Health Information Exchange (HIE) exhibits remarkable benefits for patient care such as improving healthcare quality and expediting coordinated care. The Office of the National Coordinator (ONC) for Health Information Technology is seeking patient-centric HIE designs that shift data ownership from providers to patients. There are multiple barriers to patient-centric HIE in the current system, such as security and privacy concerns, data inconsistency, timely access to the right records across multiple healthcare facilities. After investigating the current workflow of HIE, this paper provides a feasible solution to these challenges by utilizing the unique features of blockchain, a distributed ledger technology which is considered "unhackable". Utilizing the smart contract feature, which is a programmable self-executing protocol running on a blockchain, we developed a blockchain model to protect data security and patients' privacy, ensure data provenance, and provide patients full control of their health records. By personalizing data segmentation and an "allowed list" for clinicians to access their data, this design achieves patient-centric HIE. We conducted a large-scale simulation of this patient-centric HIE process and quantitatively evaluated the model's feasibility, stability, security, and robustness.

An interdisciplinary effort is needed for solving the fake news crisis, because the solutions depend not only on AI, but also on social mechanisms. In this paper, we propose an AI blockchain platform to build a strong collaboration among AI blockchain researchers and news media to advance the research fighting against fake news. This platform will provide journalists with blockchain crowd-sourced and AI validated factual data on emerging news. This platform will gather blockchain traced data and AI tools that can provide pointers to the original data sources, news propagation path, AI analyzed experts to consult on a given topic. This will provide journalists with cheaper and reliable sources of information in the Internet social media age. So that factual-sourced reporting can outpace the spread of fake news on social media which will encourage factual news sources as a way to value and promote truth for society. The technical contributions of this paper are (1) mechanism building the factual news database, (2) mechanism generating the news blockchain supply chain graph, and (3) AI blockchain based crowd sourcing fake news ranking mechanisms (4) AI blockchain platform for trusting news ecosystem. (5) reviewing the state of fake news research from the technology and social aspects, and providing list of key research issues and technical challenges.

This paper proposes a blockchain platform architecture for clinical trial and precision medicine and discusses various design aspects and provides some insights in the technology requirements and challenges. We identify 4 new system architecture components that are required to be built on top of traditional blockchain and discuss their technology challenges in our blockchain platform: (a) a new blockchain based general distributed and parallel computing paradigm component to devise and study parallel computing methodology for big data analytics, (b) blockchain application data management component for data integrity, big data integration, and integrating disparity of medical related data, (c) verifiable anonymous identity management component for identity privacy for both person and Internet of Things (IoT) devices and secure data access to make possible of the patient centric medicine, and (d) trust data sharing management component to enable a trust medical data ecosystem for collaborative research.

Information technology has irrevocably transformed businesses, education, healthcare, banking, government, and many other key sectors. It is impacting nearly every aspect of human activities in practically all fields redefining the way we live, communicate, socialize, collaborate, and transact. It also continues to leap ahead offering new capabilities and extending its utility. Adoption of IT in practice, however, presents several technical, organisational and societal challenges. This panel will present a snapshot status of IT in practice in the developed and emerging world, outline IT prospects and discuss challenges that IT sector, businesses and government must address to realize the fuller potential of IT. It will also provide directions for further research and development.

While early emphasis of Infrastructure as a Service (IaaS) clouds was on providing resource elasticity to end users, providers are increasingly interested in over-committing their resources to maximize the utilization and returns of their capital investments. In principle, over-committing resources hedges that users — on average — only need a small portion of their leased resources. When such hedge fails (i.e., resource demand far exceeds available physical capacity), providers must mitigate this provider-induced overload, typically by migrating virtual machines (VMs) to underutilized physical machines. Recent works on VM placement and migration assume the availability of target physical machines [1], [2]. However, in an over-committed cloud data center, this is not the case. VM migration can even trigger cascading overloads if performed haphazardly. In this paper, we design a new VM migration algorithm (called Scattered) that minimizes VM migrations in over-committed data centers. Compared to a traditional implementation, our algorithm can balance host utilization across all time epochs. Using real-world data traces from an enterprise cloud, we show that our migration algorithm reduces the risk of overload, minimizes the number of needed migrations, and has minimal impact on communication cost between VMs.

Despite the success High Performance Computing (HPC) across a number of application domains, the adoption of HPC resources and applications is still limited, primarily due to its high capital cost, system complexity, application availability, and service delivery model. Recently, several research efforts have shown that the emerging Cloud Com-puting service model can improve on-demand access to HPC capacity as utility. This paper introduces a framework for on-demand composing and deploying available HPC applica-tions as services on HPC clouds. The composition is enabled by an ontology that describes dependencies and relationships among HPC software and resources.

