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 when only one of the two packets are received, using the redundancy provided by the PLC mechanism of the G.711 codec, the voice can still be recreated with some loss in quality. As soon as the loss of one of the routing paths is detected at the receiver, the overlay network computes an alternate path to route the call. Because the overlay network is small, we can use more aggresive routing reconfiguration mechanism which in turn result to considerably lower routing convergence times (compared to the current BGP protocol).
- Andreas Terzis, JHU.
- Claudiu Danilov, JHU.
- Yair Amir, JHU.
- Stuart Goose, Siemens.
- David Hedqvist, Siemens.
- Y. Amir, Claudiu Danilov, S. Goose, D. Hedqvist, A. Terzis. 1-800-OVERLAYS: Using Overlay Networks to Improve VoIP Quality. Appeared in NOSSDAV 2005. PDF
- Y. Amir, C. Danilov, S. Goose, D. Hedqvist, A. Terzis. An Overlay Architecture for High Quality VoIP Streams Appeared in the IEEE Transactions on Multimedia. Volume 8, Issue 6, Dec 2006. PDF