In recent years, with the rapid development of Internet technology, live streaming e-commerce has gradually become the focus of public attention and has occupied an increasingly important position in consumers' daily lives. The significant difference between online retail and offline retail is whether consumers can directly touch the products, and the perception of the products plays a key role in consumers' purchasing decisions. Therefore, how to display products effectively is crucial for online retailers. This study aims to explore how the product display method of live streaming e-commerce affects consumers' impulsive buying behaviour, and collect consumers' views on live streaming e-commerce through questionnaires. The results show that improving the vividness and interactivity of product displays significantly enhances virtual touch, which in turn promotes impulsive buying behaviour. However, there is a positive correlation between virtual touch and perceived risk, and perceived risk fails to significantly inhibit impulsive buying behaviour, but instead shows a positive correlation, that is, high perceived risk increases impulsive buying behaviour. The conclusions of this study provide new insights into the future interaction model between online retailers and consumers, promote two-way communication, and improve communication efficiency. Through this study, online retailers can provide consumers with an experience closer to offline shopping when using live streaming e-commerce, thereby more effectively prompting consumers to make purchasing decisions.

Distributed Real-Time Media Processing refers to classes of highly distributed, delay no-tolerant applications that account for the majority of the data traffic generated in the world today. Real-Time audio/video conferencing and live content streaming are of particular research interests as technology forecasts predict video traffic surpassing every other type of data traffic in the world in the near future. Live streaming refers to applications in which audio/video streams from a source need to be delivered to a set of geo-distributed destinations while maintaining low latency of stream delivery. Real-time conferencing platforms are application platforms that implement many-to-many audio/video real-time communications. Both of these categories exhibit high sensitivity to both network state (latency, jitter, packet loss, bit rate) as well as stream processing backend load profiles (latency and jitter introduced as Cloud processing of media packets). This thesis addresses enhancing real-time media processing both at the network level parameters as well as Cloud optimisations. We provide a novel, bandwidth management algorithm, for cloud services sharing the same network infrastructure, which provides a 2x improvement in system stability. Further examining network impact on cloud services, we provide a novel hybrid Cloud-Network distributed Cloud architecture to enable locality aware, application enhancements. This architecture led to a multi-cloud management overlay algorithm that maintains low management overhead on large scale cloud deployments. On the application level we provide a study of Media Quality parameters for a WebRTC enabled Media Cloud back-end, and provide patterns of quality metrics with respect to back-end stream load and network parameters. Additionally we empirically show that a "minimal load" algorithm for stream allocation, outperforms other Rotational, or Static Threshold based algorithms. / El procesamiento de medios en tiempo real distribuido se refiere a clases de aplicaciones altamente distribuidas,no tolerantes al retardo, que representan la mayoría del tráfico de datos generado en el mundo actual. Las conferenciasde audio y video en tiempo real y la transmisión de contenido en vivo tienen especial interés en investigación, ya quela prospectiva tecnológica estima que el tráfico de video supere a cualquier otro tipo de tráfico de datos en el futurocercano. La transmisión envivo se refiere a aplicaciones en las que flujos de audio/vídeo de una fuente se han de entregara un conjunto de destinos en lugares geográficos diferentes mientras se mantiene baja la latencia de entrega del flujo(como por ejemplo la cobertura de eventos en vivo). Las plataformas de conferencia en tiempo real son plataformasde  aplicación  que  implementan  comunicaciones  de  audio/video  en  tiempo  real  entre  muchos  participantes.  