In this paper, we study the performance of the 3GPP EVS codec when this codec is used in conjunction with 100% application-layer redundancy. The objective of this work is to investigate potential performance gains for Voice over LTE (VoLTE) in bad coverage scenarios. Voice quality for the EVS codec operated in the 9.6-24.4 kbit/s bit range in super-wideband (SWB) is evaluated at different packet loss rates (PLR), using objective and subjective methods (ITU-T P.863 and P.800 ACR). Results show that EVS at 9.6 kbit/s with 100% applicationlayer redundancy has significantly higher packet loss resilience in degraded channel conditions ( 3% PLR), for an overall bit rate (around 2⇥9.6 kbit/s) compatible with VoLTE (assuming a VoLTE bearer configured to a maximum rate of 24.4 kbit/s). We also discuss the relative merit of the partial redundancy mode in the EVS codec at 13.2 kbit/s, known as the channel-aware mode (CAM), and possible RTP/RTCP signaling methods to trigger the use of application-layer redundancy.
International audienceIn this paper we evaluate metrics specified in 3GPP to characterize trade-offs between delay and quality of Voice over LTE (VoLTE) mobile phones in various network conditions. We report test results on clock accuracy, terminal delay in uplink and downlink under error-free conditions, as well as delay and quality in the presence of packet losses and network jitter. We discuss how the underlying methodology intended for delay testing can be extended to evaluate de-jitter buffer performance using a black-box approach, and how to model VoLTE packet delay/loss characteristics in a realistic way
International audienceIn this paper, we present experimental results on WebRTC voice quality as a function of LTE radio coverage. Different LTE radio conditions were tested by varying the radio path loss (with fast fading) in controlled test lab conditions. Voice quality was evaluated at different speech coding bit rates in terms of POLQA score and mouth-to-ear delay. Results show that, with acknowledged mode data transmission over LTE, degrading radio 8 coverage translated mainly into increasing end-to-end delay and 9 jitter, with virtually no packet loss up to the coverage limit. Voice 10 quality is analyzed by studying the influence of the specific jitter 11 buffer used in WebRTC endpoints
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