An Overlay Architecture for High-Quality VoIP Streams

The cost savings and novel features associated with voice over IP (VoIP) are driving its adoption by service providers. Unfortunately, the Internet's best effort service model provides no quality of service guarantees. Because low latency and jitter are the key requirements for supporting high-quality interactive conversations, VoIP applications use UDP to transfer data, thereby subjecting themselves to quality degradations caused by packet loss and network failures. In this paper, we describe an architecture to improve the performance of such VoIP applications. Two protocols are used for localized packet loss recovery and rapid rerouting in the event of network failures. The protocols are deployed on the nodes of an application-level overlay network and require no changes to the underlying infrastructure. Experimental results indicate that the architecture and protocols can be combined to yield voice quality on par with the public switched telephone network

I. Introduction

Although subscribers are accustomed to the consistent voice quality and high reliability provided by the traditional Public Switched Telephone Network (PSTN), the promise of a single converged IP network to carry voice and data—and the cost savings therein—motivates the adoption of voice-over-IP (VoIP) technologies. However, customers expect VoIP to meet and exceed the standard of quality long offered by the Public Switched Telephone Network (PSTN). Providing IP solutions that meet the stringent requirements of high-quality and reliable voice communication poses a non-trivial challenge. Delays of 100–150 ms and above are detectable by humans and can impair the interactivity of conversations. By comparison, humans are far less tolerant of audio degradation than of video degradation. Hence, to meet these requirements it is crucial to minimize primarily the network latency and secondarily packet loss as much as possible. To minimize latency, contemporary VoIP solutions use UDP as the transport protocol. However this has the potential to expose VoIP packets to packet losses and equipment failures. Although the Internet can offer reasonable quality (relatively low loss and good stability) for the majority of VoIP streams, it has been shown [1] –[3] that it remains vulnerable to occasional bursts of high loss and link failures that preclude it from delivering the constant, high-quality service demanded for telephony.