Performance Analysis of Adaptive Source Rate Control Algorithm (ASRC) for VoIP

Author

Usman, Muhammad ; Sheikh, Noor Muhammad

Author_Institution

Dept. Electr. Eng., Univ. Of Eng. & Technol., Lahore

fYear

2005

fDate

21-24 Nov. 2005

Firstpage

1

Lastpage

7

Abstract

The quality of VoIP is highly degraded by network dynamics like congestion of links, routing delays, packet loses etc. By changing the source-encoding rate adaptively with network dynamics, a much better end-to-end quality of service can be achieved. This paper discusses some already existing techniques for source rate control, highlighting their limitations and presents a recursive algorithm which changes the encoding rate of voice transmitting source to achieve optimum QoS for VoIP under randomly varying network conditions. The algorithm is tested for real time VoIP transmission and the results are compared with PCM Mu-law and G.728 fixed rate codecs.

Keywords

Internet telephony; quality of service; telecommunication congestion control; telecommunication network routing; VoIP quality; adaptive source rate control; link congestion; network dynamics; packet loses; performance analysis; quality of service; recursive algorithm; routing delays; source encoding rate; voice transmitting source; Adaptive control; Codecs; Degradation; Encoding; Performance analysis; Phase change materials; Programmable control; Quality of service; Routing; Testing;

fLanguage

English

Publisher

ieee

Conference_Titel

TENCON 2005 2005 IEEE Region 10

Conference_Location

Melbourne, Qld.

Print_ISBN

0-7803-9311-2

Electronic_ISBN

0-7803-9312-0

Type

conf

DOI

10.1109/TENCON.2005.301158

Filename

4085026