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An Adaptive Codec Switching Scheme for SIP-Based VoIP

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Part of the Lecture Notes in Computer Science book series (LNCCN,volume 7469)


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.


  • VoIP communication
  • Codec Switching
  • SIP

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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.

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  • Publisher Name: Springer, Berlin, Heidelberg

  • Print ISBN: 978-3-642-32685-1

  • Online ISBN: 978-3-642-32686-8

  • eBook Packages: Computer ScienceComputer Science (R0)