The Direct Adjustment Algorithm: A TCP-Friendly Adaptation Scheme

Sisalem, Dorgham; Schulzrinne, Henning

doi:10.1007/3-540-39939-9_6

Cited by 77 publications

(101 citation statements)

References 12 publications

Supporting

Mentioning

Contrasting

Unclassified

Order By: Relevance

“…The main disadvantage of this type of mechanisms is the lack of flexibility related to the TCP window dynamics [16]. Unlike window-based mechanisms, AIMD mechanisms [18] [19] are rate-based congestion control mechanisms that are not applied to a congestion window. The Rate Adaptation Protocol [18] implements an AIMD algorithm based on regular acknowledgments sent by the receiver.…”

Section: Rate and Loss Control For Multimedia Applicationsmentioning

confidence: 99%

“…The Rate Adaptation Protocol [18] implements an AIMD algorithm based on regular acknowledgments sent by the receiver. In [19], the authors propose an end-to-end rate adaptation scheme that adjusts the transmission rate of multime-dia applications to the congestion level of the network. Based on the estimation of the loss rate and the RTT obtained from the regular information of RTCP [8] reports, the sender increases the transmission rate during underload periods and reduces this rate during congestion periods, while avoiding an aggressive adaptation behavior.…”

Section: Rate and Loss Control For Multimedia Applicationsmentioning

confidence: 99%

See 1 more Smart Citation

Adaptive VoIP Transmission over Heterogeneous Wired/Wireless Networks

Trad

Afifi

2004

Interactive Multimedia and Next Generation Networks

View full text Add to dashboard Cite

Abstract. In this paper, we present an adaptive architecture for the transport of VoIP traffic over heterogeneous wired/wireless Internet environments. This architecture uses a VoIP gateway associated with an 802.11e QoS enhanced access point (QAP) to transcode voice flows before their transmissions over the wireless channel. The instantaneous bit rate is determined by a control mechanism based on the estimation of channel congestion state. Our mechanism dynamically adapts audio codec bit rate using a congestion avoidance technique so as to preserve acceptable levels of quality. A case study presenting the results relative to an adaptive system transmitting at bit rates typical of G.711 PCM (64 kbit/s) and G.726 ADPCM (40, 32, 24 and 16 kbit/s) speech coding standards illustrates the performance of the proposed framework. We perform extensive simulations to compare the performance between our adaptive audio rate control and TFRC mechanism. The results show that the proposed mechanism achieves better voice transmission performance, especially when the number of stations is fairly large.

show abstract

Section: Rate and Loss Control For Multimedia Applicationsmentioning

confidence: 99%

Section: Rate and Loss Control For Multimedia Applicationsmentioning

confidence: 99%

Adaptive VoIP Transmission over Heterogeneous Wired/Wireless Networks

Trad

Afifi

2004

Interactive Multimedia and Next Generation Networks

View full text Add to dashboard Cite

show abstract

“…The loss-delay based adaptation algorithm (LDA) was first proposed in 1998 [37] and extended in 2000 [39]. The exten sion is indicated by the name change from LDA to LDA+.…”

Section: Enhanced Loss-delay-based Adaptation Algorithmmentioning

confidence: 99%

Using MPEG-21 for cross-layer multimedia content adaptation

Kofler

Seidl

Timmerer

et al. 2008

SIViP

View full text Add to dashboard Cite

This paper presents a cross-layer modelformulated using interoperable description formats-for the adaptation of scalable H.264/MPEG-4 AVC (i.e., SVC) content in a video streaming system operating on a Wireless LAN access network without QoS mechanisms. SVC content adaptation on the server takes place on the application layer using an adaptation process compliant with the MPEG-21 Digital Item Adaptation (DIA) standard, based on input comprised of MPEG-21 DIA descriptions of content and usage environment parameters. The latter descriptions integrate information from different layers, e.g., device characteristics and packet loss rate, in an attempt to increase the interoperability of this cross-layer model, thus making it applicable to other models. For the sake of deriving model parameters, performance measurements from two wireless access point models were taken in account. Throughout the investigation it emerged that the behavior of the system strongly depends on the access point. Therefore, we investigated the use of endto-end-based rate control algorithms for steering the content adaptation. Simulations of rate adaptation algorithms were subsequently performed, leading to the conclusion that a TFRC-based adaptation technique (TCP-Friendly Rate Control) performs quite well in adapting to limited bandwidth and varying network conditions. In the paper we demonstrate how TFRC-based content adaptation can be realized using MPEG-21 tools.

show abstract

“…Many research projects have studied quality adaptive streaming in relationship with TCP-friendly congestion control [36,27,4,28,9,19,31,32,20]. A common idea amongst them is to let the transport protocol and its congestion control dictate the appropriate sending rate.…”

Section: Introductionmentioning

confidence: 99%

Quality-adaptive media streaming by priority drop

Krasic¹,

Walpole²,

Feng³

2003

Proceedings of the 13th International Workshop on Network and Operating Systems Support for Digital Audio and Video - NOSSDAV

View full text Add to dashboard Cite

This paper presents a general design strategy for streaming media applications in best effort computing and networking environments. Our target application is video on demand using personal computers and the Internet. In this scenario, where resource reservations and admission control mechanisms are not generally available, effective streaming must be able to adapt in a responsive and graceful manner. The design strategy we propose is based on a single simple idea, priority data dropping, or priority drop for short. We evaluate the efficacy of priority drop as an adaptation tool in the video and networking domains. We show how common video compression formats can be extended to support priority drop, thereby becoming streaming friendly. In particular, we demonstrate that priority-drop video allows adaptation over a wide range of rates and with fine granularity, and the adaptation is tailorable through declarative adaptationpolicy specifications. Our technical contribution with respect to video is to show how to express adaptation policies and how to do priority-mapping, an automatic translation from adaptation policies to priority assignments on the basic units of video. For the networking domain, we present priority-progress streaming, a real-time best-effort streaming protocol. We have implemented and released a prototype video streaming system that incorporates prioritydrop video, priority mapping, and priority-progress streaming. The system has the following advantages: it maintains timeliness of the stream in the face of rate fluctuations in the network, it attempts to utilize available bandwidth fully so as to maximize the average video quality, it starts video display quickly after the user initiates the stream, and it limits the number of quality changes that occur. In summary, we will show that priority-drop is very effective: a single video source can be streamed across a wide range of network bandwidths, on networks saturated with competing *

show abstract

The Direct Adjustment Algorithm: A TCP-Friendly Adaptation Scheme

Cited by 77 publications

References 12 publications

Adaptive VoIP Transmission over Heterogeneous Wired/Wireless Networks

Adaptive VoIP Transmission over Heterogeneous Wired/Wireless Networks

Using MPEG-21 for cross-layer multimedia content adaptation

Quality-adaptive media streaming by priority drop

Contact Info

Product

Resources

About