Building self-optimized communication systems based on applicative cross-layer information

This article proposes the Implicit Packet Meta Header(IPMH) as a standard method to compute and represent common QoS properties of the Application Data Units (ADU) of multimedia streams using legacy and proprietary streams’ headers (e.g. Real-time Transport Protocol headers). The use of IPMH by mechanisms located at different layers of the communication architecture will allow implementing fine per-packet selfoptimization of communication services regarding the actual application requirements. A case study showing how IPMH is used by error control mechanisms in the context of wireless networks is presented in order to demonstrate the feasibility and advantages of this approach

Cross layer techniques for flexible transport protocol using UDP-Lite over a satellite network

Traditional real-time multimedia and streaming services have utilised UDP over RTP. Wireless transmission, by its nature, may introduce a variable, sometimes high bit error ratio. Current transport layer protocols drop all corrupted packets, in contrast, protocols such as UDP-Lite allow error-resilient applications to be supported in the networking stack. This paper presents experimental quantitative performance metrics using H.264 and UDP Lite for the next generation transport of IP multimedia, and discusses the architectural implications for enhancing performance of a wireless and/or satellite environment

Table of Contents

This memo provides information for the Internet community. It does not specify an Internet standard of any kind. Distribution of this memo is unlimited. Copyright Notice Copyright (C) The IETF Trust (2007). The first RFC that describes an RTP payload format for ITU Telecommunication Standardization Sector (ITU-T) recommendation H.263 is RFC 2190. This specification discusses why to move RFC 2190 t

Evaluating Video Streaming over GPRS/UMTS networks: A practical case.

This is a pre-print version of the paper A. Diaz, P. Merino, L. Panizoand A. M. Recio, ”Evaluating Video Streaming Over GPRS/UMTS Net works: A Practical Case,” 2007 IEEE 65th Vehicular Technology Con ference - VTC2007-Spring, Dublin, Ireland, 2007, pp. 624-628, doi: 10.1109/VETECS.2007.139.In this paper, we focus on analyzing video streaming service performance on real networks.We propose a non intrusive methodology based on mobile devices as clients, instead of using them as modems. Our objective is to provide a more realistic test environment using actual mobile devices in real conditions of network load and radio propagation while taking into account the intrinsic mobility of mobile subscribers. This solution allows us to follow the end to end performance even when handover between different access technologies is performed. Using this methodology we carry out a study of video streaming behavior over GPRS and UMTS networks. Outstanding results related with delays, jitter, lost packets and sequence errors have been obtained. Also other conclusions about video quality, such as PSNR, have been achieved. Moreover, we analyze the impact of mobility issues such as handover or cell reselection.Work partially supported by projects TIN 2005-09405-C02-01 and SMEPP IST-5-033563-STP Universidad de Málaga. Campus de Excelencia Internacional Andalucía Tech

Seminario sullo Standard MPEG-4: utilizzo ed aspetti implementativi

Una delle tecnologie chiave che hanno permesso il grande sviluppo della televisione digitale è la compressione video. La tecnologia di codifica video nota come MPEG-2, sviluppata nei primi anni novanta, è diventata lo standard di trasmissione DTV (Digital TV) sia satellitare sia terrestre in quasi tutti i paesi del mondo. Da allora la velocità dei microprocessori e le capacità di memoria dei dispositivi hardware per la codifica e la decodifica sono migliorate significativamente rendendo possibile lo sviluppo e l’implementazione di algoritmi di codifica innovativi in grado di abbattere significativamente i limiti di compressione dello standard MPEG-2. Tali innovazioni, sfociate nel 2003 nello standard MPEG-4 AVC (Advanced Video Coding), non hanno permesso di mantenere la compatibilità all’indietro con l’MPEG-2, e questo ha inizialmente costituito un limite alla loro introduzione nei sistemi di trasmissione DTV. Tuttavia, negli ultimi anni la codifica MPEG-4 AVC si è diffusa rapidamente, è stata adottata dal progetto DVB, recentemente dall’ATSC, ed è lo standard di codifica nell’IPTV. L’obiettivo di questo seminario, che si articola in due giornate, è quello di presentare lo standard di codifica MPEG-4 AVC con particolare attenzione agli aspetti implementativi del livello di codifica video.2008-11-18Sardegna Ricerche, Edificio 2, Località Piscinamanna 09010 Pula (CA) - ItaliaSeminario sullo Standard MPEG-4: utilizzo ed aspetti implementativ

Robust error detection methods for H.264/AVC videos

The 3rd generation of mobile systems is mainly focused on enabling multimedia services such as video streaming, video call and conferencing. In order to achieve this, the Universal Mobile Telecommunications System (UMTS), is the standard that has been developed by the 3rd Generation Partnership ect (3GPP) in Europe, including the baseline profile of H.264/AVC in the specification. With the union of both technologies a great improvement on video transmission over mobile networks, and even modification of the user habits towards the use of the mobile phone is expected. Nevertheless, video transmission has always been related to wired networks and unfortunately the migration to wireless networks is not as easy as it seems. In real time applications the delay is a critical constraint. Usually, transmission protocols without delivery warranties, like the User Network Protocol (UDP) for IP based networks, are used. This works under the assumption that in real time applications dropped packets are preferable to delayed packets. Moreover, in UMTS the network needs to be treated in a different way, thus the wireless channel is a prone error channel due to its high time variance. Typically, when transmitting video, the receiver checks whether the information packet is corrupted (by means of a checksum) or if its temporal mark exceeds the specified delay. This approach is suboptimal, due to the fact that perhaps the video information is not damaged and could still be used. Instead, residual redundancy on the video stream can be used to locate the errors in the corrupted packet, increasing the granularity of the typical upper-layer checksum error detection. Based on this, the amount of information previous to the error detection can be decoded as usually. The aim of this thesis is to combine some of the more effective methods concretely, Syntax check, Watermarking and Checksum schemes have been reformulated, combined and simulated

Robust H.263+ video for real-time Internet applications

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Decoding H.264/AVC using prior information and source constraints

The H.264/AVC standard employs a number of errorresilient mechanisms to correct transmission errors. These methods assume a packet-loss scenario, where all the macroblocks (MBs) contained within a corrupted slice are dropped and concealed. However, most of the MBs contained within corrupted slices provide minimal (if any) visual distortions and therefore concealing them causes a superfluous drop in the quality of the recovered video content. This paper presents a novel error control mechanism which employs prior information and residual source-redundancy to recover the most-likelihood feasible H.264/AVC bitstream. Simulation results show that the algorithm recovers a number of corrupted sequences and achieves overall Peak Signal-to-Noise Ratio (PSNR) gains between 1dB and 2dB over the standard. The proposed solution is compatible with the H.264/AVC with no additional bandwidth requirements.peer-reviewe

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