SEAMLESS VOIP SERVICE USING CENTRALIZED CODEC ADJUSTMENT: A FRAMEWORK IN WIRELESS NETWORK
DOI:
https://doi.org/10.11113/jt.v77.6486Keywords:
Codec adjustment, centralized codec adjustmentAbstract
The evolution of wireless technology leads VoIP application to be a good preferred choice among mobile users. Customers can having a VoIP service while being mobile regardless of location, terminal, and it is independent of network access. It is believed to be an alternative to the PSTN and its service is expected to be as good as PSTN. In essence, QoS plays a crucial part in order to maintain the seamless service during the VoIP session. As wireless networks are prone to channel error and VoIP application is known to be a delay-sensitive application, network traffic congestion could easily affect the voice quality of VoIP. Therefore, the codec selection within the VoIP application must be improved so that the codec can operate at various bit rates in order to cater for different traffic conditions. In this paper, we explain how centralize codec switching can help user to communicate seamlessly. We use WiFi network traffic as an example scenario.Â
References
J. Fitzpatrick et al. 2006. An Approach to Transport Layer Handover of VoIP over WLan. Consumer Communications and Networking Conference. 2: 1093-1097.
J. K. Mupalla et al. 2002. SIP: Session Initiation Protocol. IETF RFC3261.
S. Antoniou. 2008. VoIP: How to Effectively Encapsulate Voice in IP Packets. http://www.trainsignaltraining.com/cisco-voip-voice-encapsulation/2008-03-31/.
M. Handley et al. 1998. SDP: Session Description Protocol. IETF RFC 2327.
S. Burner et al. Voice Over IP 101: Understanding VoIP Networks. www.juniper.net, USA.
http://www.webopedia.com/TERM/G/G_7xx.html.
B. K. Singh and N. Gupta. 2000. Congestion and Decongestion in a Communication Network. Physical Review E. 71.
Rajina R. Mohamed, K.N. Choong, Usman Sarwar. 2012. Session Switching between IP/PSTN Connectivity for Voice Application. Proc. 2012. Third International Conference on Intelligent Systems Modelling and Simulation.
M. Cardenete et al. 2007. VoIP Perfromance in SIP-Based Vertical Handovers Between WLAN And GPRS/UMTS Networks. IEEE Communication Society 2007 Proceedings. 1973-1978.
A. M. Amin. 2005. VoIP Performance Measurement Using QoS Parameters. The Second International Conference on Innovations in Information Technology.
San qi-Li. Study of Packet Lossin a Packet Switched Voice system. http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=13803.
M. Hassan et al. 2006. Variable Packet Size of IP packets for VoIP Transmission. Proceeding of the 24th IASTED International Conference on Internet Multimedia System and Application.136-141.
T. Koistinen. Protocol Overview: RTP and RTCP. Nokia Telecommunication.
ITU-T Recommendation P.800. Subjective Quality Test based on Mean Opinion Scores (MOS).
Nadeem Unuh. Mean Opinion Score – A Measure of Voice Quality. http://voip.about.com/od/voipbasics/a/MOS.htm.
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