Video streaming in unstructured peer-to-peer networks

With the increase in demand for streaming media capabilities across the Internet, the focus has shifted from traditional client-server to peer-to-peer approaches. Content Distribution Networks (CDNs) have also recently moved from web acceleration to media streaming. P2P CDNs can be used both as a delivery mechanism and as an independent network. However, media streaming poses different challenges from traditional content distribution, such as in-order distribution; and p2p networks use more traffic, and lack QoS control and measurement. In addition, constraints like a high churn rate and small upload bandwidths can affect the video playback at the peers. We find that certain strategies can be used to optimize the streaming experience at the receiving nodes, while also being scalable and robust to churn. This project presents the experimental results of MPEG-4 video streaming using different approaches in unstructured p2p networks

Network coding meets multimedia: a review

While every network node only relays messages in a traditional communication system, the recent network coding (NC) paradigm proposes to implement simple in-network processing with packet combinations in the nodes. NC extends the concept of "encoding" a message beyond source coding (for compression) and channel coding (for protection against errors and losses). It has been shown to increase network throughput compared to traditional networks implementation, to reduce delay and to provide robustness to transmission errors and network dynamics. These features are so appealing for multimedia applications that they have spurred a large research effort towards the development of multimedia-specific NC techniques. This paper reviews the recent work in NC for multimedia applications and focuses on the techniques that fill the gap between NC theory and practical applications. It outlines the benefits of NC and presents the open challenges in this area. The paper initially focuses on multimedia-specific aspects of network coding, in particular delay, in-network error control, and mediaspecific error control. These aspects permit to handle varying network conditions as well as client heterogeneity, which are critical to the design and deployment of multimedia systems. After introducing these general concepts, the paper reviews in detail two applications that lend themselves naturally to NC via the cooperation and broadcast models, namely peer-to-peer multimedia streaming and wireless networkin

Live Streaming with Gossip

Peer-to-peer (P2P) architectures have emerged as a popular paradigm to support the dynamic and scalable nature of distributed systems. This is particularly relevant today, given the tremendous increase in the intensity of information exchanged over the Internet. A P2P system is typically composed of participants that are willing to contribute resources, such as memory or bandwidth, in the execution of a collaborative task providing a benefit to all participants. File sharing is probably the most widely used collaborative task, where each participant wants to receive an individual copy of some file. Users collaborate by sending fragments of the file they have already downloaded to other participants. Sharing files containing multimedia content, files that typically reach the hundreds of megabytes to gigabytes, introduces a number of challenges. Given typical bandwidths of participants of hundreds of kilobits per second to a couple of megabits per second, it is unacceptable to wait until completion of the download before actually being able to use the file as the download represents a non negligible time. From the point of view of the participant, getting the (entire) file as fast as possible is typically not good enough. As one example, Video on Demand (VoD) is a scenario where a participant would like to start previewing the multimedia content (the stream), offered by a source, even though only a fraction of it has been received, and then continue the viewing while the rest of the content is being received. Following the same line of reasoning, new applications have emerged that rely on live streaming: the source does not own a file that it wants to share with others, but shares content as soon as it is produced. In other words, the content to distribute is live, not pre-recorded and stored. Typical examples include the broadcasting of live sports events, conferences or interviews. The gossip paradigm is a type of data dissemination that relies on random communication between participants in a P2P system, sharing similarities with the epidemic dissemination of diseases. An epidemic starts to spread when the source randomly chooses a set of communication partners, of size fanout, and infects them, i.e., it shares a rumor with them. This set of participants, in turn, randomly picks fanout communication partners each and infects them, i.e., share with them the same rumor. This paradigm has many advantages including fast propagation of rumors, a probabilistic guarantee that each rumor reaches all participants, high resilience to churn (i.e., participants that join and leave) and high scalability. Gossip therefore constitutes a candidate of choice for live streaming in large-scale systems. These advantages, however, come at a price. While disseminating data, gossip creates many duplicates of the same rumor and participants usually receive multiple copies of the same rumor. While this is obviously a feature when it comes to guaranteeing good dissemination of the rumor when churn is high, it is a clear disadvantage when spreading large amounts of multimedia data (i.e., ordered and time-critical) to participants with limited resources, namely upload bandwidth in the case of high-bandwidth content dissemination. This thesis therefore investigates if and how the gossip paradigm can be used as a highly effcient communication system for live streaming under the following specific scenarios: (i) where participants can only contribute limited resources, (ii) when these limited resources are heterogeneously distributed among nodes, and (iii) where only a fraction of participants are contributing their fair share of work while others are freeriding. To meet these challenges, this thesis proposes (i) gossip++: a gossip-based protocol especially tailored for live streaming that separates the dissemination of metadata, i.e., the location of the data, and the dissemination of the data itself. By first spreading the location of the content to interested participants, the protocol avoids wasted bandwidth in sending and receiving duplicates of the payload, (ii) HEAP: a fanout adaptation mechanism that enables gossip to adapt participants' contribution with respect to their resources while still preserving its reliability, and (iii) LiFT: a protocol to secure high-bandwidth gossip-based dissemination protocols against freeriders