As IP telephony becomes a critical part of the enterprise network infrastructure, there is an increasing need for tools to help manage VoIP systems. In this paper, we introduce DA VINCI, a tool to manage VoIP call routing - the core of VoIP configurations that determines how calls are routed inside the enterprise or between the enterprise and external networks. We first establish the complexity involved in call routing design and configuration by studying characteristics of nine large enterprise VoIP networks. We then develop methodologies to improve and simplify call routing configurations by (i) identifying common design patterns, i.e., shared resource routing and local resource routing, so that administrators can better understand and reuse existing call routing designs, and (ii) removing unused and superfluous configurations from dial plan designs while retaining the correctness of operations. All of our techniques are incorporated in the DA VINCI tool that has been deployed in more than ten enterprise networks to help administrators streamline VoIP maintenance through extraction, analysis, simulation, and configuration of call routing.

In this work we study the actual requirements of a multimedia chat system for helpdesks by quantifying the verbal imagery communication in 115 real transcripts. The data shows that in chats which use verbal imagery, it corresponds to about 25% of the total time, most of it (75%) coming from the agent. These results clearly show that multimedia support for contact center chat systems is called for. But unlike in personal chat systems, the existing 3rdparty chat services for helpdesk centers are text-based only. To circumvent this issue, an innovative, practical SOA-based chat system architecture is proposed in this paper that creates multimedia chat services by decomposition and re-composition of the services from an existing 3rdparty text-based helpdesk chat product. Our SOA architecture implements uniquely a session-based interactive mash up mechanism and bundles it to the previous operations in the same chat session. However, to be effective in the context of technical support, a multimedia chat system must be provided with user and agent interfaces which address the distinctive aspects of technical support communication and imagery use in the conversations. We present an interface prototype that address those issues through the use of persistent imagery, linking instructions and imagery, use of common images,and easy screen sharing.

Multiparty voice-over-IP (MVolP) services allow a group of people to freely communicate with each other via the Internet, which have many important applications such as online gaming and teleconferencing. In this paper, we present a peer-to-peer MVolP system called peerTalk. Compared to traditional approaches such as server-based mixing, peerTalk achieves better scalability and failure resilience by dynamically distributing the stream processing workload among different peers. Particularly, peerTalk decouples the MVolP service delivery into two phases: mixing phase and distribution phase. The decoupled model allows us to explore the asymmetric property of MVolP services (for example, distinct speaking/listening activities and unequal inbound/outbound bandwidths) so that the system can better adapt to distinct stream mixing and distribution requirements. To overcome arbitrary peer departures/ failures, peerTalk provides lightweight backup schemes to achieve fast failure recovery. We have implemented a prototype of the peerTalk system and evaluated its performance using both a large-scale simulation testbed and a real Internet environment. Our initial implementation demonstrates the feasibility of our approach and shows promising results: peerTalk can outperform existing approaches such as P2P overlay multicast and coupled distributed processing for providing MVolP services.

VoIP applications require the ability to identify speakers in real time. This paper presents compressed speaker recognition (CSR), an innovative approach to perform speaker recognition directly from the compressed voice packets. CSR performs online speaker recognition from live packet streams of compressed voice packets by performing fast clustering over a defined subset of the features available in each compressed voice packet. Our experimental results show that CSR is highly scalable and accurate across a broad range of speakers

Multi-party voice-over-IP (MVoIP) services provide economical and natural group communication mechanisms for many emerging applications such as on-line gaming, distance collaboration, and tele-immersion. In this paper, we present a novel peer-to-peer (P2P) stream processing system called peerTalk to provide resource-efficient and failure-resilient MVoIP services. Different from previous work, our solution is fully distributed and self-organizing without requiring specialized servers or IP multicast support. Particularly, we decouple the stream processing in MVoIP services into two phases: (1) aggregation phase that mixes audio streams from active speakers into a single stream; and (2) distribution phase that distributes the mixed audio stream to all listeners. The decoupled model allows us to optimize and adapt the P2P stream mixing and distribution processes separately. Specifically, we can adaptively spread stream mixing workload among resource-constrained peer hosts according to current speaking activities. We have implemented a prototype of the peerTalk system and conducted experiments in real-world wide-area networks. The results show that peerTalk can achieve lower resource contention and better service quality than previous common solution.

