Quality of service: delivering QoS on the Internet and in corporate networks
Quality of service: delivering QoS on the Internet and in corporate networks
Multimedia Communications
Voice over Data Networks: Covering IP and Frame Relay
Voice over Data Networks: Covering IP and Frame Relay
Assessing Voice Quality in Packet-Based Telephony
IEEE Internet Computing
A Survey of Error-Concealment Schemes for Real-Time Audio and Video Transmissions over the Internet*
MSE '00 Proceedings of the 2000 International Conference on Microelectronic Systems Education
Parameter interpolation to enhance the frame erasure robustness of CELP coders in packet networks
ICASSP '01 Proceedings of the Acoustics, Speech, and Signal Processing, 200. on IEEE International Conference - Volume 02
Modeling QoS parameters of VoIP traffic with multifractal and Markov models
ICA3PP'11 Proceedings of the 11th international conference on Algorithms and architectures for parallel processing - Volume Part II
Hi-index | 0.24 |
VoIP calls transferred over dedicated bandwidth or QoS capable networks is a cost-effective alternative for PSTN in large enterprises. However, the calls made over the best effort network, such as the global Internet, suffer packet loss and jitter. In some VoIP-codecs, such as ITU G.723.1 and G.729a, there are built-in recovery mechanisms for concealing packet-based errors in the audio/speech stream. These recovery mechanisms can conceal up to 5% packet losses without significant quality degradation, as shown in this article. The 5% quality degradation is approximately within 0.5 MOS scale when compared to the original signal. Beyond 5%, the speech quality will drop gradually. The overall quality of MOS scale 3 can be maintained even with 14-17% packet loss rates. The influence of delay variation or jitter cannot be eliminated with the concealment algorithms unless the jitter time exceeds packet loss indication delay. The influence of jitter is not critical below 20ms but beyond 20ms limit its influence will decrease the speech quality very steeply. This suggests that packet losses can be recovered in normal conditions, but the influence of jitter must be eliminated somehow. The interleaving and piggybacking-based stream manipulation enhances speech quality in packet dropout situations. The guaranteed delay over the whole Internet would enhance the possibilities of VoIP to achieve success.