User-Oriented QoS in Packet Video Delivery

We focus on packet video delivery, with an emphasis on the quality of service perceived by the end-user. A video signal passes through several subsystems, such as the source coder, the network and the decoder. Each of these can impair the information, either by data loss or by introducing delay. We describe how each of the subsystems can be tuned to optimize the quality of the delivered signal, for a given available bit rate in the network. The assessment of end-user quality is not trivial. We present recent research results, which rely on a model of the human visual system

Supporting real time video over ATM networks

Includes bibliographical references.In this project, we propose and evaluate an approach to delimit and tag such independent video slice at the ATM layer for early discard. This involves the use of a tag cell differentiated from the rest of the data by its PTI value and a modified tag switch to facilitate the selective discarding of affected cells within each video slice as opposed to dropping of cells at random from multiple video frames

Distributed multimedia systems

A distributed multimedia system (DMS) is an integrated communication, computing, and information system that enables the processing, management, delivery, and presentation of synchronized multimedia information with quality-of-service guarantees. Multimedia information may include discrete media data, such as text, data, and images, and continuous media data, such as video and audio. Such a system enhances human communications by exploiting both visual and aural senses and provides the ultimate flexibility in work and entertainment, allowing one to collaborate with remote participants, view movies on demand, access on-line digital libraries from the desktop, and so forth. In this paper, we present a technical survey of a DMS. We give an overview of distributed multimedia systems, examine the fundamental concept of digital media, identify the applications, and survey the important enabling technologies.published_or_final_versio

Designing new network adaptation and ATM adaptation layers for interactive multimedia applications

Multimedia services, audiovisual applications composed of a combination of discrete and continuous data streams, will be a major part of the traffic flowing in the next generation of high speed networks. The cornerstones for multimedia are Asynchronous Transfer Mode (ATM) foreseen as the technology for the future Broadband Integrated Services Digital Network (B-ISDN) and audio and video compression algorithms such as MPEG-2 that reduce applications bandwidth requirements. Powerful desktop computers available today can integrate seamlessly the network access and the applications and thus bring the new multimedia services to home and business users. Among these services, those based on multipoint capabilities are expected to play a major role.    Interactive multimedia applications unlike traditional data transfer applications have stringent simultaneous requirements in terms of loss and delay jitter due to the nature of audiovisual information. In addition, such stream-based applications deliver data at a variable rate, in particular if a constant quality is required.    ATM, is able to integrate traffic of different nature within a single network creating interactions of different types that translate into delay jitter and loss. Traditional protocol layers do not have the appropriate mechanisms to provide the required network quality of service (QoS) for such interactive variable bit rate (VBR) multimedia multipoint applications. This lack of functionalities calls for the design of protocol layers with the appropriate functions to handle the stringent requirements of multimedia.    This thesis contributes to the solution of this problem by proposing new Network Adaptation and ATM Adaptation Layers for interactive VBR multimedia multipoint services.    The foundations to build these new multimedia protocol layers are twofold; the requirements of real-time multimedia applications and the nature of compressed audiovisual data.    On this basis, we present a set of design principles we consider as mandatory for a generic Multimedia AAL capable of handling interactive VBR multimedia applications in point-to-point as well as multicast environments. These design principles are then used as a foundation to derive a first set of functions for the MAAL, namely; cell loss detection via sequence numbering, packet delineation, dummy cell insertion and cell loss correction via RSE FEC techniques.    The proposed functions, partly based on some theoretical studies, are implemented and evaluated in a simulated environment. Performances are evaluated from the network point of view using classic metrics such as cell and packet loss. We also study the behavior of the cell loss process in order to evaluate the efficiency to be expected from the proposed cell loss correction method. We also discuss the difficulties to map network QoS parameters to user QoS parameters for multimedia applications and especially for video information. In order to present a complete performance evaluation that is also meaningful to the end-user, we make use of the MPQM metric to map the obtained network performance results to a user level. We evaluate the impact that cell loss has onto video and also the improvements achieved with the MAAL.    All performance results are compared to an equivalent implementation based on AAL5, as specified by the current ITU-T and ATM Forum standards.    An AAL has to be by definition generic. But to fully exploit the functionalities of the AAL layer, it is necessary to have a protocol layer that will efficiently interface the network and the applications. This role is devoted to the Network Adaptation Layer.    The network adaptation layer (NAL) we propose, aims at efficiently interface the applications to the underlying network to achieve a reliable but low overhead transmission of video streams. Since this requires an a priori knowledge of the information structure to be transmitted, we propose the NAL to be codec specific.    The NAL targets interactive multimedia applications. These applications share a set of common requirements independent of the encoding scheme used. This calls for the definition of a set of design principles that should be shared by any NAL even if the implementation of the functions themselves is codec specific. On the basis of the design principles, we derive the common functions that NALs have to perform which are mainly two; the segmentation and reassembly of data packets and the selective data protection.    On this basis, we develop an MPEG-2 specific NAL. It provides a perceptual syntactic information protection, the PSIP, which results in an intelligent and minimum overhead protection of video information. The PSIP takes advantage of the hierarchical organization of the compressed video data, common to the majority of the compression algorithms, to perform a selective data protection based on the perceptual relevance of the syntactic information.    The transmission over the combined NAL-MAAL layers shows significant improvement in terms of CLR and perceptual quality compared to equivalent transmissions over AAL5 with the same overhead.    The usage of the MPQM as a performance metric, which is one of the main contributions of this thesis, leads to a very interesting observation. The experimental results show that for unexpectedly high CLRs, the average perceptual quality remains close to the original value. The economical potential of such an observation is very important. Given that the data flows are VBR, it is possible to improve network utilization by means of statistical multiplexing. It is therefore possible to reduce the cost per communication by increasing the number of connections with a minimal loss in quality.    This conclusion could not have been derived without the combined usage of perceptual and network QoS metrics, which have been able to unveil the economic potential of perceptually protected streams.    The proposed concepts are finally tested in a real environment where a proof-of-concept implementation of the MAAL has shown a behavior close to the simulated results therefore validating the proposed multimedia protocol layers

