Video-on-demand streaming從90年代就在適當的科技下成為網際網路流量的龍頭，故宮教育頻道在技術上採用TCP-based video streaming，但Video streaming使得負載較一般的伺服器繁重。透過負載平衡來分散負載是可行的方式之一，但在TCP-based video streaming的技術下，還是可能遭遇服務品質之相關議題。而服務品質要求 (Quality of Service, QoS)是用來對使用者適配的服務品質表現，做具體之等級劃分的管理工具，本研究以故宮教育頻道 (iPalace Video Channel; iPalace)為標的，針對其DNS Load Balancing搭配Round Robin的運作背景，提出VM Side QoS及Client Side QoS，在VM Side設定Load rate作為指標、在Client Side則以撥放器的表現為主，設定Initial Buffering Time、Mean Rebuffering Duration以及Rebuffering Frequency共三個指標，並搭配流量的實驗操作，劃分Regular及Class情境來蒐集資料。 在分析後發現，在環境較為一致的Class情境中，Mean Rebuffering Duration與Load Rate及Rebuffering Frequency與Load Rate具有低度正相關。未來，若能在環境及客戶端的變異性有較多的控制及掌握，則可讓資料在蒐集上更為正確，讓後續的分析及探討與實際狀況更為相符。

近十年來，由於無線網路的普及與人們對於影像串流服務的需求愈高，導致人們迫切需要更好的服務品質。但在IEEE 802.11無線網路中，本身的設計並非針對影像串流來設計，為讓影像串流能在無線網路更有效率，我們必須重新設計適合影像串流的無線網路。本研究首先探討IEEE 802.11 MAC層DCF (Distributed Coordination Function)與802.11e EDCA (Enhanced Distributed Channel Access)機制的潛在問題。由於DCF與EDCA並未特別對影像串流做設計，會導致具有時延(delay time)限制的影像封包等待過久造成失效，卻仍繼續傳送。本研究提出幾個有效方法，改善原本IEEE 802.11無線網路對影像串流傳輸效能不彰的現象。我們將利用跨層設計使MAC層能取得影像串流封包資訊，並改善DCF與EDCA的重傳(retransmission)機制，使用time limit與retry limit混和設計節省不必要的等待時間，並使用single-video multi-level queue改善傳輸效能。最後本研究將利用網路模擬器NS-2 (Network Simulater ver. 2)與影像串流測試實驗架構myEvalvid-NT作不同效能的驗證比較並評估我們所提出方法的有效性。 / Over the past decade, wireless network access and video streaming services have become more popular than ever. People are eager to have better quality of video streaming services over wireless network. However, IEEE 802.11 DCF and IEEE 802.11e EDCA are not specifically designed for video streaming. This leads to the problem of transmitting overdue video packets and thus degrades both the network performance and video quality. In this paper, we propose a hybrid design framework to improve the quality of video streaming. This framework consists of a MAC-centric cross-layer architecture to allow MAC-layer to retrieve video streaming packet information (slice type and transmission deadline), a retransmission mechanism of hybrid retransmission deadline and retry limit to save unnecessary packet waiting time, and a single-video multi-level queue to prioritize I/P/B slice delivery. Simulations show that the proposed methodology outperforms IEEE 802.11e, IEEE 802.11e Timebase and IEEE 802.11e MultiQ in packet loss rate, invalid packet ratio, lost and invalid packet ratio, delay time, jitter, and PSNR.