Session initiation protocol (SIP) is being widely adopted for VoIP, IM and other collaborative applications due to its simple yet rich functional design. However, one of the main drawbacks has been its per-application deployment (each application using its own SIP stack), leading to narrowly focused development of SIP based services. In this paper, we propose a client-side SIP service and supporting network infrastructure blocks that provide unified mechanisms to execute generic SIP functions. The composition of these building blocks allows for creating richer applications, e.g. a conferencing server coupled with a gaming server provides dynamic conferencing between current occupants of a game room. The main feature of our framework is its availability to all applications including the ones not inherently based on SIP. Also, the SIP service API is designed to be extensible and in addition to providing novel higher level functional primitives like adhoc conferencing and seamless transition of sessions, it also exports a low level interface for specialized applications. Another feature of the service is that it allows a user to plug-in an end device of his/her choice on a per-session basis. We demonstrate the richness of the API by describing prototypes for enhancing various applications as well as new converged applications.

Ubiquitous IP telephony has become a feasible Internet service, and it is expected to meet the quality standards of traditional telephone services. The work presents a distributed voice-over-IP (VoIP) conferencing system called Venus that is implemented as a composable application-level service overlay network. Compared to the traditional centralized approach, Venus achieves better scalability and resource utilization by efficiently aggregating resources across distributed voice mixers. Moreover, Venus provides multi-constrained quality-of-service (QoS) provisioning by establishing each conferencing session based on multiple QoS constraints (e.g., delay, loss rate) and resource requirements (e.g., bandwidth, audio channels). Venus provides a failure resilient VoIP conferencing service by leveraging the fast failure recovery capability of the application-level service overlay network. Large-scale simulation results illustrate the efficiency of the Venus system.

We present a client/server-based un-intrusive collaborative multimedia application - called eMeeting - that extracts and archives salient visual information from seminars, while allowing interaction among presenters and the audience in real time. The application uses APIs of the Lotus Sametime/spl reg/ system such as secure login, directory, meeting and multimedia services, to which we have added our own extensions. These extensions use the binary messaging channel of Sametime and allow us to add stroke-based chat as well as stroke-based annotations of slides. This permits communication among all clients. In parallel with the services just mentioned and in the same Java application, users of eMeeting are presented with slides that the system extracts from the presenter's visual material using an automatic recognition algorithm that classifies video frames as slides. At this time we assume that the system may or may not have access to the visual content in electronic form. Users can follow the seminar in the form of presenter's video, and as a series of extracted images. The server transmits the slides cyclically to each user using IP multicast giving each client the opportunity to select any previous slide. Thus, late joiners can browse slides that have been accumulated by the system.

10.1109/ICME.2001.1237748

The widely popular World Wide Web along with advances in desktop computers has brought the world into a new age of computing and communications. Low bit-rate video applications across the Internet are quickly emerging. The ITU-T H.263 standard was designed for low bit-rate video conferencing across phone lines and is an ideal candidate to be extended for Internet video applications. This paper focuses on the error robustness issue of compressed H.263 video streams when transmitted over the Internet. The traditional approach of inserting intra-coded frames increases the error resilience at the expense of bursty output traffic, lower picture quality, and uneven frame dropping. By extending the macroblock force update feature of the H.263 standard, we developed a scheme that complies with the standard and increases the robustness of the video stream. This macroblock updating scheme analyzes the temporal dependencies of macroblocks in successive frames and selectively updates the macroblocks which have the most impact on later frames. The performance evaluation of the proposed technique demonstrates that it achieves a good balance between error recovery speeds and video quality.

The Internet explosion is driving the need for new collaboration tools which will enable two or more users to share data, audio, and video. The real-time packet-based solutions which are emerging differ considerably from the circuit-switch solutions which have existed for some time now. In this paper, we present one such packet-based approach, the Multimedia Multiparty Teleconferencing (MMT) system, which was fully implemented as a research prototype. Using MMT as an example, we address some of the fundamental issues related to videoconferencing systems in a packet-based environment, and discuss the differences with the traditional circuit-switch approaches, namely, the ITU H.320 standard. In particular, MMT is a distributed solution, while H.320 is centralized. The use of multicast and a novel video-mixing technique to facilitate the distributed solution are presented. Furthermore, MMT audio and video streams are susceptible to congestion and packet loss in the shared media packet-based environment, while H.320 streams use dedicated connections. As such, synchronization, error resilience, and dynamic rate control schemes for the packet-based system are presented.

This paper examines the issues involved in the design of conference servers that support multiparty, multimedia conferences. These servers, called multipoint control units (MCUs) in the telephony world, coordinate the distribution of audio, video, and data streams amongst the multiple participants in a videoconference. The MCU is responsible for the processing of video and audio so that a conference participant can hear and see one or more of the other participants in the conference. It is also responsible for handling and forwarding the data streams from the participants. This paper presents different approaches to the design of an MCU to implement these functions. It also describes the design of a related device-a transcoding gateway that enables conferencing between participants using different video/audio equipment.>