Ambascategorías presentan una alta sensibilidad tanto al estado de la red (latencia, jitter, pérdida de paquetes, velocidad debits) como a los perfiles de carga de la infraestructura de procesamiento de flujo (latencia y jitter introducidos duranteel procesamiento en la nube de paquetes de datos multimedia). Esta tesis trata de mejorar el procesamiento de datosmultimedia en tiempo real tanto en los parámetros de nivel de red como en las optimizaciones en la nube.En este contexto, investigamos si los recursos de la red se podían controlar a nivel de servicio para aumentar laeficiencia y el rendimiento de la red, así como cuantificar el impacto del recurso compartido de la red en la calidad delservicio. Los recursos de red compartidos afectan el rendimiento del servicio en la nube y, por lo tanto, optimizandoo intercambiando recursos de red pueden mejorar el rendimiento del servicio en la nube. Esta posible degradación delrendimiento se debe a la infraestructura de red compartida no regulada (la asignación de recursos de ancho de banda noes consciente de los objetivos del acuerdo de nivel de servicio (SLO) y de comportamiento). Gestionando el ancho debanda de la red a través de control predictivo, permitimos un mejor uso de los recursos de red disponibles y menoresviolaciones de SLO, logrando una mayor estabilidad del sistema por al menos un factor de 2.Las redes de acceso (AN) (extremo, red principal) de los ISP, transportistas y redes comunitarias no tienen unainfraestructura de nube de propósito general, mientras que los proveedores de recursos de Internet proporcionan bajodemanda recursos de la nube. Encontramos una oportunidad para la unificación de los recursos dentro de un AN y fueracon el fin de proporcionar una oferta de nube unificada a través de una federación de nubes y proporcionar movilidad delservicio hacia los usuarios para optimizar la localidad. Este trabajo de investigación proporciona una nueva arquitecturade red híbrida y nube federada que proporciona una infraestructura de red extendida con un despliegue en nube a granescala, incorporándolo directamente a la infraestructura de red. La nueva arquitectura multi-nube permite a los serviciosllegar a un compromiso entre localidad respecto al usuario o el rendimiento en tiempo de ejecución optimizando asípara latencia para conseguir la asignación óptima de recursos de aplicaciones en tiempo real. Para  optimizar  la  latencia  en  las  aplicaciones  de  transmisión  en  vivo  se  propuso  un  nuevo  algoritmo  desuperposición  de  multi-nube  autogestionado  basado  en  una  topología  de  gradiente  en  la  que  cada  nube  de  unaaplicación de transmisión de flujos optimiza la proximidad del cliente a la fuente. El modelo de aplicación se separa enun  diseño  de  dos  capas,  el  back-end  de  entrega  multi-nube  y  los  clientes  de  flujo.  El  backend  de  gradienteautorregulado minimiza la carga de tráfico creando un árbol de expansión mínimo a través de las nubes que se utilizapara  el  enrutamiento  de  cada  flujo.  El  algoritmo  propuesto  tiene  una  tasa  de  convergencia  muy  rápida  en  losdespliegues de nube a gran escala, y no resulta afectado por la rotación de recursos de la nube, así como proporcionauna mayor estabilidad de la transmisión en vivo.En este trabajo ofrecemos un análisis de calidad de los medios de comunicación y mejoras de los emisores deflujo en la nube en tiempo real, así como estrategias de asignación para mejorar el rendimiento de nivel de serviciode las plataformas de comunicación de Web en tiempo real. Los patrones de calidad de los medios están fuertementeinfluenciados por el rendimiento del procesamiento en la nube, y por lo tanto, al ajustar este aspecto, podemos controlarla calidad de los medios. En particular, demostramos empíricamente que a medida que los tamaños de sesión aumentan,la difusión simultánea supera la codificación de capa única. Además, introducimos un algoritmo de asignación de flujopara minimizar los picos de carga en los retransmisores de flujos en la nube y comparamos el comportamiento devarias políticas de asignación de flujos. Con la mínima información y el requisito de asignación de sesión de un únicoservidor, la política de asignación de carga mínima se comporta bastante mejor que otros algoritmos basados en unumbral rotativo o estático. / Distribuerad realtidshantering av mediadata syftar på klasser av starkt distribuerade tillämpningar som inte tolererar fördröjningar, och som utgör majoriteten av datatrafiken som genereras i världen idag. Audio/video-konferenser i realtid och överföring av innehåll "live" är av speciellt intresse för forskningen eftersom teknikprognoser förutser att videotrafiken kommer att kraftigt dominera över all annan datatrafik i den nära framtiden. "Live streaming" syftar på tillämpningar i vilka audio/video strömmar från en källa och behöver distribueras till en mängd av geografisk distribuerade destinationer medan överföringen bibehåller låg latens i leveransen av det