General Information
    • ISSN: 1793-8201 (Print), 2972-4511 (Online)
    • Abbreviated Title: Int. J. Comput. Theory Eng.
    • Frequency: Quarterly
    • DOI: 10.7763/IJCTE
    • Editor-in-Chief: Prof. Mehmet Sahinoglu
    • Associate Editor-in-Chief: Assoc. Prof. Alberto Arteta, Assoc. Prof. Engin Maşazade
    • Managing Editor: Ms. Mia Hu
    • Abstracting/Indexing: Scopus (Since 2022), INSPEC (IET), CNKI,  Google Scholar, EBSCO, etc.
    • Average Days from Submission to Acceptance: 192 days
    • E-mail: ijcte@iacsitp.com
    • Journal Metrics:

Editor-in-chief
Prof. Mehmet Sahinoglu
Computer Science Department, Troy University, USA
I'm happy to take on the position of editor in chief of IJCTE. We encourage authors to submit papers concerning any branch of computer theory and engineering.

IJCTE 2015 Vol.7(1): 62-65 ISSN: 1793-8201
DOI: 10.7763/IJCTE.2015.V7.931

Practical Field Overview Voice Quality of RTP Packet Size Analyze on Codec G729 Annexb = no in Low Bandwidth Area of Bangladesh

N. A. Shafi, Al Kawser, and O. Farrok

Abstract—In VoIP applications, packet loss causes a major impact on perceived speech quality. This impact is affected by some factors like packet loss locations, loss size and loss pattern. In this paper, we have investigated using perceptual-based objective measurement methods the impact on loss location on perceived speech quality and the relationships between convergence time and loss location for two different codecs (G.729 and G.723). Experimental result shows that loss location has a severe effect on perceived speech quality. The convergence time rely on the speech content e.g. voiced/unvoiced. In terms of unvoiced segments, the convergence time is stable as in voiced phases it varies. But it associate degree bound at the top of the segment. Our method allows a more accurate measurement of the exact effect of packet loss on perceived speech quality. As most of the internet subscribers of Asian countries use very low internet bandwidth. Hence, the goal of this analysis is to propose to change some parameters of this system so that the quality of voice may be kept in a tolerable limit using only 5kbps to 5.5kbps where this codec uses at least 6.4kbps. In a real life environment it is tested practically in research that it is possible to transmit voice satisfactorily using 5.3 kbps by changing some parameter described in this paper.

Index Terms—G.723, G.729, annexb = no, RTP, UDP.

N. A. Shafi is with the Network Operations Center of Inspire Systems Limited in Dhaka, Bangladesh (e-mail: nahid_apee@yahoo.com).
Al Kawser is with the Network Operations Center of Inspire Systems Limited in Dhaka, Bangladesh (e-mail: kawser.telecom@gmail.com).
O. Farrok is with the Department of Electrical and Electronic Engineering, Ahsanullah University of Science and Technology (AUST) (e-mail: omarruet@gmail.com).

[PDF]

Cite:N. A. Shafi, Al Kawser, and O. Farrok, "Practical Field Overview Voice Quality of RTP Packet Size Analyze on Codec G729 Annexb = no in Low Bandwidth Area of Bangladesh," International Journal of Computer Theory and Engineering vol. 7, no. 1, pp. 62-65, 2015.


Copyright © 2008-2024. International Association of Computer Science and Information Technology. All rights reserved.