Cabling

Voip Cabbling uses the same structured wiring used for data category 5 (CAT 5E) or category 6 (CAT 6) and modular sockets. The tipical standard define how to lay the cabling in a star formation, such that all outlets terminate at a central patch pannel or a data switch which creates the Local Area Network (LAN).
Eah IP Phone is a network member which receives addressing from the same sources as any computer in the network such as DHCP.
By using the same protocol (IP) as the rest of the members ot the network we have completed the idea that voice gets converted into IP Packets and transported the same way the rest of the packets in the networks.

PBX

The PBX is a server which runs a software like Asterisk which has all the rules for behavior of the telephony system such as Extensions, dial plans, outbound routes, Voicemail, etc. this Network member receive the packets generated by the endpoints or extensions called IP phones.
The packets originated on a local PBX deployment travel within the same Local Area Network which usually has plenty of bandwidth and most of the times very few motives to loose packets or getting affected by Jitter.
This means that conversations between extensions or IP phones withint the same LAN will reseemble perfect audio quality.

CODECS

When voice is modulated into packets or converted back to analog signal the process is know as Coding/Decoding from which the contraction CODEC was created.
There are different technologies and algorithms for this process and each one as a flavor name, the most used is know as Phase Code Modulation U-law or PCMU also know as g711ulaw.
Analog voice by default is assigned 64Kbps of bandwidth, when we apply the CODEC algorithm which for PCMU is uncompressed this gets added header and other protocol requirements growing to some 94kbps per voice channel. Passing this size of traffic ofver a best-effor or public internet network will almost waranty packet loss or jitter along the hops from provider to provider prior arriving to destination This is what affects the audio quality on Internet calls, however g711 is not the right codec to send packets over public internet and some great advantages can be found on compression through algorithms like g729 which brings the bandwidth from 94kbps down to 13kbps per channel, or g723 which uses even less bandwidth 9kbps, so technology always presents the right choice to deal with different type of events which have to be taken in count during the design of VoIP deployments.