strömmade datat (som ett exempel kan nämnas "live"-täckning av händelser). Konferensplattformar för realtidsdata är tillämpningsplattformar som implementerar realtidskommunikation av audio/video-data av typen "många-till-många". Båda dessa kategorier uppvisar hög känslighet för såväl nätverkets tillstånd (latens, jitter, paketförluster, bithastighet) och lastprofiler av ström bearbetning "back-end" (latens och jitter introducerat som Cloud-hantering av mediadatapaket). Denna avhandling adresserar förbättringar inom realtidshantering av mediainnehåll både med avseende på nätverksnivåns parametrar och optimeraringar för molninfrastrukturen. I detta sammanhang har vi undersökt huruvida nätverksresurserna kan kontrolleras på servicenivån i syfte att öka nätverkets effektivitet och prestanda, och även att kvantifiera påverkan av den delade nätverksresursen på servicekvaliteten. Delade nätverksresurser påverkar molntjänstens prestanda och dessa kan genom en optimering eller handel med nätverksresurser förbättra molntjänstens prestanda. Denna potentiella prestandadegradering beror på en oreglerad delad nätverksinfrastruktur (allokeringen av bandbredd är inte medveten om prestanda och mål för servicenivån). Genom att mediera nätverkets bandbredd genom prediktiv kontroll, möjliggör vi ett bättre utnyttjande av de tillgängliga nätverksresurserna och en lägre grad av avvikelser mot SLO, vilket leder till en ökad stabilitet med åtminstone en faktor 2. Accessnätverken (AN) (edge, kärnnätverk) hos ISP, bärare och lokala nätverk har ingen generell molninfrastruktur, medan s.k. "Internet Resource Providers" erbjuder resurser för molntjänster "on demand". Vi ser en möjlighet till ensande av resurserna inuti ett AN och utanför i syfte att erbjuda ett samlat molntjänsterbjudande genom s.k. "Cloud Federation" och erbjuder tjänstemobilitet för användarna för att optimera lokaliteten. Denna forskningansats erbjuder ett nytt hybrid nätverk med Federated Cloud arkitektur vilken ger en utvidgad nätverksinfrastruktur med en storskalig användning av molntjänster, som direkt inkorporerar denna i nätverksinfrastrukturen. Den nyskapande "Multi-Cloud"-arkitekturen möjliggör för tjänster att balansera lokalitet för användaren mot run-time-prestanda och därigenom optimera för latens mot optimal resursallokering för realtidstillämpningar. För att optimera latensen i "live streaming"-tillämpningar föreslås en nyskapande självstyrd "multi-Cloud-overlay"-algorithm baserad på gradienttopologi i vilken varje moln för en tillämpning inom "stream broadcasting" optimerar klientens närhet till källan. Tillämpningsmodellen separeras i en tvålagersdesign, "multi-cloud delivery back-end" och "stream clients". Denna självreglerande gradientbaserade "back-end" minimerar trafiklasten genom att skapa ett minimalt spännande träd genom molnen som används för routing av strömmarna. Den föreslagna algoritmen har en mycket snabb konvergenshastighet vid större moln, och påverkas inte av "churn" hos molnresursen liksom att den erbjuder ökad motståndskraft hos "live"-strömmen. I detta arbete erbjuder vi mediakvalitetsanalys och förstärkning av realtidsmolnets "forwarders", liksom även allokeringsstrategier för att förstärka servicenivåprestanda hos "Web Real-Time Communication"-plattformar. Mediakvalitetsmönster påverkas kraftigt av molnets bearbetsningsprestanda, och således kan vi genom att påverka denna aspekt kontrollera mediakvaliteten. Specifikt demonstrerar vi empiriskt att efterhand som sessionsstorlekarna ökar, så utklassar simulcast enlagersinkodning. Dessutom introducerar vi en strömallokeringsalgoritm för att minimera "load spikes" hos "Cloud stream forwarders" och jämför beteendet hos olika strömallokeringspolicys. Med enbart minimal information och allokeringsbehoven hos en enskild serversession beter sig den minimala lastbalansallokeringspolicyn tydligt bättre än andra "rotational"- eller "static threshold"-baserade algoritmer. / <p>QC 20170425</p>

According to the literature there are still very few researches focusing on the Online Live Instruction in e-Learning. By Integrating IP-Based Video Conferencing System and Full Screen Capture with Live Streaming Broadcast System into an LMS, instructors can provide Online Live Instruction. In this kind of environment, teachers and students can interact together like in conventional classroom. This study explored the process of Online Live Instruction, solutions for the encountered problems are also provided. SOP (Standardized Operation Procedures) are developed on how to conduct Course Material Lecturing and Group Discussion & Summary Report activities. Suggestions about Online Live Instruction for teachers, teaching assistants and students are also included.

