Abstract
Contemporary Voice-Over-IP (VoIP) systems typically negotiate only one codec for the entire VoIP session life time. However, as different codecs perform differently well under certain network conditions like delay, jitter or packet loss, this can lead to a reduction of quality if those conditions change during the call. This paper makes two core contributions: First, we compare the speech quality of a set of standard VoIP codecs given different network conditions. Second, we propose an adaptive end-to-end based codec switching scheme that fully conforms to the SIP standard. Our evaluation with a real-world prototype based on Linphone shows that our codec switching scheme adapts well to changing network conditions, improving overall speech quality.
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© 2012 Springer-Verlag Berlin Heidelberg
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Aktas, I., Schmidt, F., Weingärtner, E., Schnelke, CJ., Wehrle, K. (2012). An Adaptive Codec Switching Scheme for SIP-Based VoIP. In: Andreev, S., Balandin, S., Koucheryavy, Y. (eds) Internet of Things, Smart Spaces, and Next Generation Networking. ruSMART NEW2AN 2012 2012. Lecture Notes in Computer Science, vol 7469. Springer, Berlin, Heidelberg. https://doi.org/10.1007/978-3-642-32686-8_32
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DOI: https://doi.org/10.1007/978-3-642-32686-8_32
Publisher Name: Springer, Berlin, Heidelberg
Print ISBN: 978-3-642-32685-1
Online ISBN: 978-3-642-32686-8
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