On retransmission-based error control for continuous media traffic in packet-switching networks

Dempsey, Bert J.; Liebeherr, Jörg; Weaver, Alfred C.

doi:10.1016/0169-7552(95)00051-8

Cited by 62 publications

(27 citation statements)

References 17 publications

Supporting

Mentioning

Contrasting

Order By: Relevance

“…Especially, streamed packets may be dropped at a router/switch due to congestion or delayed reception. On the one hand retransmission of lost packets is one of the basic mechanisms to mitigate network failures [48], but, on the other hand, this mechanism is not a feasible solution for delay-sensitive applications due to playback constraints. Frequent retransmissions may lead to severe network congestion, which is undesirable for the Internet streaming.…”

Section: Adaptive Video Streamingmentioning

confidence: 99%

Adaptive delivery of immersive 3D multi-view video over the Internet

Özçinar

Ekmekcioǧlu

ăźAlić

et al. 2016

Multimed Tools Appl

View full text Add to dashboard Cite

Section: Adaptive Video Streamingmentioning

confidence: 99%

Adaptive delivery of immersive 3D multi-view video over the Internet

Özçinar

Ekmekcioǧlu

ăźAlić

et al. 2016

Multimed Tools Appl

View full text Add to dashboard Cite

“…for #ow control and error handling) is not desirable due to the potential implosion of messages at the sender [2,34,35]. However, simple re-transmission techniques have been proven e!ective for many unicast and multicast multimedia applications [2,10,22,34]. Consequently, we employ a re-transmission method for the recovery of lost packets.…”

Section: Client Complexity and Client-server Communication Issuesmentioning

confidence: 99%

“…It has been well established in many published works that re-transmission based lost packet recovery is a viable approach for continuous media communication over packet networks [2,10,22,34]. For these applications, it has been popular to employ a negative automatic repeat request (NACK) in conjunction with re-transmission of the lost packet.…”

Section: Introductionmentioning

confidence: 99%

Scalable Internet video using MPEG-4

Radha

Chen

Parthasarathy

et al. 1999

Signal Processing: Image Communication

View full text Add to dashboard Cite

Real-time streaming of audio-visual content over Internet Protocol (IP) based networks has enabled a wide range of multimedia applications. An Internet streaming solution has to provide real-time delivery and presentation of a continuous media content while compensating for the lack of Quality-of-Service (QoS) guarantees over the Internet. Due to the variation and unpredictability of bandwidth and other performance parameters (e.g. packet loss rate) over IP networks, in general, most of the proposed streaming solutions are based on some type of a data loss handling method and a layered video coding scheme. In this paper, we describe a real-time streaming solution suitable for non-delay-sensitive video applications such as video-on-demand and live TV viewing.The main aspects of our proposed streaming solution are:1. An MPEG-4 based scalable video coding method using both a prediction-based base layer and a "ne-granular enhancement layer;2. An integrated transport-decoder bu!er model with priority re-transmission for the recovery of lost packets, and continuous decoding and presentation of video. In addition to describing the above two aspects of our system, we also give an overview of a recent activity within MPEG-4 video on the development of a "ne-granular-scalability coding tool for streaming applications. Results for the performance of our scalable video coding scheme and the re-transmission mechanism are also presented. The latter results are based on actual testing conducted over Internet sessions used for streaming MPEG-4 video in realtime.Published by

show abstract

“…As a result they do not unnecessarily waste bandwidth when there is no packet loss, and they can easily adapt to changes in the loss rates. Thus studies have examined soft ARQ for both unicast [24,25] and multicast [2,26,27] streaming multimedia. One way our work differs from all of these is that we assume there is an overall transmission rate limit, so that a retransmission of one message can come at the expense of the first transmission of another; these other works assume that enough bandwidth is available for any retransmissions the sender decides to send, and instead focus on soft ARQ protocol issues.…”

Section: Future and Related Workmentioning

confidence: 99%

Soft ARQ for Layered Streaming Media

Podolsky¹,

McCanne²,

Vetterli³

1998

View full text Add to dashboard Cite

Abstract.A growing and important class of traffic in the Internet is so-called "streaming media," in which a server transmits a packetized multimedia signal to a receiver that buffers the packets for playback. This playback buffer, if adequately sized, counteracts the adverse impact of delay jitter and reordering suffered by packets as they traverse the network, and if large enough also allows lost packets to be retransmitted before their playback deadline expires. We call this framework for retransmitting lost streaming-media packets "soft ARQ" since it represents a relaxed form of Automatic Repeat reQuest (ARQ). While state-of-the-art media servers employ such strategies, no work to date has proposed an optimal strategy for delay-constrained retransmissions of streaming media-specifically, one which determines what is the optimal packet to transmit at any given point in time. In this paper, we address this issue and present a framework for streaming media retransmission based on layered media representations, in which a signal is decomposed into a discrete number of layers and each successive layer provides enhanced quality. In our approach, the source chooses between transmitting (1) newer but critical coarse information (e.g., a first approximation of the media signal) and (2) older but less important refinement information (e.g., added details) using a decision process that minimizes the expected signal distortion at the receiver. To arrive at the proper mix of these two extreme strategies, we derive an optimal strategy for transmitting layered data over a binary erasure channel with instantaneous feedback. To provide a quantitative performance comparison of different transmission policies, we conduct a Markov-chain analysis, which shows that the best transmission policy is time-invariant and thus does not change as the frames' layers approach their expiration times.

show abstract

On retransmission-based error control for continuous media traffic in packet-switching networks

Cited by 62 publications

References 17 publications

Adaptive delivery of immersive 3D multi-view video over the Internet

Adaptive delivery of immersive 3D multi-view video over the Internet

Scalable Internet video using MPEG-4

Soft ARQ for Layered Streaming Media

Contact Info

Product

Resources

About