A source and channel rate adaptation algorithm for AMR in VoIP using the Emodel

Matta, Johnny; Pépin, Christine; Lashkari, Khosrow; Jain, Ravi

doi:10.1145/776337.776338

Cited by 8 publications

(7 citation statements)

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“…Three different types of degradation may occur in the last-hop wireless link: packet loss due to congestion, delay due to congestion and packet loss due to bit errors at the link layer. Although a lot of voice codec can tolerate some small loss without severe degradation, most of them operate under preset schemes for data and channel code rates making them vulnerable to the varying conditions on wired and wireless IP-based hops [17]. Some kind of adaptation is therefore needed to dynamically adapt the codec bit rate to the changing wireless network conditions so as to preserve acceptable levels of reliability and quality.…”

Section: Problem Statementmentioning

confidence: 99%

Adaptive VoIP Transmission over Heterogeneous Wired/Wireless Networks

Trad

Afifi

2004

Interactive Multimedia and Next Generation Networks

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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.

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Section: Problem Statementmentioning

confidence: 99%

Adaptive VoIP Transmission over Heterogeneous Wired/Wireless Networks

Trad

Afifi

2004

Interactive Multimedia and Next Generation Networks

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“…In recent years, researchers have proposed a number of methods approaching this issue from a user-centric view (i.e., adapting the sending rate of voice calls based on user satisfaction [2] [3]). Rate adaptation mechanisms ramp up the sending rate quickly when the available bandwidth is sufficient, and carefully tune up or down the sending rate when the network becomes congested.…”

Section: Introductionmentioning

confidence: 99%

Modeling the qoe of rate changes in SKYPE/SILK VoIP calls

Chen

Chu

Yeh

et al. 2012

Proceedings of the 20th ACM International Conference on Multimedia

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The effective end-to-end transport of delay-sensitive voice data has long been a problem in multimedia networking. One of the major issues is determining the sending rate of real-time VoIP streams such that the user experience is maximized per unit network resource consumed. A particularly interesting complication that remains to be addressed is that the available bandwidth is often dynamic. Thus, it is unclear whether a marginal increase warrants better user experience. If a user naively tunes the sending rate to the optimum at any given opportunity, the user experience could fluctuate.To investigate the effects of magnitude and frequency of rate changes on user experience, we recruited 127 human participants to systematically score emulated Skype calls with different combinations of rate changes, including varying magnitude and frequency of rate changes. Results show that 1) the rate change frequency affects the user experience on a logarithmic scale, echoing Weber-Fechner's Law [1], 2) the effect of rate change magnitude depends on how users perceive the quality difference, and 3) this study derives a closed-form model of user perception for rate changes for Skype calls.

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“…Let W (p ′ ) denote the energy consumed by the typical node when it transmits with MAP p ′ . In practice, in many applications, the utility related to goodput is not a linear function of the MAP, see for example [3], [5]. Keeping this and the remarks made above in view, we shall be interested in a utility of the form…”

Section: Rate Control: the Non-cooperative Casementioning

confidence: 99%

Stochastic geometry based medium access games

Hanawal¹,

Altman²,

Baccelli³

2012

2012 Proceedings IEEE INFOCOM

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Abstract-This paper studies the performance of Mobile Ad hoc Network (MANET) when the nodes, that form a Poisson point process, selfishly choose their Medium Access Probability (MAP). We introduce a pricing scheme based on the transmission energy requirements and compute the symmetric Nash equilibrium MAP in closed form. It is shown that by appropriately pricing the nodes the selfish behavior of the nodes can be used to achieve the social optimum at equilibrium. Goodput and delay are considered as the performance metric that each node is interested in optimizing taking into account the cost it incurs. The Price of anarchy is then analyzed for these games. When utility with delay is considered, we bound the price of anarchy and study the effect of system parameters. When utility with goodput is considered it is shown that PoA is infinite at the price that achieves global optimal density of success.

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A source and channel rate adaptation algorithm for AMR in VoIP using the Emodel

Cited by 8 publications

References 0 publications

Adaptive VoIP Transmission over Heterogeneous Wired/Wireless Networks

Adaptive VoIP Transmission over Heterogeneous Wired/Wireless Networks

Modeling the qoe of rate changes in SKYPE/SILK VoIP calls

Stochastic geometry based medium access games

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