Due to the growing demand for VoIP (Voice over Internet Protocol) services, researches on VoIP design have attained more and more attention. Compared with a traditional voice service - PSTN (Public Switched Telephony Network), VoIP is able to provide lower cost and more flexibility. However, there are still many challenging issues to guarantee a consistent and good quality of voice connection over the best-effort Internet. In this work, we use a measurement-based approach to do quantitative evaluation of two most popular VoIP applications, i.e., MSN and Skype. In general, Skype performs better than MSN --- it shows that an up to 47% improvement in the overall throughput and an up to 50% improvement in the MOS. Such performance improvement for Skype is due to its (a) rate control mechanism, (b) voice codec, (c) error-resilience mechanism, and (d) better relaying mechanism to go through NAT servers or firewalls. We believe that this study can be of great use in designing a better voice service in current or next-generation heterogeneous networks.
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