Performance analysis of an ATM network with multimedia traffic: a simulation study

Traffic and congestion control are important in enabling ATM networks to maintain the Quality of Service (QoS) required by end users. A Call Admission Control (CAC) strategy ensures that the network has sufficient resources available at the start of each call, but this does not prevent a traffic source from violating the negotiated contract. A policing strategy (User Parameter Control (UPC)) is also required to enforce the negotiated rates for a particular connection and to protect conforming users from network overload. The aim of this work is to investigate traffic policing and bandwidth management at the User to Network Interface (UNI). A policing function is proposed which is based on the leaky bucket (LB) which offers improved performance for both real time (RT) traffic such as speech and video and non-real time (non-RT) traffic, mainly data by taking into account the QoS requirements. A video cell in violation of the negotiated bit rate causes the remainder of the slice to be discarded. This 'tail clipping' provides protection for the decoder from damaged video slices. Speech cells are coded using a frequency domain coder, which places the most significant bits of a double speech sample into a high priority cell and the least significant bits into a high priority cell. In the case of congestion, the low priority cell can be discarded with little impact on the intelligibility of the received speech. However, data cells require loss-free delivery and are buffered rather than being discarded or tagged for subsequent deletion. This triple strategy is termed the super leaky bucket (SLB). Separate queues for RT and non-RT traffic, are also proposed at the multiplexer, with non pre-emptive priority service for RT traffic if the queue exceeds a predetermined threshold. If the RT queue continues to grow beyond a second threshold, then all low priority cells (mainly speech) are discarded. This scheme protects non-RT traffic from being tagged and subsequently discarded, by queueing the cells and also by throttling back non-RT sources during periods of congestion. It also prevents the RT cells from being delayed excessively in the multiplexer queue. A simulation model has been designed and implemented to test the proposal. Realistic sources have been incorporated into the model to simulate the types of traffic which could be expected on an ATM network. The results show that the S-LB outperforms the standard LB for video cells. The number of cells discarded and the resulting number of damaged video slices are significantly reduced. Dual queues with cyclic service at the multiplexer also reduce the delays experienced by RT cells. The QoS for all categories of traffic is preserved

Performance of Digitally Compressed (MPEG) Picture Transmission via Real Transmission Systems

