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Using overlay networks to improve VoIP performance VoIP is widely used today, given its lower cost to service providers and end users. However,
 

Summary: Using overlay networks to improve VoIP performance
VoIP is widely used today, given its lower cost to service providers and end users. However,
because VoIP calls use UDP, they are affected by packet loss due to congestion and routing
protocol reconfigurations. These losses result in lower user-perceived call quality. Even
relatively small packet loss (i.e. ~1-2%) can degrade the quality of VoIP calls below what is
called PTSN level quality. Moreover, these packet losses can not be repaired by end-to-end
retransmissions because interactive VoIP calls have very tight delay budgets (e.g. 100-150
msec one-way delay, a large portion of which is consumed by packetization and compression).
To address the problem of congestive packet loss, we propose an approach that routes VoIP
calls through a network of application-level routers. These routers form an overlay network used
to forward the callers' traffic. The virtual links between overlay routers implement a limited form
of reliability, whereby the receiving router uses NACKs to indicate packet loss, while the
application router on the other side of the virtual link retransmits lost packets as long as they do
not exhaust their delay budget. We have shown that this limited hop-by-hop retransmission
scheme is able to provide PSTN-level quality even under packet loss conditions worse than
those reported on the Internet today.

We reduce the impact of packet loss due to link failures and routing convergence delay by
forwarding packets along two non-overlapping overlay paths. As long as the receiver correctly
receives packets from both paths the call can be played back at full quality. However, even

  

Source: Amir, Yair - Department of Computer Science, Johns Hopkins University

 

Collections: Computer Technologies and Information Sciences