Actuellement, nous constatons une augmentation de demande de services vidéo sur les réseaux P2P. Ces réseaux jouent un rôle primordial dans la transmission de contenus multimédia à grande échelle pour des clients hétérogènes. Cependant, le déploiement de services vidéo temps réel sur les réseaux P2P a suscité un grand nombre de défis dû à l’hétérogénéité des terminaux et des réseaux d’accès, aux caractéristiques dynamiques des pairs, et aux autres problèmes hérités des protocoles TCP/IP, à savoir les pertes de paquets, les délais de transfert et la variation de la bande passante de bout-en-bout. Dans ce contexte, l’objectif de cette thèse est d’analyser les différents problèmes et de proposer un mécanisme de transport vidéo temps réel avec le provisionnement de la qualité de Service (QoS). Ainsi, nous proposons trois contributions majeures. Premièrement, il s’agit d’un mécanisme de streaming vidéo adaptatif permettant de sélectionner les meilleurs pair émetteurs. Ce mécanisme permet de structurer les pairs dans des réseaux overlay hybrides avec une prise en charge des caractéristiques sémantiques du contenu et des caractéristiques physiques du lien de transport. Nous présentons ensuite un mécanisme d’ordonnancement de paquets vidéo combiné à une commutation de pairs et/ou de flux pour assurer un transport lisse. Finalement, une architecture de collaboration entre les applications P2P et les fournisseurs de services / réseaux est proposée pour supporter un contrôle d’admission de flux. / There is an increasing demand for scalable deployment of real-time multimedia streaming applications over Internet. In this context, Peer-to-Peer (P2P) networks are playing an important role for supporting robust and large-scale transmission of multimedia content to heterogeneous clients. However, the deployment of real-time video streaming applications over P2P networks arises lot of challenges due to heterogeneity of terminals and access networks, dynamicity of peers, and other problems inherited from IP network. Real-time streaming applications are very sensitive to packet loss, jitter / transmission delay, and available end-to-end bandwidth. These elements have key importance in QoS provisioning and need extra consideration for smooth delivery of video streaming applications over P2P networks. Beside the abovementioned issues, P2P applications lack of awareness in constructing their overlay topologies and do not have any explicit interaction with service and network providers. This situation leads to inefficient utilization of network resources and may cause potential violation of peering agreements between providers. The aim of this thesis is to analyze these issues and to propose an adaptive real-time transport mechanism for QoS provisioning of Scalable Video Coding (SVC) applications over P2P networks. Our contributions in this dissertation are threefold. First, we propose a hybrid overlay organization mechanism allowing intelligent organization of sender peers based on network-awareness, media- awareness, and quality-awareness. This overlay organization is further used for an appropriate selection of best sender peers, and the efficient switching of peers to ensure a smooth video delivery when any of the sender peers is no more reliable. Second, we propose a packet video scheduling mechanism to assign different parts of the video content to specific peers. Third, we present a service provider driven P2P network framework that enables effective interaction between service / network providers and P2P applications to perform QoS provisioning mechanism for the video streaming.

Made available in DSpace on 2016-04-04T18:31:20Z (GMT). No. of bitstreams: 1 Edizon Eduardo Basseto Junior.pdf: 3122282 bytes, checksum: f4c584d20c71eb7d623b23f592a8a4c2 (MD5) Previous issue date: 2008-02-08 / The goal of this work is to propose a metric to analyze the quality of videos transmitted through wireless networks. A wireless network is a degenerative means to transmit data given its huge instability, which affects transmitted videos. The present proposal adds a technical network parameter, known as jitter, to clients opinion. For these studies a video was streamed through a RF emulation testbed, which can emulate the IEEE wireless networks (WLAN) 802.11b and 802.11g standards. The experiments were done by streaming a DVD, with known characteristics, from a server to a workstation, passing through the testbed. This is capable to concentrate, in a controlled way, the wireless signal into a coaxial cable. With the possibility to control the signal level of the carrier, twelve scenarios were developed and studied: video transmissions with maximum and minimum signal levels, and also with the signal level characterized as Rayleigh distribution, were done for each of the two IEEE standards measures. Under the same signal conditions, I analyzed the video streaming with UDP traffic sent together. The results confirm the action of jitter in the degeneration of the received video, which justifies the use of jitter in the proposed metric. The results also show that the 802.11g networks work a little better than 802.11b for video streaming, but none of them showed a good performance when the UDP traffic was sent together. A questionnaire has also been developed, with the purpose of analyzing the streaming subjectively. Those two factors (jitter and subjective analysis) result in an equation that can be applied to a MOS scale making it possible to numerically define the quality of the video transmission. / O objetivo deste trabalho ? propor uma m?trica de qualidade para v?deos transmitidos em redes sem fio. As redes sem fio s?o um meio de transmiss?o muito degenerativo devido ? sua instabilidade, o que afeta um v?deo transmitido. A presente proposta soma ? opini?o dos espectadores um par?metro t?cnico da rede, conhecido por jitter, e a taxa de dados do v?deo transmitido. Para os estudos realizados, um v?deo foi transmitido atrav?s de uma bancada de emula??o de r?dio freq??ncia (RF) que simula a transmiss?o das redes sem fios (WLAN) padr?es IEEE 802.11b e 802.11g. Os experimentos foram realizados a partir do streaming de um DVD, com caracter?sticas conhecidas, de um servidor para uma esta??o, atrav?s da bancada que confina o sinal transmitido de maneira controlada em um cabo coaxial. Tendo-se essa possibilidade de controlar o n?vel do sinal da portadora, foram estudadas doze situa??es. Para cada um dos padr?es IEEE foram realizadas medidas com a transmiss?o de v?deo e um n?vel m?ximo de sinal, outra com o n?vel m?nimo e uma situa??o que tem o n?vel de sinal caracterizado por uma distribui??o de Rayleigh. Para essas mesmas condi??es de sinal foi estudada a transmiss?o do v?deo mais um tr?fego UDP concorrente. Os resultados confirmam a atua??o do jitter na degenera??o do v?deo recebido, justificando sua utiliza??o na m?trica proposta. Os resultados demonstram tamb?m que existe uma pequena vantagem das redes 802.11g na transmiss?o do streaming, por?m nenhuma das redes demonstrou um desempenho satisfat?rio na presen?a de tr?fego UDP transmitido concorrentemente. Na proposta tamb?m ? desenvolvido um question?rio com a finalidade de analisar subjetivamente o streaming, gerando uma equa??o que pode ser aplicada ? escala MOS, podendo assim definir numericamente a qualidade de v?deo transmitido.

Le trafic vidéo s’est considérablement accru et est prévu de doubler pour représenter 82% du trafic Internet d’ici 2021. Une telle croissance surcharge les fournisseurs de services Internet (ISP), nuisant à la Qualité d’Expérience (QoE) perçue par les utilisateurs. Cette thèse vise à améliorer la QoE des utilisateurs de streaming vidéo sans hypothèse de changement d’infrastructure physique des opérateurs. Pour cela, nous combinons les technologies de caching réseau, de streaming HTTP adaptatif (HAS), et de transport multipath. Nous explorons d’abord l’interaction entre HAS et caching, pour montrer que les algorithmes d’adaptation de qualité vidéo ont besoin de savoir qu’il y a un cache et ce qui y est stocké, et proposons des algorithmes bénéficiant de cette connaissance. Concluant sur la difficulté d’obtenir la connaissance de l’état du cache, nous étudions ensuite un système de distribution vidéo à large échelle, où les caches sont représentés par un réseau de distribution du contenu (CDN). Un CDN déploie des caches à l’intérieur des réseaux des ISP, et dispose de ses propres serveurs externes. L’originalité du problème vient de l’hypothèse que nous faisons que l’utilisateur est simultanément connecté à 2 ISP. Ceci lui permet d’accéder en multipath aux serveurs externes aux ISP (pouvant ainsi accroître le débit mais chargeant plus les ISP), ou streamer le contenu depuis un cache plus proche mais avec un seul chemin. Ce désaccord entre les objectifs du CDN et de l’ISP conduit à des performances sous-optimales. Nous développons un schéma de collaboration entre ISP et CDN qui permet de nous rapprocher de l’optimal dans certains cas, et discutons l’implémentation pratique. / Video traffic volume grew considerably in recent years and is forecasted to reach 82% of the total Internet traffic by 2021, doubling its net volume as compared to today. Such growth overloads Internet Service Providers' networks (ISPs), which negatively impacts users' Quality of Experience (QoE). This thesis attempts to tackle the problem of improving users' video QoE without relying on network upgrades. For this, we have chosen to combine such technologies as in-network caching, HTTP Adaptive Streaming (HAS), and multipath data transport. We start with exploration of interaction between HAS and caching; we confirm the need of cache-awareness in quality adaptation algorithms and propose such an extension to a state-of-the-art optimisation-based algorithm. Concluding on the difficulty of achieving cache-awareness, we take a step back to study a video delivery system on a large scale, where in-network caches are represented by Content Delivery Networks (CDNs). They deploy caches inside ISPs and dispose of their own outside video servers. As a novelty, we consider users to have a simultaneous connectivity to several ISP networks. This allows video clients either to access outside multipath servers with aggregate bandwidth (which may increase their QoE, but will also bring more traffic into ISP), or stream their content from a closer cache through only single connectivity (bringing less traffic into ISP). This disagreement in ISP and CDN objectives leads to suboptimal system performance. In response to this, we develop a collaboration scheme between two actors, performance of which can approach optimal boundary for certain settings, and discuss its practical implementation.

