Active queue management with adaptive codec bit rate variation in WLAN for efficient VoIP performance

Abstract

VoIP over Wireless LANs is greatly aff ected by path loss, RF interference and other sources of signal attenuation in addition to network congestion. The primary factors involved in effective real-time communication, namely delay and loss, must be within certain controlled limits in such a scenario. Wireless LAN Access Points must, therefore, act in tune with the codecs to ensure high quality of the ongoing VoIP sessions. In this paper, we analyze the various codec parameters with respect to wireless LAN APs. Thereafter, we develop an optimization algorithm based on proactive strategy to keep loss and latency within tolerable limits. Finally, we implement the algorithm in our test-bed to provide optimum Voice over IP (VoIP) performance.

Determination of bit-rate adaptation thresholds for the Opus codec for VoIP services

“…Moller et al have shown in [16] that switching from one codec to another can improve the speech quality, if the former codec was suffering a high packet loss rate and if the latter codec significantly reduces the packet loss impact on the speech quality. Furthermore, in [17], the authors note that decreasing the transmission rate by increasing packet size is not an effective method of ensuring voice quality, possibly due to the fact that more voice data is lost when a packet is lost. To combat this, the authors proposed a similar switching method to [16].…”