With camera, smart phone, and wireless connection to a worldwide distribution source on a single device that fits in your pocket, now billions of citizens are able to become sousveillant micro-media activist – in real time. This case study investigates purposive texts in detail from over 50 hours of live and archived streaming video webcasts taken from geographically diverse sites. The goal is to explore how this tool is being used by videographers in a complex 21st century social movement. My sample video texts were gathered in late February and early March 2012 as the Occupy Movement stirred to life after a relatively quiet winter (from the corporate media’s point-of-view). In this project, I examine how Occupy’s use of live-streaming video combines “mediated co-presence” (Giddens 1984; Ito 2005) with “networked autonomy” (Castells 2011) to represent the ambivalent face of a complex, postmodern movement for social justice. / text

The proliferation of new devices (such as smartphones and tablets) promotes new multimedia services (e.g. user-generated live video broadcasting), as well as new streaming techniques (e.g. rate-adaptive streaming). As a matter of fact, scientists observe a formidable, sustainable growth of Internet traffic related to video streaming. Yet, network infrastructures struggle to cope with this growth and it is now frequent that a delivery network is insufficiently provisioned. Such underprovisioning problem is more severe for live videos due to its real-time requirement. In this thesis, we focus on bandwidth efficient video delivery solutions for live streaming in underprovisioned video delivery networks. Specifically, we have two main contributions: (1) a user-generated live videos sharing system based on peer-to-peer (P2P) technique, and (2) a live rate-adaptive streaming system based on Content Delivery Network (CDN). First of all, we built an multioverlay P2P video sharing system which allows Internet users to broadcast their own live videos. Typically, such a system consists of multiple P2P live video streaming systems, and faces the problem of finding a suitable allocation of peer upload bandwidth. We designed various bandwidth allocation algorithms for this problem and showed how optimal solutions can be efficiently computed. Then, we studied the problem of delivering live rate-adaptive streams in the CDN. We identified a discretize streaming model for multiple live videos in modern CDNs. We formulated a general optimization problem through Integer Linear Programming (ILP) and showed that it is NP-complete. Further, we presented a fast, easy to implement, and near-optimal algorithm with approved approximation ratios for a specific scenario. This work is the first step towards streaming multiple live rate-adaptive videos in CDN and provides a fundamental theoretical basis for deeper investigation. Last, we further extended the discretized streaming model into an user-centric one which maximizes the overall satisfaction of an user population. Further, we presented a practical system, which efficiently utilizes CDN infrastructure to deliver live video streams to viewers in dynamic and large-scale CDNs. The benefits of our approaches on reducing the CDN infrastructure capacity is validated through a set of realistic trace-driven large-scale simulations. All in one, this thesis explores bandwidth efficient live video delivery solutions in underprovisioned delivery network for multiple streaming technologies. The aim is to maximally utilize the bandwidth of relay nodes (peers in P2P and forwarding equipments in CDN) to achieve an optimization goal.