Au cours de cette thèse, nous nous sommes d'abord intéressés à l'optimisation des ressources et à la modulation adaptative dans les systèmes SC-FDMA (Single Carrier Frequency Division Multiple Access). Ce problème d'optimisation est combinatoire à complexité de calcul exponentielle. Afin de pallier à cette difficulté, nous avons utilisé la théorie de la dualité canonique, grâce à laquelle, la complexité du problème d'optimisation devient polynômiale et cela en constitue une amélioration remarquable. L'approche proposée est très proche de la solution optimale. Nous avons ensuite étudié la problématique complexe de l'allocation de ressources pour le "Streaming Vidéo" dans les réseaux sans fil, où il est nécessaire d'assurer une transmission vidéo de haute qualité en présence de canaux et de brouillages variables au cours du temps. Dans ce contexte, nous avons proposé une nouvelle méthode d'allocation de puissance conjointement à l'adaptation du débit vidéo. Pour ce faire, nous avons adopté une approche de la théorie de contrôle, intitulée "Risk-Sensitive Control". Nous avons dédié la troisième partie de la thèse à la conception d'une nouvelle stratégie "best-M" pour le renvoi du CQI (Channel Quality Indicator) pour les systèmes multi-utilisateurs et multi-porteuses. En générale, l'erreur d'estimation du CQI ainsi que son délai de renvoi sont gérés au niveau de la station de base. Notre nouvelle stratégie "best-M" suppose que la gestion de ces problèmes est confiée aux utilisateurs. De ce fait, la performance du système se trouve améliorée sans que son débit de signalisation ne soit augmenté en voix montante. / Adaptive resource allocation in wireless communication systems is crucial in order to support the diverse QoS needs of the services and optimize resource utilization. The design of resource allocation schemes should consider the service type for which it is intended. Moreover, due to feedback delay and channel estimation error, the Channel Quality Indicator (CQI) reported to the transmitter may not be a perfect measure of the channel quality and its use for resource allocation may severely degrade the systems performance. In this thesis, we study resource allocation and CQI reporting for wireless networks while taking the aforementioned factors into consideration. First, we consider resource allocation and adaptive modulation in uplink SC-FDMA systems. This is a combinatorial problem whose optimal solution is exponentially complex. We use canonical duality theory to derive a polynomial complexity resource allocation algorithm that provides a nearly optimal solution to the problem. Then, we focus on resource allocation for video streaming in wireless networks with time-varying interference. To this end, by using risk-sensitive control approach, we develop a cross-layer optimization framework that performs power control at the PHY/MAC layer and rate adaptation at the APPLICATION layer jointly and provides fairness among nodes. Finally, by using stochastic control and game theory, we design a robust best-M CQI reporting scheme for multi-carrier and multi-user systems which takes into account the impact of feedback delay and error in CQI computation. Performing resource allocation on the basis of the proposed CQI reporting can significantly improve the system performance.

使用無線網路時常會遇到傳送訊號衰減、通訊死角或氣候不佳等原因，造成傳輸失敗。高畫質影像H.264/MPEG-4 編碼考慮影音封包具有不同重要性，對影音封包須制定不同的重傳機制。本研究改善WiMAX2中HARQ（Auto Repeat Request）的流程，將高畫質影像藉由TCP-Like方式傳輸影像，辨識影音封包資訊，確保GOP-Tree中＂骨幹＂節點的抵達率並利用Lagrange Method of Multiplier方法，對GOP-Tree中非骨幹節點提出合適的Trade-off。IEEE 802.16m標準支援Mobile Multi-hop Relay （MMR）的架構，針對目前基地台覆蓋範圍不足及遮蔽效應等問題提出中繼傳輸站(Relay Station，RS）來提升網路吞吐量並增進無線通訊品質。但也因此需要有較多的連線頻道及頻寬。然而此方式不能保證所有頻道都處於良好傳輸狀態，此而常有相當高的資料傳送失敗率。 混和自動重傳要求（Hybrid Automatic Repeat reQuest, HARQ）的提出就是用來確保資料傳遞成功的機制。HARQ早在IEEE 802.16-2004就被提出，然而我們發現HARQ機制在IEEE 802.16m效能不佳，更無法保證QoS (Quality of Service）的服務品質要求。在本研究中我們改善Chen [4]所提出的Dynamic Pre-allocation HARQ（DP-HARQ）機制，考慮當slot分配不足，封包在發生非連續錯誤時造成延遲時間增加等問題。我們所提出的方法加入具服務品質感知（QoS aware）的機制，來降低整體延遲時間（delay）並能有效提高吞吐量（throughput）。在模擬實驗中，我們以具二個hop，二個BS及十個SS以上的模擬環境，對不同方法做不同效能的評比，並以PSNR值檢驗所提出之方法是否能提升畫質，以確保影像傳輸品質。我們透過實驗數據證明提出的方法優於IEEE 802.16m的HARQ機制。 / This research focus on running high definition video over WiMAX2.It is usually that transmission fails through wireless network as it encounters factors like (transmission) signal attenuation, dead zone, bad weather, etc. The high definition video coding in H.264/MPEG-4 involves different degress of important video frames considering the video package. It defines different re-transmission mechanisms for video package. This research starts with the enhancement on the process of HARQ （Hybrid Auto Repeat Request）over WiMAX2, transmitting high definition video signals by“TCP-Like”protocol and indentifying the video package information to ensure a stable arrival rate of the “backbone” nodes in GOP-Tree. Furthermore, we apply Lagrange’s Method of Multiplier and propose an appropriate customization to the non-backbone part in GOP-Tree. Though the IEEE 802.16 standards has the concept of Mobile Multi-hop Relay（MMR）, yet there are problems to be resolved and improved when it comes to reality due to the facts like coverage deficiency of base stations and shadow effect. The IEEE 802.16m standard uses Relay Station (RS）to enhance network traffic throughput and improve QoS over wireless network.Relay transmission stations are set for data transmission in Mobile Multi-hop Relay (MMR),thus it requires more connection channels and bandwidth. The thesis concludes with performance comparison under the simulation environments of 2 hops, 2 BSs and 10 SSs, evaluation of the methodology with PSNR for enhancing the QoS of HD video transmission.The proposed method is proved to surpass the HARQ methodology in IEEE 802.16m.

Video streaming is widely recognized as the next Internet killer application. It was not one of the Internet's original target applications and its protocols (TCP in particular) were tuned mainly for e efficient bulk file transfer. As a result, a significant effort has focused on the development of UDP-based special protocols for streaming multimedia on the Internet. Recently, there has been a shift in video streaming from UDP to TCP, and specifically to HTTP. HTTP streaming provides a very attractive platform for video distribution on the Internet mainly because it can utilize all the current Internet infrastructure. In this thesis we make the argument that the marriage between HTTP streaming and the current Internet infrastructure can create many problems and inefficiencies. In order to solve these issues, we provide a set of techniques and protocols that can help both the network and end-hosts to make better decisions to improve video streaming quality. The thesis makes the following contributions: - We conduct a characterization study of popular commercial streaming services on mobile platforms. Our study shows that streaming services make different design decisions when implementing video players on different mobile platforms. We show that this can lead to several inefficiencies and undesirable behaviors specially when several clients compete for bandwidth in a shared bottleneck link. - Fairness between traffic flows has been preserved on the Internet through the use of TCP. However, due to the dynamics of adaptive video players and the lack of standard client adaptation techniques, fairness between multiple competing video flows is still an open issue of research. Our work extends the definition of standard bitrate fairness to utility fairness where utility is the Quality of Experience (QoE) of a video stream. We define QoE max-min fairness for a set of adaptive video flows competing for bandwidth in a network and we develop an algorithm that computes the set of bitrates that should be assigned to each stream to achieve fairness. We design and implement a system that can apply QoE fairness in home networks and evaluate the system on a real home router. - A well known problem that has been associated with TCP traffic is the buffer bloat problem. We use an experimental setup to show that adaptive video flows can cause buffer bloat which can significantly harm time sensitive applications sharing the same bottleneck link with video traffic. In addition, we develop a technique that can be used by video players to mitigate this problem. We implement our technique in a real video player and evaluate it on our testbed. - With the increasing popularity of video streaming on the Internet, the amounts of traffic on the peering links between video streaming providers and Internet Service Providers (ISPs) have become the source of many disputes. Hybrid CDN/P2P streaming systems can be used to reduce the amounts of traffic on the peering links by leveraging users upload bandwidth to redistribute some of the load to other peers. We develop an analysis for hybrid CDN/P2P systems that broadcast live adaptive video streams. The analysis helps the CDN to make better decisions to optimize video quality for its users.

