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

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.