dOTM: a mechanism for distributing centralized multi-party video conferencing in the cloud

One of the key factors for a given application to take advantage of cloud computing is the ability to scale in an efficient, fast and reliable way. In centralized multi-party video conferencing, dynamically scaling a running conversation is a complex problem. In this paper we propose a methodology to divide the Multipoint Control Unit (the video conferencing server) into more simple units, broadcasters. Each broadcaster receives the media from a participant, processes it and forwards it to the rest. These broadcasters can be distributed among a group of CPUs. By using this methodology, video conferencing systems can scale in a more granular way, improving the deployment

Service Oriented Architecture for VoIP Conferencing

Voice/Video over IP (VoIP) systems to date have been either highly centralized or dependent on the IP multicast in nature. Global Multimedia Collaboration System is a scalable, integrated and service-oriented VoIP conferencing system, based on the XGSP collaboration framework and NaradaBrokering messaging middleware. This system can provide media and session services to heterogeneous endpoints such as H.323, SIP, Access Grid, RealPlayer as well as cellular phone. In this paper, we address the challenges of scalability, interoperablity and heterogeneity in massive VoIP conferencing system. We believe that our approach opens up new opportunities for leveraging classic VoIP systems by using new technologies in service-oriented computing

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

Design and Implementation of Audio/Video Collaboration System Based on Publish/subscribe Event Middleware

In this paper we present our A/V collaboration system based on our XGSP collaboration framework and NaradaBrokering messaging middleware. Using publish/subscribe event model, this system can provide videoconferencing services to heterogeneous endpoints such as H.323, SIP and Access Grid. This paper discusses the common a/v collaboration model shared by all kinds of A/V conferencing clients and introduces the details about how to implement such a model based on publish/subscribe event middleware

A novel P2P and cloud computing hybrid architecture for multimedia streaming QoS cost functions

Since its appearance, peer-to-peer technology has given raise to various multimedia streaming applications. Today, cloud computing offers different service models as a base for successful end user applications. In this paper we propose joining peer-to-peer and cloud computing into new architectural realization of a distributed cloud computing network for multimedia streaming, in a both centralized and peer-to-peer distributed manner. This architecture merges private and public clouds and it is intended for a commercial use, but in the same time scalable to offer the possibility of non-profitable use. In order to take advantage of the cloud paradigm and make multimedia streaming more efficient, we introduce APIs in the cloud, containing build-in functions for automatic QoS calculation, which permits negotiating QoS parameters such as bandwidth, jitter and latency, among a cloud service provider and its potential clients

A scalable WebRTC platform based on open technologies

International Conference on Computer, Information and Telecommunication Systems (7th, 2018, Colmar, Francia

Multi-Stream Management for Supporting Multi-Party 3D Tele-Immersive Environments

Three-dimensional tele-immersive (3DTI) environments have great potential to promote collaborative work among geographically distributed participants. However, extensive application of 3DTI environments is still hindered by problems pertaining to scalability, manageability and reliance of special-purpose components. Thus, one critical question is how to organize the acquisition, transmission and display of large volume real-time 3D visual data over commercially available computing and networking infrastructures so that .everybody. would be able to install and enjoy 3DTI environments for high quality tele-collaboration. In the thesis, we explore the design space from the angle of multi-stream Quality-of-Service (QoS) management to support multi-party 3DTI communication. In 3DTI environments, multiple correlated 3D video streams are deployed to provide a comprehensive representation of the physical scene. Traditional QoS approach in 2D and single-stream scenario has become inadequate. On the other hand, the existence of multiple streams provides unique opportunity for QoS provisioning. We propose an innovative cross-layer hierarchical and distributed multi-stream management middleware framework for QoS provisioning to fully enable multi-party 3DTI communication over general delivery infrastructure. The major contributions are as follows. First, we introduce the view model for representing the user interest in the application layer. The design revolves around the concept of view-aware multi-stream coordination, which leverages the central role of view semantics in 3D video systems. Second, in the stream differentiation layer we present the design of view to stream mapping, where a subset of relevant streams are selected based on the relative importance of each stream to the current view. Conventional streaming controllers focus on a fixed set of streams specified by the application. Different from all the others, in our management framework the application layer only specifies the view information while the underlying controller dynamically determines the set of streams to be managed. Third, in the stream coordination layer we present two designs applicable in different situations. In the case of end-to-end 3DTI communication, a learning-based controller is embedded which provides bandwidth allocation for relevant streams. In the case of multi-party 3DTI communication, we propose a novel ViewCast protocol to coordinate the multi-stream content dissemination upon an end-system overlay network

Journal of Telecommunications and Information Technology, nr 3

kwartalni

Livenet: A low-latency video transport network for large-scale live streaming

Low-latency live streaming has imposed stringent latency requirements on video transport networks. In this paper we report on the design and operation of the Alibaba low-latency video transport network, LiveNet. LiveNet builds on a flat CDN overlay with a centralized controller for global optimization. As part of this, we present our design of the global routing computation and path assignment, as well as our fast data transmission architecture with fine-grained control of video frames. The performance results obtained from three years of operation demonstrate the effectiveness of LiveNet in improving CDN performance and QoE metrics. Compared with our prior state-of-The-Art hierarchical CDN deployment, LiveNet halves the CDN delay and ensures 98% of views do not experience stalls and that 95% can start playback within 1 second. We further report our experiences of running LiveNet over the last 3 years