Enhanced adaptive RTCP-based inter-destination multimedia synchronization approach for distributed applications

[EN] Newer social multimedia applications, such as Social TV or networked multi-player games, enable independent groups (or clusters) of users to interact among themselves and share services within the context of simultaneous media content consumption. In such scenarios, concurrently synchronized playout points must be ensured so as not to degrade the user experience on such interaction. We refer to this process as Inter-Destination Multimedia Synchronization (IDMS). This paper presents the design, implementation and evaluation of an evolved version of an RTCP-based IDMS approach, including an Adaptive Media Playout (AMP) scheme that aims to dynamically and smoothly adjust the playout timing of each one of the geographically distributed consumers in a specific cluster if an allowable asynchrony threshold between their playout states is exceeded. For that purpose, we previously had also to develop a full implementation of RTP/RTCP protocols for NS-2, in which we included the IDMS approach as an optional functionality. Simulation results prove the feasibility of such IDMS and AMP proposals, by adopting several dynamic master reference selection policies, to maintain an overall synchronization status (within allowable limits) in each cluster of participants, while minimizing the occurrence of long-term playout discontinuities (such as skips/pauses) which are subjectively more annoying and less tolerable to users than small variations in the media playout rate.This work has been financed, partially, by Universitat Politecnica de Valencia (UPV), under its R&D Support Program in PAID-05-11-002-331 Project and in PAID-01-10. Authors also would like to thank the anonymous reviewers that helped to significantly improve the quality of the paper with their constructive comments.Montagud, M.; Boronat, F. (2012). Enhanced adaptive RTCP-based inter-destination multimedia synchronization approach for distributed applications. Computer Networks. 56(12):2912-2933. https://doi.org/10.1016/j.comnet.2012.05.00329122933561

An adaptive algorithm for Internet multimedia delivery in a DiffServ environment.

To meet the Quality of Service (QoS) requirements of multimedia applications and to reduce the network congestion, several service models and mechanisms have been proposed. Among these, Differentiated Service (DiffServ) architecture has been considered as a scalable and flexible QoS architecture for the Internet. DiffServ provides class-based QoS guarantees. Applications in different classes receive different QoS and are priced differently. If network congestion occurs, DiffServ may not be able to guarantee the QoS for the application. Thus, the QoS may not reflect the price paid for the service. A problem of considerable economic and research importance is how to achieve a good price and quality tradeoff even at times of congestion. This thesis presents an Adaptive Class Switching Algorithm (ACSA) which intends to provide good quality with good price for real-time multimedia applications in a DiffServ environment. The ACSA algorithm combines the techniques of Real-time Transport Protocol (RTP), DiffServ, and Adaptation together. It also takes both QoS and price into account to provide users a good QoS with a good price. The algorithm dynamically selects the most suitable class based on both the QoS feedback received and the highest user utility. The user utility is a function of quality, price, and the weight which reflects the relative sensitivity to quality and price. The class with the highest user utility is the class that provides the best quality and price tradeoff. The QoS feedback is conveyed by RTP\u27s Control Protocol (RTCP) Receiver Reports. The results of simulation demonstrate that ACSA can react fast to the current class state in the network and reflects the best QoS and price tradeoff. It always seeks to find a class which provides the highest user utility except when the Internet is congested and the required QoS in all classes can not be satisfied. If this happens, the real-time multimedia flow chooses Best-Effort class with no payment. (Abstract shortened by UMI.) Paper copy at Leddy Library: Theses & Major Papers - Basement, West Bldg. / Call Number: Thesis2005 .F46. Source: Masters Abstracts International, Volume: 44-01, page: 0389. Thesis (M.Sc.)--University of Windsor (Canada), 2005

A More Realistic RTP/RTCP-Based Simulation Platform for Video Streaming QoS Evaluation

