1 for more information on how T1s and multiplexing in telephony works. When large companies started installing data networks, they discovered that main- taining (and paying for) a data network and a phone network was prohibitively expen- sive. Someone had the idea that a phone call could be digitized, packetized, and sent over a data network using IP. If they could converge two networks into one, the com- panies could significantly lower recurring costs. Yeah, the technology is cool and all, but the execs made VoIP mainstream because it allowed them to save money. There is a problem, though, and that is call quality. Without getting too far into a very deep topic, suffice it to say that people were used to flawless quality when they picked up a phone and made a call. You may not realize it now, but phones used to sound good. Cell phones have significantly lowered our expectations for call quality. In the days before cell phones and VoIP, call quality was remarkably good, a fact upon which old-school telephony guys will happily ex- pound when VoIP users complain. To even approach the call quality of a traditional analog telephone, a digitized, packe- tized voice call must be sent quickly and reliably. Additionally, the packets must arrive in the order in which they were sent. Unfortunately, those requirements aren't all easily met with IP. For VoIP call packets to be sent quickly, UDP is used for speed, but UDP does not include any error checking, nor does it guarantee that packets arrive in order. To overcome these shortcomings, QoS is used to make sure that VoIP call packets are sent before any other packets and, once sent, are not dropped. To make things even more interesting, there must be a stream in each direction, since voice calls are full duplex. While Real-Time Protocol (RTP) does include timestamps and sequence numbers SO that packet loss and packet order problems can be detected, it does not work to resolve these issues if they are discovered. Communication between two endpoints is set up via call cont