Inter-Domain Integration of Services and Service Management

The evolution of the global telecommunications industry into an open services market presents developers of telecommunication service and management systems with many new challenges. Increased competition, complex service provision chains and integrated service offerings require effective techniques for the rapid integration of service and management systems over multiple organisational domains. These integration issues have been examined in the ACTS project Prospect by developing a working set of integrated, managed telecommunications services for a user trial. This paper presents the initial results of this work detailing the technologies and standards used, the architectural approach taken and the application of this approach to specific services

Analysis and implementation of the Large Scale Video-on-Demand System

Next Generation Network (NGN) provides multimedia services over broadband based networks, which supports high definition TV (HDTV), and DVD quality video-on-demand content. The video services are thus seen as merging mainly three areas such as computing, communication, and broadcasting. It has numerous advantages and more exploration for the large-scale deployment of video-on-demand system is still needed. This is due to its economic and design constraints. It's need significant initial investments for full service provision. This paper presents different estimation for the different topologies and it require efficient planning for a VOD system network. The methodology investigates the network bandwidth requirements of a VOD system based on centralized servers, and distributed local proxies. Network traffic models are developed to evaluate the VOD system's operational bandwidth requirements for these two network architectures. This paper present an efficient estimation of the of the bandwidth requirement for the different architectures.Comment: 9 pages, 8 figure

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

Improving The Efficiency Of Video Transmission In Computer Networks

In-depth examination of current techniques for enhancing the efficiency of video transmission over digital networks is provided in this study. Due to the growing need for high-quality video content, optimizing video transmission is an important area of research. This review categorizes and in-depth examines a range of methods proposed in the literature to enhance video transmission effectiveness. ABR, DNN architecture, adaptive streaming, Quality of Service (QoS), error resilience, congestion control, video compression, and hardware acceleration for video provisioning are just a few of the cutting-edge techniques that are covered in the discussion, which ranges from the more traditional to the cutting-edge. This essay provides a methodical evaluation of the numerous tactics that are available, along with an analysis of their guiding principles, advantages, and disadvantages. The paper also offers a comparative analysis of various approaches, highlighting trends, gaps, and potential future research directions in this crucial domain, all of which help to create more efficient video compression and transmission paradigms in computer networks

Quality of Service improvements for real time multimedia applications using next generation network architectures and blockchain in Internet Service Provider cooperative scenario

Real time communications are becoming part of our daily life, requiring constrained requisites with the purpose of being enjoyed in harmony by end users. The factors ruling these requisites are Quality of Service parameters of the users' Internet connections. Achieving a satisfactory QoS level for real time communications depends on parameters that are strongly influenced by the quality of the network connections among the Internet Service Providers, which are located in the path between final users and Over The Top service providers that are supplying them with real time services. Final users can be: business people having real time videoconferences, or adopting crytpocurrencies in their exchanges, videogamers playing online games together with others residing in other countries, migrants talking with their relatives or watching their children growing up in their home countries, people with disabilities adopting tecnologies to help them, doctors performing remote surgeries, manufacturers adopting augmented reality devices to perform dangerous tasks. Each of them performing their daily activities are requiring specific QoS parameters to their ISPs, that nowadays seem to be unable to provide them with a satisfactory QoS level for these kinds of real time services. Through the adoption of next generation networks, such as the Information Centric Networking, it would be possible to overcome the QoS problems that nowadays are experienced. By adopting Blockchain technologies, in several use cases, it would be possible to improve those security aspects related to the non-temperability of information and privacy. I started this thesis analyzing next generation architectures enabling real time multimedia communications. In Software Defined Networking, Named Data Networking and Community Information Centric Networking, I highlighted potential approaches to solve QoS problems that are affecting real time multimedia applications. During my experiments I found that applications able to transmit high quality videos, such as 4k or 8k videos, or to directly interact with devices AR/VR enabled are missing for both ICN approaches. Then I proposed a REST interface for the enforcing of a specific QoS parameter, the round trip time (RTT) taking into consideration the specific use case of a game company that connects with the same telecommunication company of the final user. Supposing that the proposed REST APIs have been deployed in the game company and in the ISP, when one or more users are experiencing lag, the game company will try to ask the ISP to reduce the RTT for that specific user or that group of users. This request can be done by performing a call to a method where IP address(es) and the maximum RTT desired are passed. I also proposed other methods, through which it would be possible to retrieve information about the QoS parameters, and exchange, if necessary, an exceeding parameter in change of another one. The proposed REST APIs can also be used in more complex scenarios, where ISPs along the path are chained together, in order to improve the end to end QoS among Over The Top service provider and final users. To store the information exchanged by using the proposed REST APIs, I proposed to adopt a permissioned blockchain, analizying the ISPs cooperative use case with Hyperledger Fabric, where I proposed the adoption of the Proof of Authority consensus algorithm, to increase the throughput in terms of transactions per second. In a specific case that I examined, I am proposing a combination of Information Centric Networking and Blockchain, in an architecture where ISPs are exchanging valuable information regarding final Users, to improve their QoS parameters. I also proposed my smart contract for the gaming delay use case, that can be used to rule the communication among those ISPs that are along the path among OTT and final users. An extension of this work can be done, by defining billing costs for the QoS improvements

Multiparty/Multimedia Conferencing in Mobile Ad-Hoc Networks for Improving Communications between Firefighters