Enhanced Multimedia Exchanges over the Internet

Although the Internet was not originally designed for exchanging multimedia streams, consumers heavily depend on it for audiovisual data delivery. The intermittent nature of multimedia traffic, the unguaranteed underlying communication infrastructure, and dynamic user behavior collectively result in the degradation of Quality-of-Service (QoS) and Quality-of-Experience (QoE) perceived by end-users. Consequently, the volume of signalling messages is inevitably increased to compensate for the degradation of the desired service qualities. Improved multimedia services could leverage adaptive streaming as well as blockchain-based solutions to enhance media-rich experiences over the Internet at the cost of increased signalling volume. Many recent studies in the literature provide signalling reduction and blockchain-based methods for authenticated media access over the Internet while utilizing resources quasi-efficiently. To further increase the efficiency of multimedia communications, novel signalling overhead and content access latency reduction solutions are investigated in this dissertation including: (1) the first two research topics utilize steganography to reduce signalling bandwidth utilization while increasing the capacity of the multimedia network; and (2) the third research topic utilizes multimedia content access request management schemes to guarantee throughput values for servicing users, end-devices, and the network. Signalling of multimedia streaming is generated at every layer of the communication protocol stack; At the highest layer, segment requests are generated, and at the lower layers, byte tracking messages are exchanged. Through leveraging steganography, essential signalling information is encoded within multimedia payloads to reduce the amount of resources consumed by non-payload data. The first steganographic solution hides signalling messages within multimedia payloads, thereby freeing intermediate node buffers from queuing non-payload packets. Consequently, source nodes are capable of delivering control information to receiving nodes at no additional network overhead. A utility function is designed to minimize the volume of overhead exchanged while minimizing visual artifacts. Therefore, the proposed scheme is designed to leverage the fidelity of the multimedia stream to reduce the largest amount of control overhead with the lowest negative visual impact. The second steganographic solution enables protocol translation through embedding packet header information within payload data to alternatively utilize lightweight headers. The protocol translator leverages a proposed utility function to enable the maximum number of translations while maintaining QoS and QoE requirements in terms of packet throughput and playback bit-rate. As the number of multimedia users and sources increases, decentralized content access and management over a blockchain-based system is inevitable. Blockchain technologies suffer from large processing latencies; consequently reducing the throughput of a multimedia network. Reducing blockchain-based access latencies is therefore essential to maintaining a decentralized scalable model with seamless functionality and efficient utilization of resources. Adapting blockchains to feeless applications will then port the utility of ledger-based networks to audiovisual applications in a faultless manner. The proposed transaction processing scheme will enable ledger maintainers in sustaining desired throughputs necessary for delivering expected QoS and QoE values for decentralized audiovisual platforms. A block slicing algorithm is designed to ensure that the ledger maintenance strategy is benefiting the operations of the blockchain-based multimedia network. Using the proposed algorithm, the throughput and latency of operations within the multimedia network are then maintained at a desired level

