Adaptive Speech Quality Management in Voice-over-IP Communications

IEEE 802.11 is the most popular standard for WLAN networks. It offers different physical transmission rates. This paper focuses on this multi transmission rate of 802.11 WLANs and its effect on speech quality. In non-adaptive systems, when the physical layer switches from a higher transmission rate to a lower one, different than the one that the VoIP flow needs, the switching may result in congestion, high delay and packet loss, and consequently speech quality degradation. However, there are some algorithms that adapt the transmission parameters according to the channel conditions. In this study we demonstrate how choosing parameter (different codec and packet size) can affect the voice quality, network delay and packet loss. Further, this study presents a comparison between adaptive and non-adaptive methods. The adaptive method has also been evaluated for different congestion level from perceived speech quality point of view.

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“…The comprehensive adaptive speech management for VoIP proposed by [21] has considered fluctuation in general network not specially for WLAN.…”

Section: Codec Bit-rate and Packet Size Approachmentioning

confidence: 99%

Adaptive QoS control approaches for VoIP over multi-rate WLANs

Kazemitabar

Said

2012

2012 International Conference on Computer &Amp; Information Science (ICCIS)

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“…There exists a number of VoIP quality adaptation schemes in the literature, the most recent and relevant one has been proposed by Myakotnykh and Thompson in [4]. The scheme uses two parameters, instantaneous quality level computed by E-model [2] and perceptual metric which estimates the integral speech quality as inputs to adaptation decision.…”

Section: Related Workmentioning

confidence: 99%

Feedback-Free Early VoIP Quality Adaptation Scheme in Next Generation Networks

Mkwawa

Jammeh

Sun

et al. 2010

2010 IEEE Global Telecommunications Conference GLOBECOM 2010

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The Next Generation Networks (NGNs) were proposed to provide ubiquitous access to multimedia services over the Internet. With the best effort nature of the Internet, the quality of real time multimedia services such as voice and video can rapidly degrade due to network impairments.In this paper, we propose a novel feedback-free Early VoIP Quality Adaptation Scheme (EVQAS) which can adapt voice send bit rate (e.g. Adaptive Multi-Rate (AMR) codec mode) automatically according to the buildup of queuing delay in the network. This differs from existing adaptation methods which adapt send bit rate according to either packet loss information or Perceived Quality of Service (PQoS) metrics (e.g. Mean Opinion Score (MOS) ) derived from packet loss and delay. The proposed scheme can avoid congested packet loss which has an obvious impact on perceived voice quality. We implemented this adaptation mechanism in open Android mobile and tested in Open IMS Core testbed. Preliminary results show that the early adaptation mechanism can effectively release congestion before packet loss occurs. This has increased overall perceived voice quality when compared with existing adaptation method. This lightweight feedback-free early adaptation mechanism can be easily applied to other mobile devices to improve VoIP quality in current NGN VoIP applications.

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“…On the other hand, since no specific codec can work well in all network conditions (Karapantazis & Pavlidou, 2009), developing codec adaptive techniques have been proposed. Although developing this mechanism is still in its early stage (Myakotnykh & Thompson, 2009), different adapting codec rate schemes were proposed particularly for real-time applications in wired, wireless, or WLAN networks. Codec rate adaptation technique is defined as a technique that adjusts codec parameters or changes it to another codec with lower rate when the network gets congested.…”

Section: A New Cross-layer Signaling Approach For Voip-oriented Pshp mentioning

confidence: 99%

“…Codec rate adaptation technique is defined as a technique that adjusts codec parameters or changes it to another codec with lower rate when the network gets congested. Codec parameters that can be considered in this technique are: packet size and compression rate (Myakotnykh & Thompson, 2009). Besides, adaptive rate approaches have been implemented using different constant bit-rate codec or variable bit-rate codec, such as AMR (Servetti & De Martin, 2003) and Speex (Sabrina & Valin, 2008).…”

Section: A New Cross-layer Signaling Approach For Voip-oriented Pshp mentioning

confidence: 99%