[VoIP] Early audio aka non-sup call progress info

[John R. Covert]

Martin Harriss wrote:
>Customer dials into IVR, interacts with same, and if that's as far as
>the interaction goes, the call doesn't supervise and thus no money
>changes hands.  If the customer ends up being transferred to a live
>person, the call supervises when the live person answers.

This can only work reliably on services where the CALLED party
determines the carrier (e.g. 800 Service).  Forward audio is never
carried on sent-paid calls via AT&T, for example, as well as most
of the other big carriers.  There can be explicit exceptions by number,
but they are rare.

>it has the potential for causing all kinds of problems in the VOIP world

Yes, American Airlines does do this on their 800 number.  Supervision
happens well before an agent answers; it takes place as soon as the
first set of menus are traversed.  Only DTMF is passed forward.

-- Executing Dial("SIP/x28-0ace2a00", "SIP/18004337300 at voip-itp|120") in new stack
-- Called 18004337300 at voip-itp
-- SIP/voip-itp-0acff000 is making progress passing it to SIP/x28-0ace2a00

That's when the early audio path is opened up.  The recording is playing,
and I can press digits through a two level menu.  Then I'm switched to
the application desired:

-- SIP/voip-itp-0acff000 is making progress passing it to SIP/x28-0ace2a00
-- SIP/voip-itp-0acff000 answered SIP/x28-0ace2a00

And the music on hold begins.

A correctly functioning VoIP setup will handle this correctly and
process early audio outbound and DTMF inbound.  Early audio does tend
to be a problem for some providers, but any call progress information
received over the network should/must be passed back to the caller, or,
for example, intercept referrals will be lost.

/john