Towards Trouble-Free Networks for End Users

Network applications and Internet services fail all too frequently. However, end users cannot effectively identify the root cause using traditional troubleshooting techniques due to the limited capability to distinguish failures caused by local network elements from failures caused by elements located outside the local area network. To overcome these limitations, we propose a new approach, one that leverages collaboration of user machines to assist end users in diagnosing various failures related to Internet connectivity and poor network performance. First, we present DYSWIS ("Do You See What I See?"), an automatic network fault detection and diagnosis system for end users. DYSWIS identifies the root cause(s) of network faults using diagnostic rules that consider diverse information from multiple nodes. In addition, the DYSWIS rule system is specially designed to support crowdsourced and distributed probes. We also describe the architecture of DYSWIS and compare its performance with other tools. Finally, we demonstrate that the system successfully detects and diagnoses network failures which are difficult to diagnose using a single-user probe. Failures in lower layers of the protocol stack also have the potential to disrupt Internet access; for example, slow Internet connectivity is often caused by poor Wi-Fi performance. Channel contention and non-Wi-Fi interference are the primary reasons for this performance degradation. We investigate the characteristics of non-Wi-Fi interference that can severely degrade Wi-Fi performance and present WiSlow ("Why is my Wi-Fi slow?"), a software tool that diagnoses the root causes of poor Wi-Fi performance. WiSlow employs user-level network probes and leverages peer collaboration to identify the physical location of these causes. The software includes two principal methods: packet loss analysis and 802.11 ACK number analysis. When the issue is located near Wi-Fi devices, the accuracy of WiSlow exceeds 90%. Finally, we expand our collaborative approach to the Internet of Things (IoT) and propose a platform for network-troubleshooting on home devices. This platform takes advantage of built-in technology common to modern devices --- multiple communication interfaces. For example, when a home device has a problem with an interface it sends a probe request to other devices using an alternative interface. The system then exploits cooperation of both internal devices and remote machines. We show that this approach is useful in home networks by demonstrating an application that contains actual diagnostic algorithms

Robust P2P Live Streaming

Projecte fet en col.laboració amb la Fundació i2CATThe provisioning of robust real-time communication services (voice, video, etc.) or media contents through the Internet in a distributed manner is an important challenge, which will strongly influence in current and future Internet evolution. Aware of this, we are developing a project named Trilogy leaded by the i2CAT Foundation, which has as main pillar the study, development and evaluation of Peer-to-Peer (P2P) Live streaming architectures for the distribution of high-quality media contents. In this context, this work concretely covers media coding aspects and proposes the use of Multiple Description Coding (MDC) as a flexible solution for providing robust and scalable live streaming over P2P networks. This work describes current state of the art in media coding techniques and P2P streaming architectures, presents the implemented prototype as well as its simulation and validation results

Robust peer-to-peer systems

textPeer-to-peer (p2p) approaches are an increasingly effective way to deploy services. Popular examples include BitTorrent, Skype, and KaZaA. These approaches are attractive because they can be highly fault-tolerant, scalable, adaptive, and less expensive than a more centralized solution. Cooperation lies at the heart of these strengths. Yet, in settings where working together is crucial, a natural question is: "What if users stop cooperating?" After all, cooperative services are typically deployed over multiple administrative domains, and thus vulnerable to Byzantine failures and users who may act selfishly. This dissertation explores how to construct p2p systems to tolerate Byzantine participants while also incentivizing selfish participants to contribute resources. We describe how to balance obedience against choice in building a robust p2p live streaming system. Imposing obedience is desirable as it leaves little room for peers to attack or cheat the system. However, providing choice is also attractive as it allows us to engineer flexible and efficient solutions. We first focus on obedience by using Nash equilibria to drive the design of BAR Gossip, the first gossip protocol that is resilient to Byzantine and selfish nodes. BAR Gossip relies on verifiable pseudo-random partner selection to eliminate non-determinism, which can be used to game the system, while maintaining the robustness and rapid convergence of traditional gossip. A novel fair enough exchange primitive entices cooperation among selfish peers on short timescales, thereby avoiding the need for distributed reputation schemes. We next focus on tempering obedience with choice by using approximate equilibria to guide the construction of a novel p2p live streaming system. These equilibria allow us to design incentives to limit selfish behavior rigorously, yet provide sufficient flexibility to build practical systems. We show the advantages of using an [element of]-Nash equilibrium, instead of an exact Nash, to design and implement FlightPath, our live streaming system that uses bandwidth efficiently, absorbs flash crowds, adapts to sudden peer departures, handles churn, and tolerates malicious activity.Computer Science