Submitted by Renata Lopes (renatasil82@gmail.com) on 2016-05-17T12:01:58Z No. of bitstreams: 1 wagnerdealmeidajunior.pdf: 2594589 bytes, checksum: 9b2a2375eb8725cfda2a8aabadd86f22 (MD5) / Approved for entry into archive by Adriana Oliveira (adriana.oliveira@ufjf.edu.br) on 2016-06-28T14:01:42Z (GMT) No. of bitstreams: 1 wagnerdealmeidajunior.pdf: 2594589 bytes, checksum: 9b2a2375eb8725cfda2a8aabadd86f22 (MD5) / Made available in DSpace on 2016-06-28T14:01:42Z (GMT). No. of bitstreams: 1 wagnerdealmeidajunior.pdf: 2594589 bytes, checksum: 9b2a2375eb8725cfda2a8aabadd86f22 (MD5) Previous issue date: 2015-09-04 / A transmissão de conteúdo de mídia pela Internet é uma tendência que vem crescendo atualmente. Especialmente, o cenário de transmissão de vídeo sob demanda (VoD) já é bastante consolidado. Um novo desafio que surge é a transmissão de vídeo ao vivo, especialmente de conteúdo capaz de atrair um grande número de espectadores como eventos de grande escala. Neste trabalho nós caracterizamos a transmissão ao vivo pela Internet de dois eventos esportivos de grande escala. Esses eventos foram transmitidos gratuitamente por um dos maiores provedores de conteúdo da América Latina através de um portal web acessado pelo protocolo HTTP. Nos baseamos em registros de acesso dos servidores web do provedor para extrair as informações usadas nas caracterizações. Nós quantificamos o impacto da transmissão de um evento de grande escala no provedor de conteúdo e a qualidade de serviço recebida pelos usuários finais. Mostramos que os servidores de vídeo recebem carga substancial e que a carga varia significativamente entre jogos, o que dificulta o provisionamento de capacidade. Além disso apresentamos um mapa da difusão de acesso por região do Brasil com base em observações sobre a mídia recebida pelos clientes. / Streaming media on internet is currently a growing trend. In particular, video-on-demand (VoD) scenario is already quite consolidated. Although, live streaming still facing challenges, specially during large-scale events, which may attract a large number of viewers. In this work, we characterize two large-scale Internet live streaming sport events. These events were streamed free of charge through a web portal accessed via HTTP protocol by one of the largest content provider in Latin America. Our characterization relies on access logs we have obtained from the content provider’s web server. We quantify the impact of a large-scale event transmission on content provider servers. We also characterize the quality of service end users receive. Our results show that video servers receive substantial network load during a large-scale event. Moreover, this network load can vary significantly among distinct matches, which difficult capacity provisioning. In addition we present a map of the Brazilian Internet access distribution based on observations about the media content received by customers.

Peer to peer networks are becoming increasingly popular among Internet users as the downloading peers share the storage and upload bandwidth load of the system. This makes it possible for a large number of users to share a data file available at a server without the server upload bandwidth becoming a bottleneck. The P2P technology is being widely used not only for file sharing but also for video on demand, live streaming and IPTV. The delay deadlines are more stringent in live streaming and IPTV than those in file sharing as the traffic is real time. The performance perceived by a user depends upon whether the video stream is being downloaded at the streaming rate. Coolstreaming is the first large scale P2P IPTV system. We model the multi-channel Coolstreaming system via an open queueing network. The peer dynamics at a channel is modelled by a closed queueing network working at a faster rate. We compute the expected number of substreams in the overlay of New Coolstreaming which are not being received at the proper rate. The computation of the Markov chain with a very large state space is handled using the two time scale decomposition. Further we characterize the end to end delay encountered by a video stream originating from the server and received at a user of New Coolstreaming. Three factors contribute towards the delay. The first factor is the mean path length in terms of overlay hops of the partnership graph. The second factor is the mean number of routers between any two overlay peers in the network layer and the third factor is the queueing delay at a router in the Internet. The mean shortest path length in terms of overlay peers in the New Coolstreaming graph is shown to be O(logn)where nis the number of peers in the overlay. This is done by modelling the overlay by a random graph. The mean shortest path in terms of routers in the Internet’s router level topology is seen to be at most O(logNI)where NIis the number of routers in the Internet. We also discuss a method by which we can get the mean delay at a router in the Internet. Thus, the mean end to end delay in New Coolstreaming is shown to be upper bounded by O(lognlogNIE[W])where E[W]is the mean delay at a router in the Internet.

