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2005, ACM SIGCOMM Computer …
This paper introduces large scale content distribution protocols, which are capable of scaling to massive numbers of users and providing low delay end-to-end delivery. Delivery of files and static objects is described, with real-time content streaming being outside the scope of this paper. The focus is on solutions provided by the IETF Reliable Multicast Transport Working Group. More precisely, the paper explains FLUTE, ALC and the associated building blocks. Then it discusses how these components are used in the Multimedia Broadcast Multicast Service (MBMS) for 3G systems and in the IP Datacast (IPDC) service for Digital Video Broadcast for Handheld devices (DVB-H).
1996
This white paper outlines a simple protocol framework that was developed for reliable multicast dissemination of data files. It also provides some historical development and testing history. The general framework described here was originally developed and used by the Image Multicaster (IMM) application within the Internet Mbone for reliable multicast file transfer. This document describes the more general use of the protocol framework as a reliable bulk file transfer technique. We discuss the present operational modes, some performance issues, and the basic application data units (ADUs) used. This is not intended to be a detailed protocol specification document, but rather a broad description of the basic protocol features and a discussion of issues.
2005
This paper proposes a semi-reliable multicast protocol that aims to increase the quality of video streams transmitted in large-scale systems without overloading the video source and the communications network. This protocol, which is based on the IP multicast protocol and the MPEG standard, evaluates the necessity of retransmitting lost packets taking into account the capacity of the corresponding MPEG frames to improve the quality of the video stream.
2002
With the explosive growth of video applications over the Internet, many approaches have been proposed to stream video effectively over packet switched, best-effort networks. A number of these use techniques from source and channel coding, or implement transport protocols, or modify system architectures in order to deal with delay, loss, and time-varying nature of the Internet. In this paper, we propose a framework for streaming video from multiple senders simultaneously to a single receiver. The main motivation in doing so is to exploit path diversity in order to achieve higher throughput, and to increase tolerance to packet loss and delay due to network congestion. In this framework, we propose a receiver-driven transport protocol to coordinate simultaneous transmissions of video from multiple senders. Our protocol employs two algorithms: rate allocation and packet partition. The rate allocation algorithm determines sending rate for each sender to minimize the packet loss, while the packet partition algorithm minimizes the probability of packets arriving late. Using NS and actual Internet experiments, we demonstrate the effectiveness of our proposed distributed transport protocol in terms of the overall packet loss rate, and compare its performance against a naïve distributed protocol.
Streaming media on the internet has experienced rapid growth over the last few years and will continue to increase in importance as broadband technologies and authoring tools continue to improve. As the internet becomes an increasingly popular alternative to traditional communications media, internet streaming will become a significant component of many content providers' communications strategy. Internet streaming, however, poses significant challenges for content providers since it has significant distribution problems. Scalability, quality, reliability, and cost are all issues that have to be addressed in a successful streaming media offering.
2008
Content distribution in general, and multicasting in particular, over a wired network to static hosts can be realized by placing proxies and gateways at several parts of the network. However, if the end hosts are mobile over heterogeneous wireless access networks, one needs to consider many operational issues such as network detection, handoff, join and leave latency, and desired level of quality of service, as well as caching and load balancing. This article surveys a set of protocols and technologies that offer multicast-based services for streaming multimedia in a mobile environment. It also brings forth some of the issues related to mobile content distribution in the wireless Internet that may be helpful during its deployment by application service providers.
2006
Mobile multimedia services like news, weather forecasts and mobile television require efficient technologies for distributing multimedia contents simultaneously to large mobile user groups. UMTS (Universal Mobile Telecommunications System) Release 6 standardises MBMS (Multimedia Broadcast and Multicast Services) for the first time. This article gives an overview of the MBMS standard and discusses current research activities for the evolution of MBMS.
IEEE Communications Magazine, 2000
Multimedia applications over Internet are becoming popular due to the widespread deployment of broadband access. In conventional streaming architectures the client-server model and the usage of content distribution networks (CDNs) along with IP multicast were the most desirable approaches for many years. The client/server architecture, however, severely limits the number of simultaneous users in video streaming. The reason is the bandwidth bottleneck at the server side, since usually many clients request the content from the server. A CDN overcomes the same bottleneck problem by introducing dedicated servers at geographically different locations, resulting in expensive deployment and maintenance.
