CTO Lab

Background: Analog Telephone Adapters (ATA’s) are hardware devices that are used to convert analog signals to VoIP, or VoIP to Analog. They are similar in theory to a standard Modem in that PSTN analog signals are simply converted to digital signals for use by digital equipment. ATA’s are used in the VoIP world to allow standard PSTN lines to interface with H.323 or SIP based VoIP equipment and switches (FXO)- or – to allow standard Analog phone sets to accept VoIP based phone services such as Vonage – or IP PBX connections and services.

ATA’s typically function in B2BUA environments in VoIP using SIP Registration mechanisms to IP Switch equipment, which will maintain a stateful relationship while providing VoIP services to Analog handsets, Fax Machines, or TDD devices. It is also theoretically possible to configure ATA devices to utilize VoIP Trunking capabilities – which do not require a state connection to an IP Switch device or Gateway. However, testing has confirmed that switching and routing intelligence is required within the ATA device to properly treat and handle calls to route messages.

Voxeo Labs has tested several ATA devices in order to assess the ability of an ATA device to serve as a PSTN to VoIP signal conversion tool in order to allow the SIP Based Voxeo Prophecy IVR platform to interface directly to the Public Switched Telephone Network (PSTN). Prophecy is a native SIP based VoIP platform, it is necessary to convert SIP (VoIP) to analog TDM signals in order to send calls to the traditional Telephone devices.

For our test cases and potential typical user scenarios, it is presumed that our users will wish to terminate POTS (Plain Old Telephone Service) lines directly into the Voxeo Prophecy Server, in order to enable Prophecy to Receive and Send phone calls.

For purposes of this discussion, it’s necessary to define several kinds of PSTN interfaces, FXO and FXS. ATA’s typically have several kinds of interfaces. They will always have an Ethernet Interface (or several). They will potentially have a console port for CLI operations, and they will have POTS line interfaces RJ11/RJ12 that are potentially FXS or FXO. The primary difference is that an FXO (Foreign Exchange Office) port will generate dial tone if an analog handset is attached. This can be facilitated by using a regular POTS line connected to the Telephone network, or by using a PSTN Simulator. An FXS port (Foreign Exchange Station) port does not produce dial tone when attached to an Analog handset. To generate a call from the POTS handset to a computer telephony system, you need to use an FXO connection. To generate a connection to the POTS handset, you need to configure an FXS connection. In our testing, because we were concerned with SIP Trunking using IP Authentication instead of SIP Registration to an IP Gateway using Username/PW – our scenarios required FXO connectivity which was facilitated using a PSTN simulator and home PSTN service via Analog lines.

Why does all this matter anyway?

Well.. It matters because we are trying to make computers listen to – and talk – to people.

The “Modern” Telecommunications Network uses Analog technology (PSTN) in order to interface with People. Computers are able to understand VoIP (SIP in our case) natively. The trick is how to code and decode these different languages so that they can be used for IVR and VoIP purposes. In the past, we have had to utilize very expensive specialized hardware and DSP’s (Digital Signal Processor) to perform the conversion. Dialogic, BrookTrout, NMS are leading manufacturers of voice board hardware to handle these tasks. However, this solution has greatly added to the cost and complexity of automated voice platform solutions. The move to attempt to convert PSTN signals to VoIP – Cheaply – has generated interest in IP Switching – and now – simple IP conversion via ATA’s and other devices – that code and decode PSTN Signals into VoIP messages that can be interpreted by native VoIP devices.

In the next few weeks I’ll be looking into several different kinds of ATA’s in order to assess the viability of using them for small (4-12 port) premise based IVR deployments.

Hopefully this information will allow you to make the leap from PSTN to VoIP a bit less complicated.

Cheers, Chris

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