Användningen av strömmande video ökar snabbt just nu. Ett populärt konceptär adaptive bitrate streaming som går ut på att en video kodas i flera olikabithastigheter. Dessa videor tas sedan och delas upp i små filer och görstillgänglig via internet. När du vill spela upp en sådan video laddar du först hemen fil som beskriver vart filerna finns och i vilka bithastigheter de är kodade i.Mediaspelaren kan där efter börja ladda hem filerna och spela upp dom. Om defysiska förutsättningarna, som exempelvis nedladdningshastighet eller CPUbelastning,ändras under uppspelningen kan mediaspelaren enkelt byta kvalitépå videon genom att börja ladda filer av en annan bithastighet och slippa attvideon laggar. Denna rapport tar därför en närmare titt på fyra tekniker inomadaptive bitrate streaming. De som undersöks är HTTP Live Streaming,Dynamic Adaptive Streaming over HTTP, HTTP Dynamic Streaming ochSmooth Streaming med avseende på vilka protokoll som dom använder.Rapporten undersöker även hur Apple och FFmpeg har implementerat HTTPLive streaming med avseende på hur mycket data som behövs läsas i en filinnan videon kan börja spelas upp. Rapporten visar att det inte är så storaskillnader mellan de fyra teknikerna. Dock sticker Dynamic AdaptiveStreaming over HTTP ut lite genom att vara helt oberoende av vilket ljud ellervideoprotokoll som används. Rapporten visar också på en brist i specificeringenav HTTP Live Streaming då det inte är specificerat att första komplettabildrutan i videoströmmen bör ligga i början av filen. I Apples implementationbehövs upp till 30 kB data läsas innan uppspelning kan påbörjas medan iFFmpegs implementation är det ca 600 byte. / The use of streaming video is growing rapidly at the moment. A popular conceptis adaptive bitrate streaming, which is when a video gets encoded in severaldifferent bit rates. These videos are then split into small files and made availablevia the internet. When you want to play such a video, you first download afile that describes where the files are located and in what bitrates they are encodedin. The media player then begin downloading the files and play them. Ifthe physical conditions, such as the download speed or CPU load, changes duringplayback, the media player can easily change the quality of the video bystarting to downloading files of a different bit rate and avoid that the video lags.This report will take a closer look at four techniques in adaptive bitrate streaming.They examined techniques are HTTP Live Streaming, Dynamic AdaptiveStreaming over HTTP, HTTP Dynamic Streaming and Smooth Streaming andwhich protocols they use. The report also examines how Apple and FFmpeg hasimplemented HTTP Live Streaming with respect to how much data is needed toread a file before the video can begin to be played. The report shows that thereare no large differences between the four techniques. However, Dynamic AdaptiveStreaming over HTTP stood out a bit by being completely independent ofany audio or video protocols. The report also shows a shortcoming in the specificationof HTTP Live Streaming as it is not specified that the first completeframe of the video stream should be at the beginning of the file. In Apple's implementationits needed to read up to 30 KB of data before playback can bestarted while in FFmpeg's implementation its about 600 bytes.

隨著網路科技的發展，許多新興媒體隨之興起，直播服務即為其中之一，直播為現今品牌或者商家躍躍欲試的行銷溝通管道。直播服務有許多的應用方式，而當中「直播＋電商」為本研究所欲了解的應用，因直播電商近來廣受業界討論，也可看見許多品牌皆直播與消費者溝通以及導購，且許多品牌也都表示有良好的成效。直播電商即是透過直播服務在網路上進行商品的販售，其中主要組成的元素包括直播功能、直播主、直播內容以及閱聽眾，本研究希望能夠了解直播電商是如何能夠提升消費者購買意願以及對購買決策流程影響。 本研究將受訪者分為三組（前測、直播以及重播組），先讓受訪者共同觀看完直播電商節目後，再透過焦點團體訪談法深入瞭解其態度，研究結果歸納如下： 一、 購買意願 內容與直播主為直播電商關鍵成功因素，實用性價值高之內容、具專業性之直播主較能促使購買意願，而具知名度之直播主能吸引消費者注意以及提升購買意願，且當產品與直播主具有高度關聯性時較能說服消費者；直播功能對於消費者而言為附加價值，但是當消費者對於直播內容與直播主感到滿足、滿意時，直播功能能有效提升購買意願。 二、 購買決策流程 直播透過導購功能、互動功能能夠縮短消費者購買決策流程的資訊搜集以及評估方案階段。 根據研究結果，本研究提出實務建議以提供企業作為未來營運參考，亦提出研究建議作為未來研究參考。 / As the Internet Technologies continue to bloom and develop, many new media emerge. Of all, live streaming, has become the must try for brands and sellers nowadays. Among all the applications that live streaming brings, live streaming commerce, which is selling products through live streaming, is the focus of this study. The main components of live streaming commerce include live features, host, content and audiences. This study aims to understand the impact of live streaming commerce on consumers’ purchase intention and purchase decision-making process. To conduct focus group interviews, the study divided participants into three groups: 3 people in the pre-test, 5 people watching livestream, and 5 people watching video (same content as in the previous group but not live). Results and conclusions are summarized as below: The key successful factors of live streaming commerce are content and host. The content with high utilitarian value and the expert hosts are both significant to consumers’ purchase intention. Live streaming with popular hosts could capture consumers’ attention and enhance their purchase intention. Moreover, the live streaming hosts are more likely to persuade consumers into purchasing if the characteristics of live streaming hosts are strongly relevant to the products. Consumers regard functions of live streaming as additional value. Live streaming could effectively increase consumer purchase intention when the consumers are satisfied with the content and the performance of host in the live streaming. Live streaming commerce shortens the consumer decision making process through features of real- time text chatting and shopping guide. According the results, this study presents the practical recommendations to provide enterprises as a future operational reference, and propose research recommendations as a future study reference.