In current practice, firefighters’ communications systems are verbal, using a simplex Radio Frequency (RF) system (walkie-talkie). They use a push-to-talk mechanism in which only one person can talk at any time and all other firefighters will hear the messages. They use special codes (e.g. 1008, 1009, etc.) to express their current situation. Firefighters of the same team need to be in visual contact with each other at all times. This RF system does not support other functionalities (e.g. video communications, conference calls). In addition, because communication between firefighters is a flat structure, private communications is not possible. Mobile Ad-Hoc Networks (MANETs) are infrastructure-less and self-organized wireless networks of mobile devices, which are not based on any centralized control. MANETs are suitable for the hosting of a wide range of applications in emergency situations, such as natural or human-induced disasters, and military and commercial settings. Multimedia conferencing is an important category of application that can be deployed in MANETs. This includes well-known sets of applications, such as audio/video conferencing, data communications, and multiplayer games. Conferencing can be defined as the conversational exchange of data content between several parties. Conferencing requires, at the very least, the opening of two sessions: a call signaling session, and a media handling session. Call signaling is used to set up, modify, and terminate the conference. Media handling is used to cover the transportation of the media, and to control/manage the media mixers and media connections. So far, very little attention has been devoted to the firefighters’ communication system. In the present work, we focus on building a new communication system for firefighters using multimedia conferencing/sub-conferencing in MANETs. The background information for the firefighters’ current communications system and MANETs, along with the multimedia conferencing, is provided. The limitations of this system are determined, and the requirements are derived to determine the functionalities of a better communication system that will overcome current limitations. We have proposed a cluster-based signaling architecture that meets our requirements. We have also identified a state-of-the-art media handling and mixing system that meets most of our requirements, and have adapted it to inter-work with our signaling system. We have implemented the proposed architecture using SIP signaling protocol. Performance measurements have been performed on the prototype. Through experiments, we have found that the new multimedia communication system is a very promising approach to solve the current firefighters’ communication problems

Media handling for conferencing in MANETs

Mobile Ad hoc NETworks (MANETs) are formed by devices set up temporarily to communicate without using a pre-existing network infrastructure. Devices in these networks are disparate in terms of resource capabilities (e.g. processing power, battery energy). Multihop Cellular Networks (MCNs) incorporate multihop mobile ad-hoc paradigms into 3G conventional single-hop cellular networks. Conferencing, an essential category of applications in MANETs and MCNs, includes popular applications such as audio/video conferencing. It is defined as an interactive multimedia service comprising online exchange of multimedia content among several users. Conferencing requires two sessions: a call signaling session and a media handling session. Call signaling is used to set up, modify, and tear down conference sessions. Media handling deals with aspects such as media transportation, media mixing, and transcoding. In this thesis, we are concerned with media handling for conferencing in MANETs and MCNs. We propose an architecture based on two overlay networks: one for mixing and one for control. The first overlay is composed of nodes acting as mixers. Each node in the network has a media connection with one mixer in the first overlay. A novel distributed mixing architecture that minimizes the number of mixers in end-to-end paths is proposed as an architectural solution for this first overlay. A sub-network of nodes, called controllers, composes the second overlay. Each controller controls a set of mixers, and collectively, they manage and control the two-overlay network. The management and control tasks are assured by a media signaling architecture based on an extended version of Megaco/H.L248. The two-overlay network is self-organizing, and thus automatically assigns users to mixers, controls mixers and controllers, and recovers the network from failures. We propose a novel self-organizing scheme that has three components: self-growing, self-shrinking and self-healing. Self-growing and self-shrinking use novel workload balancing schemes that make decisions to enable and disable mixers and controllers. The workload balancing schemes use resources efficiently by balancing the load among the nodes according to their capabilities. Self-healing detects failed nodes and recovers the network when failures of nodes with responsibilities (mixers and controllers) occur. Detection of failed nodes is based on a novel application-level failure detection architecture. A novel architecture for media handling in MCNs is proposed. We use mediator concepts to connect the media handling entities of a MANET with the media entities of a 3G cellular network. A media mediator assures signaling and media connectivity between the two networks and acts as a translator of the different media handling protocols

Enterprise network convergence: path to cost optimization

During the past two decades, telecommunications has evolved a great deal. In the eighties, people were using television, radio and telephone as their communication systems. Eventually, the introduction of the Internet and the WWW immensely transformed the telecommunications industry. This internet revolution brought about a huge change in the way businesses communicated and operated. Enterprise networks now had an increasing demand for more bandwidth as they started to embrace newer technologies. The requirements of the enterprise networks grew as the applications and services that were used in the network expanded. This stipulation for fast and high performance communication systems has now led to the emergence of converged network solutions. Enterprises across the globe are investigating new ways to implement voice, video, and data over a single network for various reasons – to optimize network costs, to restructure their communication system, to extend next generation networking abilities, or to bridge the gap between their corporate network and the existing technological progress. To date, organizations had multiple network services to support a range of communication needs. Investing in this type of multiple communication infrastructures limits the networks ability to provide resourceful bandwidth optimization services throughout the system. Thus, as the requirements for the corporate networks to handle dynamic traffic grow day by day, the need for a more effective and efficient network arises. A converged network is the solution for enterprises aspiring to employ advanced applications and innovative services. This thesis will emphasize the importance of converging network infrastructure and prove that it leads to cost savings. It discusses the characteristics, architecture, and relevant protocols of the voice, data and video traffic over both traditional infrastructure and converged architecture. While IP-based networks present excellent quality for non real-time data networking, the network by itself is not capable of providing reliable, quality and secure services for real-time traffic. In order for IP networks to perform reliable and timely transmission of real-time data, additional mechanisms to reduce delay, jitter and packet loss are required. Therefore, this thesis will also discuss the important mechanisms for running real-time traffic like voice and video over an IP network. Lastly, it will also provide an example of an enterprise network specifications (voice, video and data), and present an in depth cost analysis of a typical network vs. a converged network to prove that converged infrastructures provide significant savings

Signaling for Internet Telephony

Internet telephony must offer the standard telephony services.However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network(PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services