An approach for improving performance of aggregate voice-over-IP traffic

摘要

The emerging popularity and interest in Voice-over-IP (VoIP) has been accompaniedby customer concerns about voice quality over these networks. The lack of anappropriate real-time capable infrastructure in packet networks along with the threats ofdenial-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 areasonable protection for VoIP calls against packet losses. Traditionally, each voice callemploys 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 suitablelinear-time encoding for the aggregated voice traffic, considerable quality improvementcan be achieved with little redundancy. We show that it is possible to achieve ratescloser to channel capacity as more calls are combined with very small output loss rateseven in the presence of significant packet loss rates in the network. The advantages ofthe proposed scheme far exceed similar or other coding techniques applied to individualvoice calls.