Influence of codecs on adaptive jitter buffer algorithm
Date
2007-04-27Author
Hirannaiah, Radhika M.
Jasti, Amarnath
Pendse, Ravi
Metadata
Show full item recordCitation
Hirannaiah, Radhika M., Jasti, Amarnath & Pendse, Ravi.(2007). Influence of codecs on adaptive jitter buffer algorithm . In Proceedings : 3rd Annual Symposium : Graduate Research and Scholarly Projects. Wichita, KS : Wichita State University, p.195-196
Abstract
Transmitting real-time audio or video applications
over the Internet is a challenge in the current networking
technology. The motivation for deploying real-time
applications includes the reduction in voice communication
overheads and the enhancement of services. The integration
of voice, video, and data encounters a variable amount of
jitter and delay. Typically packet loss ranges from 0% to 20%
and one-way delay from 5 to 500 milliseconds [1] [2].
Reducing jitter delay involves buffering of audio packets at
the receiver so that the packets arrive sequentially on time at
the destination. Adaptive jitter buffering at the receiver
improves the quality of voice connections on the Internet. In
this study, a simulation model was proposed to further
enhance the existing jitter buffer model to change the voice
codecs dynamically. Voice codecs were changed from higher
bit rate to lower bit rate during an established call session
based on jitter buffer value. The proposed model reduced the
packet loss thereby improving the call performance during
the on-going call session.
Description
Paper presented to the 3rd Annual Symposium on Graduate Research and Scholarly Projects (GRASP) held at the Hughes Metropolitan Complex, Wichita State University, April 27, 2007.
Research completed Department of Electrical and Computer Engineering, College of Engineering.