The emerging popularity and interest in Voice-over-IP (VoIP) has been accompanied
by customer concerns about voice quality over these networks. The lack of an
appropriate real-time capable infrastructure in packet networks along with the threats of
denial-of service (DoS) attacks can deteriorate the service that these voice calls receive.
And these conditions contribute to the decline in call quality in VoIP applications;
therefore, error-correcting/concealing techniques remain the only alternative to provide a
reasonable protection for VoIP calls against packet losses. Traditionally, each voice call
employs its own end-to-end forward-error-correction (FEC) mechanisms. In this paper,
we show that when VoIP calls are aggregated over a provider's link, with a suitable
linear-time encoding for the aggregated voice traffic, considerable quality improvement
can be achieved with little redundancy. We show that it is possible to achieve rates
closer to channel capacity as more calls are combined with very small output loss rates
even in the presence of significant packet loss rates in the network. The advantages of
the proposed scheme far exceed similar or other coding techniques applied to individual
voice calls.
| Identifer | oai:union.ndltd.org:tamu.edu/oai:repository.tamu.edu:1969.1/4216 |
| Date | 30 October 2006 |
| Creators | Al-Najjar, Camelia |
| Contributors | Reddy, A. L. Narasimha |
| Publisher | Texas A&M University |
| Source Sets | Texas A and M University |
| Language | en_US |
| Detected Language | English |
| Type | Book, Thesis, Electronic Thesis, text |
| Format | 314200 bytes, electronic, application/pdf, born digital |
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