How It Works
IP telephony describes any solution that digitizes voice and transports it over an IP network. Because the VoIP connection is packet-switched, data and multiple voice streams can share a single IP connection.
A phone call requires more than voice transport. The call setup process must produce a dial tone, accept and process a phone number, generate a ring at the other end if appropriate, produce a ring or busy signal at your handset, and connect the call when the receiving party picks up.
The ITU (International Telecommunication Union) and the IETF (Internet Engineering Task Force) have established protocols for simulating call setup and takedown. Most IP telephony products use ITU H.323 or IETF SIP (Session Initiation Protocol). SIP comes from the data networking world and is somewhat lightweight; H.323 comes from the telecom world and is a heavy-duty collection of protocols for call setup and other functions. Each protocol has its supporters, but both standards work, and most vendors make their products interoperable.
After the initial call setup, the actual encoding and exchange of data occur. A software or hardware codecin the IP handset, the PC, or the serversamples the analog voice signal, converts it to a bitstream, compresses the data, and outputs the result at a certain defined rate. The data is then packetized, sent across the network, reassembled, decompressed, and converted back to an analog voice signal.
The ITU defines a number of standard codecs. Their data rates, levels of compression, and delay characteristics are different, but their names all start with G. They include G.711u (used in North America) and G.711a (used outside North America), G.729, and G.723.1. G.711u is used by the PSTN (Public Switched Telephone Network); G.723.1, G.726, and G.729 tend to be used more in the VoIP world.
Finally, a voice gateway connects the IP telephony network to any other type of network, including a PBX or the PSTN. There are almost always one or more gateways in a VoIP configuration.