Multimedia Teleservices Modelled with the OSI Application Layer Structure

This paper looks into the communications capabilities that are required by distributed multimedia applications to achieve relation preserving information exchange. These capabilities are derived by analyzing the notion of information exchange and are embodied in communications functionalities. To emphasize the importance of the users' view, a top-down approach is applied. The (revised) OSI Application Layer Structure (OSI-ALS) is used to model the communications functionalities and to develop an architecture for composition of multimedia services with these functionalities. This work may therefore be considered an exercise to evaluate the suitability of OSI-ALS for composition of multimedia teleservices

MHEG: An Interchange Format for Interactive Multimedia Presentations

This paper surveys the upcoming MHEG standard, as under development by the Multimedia Hypermedia Experts Group of ISO. It's scope is to define a system-independent encoding of the structure information which can be used for storing, exchanging, and executing of multimedia presentations in final form. We use a running example to describe the building blocks of an MHEG encoded presentation. To contribute our experience using MHEG in a networked multimedia kiosk environment, we present a complete MHEG run-time environment

Service composition based on SIP peer-to-peer networks

Today the telecommunication market is faced with the situation that customers are requesting for new telecommunication services, especially value added services. The concept of Next Generation Networks (NGN) seems to be a solution for this, so this concept finds its way into the telecommunication area. These customer expectations have emerged in the context of NGN and the associated migration of the telecommunication networks from traditional circuit-switched towards packet-switched networks. One fundamental aspect of the NGN concept is to outsource the intelligence of services from the switching plane onto separated Service Delivery Platforms using SIP (Session Initiation Protocol) to provide the required signalling functionality. Caused by this migration process towards NGN SIP has appeared as the major signalling protocol for IP (Internet Protocol) based NGN. This will lead in contrast to ISDN (Integrated Services Digital Network) and IN (Intelligent Network) to significantly lower dependences among the network and services and enables to implement new services much easier and faster. In addition, further concepts from the IT (Information Technology) namely SOA (Service-Oriented Architecture) have largely influenced the telecommunication sector forced by amalgamation of IT and telecommunications. The benefit of applying SOA in telecommunication services is the acceleration of service creation and delivery. Main features of the SOA are that services are reusable, discoverable combinable and independently accessible from any location. Integration of those features offers a broader flexibility and efficiency for varying demands on services. This thesis proposes a novel framework for service provisioning and composition in SIP-based peer-to-peer networks applying the principles of SOA. One key contribution of the framework is the approach to enable the provisioning and composition of services which is performed by applying SIP. Based on this, the framework provides a flexible and fast way to request the creation for composite services. Furthermore the framework enables to request and combine multimodal value-added services, which means that they are no longer limited regarding media types such as audio, video and text. The proposed framework has been validated by a prototype implementation

Video Quality Prediction for Video over Wireless Access Networks (UMTS and WLAN)

Transmission of video content over wireless access networks (in particular, Wireless Local Area Networks (WLAN) and Third Generation Universal Mobile Telecommunication System (3G UMTS)) is growing exponentially and gaining popularity, and is predicted to expose new revenue streams for mobile network operators. However, the success of these video applications over wireless access networks very much depend on meeting the user’s Quality of Service (QoS) requirements. Thus, it is highly desirable to be able to predict and, if appropriate, to control video quality to meet user’s QoS requirements. Video quality is affected by distortions caused by the encoder and the wireless access network. The impact of these distortions is content dependent, but this feature has not been widely used in existing video quality prediction models. The main aim of the project is the development of novel and efficient models for video quality prediction in a non-intrusive way for low bitrate and resolution videos and to demonstrate their application in QoS-driven adaptation schemes for mobile video streaming applications. This led to five main contributions of the thesis as follows:(1) A thorough understanding of the relationships between video quality, wireless access network (UMTS and WLAN) parameters (e.g. packet/block loss, mean burst length and link bandwidth), encoder parameters (e.g. sender bitrate, frame rate) and content type is provided. An understanding of the relationships and interactions between them and their impact on video quality is important as it provides a basis for the development of non-intrusive video quality prediction models.(2) A new content classification method was proposed based on statistical tools as content type was found to be the most important parameter. (3) Efficient regression-based and artificial neural network-based learning models were developed for video quality prediction over WLAN and UMTS access networks. The models are light weight (can be implemented in real time monitoring), provide a measure for user perceived quality, without time consuming subjective tests. The models have potential applications in several other areas, including QoS control and optimization in network planning and content provisioning for network/service providers.(4) The applications of the proposed regression-based models were investigated in (i) optimization of content provisioning and network resource utilization and (ii) A new fuzzy sender bitrate adaptation scheme was presented at the sender side over WLAN and UMTS access networks. (5) Finally, Internet-based subjective tests that captured distortions caused by the encoder and the wireless access network for different types of contents were designed. The database of subjective results has been made available to research community as there is a lack of subjective video quality assessment databases.Partially sponsored by EU FP7 ADAMANTIUM Project (EU Contract 214751

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

Framework for Automated Functional Tests within Value-Added Service Environments

Full version unavailable due to 3rd party copyright restrictions.Recent years have witnessed that standard telecommunication services evolved more and more to next generation value-added services. This fact is accompanied by a change of service characteristics as new services are designed to fulfil the customer’s demands instead of just focussing on technologies and protocols. These demands can be very specific and, therefore, diverse potential service functionalities have to be considered by the service providers. To make matters worse for service providers, a fast transition from concept to market product and low price of a new service is required due to the increasing competition in the telecommunication industry. Therefore, effective test solutions need to be developed that can be integrated in current value-added service development life-cycles. Besides, these solutions should support the involvement of all participating stakeholders such as the service provider, the test developers as well as the service developers, and, in order to consider an agile approach, also the service customer. This thesis proposes a novel framework for functional testing that is based on a new sort of description language for value-added services (Service Test Description). Based on instances of the Service Test Description, sets of reusable test components described by means of an applied Statecharts notation are automatically selected and composed to so-called behaviour models. From the behaviour models, abstract test cases can be automatically generated which are then transformed to TTCN-3 test cases and then assembled to an Executable Test Suite. Within a TTCN-3 test system, the Executable Test Suite can be executed against the corresponding value-added service referred to as System Under Test. One benefit of the proposed framework is its application within standard development life-cycles. Therefore, the thesis presents a methodology that considers both service development and test development as parallel tasks and foresees procedures to synchronise the tasks and to allow an agile approach with customer involvement. The novel framework is validated through a proof-of-concept working prototype. Example value-added services have been chosen to illustrate the whole process from compiling instances of the Service Test Description until the execution of automated tests. Overall, this thesis presents a novel solution for service providers to improve the quality of their provided value-added services through automated functional testing procedures. It enables the early involvement of the customers into the service development life-cycle and also helps test developers and service developers to collaborate