterogeneous individual networks such as WPAN, WLAN, WMAN and Cellular network, adopting the IP as common infrastructural protocol and providing virtually always-connected network. Furthermore, there are many devices which enable easy acquisition and storage of information as pictures, movies, emails, etc. Therefore, the information overload and divergent content’s characteristics make it difficult for users to handle their data in manual way. Consequently, there is a need for personalised automatic services which would enable data exchange across heterogeneous network and devices. To support these personalised services, user centric approaches for data delivery across the heterogeneous network are also required. In this context, this thesis proposes Smart PIN - a novel performance and cost-oriented context-aware Personal Information Network. Smart PIN's architecture is detailed including its network, service and management components. Within the service component, two novel schemes for efficient delivery of context and content data are proposed: Multimedia Data Replication Scheme (MDRS) and Quality-oriented Algorithm for Multiple-source Multimedia Delivery (QAMMD). MDRS supports efficient data accessibility among distributed devices using data replication which is based on a utility function and a minimum data set. QAMMD employs a buffer underflow avoidance scheme for streaming, which achieves high multimedia quality without content adaptation to network conditions. Simulation models for MDRS and QAMMD were built which are based on various heterogeneous network scenarios. Additionally a multiple-source streaming based on QAMMS was implemented as a prototype and tested in an emulated network environment. Comparative tests show that MDRS and QAMMD perform significantly better than other approaches

Enhanced Multimedia Exchanges over the Internet

Although the Internet was not originally designed for exchanging multimedia streams, consumers heavily depend on it for audiovisual data delivery. The intermittent nature of multimedia traffic, the unguaranteed underlying communication infrastructure, and dynamic user behavior collectively result in the degradation of Quality-of-Service (QoS) and Quality-of-Experience (QoE) perceived by end-users. Consequently, the volume of signalling messages is inevitably increased to compensate for the degradation of the desired service qualities. Improved multimedia services could leverage adaptive streaming as well as blockchain-based solutions to enhance media-rich experiences over the Internet at the cost of increased signalling volume. Many recent studies in the literature provide signalling reduction and blockchain-based methods for authenticated media access over the Internet while utilizing resources quasi-efficiently. To further increase the efficiency of multimedia communications, novel signalling overhead and content access latency reduction solutions are investigated in this dissertation including: (1) the first two research topics utilize steganography to reduce signalling bandwidth utilization while increasing the capacity of the multimedia network; and (2) the third research topic utilizes multimedia content access request management schemes to guarantee throughput values for servicing users, end-devices, and the network. Signalling of multimedia streaming is generated at every layer of the communication protocol stack; At the highest layer, segment requests are generated, and at the lower layers, byte tracking messages are exchanged. Through leveraging steganography, essential signalling information is encoded within multimedia payloads to reduce the amount of resources consumed by non-payload data. The first steganographic solution hides signalling messages within multimedia payloads, thereby freeing intermediate node buffers from queuing non-payload packets. Consequently, source nodes are capable of delivering control information to receiving nodes at no additional network overhead. A utility function is designed to minimize the volume of overhead exchanged while minimizing visual artifacts. Therefore, the proposed scheme is designed to leverage the fidelity of the multimedia stream to reduce the largest amount of control overhead with the lowest negative visual impact. The second steganographic solution enables protocol translation through embedding packet header information within payload data to alternatively utilize lightweight headers. The protocol translator leverages a proposed utility function to enable the maximum number of translations while maintaining QoS and QoE requirements in terms of packet throughput and playback bit-rate. As the number of multimedia users and sources increases, decentralized content access and management over a blockchain-based system is inevitable. Blockchain technologies suffer from large processing latencies; consequently reducing the throughput of a multimedia network. Reducing blockchain-based access latencies is therefore essential to maintaining a decentralized scalable model with seamless functionality and efficient utilization of resources. Adapting blockchains to feeless applications will then port the utility of ledger-based networks to audiovisual applications in a faultless manner. The proposed transaction processing scheme will enable ledger maintainers in sustaining desired throughputs necessary for delivering expected QoS and QoE values for decentralized audiovisual platforms. A block slicing algorithm is designed to ensure that the ledger maintenance strategy is benefiting the operations of the blockchain-based multimedia network. Using the proposed algorithm, the throughput and latency of operations within the multimedia network are then maintained at a desired level