MPEG je jedna od najpopularnijih serija standarda za video/audiokompresiju, pogodnih za različite aplikacije, ali temeljenih na sličnim principima. Ako se MPEG komprimirani videosignal treba prenijeti do udaljenog korisnika, na raspolaganju su različite mrežne tehnologije, primjerice ATM i IP, pri čemu je od interesa osigurati potrebnu kvalitetu usluge, i to uporabom najekonomičnije tehnologije na raspolaganju. Da bi se to ostvarilo, potrebno je nadzirati parametre kvalitete usluge na razini relevantne prijenosne tehnologije (kakvi su npr. varijacije kašnjenja i gubitak podatkovnih jedinica protokola kojim se koristi – okvira, ćelija ili paketa), i odabrati odgovarajuće metode za njihovo držanje na razini koju (zbog smanjenja redundancije) osjetljivi komprimirani MPEG niz može tolerirati. S tim u vezi, u ovome je radu, s jedne strane, predstavljen primjer tehnika mjerenja i testiranja prijenosnog sustava (ATM), a s druge strane, dostignuta je perceptualna kvaliteta prenesenoga MPEG signala, a zatim su doneseni odgovarajući zaključci o razini utjecaja gubitaka podataka u mreži na kvalitetu primljenog videosignala.MPEG is one of the most popular families of audio/video compression techniques, suitable for different applications but still based on similar principles. If an MPEG compressed video signal is to be transmitted to the remote user, various network technologies are available, such as eg. ATM and IP, where it is of interest to provide the needed quality of service (QoS) still using the most economical technology available. In order to accomplish that goal, it is necessary to monitor the QoS parameters of the relevant transmission technology (such as eg. delay variation and loss of protocol data units – frames, cells and packets), and then select appropriate methods for keeping the parameters values at the level that the vulnerable (due to reduction of the redundancy) compressed MPEG stream can tolerate. With this respect, in this paper, from one point of view, the example of test and measurement techniques as applied to the transmission system (ATM), and from another point of view, the achieved perceptual quality of the transmitted MPEG signal, have been considered, and the appropriate conclusions made about the degree of the impact of network packet loss on the quality of the received video signal

Decentralization of multimedia content in a heterogeneous environment

The aim of this study has been the decentralization of multimedia content in a heterogeneous environment. The environment consisted of the research networks connecting the European Organization for Nuclear Research and the Finnish University and Research Network. The European Organization for Nuclear Research produces multimedia content which can be used as studying material all over the world. The Web University pilot in the European Organization for Nuclear Research has been developing a multimedia content delivery service for years. Delivering the multimedia content requires plenty of capacity from the network infrastructure. Different content of the material can have different demands for the network. In a heterogeneous environment, like the Internet, fulfilling all the demands can be a problem. Several methods exist to improve the situation. Decentralization of the content is one of the most popular solutions. Mirroring and caching are the main methods for decentralization. Recently developed content delivery networks are using both of these techniques to satisfy the demands of the content. The practical application consisted of measurements of the network connection between the multimedia server in the European Organization for Nuclear Research and the Finnish University and Research Network, planning and building a decentralization system for the multimedia content. After the measurements, it became clear that there is n o need for decentralization of the multimedia content for users that are able to utilise the Finnish University and Research Network. There could be double today's usage, and still there would be no problems with the capacity. However, the European Organization for Nuclear Research routes all traffic that comes from outside research networks through a gateway in the USA. This affects every connection that is made from Finland: users are not able to use the international connection offered by the Finnish University and Research Network. For these users I designed and built a simple, modular and portable decentralization system

New Network and ATM Adaptation Layers for Real-Time Multimedia Applications: A Performance Study Based on Psychophysics

We present in this paper Network and ATM Adaptation Layers for real-time multimedia applications. These layers provide a robust transmission by applying per-cell sequence numbering combined with a selective Forward Error Correction (FEC) mechanism based on Burst Erasure codes. We compare their performance against a transmission over AAL5 by simulating the transport of an MPEG-2 sequence over an ATM network. Performance is measured in terms of Cell Loss Ratio (CLR) and user perceived quality. The proposed layers achieve an improvement on the cell loss figures obtained for AAL5 of about one order of magnitude under the same traffic conditions. To evaluate the impact of cell losses at the application level, we apply a perceptual quality measure to the decoded MPEG-2 sequences. From a perceptual point of view, the proposed AAL achieves a graceful quality degradation compared to AAL5 which shows a critical CLR value beyond which quality drops very fast. The application of a selective FEC achieves an even smoother image quality degradation with a small overhead