A generic algorithm for mid-call audio codec switching

Haytham Assem, Mohamed Adel, Brendan Jennings, David Malone, Jonathan Dunne, Pat O'Sullivan

Research output: Chapter in Book/Report/Conference proceedingConference contributionpeer-review

7 Citations (Scopus)

Abstract

We present and evaluate an algorithm that performs in-call selection of the most appropriate audio codec given prevailing conditions on the network path between the endpoints of a voice call. We have studied the behaviour of different codecs under varying network conditions, in doing so deriving the impairment factors for non-ITU-T codecs so that the E-model can be used to assess voice call quality for them. Moreover, we have studied the drawbacks of codec switching from the end user perception point of view; our switching algorithm seeks to minimise this impact. We have tested our algorithm on different packages that contain a selection of the most commonly used codecs: G.711, SILK, ILBC, GSM and SPEEX. Our results show that in many typical network scenarios, our switching codecs mid-call algorithm results in better Quality of Experience (QoE) than would have been achieved had the initial codec been used throughout the call.

Original languageEnglish
Title of host publicationProceedings of the 2013 IFIP/IEEE International Symposium on Integrated Network Management, IM 2013
Pages1276-1281
Number of pages6
Publication statusPublished - 2013
Event2013 IFIP/IEEE International Symposium on Integrated Network Management, IM 2013 - Ghent, Belgium
Duration: 27 May 201331 May 2013

Publication series

NameProceedings of the 2013 IFIP/IEEE International Symposium on Integrated Network Management, IM 2013

Conference

Conference2013 IFIP/IEEE International Symposium on Integrated Network Management, IM 2013
Country/TerritoryBelgium
CityGhent
Period27/05/201331/05/2013

Keywords

  • Audio Codecs
  • Codec Switching
  • E-model
  • VoIP

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