! Sure, Internet-based VoIP is cheap, but one of the things you give up is quality. Telephony circuits deliver a certain amount of quality that is simply not available when you use Internet-based VoIP. We can approach that level of quality when using private networks and QoS, but the moment your packet leaves your network and enters the great Internet cloud, it is at the mercy of (potentially) many providers, routers, and links, all of which may be seriously oversubscribed. Loss of far less than 1% of packets will cause significantly degraded audio quality on a VoIP call. Packet loss is usually a symptom of oversubscribed links, improperly con- figured (or missing) QoS, or bad hardware. Packet loss is usually described by users as choppy calls, though if the loss is significant, calls might be dropped altogether. Packet loss can also cause mechanical-sounding audio or other user complaints. Jitter Jitter has to do with timely delivery of packets. If your packets are being delivered in order, but the first five are bunched together while the next five are separated by huge FTP packets (or worse-jumbo frames), the receiving end will have trouble decoding the signal into recognizable audio. The technical term for this problem is packet-to- packet delay variation. Since this can be a microsecond issue, most modern VoIP devices incorporate some- thing called a jitter buffer to lessen its effects. Jitter buffers hold the incoming packets for a short time until they can be processed properly. This removes the delivery delay problem, but introduces new latency, since the packets sit in a buffer for a short time before being decoded. If jitter is extreme, the jitter buffer can usually be enlarged, but this often results in latency that exceeds the average user's annoyance threshold. Jitter is the gremlin of the VoIP world. If you've got jitter, it can be tough to find the culprit, though improperly configured (or missing) QoS is a common cause. How VolP Works Small-Office VolP Example In a Cisco VoIP environment, there wi