[EN] Over the last few years, the demand for real-time multimedia services has been growing progressively so that video streaming applications are expected to be dominant in future communications systems, and most of them using RTP/RTCP protocols. This paper presents an evolved tool-set for video streaming QoS evaluation in simulated environments using such protocols. We have designed a new NS-2 module with a full RTP/RTCP implementation (following strictly the RFC 3550) and we propose to combine it with additional multimedia tools to obtain an advanced simulation framework that allows the measurement of network-level QoS metrics (such as throughput, delay, jitter or loss rate) in simulation time. Besides, as the transmitted video files can be reconstructed and played out at the receiver side, the measurement of the quality of the delivered video streams, by employing the most common objective quality metrics (such as PSNR, SSIM or VQM) or subjective metrics (MOS), is also supported. By using this tool-set, researchers and practitioners can assess their novel designs (such as network protocols, routing strategies or video coding mechanisms) for such applications in heterogeneous scenarios over different network conditions. As RTCP feedback capabilities have been added, source based control techniques (such as rate adaptability or Multiple Description Coding) could be included and tested using this more realistic and powerful simulation platform.This work has been financed, partially, by Polytechnics University of Valencia (UPV), under its R&D Support Program in PAID-06-08-002-585 Project and in PAID-01-10, and by Generalitat Valenciana, under its R&D Support Program in GV 2010/009 Project.Boronat Segui, F.; Montagud Aguar, M.; Vidal Gimeno, VE. (2011). A More Realistic RTP/RTCP-Based Simulation Platform for Video Streaming QoS Evaluation. Journal of Mobile Multimedia. 7(1):66-88. http://hdl.handle.net/10251/45964S66887

Enhanced QoS for real-time multimedia delivery over the wireless link using RFID technology.

This thesis presents a Sensor Guided Wireless Adaptation Scheme (SGWAS) that works in a micromobility domain. SGWAS infers the reason of high packet loss in a realtime multimedia flow received by a mobile node in a wireless cell. Determining the reason of packet loss relies on information obtained from wireless sensors, specifically RFID devices scattered in the cell, to detect the location of the mobile node. If packet loss is due to wireless link congestion, then local rate adaptation is applied in the cell. However, if it is due to handoff or signal propagation effects, e.g. obstruction or attenuation, then rate adaptation is not performed. The source adapts its transmission rate if congestion occurs in the wired network. SGWAS helps improve the quality of service and avoids unnecessary rate adaptation. Simulation results demonstrate that SGWAS identifies the reason of high packet loss and performs rate adaptation only when needed.Dept. of Computer Science. Paper copy at Leddy Library: Theses & Major Papers - Basement, West Bldg. / Call Number: Thesis2006 .E58. Source: Masters Abstracts International, Volume: 45-01, page: 0352. Thesis (M.Sc.)--University of Windsor (Canada), 2006

Design, Development and Assessment of Control Schemes for IDMS in a Standardized RTCP-based Solution

[EN] Currently, several media sharing applications that allow social interactions between distributed users are gaining momentum. In these networked scenarios, synchronized playout between the involved participants must be provided to enable truly interactive and coherent shared media experiences. This research topic is known as Inter-Destination Media Synchronization (IDMS). This paper presents the design and development of an advanced IDMS solution, which is based on extending the capabilities of RTP/RTCP standard protocols. Particularly, novel RTCP extensions, in combination with several control algorithms and adjustment techniques, have been specified to enable an adaptive, highly accurate and standard compliant IDMS solution. Moreover, as different control or architectural schemes for IDMS exist, and each one is best suited for specific use cases, the IDMS solution has been extended to be able to adopt each one of them. Simulation results prove the satisfactory responsiveness of our IDMS solution in a small scale scenario, as well as its consistent behavior, when using each one of the deployed architectural schemes.This work has been financed, partially, by Universitat Politecnica de Valencia (UPV), under its R&D Support Program in PAID-01-10. TNO's work has been partially funded by European Community's Seventh Framework Programme (FP7/2007-2013) under Grant Agreement No. ICT-2011-8-318343 (STEER Project). CWI's work has been partially funded by the European Community's Seventh Framework Programme (FP7/2007-2013) under Grant Agreement No. ICT-2011-7-287723 (REVERIE Project).Montagud Aguar, M.; Boronat Segui, F.; Stokking, H.; Cesar, P. (2014). Design, Development and Assessment of Control Schemes for IDMS in a Standardized RTCP-based Solution. Computer Networks. 70:240-259. https://doi.org/10.1016/j.comnet.2014.06.004S2402597

