Redial: Interactive Telephony : Week 1

PSTN: Public Switched Telephone Network

The worlds "circuit" switched phone network. Originally dedicated circuits were created between the caller and the receiver and managed by an operator. The audio was transmitted as analog.

Now the only part of the PSTN system (generally) that is still analog is the connection to the home or business from the telephone network. The bulk of the audio transport is digital but still circuit switched (as opposed to using IP routing on the internet).

More Information:
Telephone System Test Board
Wikipedia: PSTN
Telephone Tribute

VoIP: Voice over IP

VoIP can stand for anything that carries voice over an IP network (such as the internet). Recently bandwidth and latency have gotten to the point where the internet can reliably be used to transmit voice.

There are many types of VoIP but we are going to concentrate on the most standard protocol, SIP (Session Initiation Protocol). SIP is similar to SMTP (Mail Transport Protocol) in that a message is sent from one party to another. In VoIP applications, the SIP message will normally define what ports, codecs and transport mechanisms are to be used to send the audio.

Asterisk: Bridging PSTN with VoIP

Asterisk is a PBX (a Private Branch Exchange). Commonly used in businesses, a PBX is something that is generally connected into the PSTN directly to a telephone company. PBX's can manage internal calls (from one handset to another within an organization) as well as manage incoming and outgoing lines.

Asterisk is the first fully functional PBX with both PSTN and VoIP capabilities (that I am aware of). Asterisk is also highly configurableable and can be scripted with a variety of languages.

Our Asterisk install doesn't hook directly into a phone network, rather we use VoIP and service from a TSP (telephony service provider) to bridge between the internet and the regular phone system.

Redial Asterisk Information