Analyzing Voice And Video Call Service Performance Over A Local Area Network

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2010Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2010Bu çalışmada, VOIP teknolojisinden ve bu teknolojiyi kablolu ve kablosuz ortamda gerçeklemenin en önemli darboğazları anlatılacaktır. Ayrıca H.323, SIP (Session Initiation Protocol), Megaco ve MGCP gibi yaygın olarak kullanılan ses iletim protokolleri ve H.261, H.263 ve H.264 gibi görüntü iletim protokollerinden bahsedilmiştir. Ses kodek seçimi ve VOIP servis kalitesine etki eden faktörleri anlatılmaktadır. Bu tezde, ses, görüntü ve veri iletişimini aynı anda bünyesinde barındıran gerçek şebekeler simüle edilecektir. Kullanıcılara rastlantısal olarak ses, görüntü ve FTP gibi birtakım uygulamalar atanmıştır. Ayrıca önerilen kablolu şebekeye, kablosuz bir şebeke ilave edilerek sonuçlar incelenecektir. Optimal servis kalitesini sağlamak için seçilen uygun kuyruklama mekanizmaları ve kodek seçimlerini içeren senaryolar incelenecek ve OPNET ile elde edilmiş simülasyon sonuçları tartışılacaktır.In this study, we present a detailed description of the VoIP and also the most common challenges of implementing voice communication into wireline or wireless networks are discussed. Common voice protocols, such as H.323, Session Initiation Protocol (SIP), Megaco, MGCP and video protocols such as H.261, H.263, H.264 are described as well. CODEC selection and factors affecting VoIP Quality of Service are analyzed. We simulate a real network which includes both voice, video and data communication simultaneously. Workstations are randomly assigned to different applications, such as voice, video and FTP. We will also implement a wireless network to our proposed system. The scenarios including selecting appropriate queuing scheme and codec selection are presented and the simulation results with OPNET are drawn.Yüksek LisansM.Sc

Runtime Adaptive System-on-Chip Communication Architecture

The adaptive system provides adaptivity both in the system-level and in the architecture-level. The system-level adaptation is provided using a runtime application mapping. The architecture-level adaptation is implemented by using several novel methodologies to increase the resource utilization of the underlying silicon fabric, i.e. sharing the Virtual Channel Buffers among different output ports. To achieve successful runtime adaptation, a runtime observability infrastructure is included

Quality-oriented adaptation scheme for multimedia streaming in local broadband multi-service IP networks

The research reported in this thesis proposes, designs and tests the Quality-Oriented Adaptation Scheme (QOAS), an application-level adaptive scheme that offers high quality multimedia services to home residences and business premises via local broadband IP-networks in the presence of other traffic of different types. QOAS uses a novel client-located grading scheme that maps some network-related parameters’ values, variations and variation patterns (e.g. delay, jitter, loss rate) to application-level scores that describe the quality of delivery. This grading scheme also involves an objective metric that estimates the end-user perceived quality, increasing its effectiveness. A server-located arbiter takes content and rate adaptation decisions based on these quality scores, which is the only information sent via feedback by the clients. QOAS has been modelled, implemented and tested through simulations and an instantiation of it has been realized in a prototype system. The performance was assessed in terms of estimated end-user perceived quality, network utilisation, loss rate and number of customers served by a fixed infrastructure. The influence of variations in the parameters used by QOAS and of the networkrelated characteristics was studied. The scheme’s adaptive reaction was tested with background traffic of different type, size and variation patterns and in the presence of concurrent multimedia streaming processes subject to user-interactions. The results show that the performance of QOAS was very close to that of an ideal adaptive scheme. In comparison with other adaptive schemes QOAS allows for a significant increase in the number of simultaneous users while maintaining a good end-user perceived quality. These results are verified by a set of subjective tests that have been performed on viewers using a prototype system

Multi-Stream Management for Supporting Multi-Party 3D Tele-Immersive Environments

Three-dimensional tele-immersive (3DTI) environments have great potential to promote collaborative work among geographically distributed participants. However, extensive application of 3DTI environments is still hindered by problems pertaining to scalability, manageability and reliance of special-purpose components. Thus, one critical question is how to organize the acquisition, transmission and display of large volume real-time 3D visual data over commercially available computing and networking infrastructures so that .everybody. would be able to install and enjoy 3DTI environments for high quality tele-collaboration. In the thesis, we explore the design space from the angle of multi-stream Quality-of-Service (QoS) management to support multi-party 3DTI communication. In 3DTI environments, multiple correlated 3D video streams are deployed to provide a comprehensive representation of the physical scene. Traditional QoS approach in 2D and single-stream scenario has become inadequate. On the other hand, the existence of multiple streams provides unique opportunity for QoS provisioning. We propose an innovative cross-layer hierarchical and distributed multi-stream management middleware framework for QoS provisioning to fully enable multi-party 3DTI communication over general delivery infrastructure. The major contributions are as follows. First, we introduce the view model for representing the user interest in the application layer. The design revolves around the concept of view-aware multi-stream coordination, which leverages the central role of view semantics in 3D video systems. Second, in the stream differentiation layer we present the design of view to stream mapping, where a subset of relevant streams are selected based on the relative importance of each stream to the current view. Conventional streaming controllers focus on a fixed set of streams specified by the application. Different from all the others, in our management framework the application layer only specifies the view information while the underlying controller dynamically determines the set of streams to be managed. Third, in the stream coordination layer we present two designs applicable in different situations. In the case of end-to-end 3DTI communication, a learning-based controller is embedded which provides bandwidth allocation for relevant streams. In the case of multi-party 3DTI communication, we propose a novel ViewCast protocol to coordinate the multi-stream content dissemination upon an end-system overlay network