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PQoS-driven VoIP service adaptation in UMTS networks
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Source International Conference On Simulation Tools And Techniques For Communications, Networks And Systems & Workshops archive
Proceedings of the 2nd International Conference on Simulation Tools and Techniques table of contents
Rome, Italy
SESSION: QoS-oriented simulation studies table of contents
Article No. 91  
Year of Publication: 2009
ISBN:978-963-9799-45-5
Authors
Nagore Bilbao  University of the Basque Country (UPV/EHU), Bilbao, Spain
Jose Oscar Fajardo  University of the Basque Country (UPV/EHU), Bilbao, Spain
Fidel Liberal  University of the Basque Country (UPV/EHU), Bilbao, Spain
Sponsors
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: ICST
Publisher
Bibliometrics
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DOI Bookmark: 10.4108/ICST.SIMUTOOLS2009.5732

ABSTRACT

This paper shows the relevance of implementing a dynamic and real-time Voice over IP (VoIP) service adaptation mechanism in order to avoid Perceived Quality of Service (PQoS) degradations. We have studied the most relevant parameters affecting the PQoS experienced by end users in VoIP services over mobile accesses. We will describe the enhancements carried out in the implementation of the simulation model in order to assess VoIP PQoS based on ITU-T E-model and introduce a real-time service adaptation method. This adaptation method involves codec and packetization changes through SIP signalling, jointly to dejittering buffer management, and would be launched upon detection of PQoS degradation, in order to enhance end users' experience.


REFERENCES

Note: OCR errors may be found in this Reference List extracted from the full text article. ACM has opted to expose the complete List rather than only correct and linked references.

 
1
ITU-T Recommendation G.107. 2007. "The E-model, a computational model for use in transmission planning".
 
2
3GPP TS 26.235. 2008. "Packet switched conversational multimedia applications; Default codecs".
 
3
3GPP TR 26.975. 2007. "Performance characterization of the Adaptative Multi-Rate (AMR) speech codec"
 
4
Poppe, F., De Vleeschauwer, D. and Petit, G. H. 2001. "Choosing the UMTS Air Interface Parameters, the Voice Packet Size and the Dejittering Delay for Voice-over-IP Call between a UMTS and a PSTN Party". IEEE INFOCOM 2001.
 
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6
Handley, M., Jacobson, V. and Perkins, C. 2006. "SDP: Session Description Protocol", IETF RFC 4566.
 
7
Johansson, I. 2006. "VoIP Shim for RTP Payload Formats". IETF draft.
 
8
ITU-T Recommendation G.113. 2001. "Transmission impairments due to speech processing. Provisional planning values for the equipment impairment factor Ie and packet-loss robustness factor Bpl"
 
9
Vázquez, E. Simulation of SIP in the 3GPP IP Multimedia Subsystem (IMS). http://www.dit.upm.es/asignaturas/opnet/
 
10
Barbaresi, A., Mantovani, A. 2007. "Performance Evaluation of Quality of VoIP Service Over UMTS-UTRAN R99". IEEE ICC '07.
 
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Collaborative Colleagues:
Nagore Bilbao: colleagues
Jose Oscar Fajardo: colleagues
Fidel Liberal: colleagues