Multimedia services and applications became the driving force in the development and widespread deployment of wireless broadband access technologies and high speed local area networks. Mobile phone service providers are offering wide range of multimedia applications over high speed wireless data networks. People can watch live TV, stream on-demand video clips and place videotelephony calls using multimedia capable mobile devices. Mobile devices will soon support capturing and displaying high definition video. Similar evolution is also occurring in the local area domain. The video receiver or storage devices were conventionally connected to display devices using cables. By using wireless local area networking (WLAN) technologies, convenient and cable-free connectivity can be achieved. Media over wireless home networks prevents the cable mess and provides mobility to portable TVs. However, there still exit challenges for improving the quality-of-service (QoS) of multimedia applications. Conventional service architectures, network structures and protocols lack to provide a robust distribution medium since most of them are not designed considering the high data rate and real-time transmission requirements of digital video. In this thesis the challenges of wireless video streaming are addressed in two main categories. Streaming protocol level issues constitute the first category. We will refer to the collection of network protocols that enable transmitting digital compressed video from a source to a receiver as the streaming protocol. The objective of streaming protocol solutions is the high quality video transfer between two networked devices. Novel application-layer video bit-rate adaptation methods are designed for handling short- and long-term bandwidth variations of the wireless local area network (WLAN) links. Both transrating and scalable video coding techniques are used to generate video bit-rate flexibility. Another contribution of this thesis study is an error control method that dynamically adjusts the forward error correction (FEC) rate based on channel bit-error rate (BER) estimation and video coding structure. The second category is the streaming service level issues, which generally surface in large scale systems. Service system solutions target to achieve system scalability and provide low cost / high quality service to consumers. Peer-to-peer assisted video streaming technologies are developed to reduce the load of video servers. Novel video file segment caching strategies are proposed for more efficient peer-to-peer collaboration.

O uso de Internet para a distribuição de fluxos de vídeo tem se mostrado uma tendência atual e traz consigo grandes desafios. O alicerce sobre qual a Internet está fundamentada, comutação por pacotes e arquitetura cliente-servidor, não proporciona as melhores condições para este tipo de serviço. A arquitetura P2P (peer-to-peer) vem sendo considerada como infraestrutura para a distribuição de fluxos de vídeo na Internet. A idéia básica da distribuição de vídeo com o suporte de P2P é a de que os vários nós integrantes da rede sobreposta distribuem e encaminham pedaços de vídeo de forma cooperativa, dividindo as tarefas, e colocando à disposição da rede seus recursos locais. Dentro deste contexto, é importante investigar o que ocorre com a qualidade do serviço de distribuição de vídeo quando a infraestrutura provida pelas redes P2P é contaminada por nós que não estejam dispostos a cooperar, já que a base desta arquitetura é a cooperação. Neste trabalho, inicialmente é feito um estudo para verificar o quanto a presença de nós não-cooperativos pode afetar a qualidade da aplicação de distribuição de fluxo de vídeo em uma rede P2P. Com base nos resultados obtidos, é proposto um mecanismo de incentivo à cooperação para que seja garantida uma boa qualidade de vídeo aos nós cooperativos e alguma punição aos nós não-cooperativos. Os testes e avaliações foram realizados utilizando-se o simulador PeerSim. / Using the Internet for video stream is becoming a trend, but it brings many challenges. The foundation upon which the Internet is based, packet switching and client-server architecture, is not suitable for this type of service. P2P (peer to peer) architecture is being considered as an infrastructure for video streams on the Internet. The basic idea is that the several members of the overlay network cooperate in the task of distributing and fowarding video chunks, making available their local resources to the network. Within this context, it is important to investigate what happens to the quality of service of the video distribution when the infrastructure provided by the P2P network is contaminated with free-riding nodes, which are not willing to cooperate, since the basis of this architecture is cooperation. In this work, study is initially carried out to check how the presence of uncooperative nodes can affect the quality of the distribution application of video streaming on a P2P network. Based on these results, a mechanism is proposed to encourage cooperation in order to be guaranteed a video with good quality to the cooperative nodes and some punishment for those uncooperative. The tests and evaluations were performed using the PeerSim simulator.