We recently received a reader's question asking why VoIP quality was different when going over the Internet vs. going over a corporate IP connection. So today, we'd like to review a few basics about how VoIP works and why VoIP quality is different when using different kinds of access and backbone networks.

First, some basics about how voice conversations get converted into a data packet for transport over a data network. In the simplest terms, speech is sampled and then "digitized" in much the same way music is converted to digital format for play on a compact disc.

Digitized voice has been around for decades, and is commonly used by phone companies even in the legacy public switched telephone network. Once digitized, the voice packet is placed into an IP data frame and is "ready to go" over any IP network - including private IP networks and the public Internet. VoIP uses routers that send the VoIP packet along to its destination.

At the IP level, VoIP uses user datagram protocol, a "send and pray" IP protocol because UDP doesn't introduce the same kind of delay that standard TCP generates. VoIP session control (a.k.a. signaling) commonly uses one of two standards including H.323 or Session Initiation Protocol to communicate with the phone network.

We'll continue our VoIP basics and further define the difference between using the Internet and an IP-VPN for VoIP next time, but for more detailed information readers can review a technology backgrounder we wrote back in 2004 by clicking here (PDF).

Steve Taylor is president of Distributed Networking Associates and publisher/editor-in-chief of Webtorials. Larry Hettick is a principal analyst at Current Analysis.