2002
Abstract To serve asynchronous requests using multicast, two categories of techniques---stream merging and periodic broadcasting---have been proposed. For sequential streaming access, where requests are uninterrupted from the beginning to the end of an object, these techniques are highly scalable: the required server bandwidth for stream merging grows logarithmically as request arrival rate, and the required server bandwidth for periodic broadcasting varies logarithmically as the inverse of start-up delay.
2001
Despite its obvious suitability for distributed multimedia applications, multicasting has not yet found widespread application. Having analyzed shortcomings of today's approaches, we devise in the GCAP project a new endto-end transport architecture for multimedia multicasting that supports partial order and partial reliability. In this paper, we argue that, at the network layer, single-source multicasting (PIM-SSM) should be chosen. Consequently, our Monomedia Multicast protocol provides, along with reliability and QoS monitoring functionality, an ALM based multicast solution referred to as TBCP (Tree Building Control Protocol), to be used as back channel for SSM, e.g. for retransmission requests. On top of the Monomedia protocol, our Multimedia Multicast protocol handles multimedia sessions composed of multiple monomedia connections: The FPTP (Fully Programmable Transport Protocol) allows applications to specify, through its API, the (global) synchronization and (individual) reliability requirements within a multimedia session. Our group management approach is focused on group integrity.
ACM SIGCOMM Computer Communication Review, 2002
We describe a new scalable application-layer multicast protocol, specifically designed for low-bandwidth, data streaming applications with large receiver sets. Our scheme is based upon a hierarchical clustering of the application-layer multicast peers and can support a number of different data delivery trees with desirable properties.We present extensive simulations of both our protocol and the Narada application-layer multicast protocol over Internet-like topologies. Our results show that for groups of size 32 or more, our protocol has lower link stress (by about 25%), improved or similar end-to-end latencies and similar failure recovery properties. More importantly, it is able to achieve these results by using orders of magnitude lower control traffic.Finally, we present results from our wide-area testbed in which we experimented with 32-100 member groups distributed over 8 different sites. In our experiments, average group members established and maintained low-latency paths and ...
Multimedia Computing and Networking 2002, 2001
With the explosive growth of video applications over the Internet, many approaches have been proposed to stream video effectively over packet switched, best-effort networks. A number of these use techniques from source and channel coding, or implement transport protocols, or modify system architectures in order to deal with delay, loss, and time-varying nature of the Internet. In this paper, we propose a framework for streaming video from multiple senders simultaneously to a single receiver. The main motivation in doing so is to exploit path diversity in order to achieve higher throughput, and to increase tolerance to packet loss and delay due to network congestion. In this framework, we propose a receiver-driven transport protocol to coordinate simultaneous transmissions of video from multiple senders. Our protocol employs two algorithms: rate allocation and packet partition. The rate allocation algorithm determines sending rate for each sender to minimize the packet loss, while the packet partition algorithm minimizes the probability of packets arriving late. Using NS and actual Internet experiments, we demonstrate the effectiveness of our proposed distributed transport protocol in terms of the overall packet loss rate, and compare its performance against a naïve distributed protocol.
ArXiv, 2013
A grid computing system is designed for solving complicated scientific and commercial problems effectively, whereas mobile computing is a traditional distributed system having computing capability with mobility and adopting wireless communications. Media and Entertainment fields can take advantage from both paradigms by applying its usage in gaming applications and multimedia data management. Multimedia data has to be stored and retrieved in an efficient and effective manner to put it in use. In this paper, we proposed an application layer protocol for delivery of multimedia data in wireless girds i.e. multimedia grid protocol (MMGP). To make streaming efficient a new video compression algorithm called dWave is designed and embedded in the proposed protocol. This protocol will provide faster, reliable access and render an imperceptible QoS in delivering multimedia in wireless grid environment and tackles the challenging issues such as i) intermittent connectivity, ii) device hetero...
1998
In order to provide guaranteed QoS multiparty collaborative multimedia applications require reliable transmission of data. The multimedia applications can vary from distributed games, shared whiteboard to interactive video conferencing. These applications often involve a large number of participants and are interactive in nature with participants dynamically joining and leaving the applications[Sudan95]. In order to provide many-to-many interaction when the number of participants is large IP multicast is a very good option for communication. IP multicast provides scalability and efficient routing but does not provide the reliability these multimedia applications may require. Though a lot of research has been done on reliable multicast transport protocol, it really seems that the only way of doing a reliable multicast is to build it for a given purpose like conference control in multimedia conferencing. This paper compares some of the available multicast transport protocols and analyses the most suitable features and functionalities provided by these protocols for a facet of conference control, floor control. The goal is to find or design a reliable multicast transport protocol which would scale to tens or hundreds of participants scattered across the Internet and deliver the control messages reliably.
