Emerging voice over IP networks dictate a need for efficient network resource management schemes to mitigate the best-effort nature of IP. Traditional quality of service and call admission control mechanisms provide network resource allocation to static voice flows with parameters fixed by a selected voice codec. Frequently, the available bandwidth is not efficiently utilized, which results in a lower-than-possible average voice quality and potential wastage of resources. In this thesis, the author proposes a dynamic voice codec determination scheme that selects codec’s on per-call basis with respect to available resources. The proposed mechanism relies on continuous monitoring of a centrally managed IP network to determine the best voice codec selections for each pair of participating voice gateways. The determination process is modeled as a simple knapsack-like problem to take the fullest advantage of available resources, while maximizing the average voice quality. Overall, the new system is shown to achieve a better average voice quality and more efficient network resource utilization, when compared to traditional application centric QoS and all admission control solutions. / Thesis (M.S.)--Wichita State University, Electrical and Computer Engineering. / "May 2006." / Includes bibliographic references (leaves 60-63)
| Identifer | oai:union.ndltd.org:WICHITA/oai:soar.wichita.edu:10057/288 |
| Date | 05 1900 |
| Creators | Osipov, Andrey E. |
| Contributors | Pendse, Ravi |
| Source Sets | Wichita State University |
| Language | en_US |
| Detected Language | English |
| Type | Thesis |
| Format | 484497 bytes, x, 66 leaves : ill., digital, PDF file., application/pdf |
| Rights | Copyright Andrey E.Osipov, 2006. All rights reserved. |
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