本文分析近年盛行的電子遊戲實況活動。從分心、不專注於電玩的現象開始，本文以「如何玩」這個提問來引導論證「嬉玩」與「遊戲」這組概念在分析上的效力，藉此，本文論證了電玩實況活動中的心智狀態與活動是游移在電玩活動與嬉鬧互動，在嚴肅遊戲與嬉戲玩耍之間游移，這樣的分神狀態之中也產生了有樂趣、具有實效的遊戲性。 從技術架構的觀點來看，以即時串流的技術為基礎，電玩實況活動結合了即時傳輸與即時互動。使用者所獲致的遊戲聆賞經驗以及互動是即時且與他人同步，便也構成了網路媒介之下的共時經驗，實況中以電玩活動為對象的起鬨也便具共在的實感。本文從聊天室起鬨、技術戲耍挪用等現象中論證了基於遊戲性旨趣的共謀嬉玩。嘲諷、干擾的對抗性是為了製造分神的條件，從中產生張力並獲致遊戲性，而包括實況主在內的所有參與者，皆是在這樣的分神狀態中嬉玩，維繫歡騰的共感情緒。 電子遊戲具有將遊戲性限定在其中的封閉性質，也因此對於電玩的遊戲經驗多有私密、個人的想像，即時串流的電玩實況活動便是打開了這樣的封閉性質。透過網路媒介將人匯聚在遊戲活動周圍，不僅共享了電玩活動本身的樂趣，也在即時串流之下共時的互動跟聆賞經驗中以遊戲為對象群聚嬉戲。這樣的網路活動之所以盛行，顯示了人對遊戲性的需求不僅是來自遊戲中，也來自與他人玩在一起的連結共感，電玩實況便是實現了這樣的需求。 / Taking the prevailing "video game live streaming" activities as research object, this thesis takes "play" and "game" as a set of core concepts, and begins the exploration into their conceptual effect. Setting out the arguments with "How to Play" question, this thesis concludes that the mentality and activities in video game live streaming are in an "in-between" condition. Between video game activity and playful disturbance, seriousness and playfulness, there are gameplay with actual effect. Taking the technical conditions into consideration, video game live streaming consists of live media transportation and live interaction. Users' experiences synchronize with each other, thus contribute to the "togetherness" mediated through the Internet, and also the carnivalistic interactions and disturbance in the live channel. From the fuss and disturbance in the chatroom to the appropriation of streaming "plugins", this thesis argues that these actions conducted by streamer and users contribute to the "distractive" scenario, and the conducts are all based on "the interest of play". Different from serious and immersive mentality which is deemed as the fundamental of playing and gaming, the gameplay in live stream is based on distraction, which is the core condition that brings tension and thus playfulness. All the users in video game live streaming, the streamer included, are connected and forged into a synchronized togetherness by this kind of distractive, carnivalistic playfulness. Digital video games have tributes of enclosure, which require concentration, seriousness and absorption. Video game live streaming opens up this kind of enclosure, and brings people in and around the video game activities. In the live channel, users experience the joy and fun of the game itself together in a synchronized way, and also act on the interest of play, creating distraction and disturbance that bring to playful experiences, and that explains why this kind of Internet activity prevails. This thesis shows that the need of gameplay comes not only from game activities themselves, but also from the desire to "play together". In a digital age, video game live streaming is just the fulfillment of this kind of desire.