Algoritmos para retransmissão de vídeo H.264 em redes sobrepostas

Dissertação (mestrado) - Universidade Federal de Santa Catarina, Centro Tecnológico, Programa de Pós-Graduação em Ciência da Computação, Florianópolis, 2010Com a popularização da banda larga os usuários domésticos passaram a consumir diversos serviços multimídia pela Internet: Telefonia IP, rádio, vídeo sob demanda. Por enquanto, em se tratando de vídeos, os principais serviços disponíveis na rede não oferecem as características presentes na TV, ou seja, não são ao vivo e necessitam de buferização, gerando portanto atraso. Eles também requerem uma grande largura de banda nos servidores, linearmente crescente com número de clientes. Através das redes sobrepostas (orientadas a dados) pode-se remediar a falta de suporte ao IP Multicast, que seria uma solução para essa classe de aplicação, sem os problemas com a largura de banda. Entretanto tais redes não especificam o tratamento das perdas de pacotes, de modo que ou estas são ignoradas, ou são recuperadas em qualquer situação. Este trabalho faz uma análise dos recursos do padrão de compressão de vídeo digital H.264 e apresenta critérios claros para retransmissão a partir dos diferentes tipos de codificação presentes nesse formato. Estabelece prioridades a partir desses tipos e define três algoritmos para a escolha de partes perdidas para a recuperação. Um deles (SeRViSO) observa uma meta fixa levando em conta as prioridades, em termos de quanto dos dados deve retransmitir. O segundo (Adaptativo) usa uma meta variável, buscando inicialmente a retransmissão de todas as partes perdidas de um segmento, mas a cada vez que a retransmissão de partes do mesmo segmento for solicitada a meta diminui. O último algoritmo ((m,k)-Firm) também tem metas fixas, mas uma para cada tipo de codificação, e não tolera perdas de partes com o mesmo tipo de codificação muito próximas. Todos eles são definidos dentro do escopo da proposta da rede sobreposta SeRViSO. As vantagens de um tratamento claro das perdas são óbvias: maior adaptação às condições da rede e perdas menos importantes, ou seja, que não redundem em propagação de erro por muitos quadros. Os resultados dos testes com um protótipo confirmam isso, de forma que a perda de partes do tipo mais importante foi consistentemente menor que a perda média

Peer Selection in Peer-to-Peer Streaming Systems

One important task of any peer-to-peer streaming system (p2p-ss) is how to choose which peers should connect to which peers. How well a p2p-ss perform this task greatly influences its performance. This thesis explores how different peer selection algorithms affect the performance of such systems. A framework for doing the comparisons of peer selection algorithms is built on top of the network simulator ns2, making it possible to later extend the simulations with new peer selection algorithms, congestion control algorithms, wireless networks, cross traffic and other. However, ns2 is a low-level simulator, hence limiting the number of peers in the simulations, because CPU-resources are limited. The simulations are limited to single-layered streams. We find that a centralized selection method, which utilizes knowledge of bandwidth capacities and routing in the network, greatly outperforms both simple random selection of peers, and selection of close peers. Even though centralized selection does not scale well, and is therefore only applicable for a limited number of peers, this shows there is much room for improvement over basic strategies