Enabling reliable and power efficient real-time multimedia delivery over wireless sensor networks

There is an increasing need to run real-time multimedia applications, e.g. battle field and border surveillance, over Wireless Sensor Networks (WSNs). In WSNs, packet delivery exhibits high packet loss rate due to congestion, wireless channel high bit error rate, route failure, signal attenuation, etc... Flooding conventional packets over all sensors redundantly provides reliable delivery. However, flooding real-time multimedia packets is energy inefficient for power limited sensors and causes severe contentions affecting reliable delivery. We propose the Flooding Zone Initialization Protocol (FZIP) to enhance reliability and reduce power consumption of real-time multimedia flooding in WSNs. FZIP is a setup protocol which constrains flooding within a small subset of intermediate nodes called Flooding Zone (FZ). Also, we propose the Flooding Zone Control Protocol (FZCP) which monitors the session quality and dynamically changes the FZ size to adapt to current network state, thus providing a tradeoff of good quality and less power consumption

Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods

Rate-control for conversational H.264 video communication in heterogeneous networks

The transmission bit rate available along a communication path in a heterogeneous network is highly variable. The wireless link quality may vary due to interference and fading phenomena and, peered with radio layer reconfiguration and link layer protection mechanisms, lead to varying error rates, latencies, and, most importantly, changes in the available bit rate. And in both fixed and wireless networks, varying amounts of cross traffic from other nodes (i.e., the total offered load on the individual links of a network path) may lead to fluctuations in queue size (reflected again in a path latency) and to congestion (reflected in packet drops from router quenes). Senders have to adapt dynamically to these network conditions and adjust their sending rate and possibly other transmission parameters (such as encoding or redundancy) to match the available bit rate while maximizing the media quality perceived at the receiver. We investigate congestion indicators and their characteristics in different multimedia environments. Taking these characteristics into account, we propose a rate-adaptation algorithm that works in the following environments: a) Mobile-Mobile, b) Internet-Internet and c) Heterogeneous, Mobile-Internet scenarios. Using metrics such as Peak Signal-to-Noise Ratio (PSNR), loss rate, bandwidth utilization and fairness, we compare the algorithm with other rate-control algorithms for conversational video communication

Performance evaluation of information and communications technology infrastructure for smart distribution network applications

This thesis was submitted for the degree of Master of Philosophy and awarded by Brunel University.Current electrical networks require secure, scalable and cost-effective Information and Communications Technology (ICT) solutions to facilitate the novel functionalities required by Smart Grids. Countries around the globe are investigating alternative energy sources to mitigate the current energy crisis and environmental issues experienced by many countries due to global warming, rapid growth of population, inefficient energy management, dwindling fossil fuel resources, etc. Therefore, alternative or renewable energy sources, such as wind, solar, hydro, combined heat and power, etc., are required to mitigate such a crisis and such sources will also need to be integrated in to the power grid in a distributed manner. Such distributed energy sources are mainly connected to the distribution networks and introduce huge challenges to the distribution network operator (DNO). Many of these challenges cannot be dealt with effectively using existing network operation mechanisms therefore the research and development of novel ICT solutions to support smart distribution network operation is required. This research investigated suitable ICT solutions to enable the Smart Grid to tackle these challenges and proposes ICT infrastructure models that can be used for simulation studies in order to investigate cost-effective, scalable and secure solutions for the DNOs. Initially, a Quality of Service (QoS) monitoring test-bed was proposed to evaluate the performance of bandwidth intensive applications, such as smart meter data transmission. Simulation studies for different communication technologies, cellular and Power Line Communication (PLC), were also carried out and the simulation models were verified using experimental test results. Finally, the modelling and analysis of smart metering infrastructure was carried out using simulation and extensive studies were performed to evaluate the data transmission rate performance for different configurations of smart meters and concentrators