Title :
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
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;
Conference_Titel :
TENCON 2005 2005 IEEE Region 10
Conference_Location :
Melbourne, Qld.
Print_ISBN :
0-7803-9311-2
Electronic_ISBN :
0-7803-9312-0
DOI :
10.1109/TENCON.2005.301158