Proceedings of The IEEE, 2004
Streaming media on the Internet has experienced rapid growth over the last few years and will continue to increase in importance as broadband technologies and authoring tools continue to improve. As the Internet becomes an increasingly popular alternative to traditional communications media, Internet streaming will become a significant component of many content providers' communications strategies. Internet streaming, however, poses significant challenges for content providers, since it has significant distribution problems. Scalability, quality, reliability, and cost are all issues that have to be addressed in a successful streaming media offering. Streaming content delivery networks (streaming CDNs) attempt to provide solutions to the bottlenecks encountered by streaming applications on the Internet. However, only a small number of them has been deployed, and little is known about the internal organization of these systems. In this paper, we discuss the design choices made during the evolution of Akamai's CDN for streaming media. In particular, we look at the design choices made to ensure the network's scalability, quality of delivered content, and reliability while keeping costs low. Performance studies conducted on the evolving system indicate that our design scores highly on all of the above categories.
2004
In this paper, we propose a new scalable application-layer protocol, specifically designed for data streaming applications with large client sets. This is based upon a control hierarchy of successive levels for the clients, has minimal overhead with constant number of messages per client, and is robust to client and network failures, making it suitable for wireless environments. The video server bandwidth utilization is also significantly reduced. We present an analysis and simulation results, showing that LEMP is near optimum in terms of performance.
Proceedings of the 8th …, 2001
In multimedia distributed applications such as videoconferencing, real time data exchange requires formal models to design and specify the Quality of Service (QoS) of such multimedia communication. The proposed multimedia multicast protocols are based on enhanced Timed Petri Nets multimedia models. These models allow synchronised data streams to be exchanged in partial order and reliability, according to the required QoS. So, using a multicast monomedia transport sub-layer, we are able to design and implement multicast multimedia transport protocols, in which the multimedia synchronisation is defined in our multimedia models. It will be shown how these protocols have been designed to support multicast communication. All these protocols must be deployed in current networks, without operator support. This deployment will be done using active networking facilities to download the required code remotely on selected nodes. Experiments have given the first implementation results on European network as defined in EU project "GCAP".
Wireless Communications and Mobile Computing, 2008
This article provides an overview of the multimedia broadcast/multicast service (MBMS) for universal mobile telecommunications system (UMTS) networks. We first outline the features of UMTS networks as defined by the 3rd generation partnership project (3GPP) in order to provide a background for the discussion to follow. We then present the overall MBMS architecture, the services that it provides to the users and the differences between the broadcast and multicast options. The implementation details of MBMS are explained in terms of the modifications needed to the network, the new signaling procedures required and the impact of MBMS on the radio part of the network. We then describe how the security architecture of UMTS is used to support content protection and key distribution for MBMS groups. We evaluate the prospects of MBMS by comparing it first with IP multicast and then with DVB-H, considered by many to be the closest competitor of MBMS. Finally, we conclude with a discussion of the technical and business challenges still faced by MBMS.
Voice over IP (VoIP) is one of the most important and complex new services that are being introduced in Internet. VoIP makes use of several different technologies like signalling, streaming of real time data, session management, etc. The development and the experimentation of video conferencing applications over IP multicast networks have contributed greatly to the maturation of some of these technologies. This article summarizes the most important topics related with IP Multicast technology and video conferecing over IP Multicast networks. After introducing IP multicast technology as a mean to support many-to-many communications, we present some of the protocols and the applications used over IP multicast service. Finally, we outline some of the problems that preclude IP multicast to be widely deployed.
Lecture Notes in Computer Science, 2011
This chapter focuses on the multimedia distribution over Internet IP under the auspices of the NoE Euro-NGI research project "Routing in Overlay Networks (ROVER)". The multimedia distribution is supported by several components such as services, content distribution chain, protocols and standards whilst Internet is used for content acquisition, management and delivery as well as an Internet Protocol Television (IPTV) infrastructure with QoS facilities. As the convergence between fixed and mobile services of wide and local area networks is also expected to take place in the home networking, this puts an extra burden on multimedia distribution, which requires the different types of wireless access solutions (e.g., WiMAX). In this context, the ROVER research project adopts the IP Multimedia Subsystem (IMS), which offers a wide range of multimedia services over a single IP infrastructure such as authentication and, for wireless services, roaming capabilities. The research project also considers overlay routing as an alternative solution for content distribution.
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