On Aug 21, 2008, at 10:37 AM, Dwight, Timothy M (Tim) wrote:
What if you want to implement those or similar services between VoIP
end
systems? For example what if you want to provide "color ringback
tones"
between VoIP end systems? There is no PSTN gateway involvement in
such
services.
The problem with early media is that it assumes that there is a need
to transfer media BEFORE a session is "established". We inherit this
assumption from the PSTN, because the PSTN billing model allows one-
way media to flow from destination to origin before the call is
"established". It works only because the stateful-everything prevents
multiple early media sessions. With SIP, and forking, we get chaos.
So the "right answer" is to have two separate but fully negotiated SIP
sessions -- one for the early media, one for the regular media. The
second session, should it occur, replaces the first (and we have a
signaling mechanism to do just that). This eliminates the whole (or at
least most of) "early media" problem. But it confuses the heck out of
bellheads who have decided to start billing based on the 200 OK, but
also want to slavishly imitate the billing sequence of PSTN. It's
stupid, broken, and can be manipulated to get vast amounts of "free
long distance" out of the system, but it is deeply entrenched.
More on this: In the model I'm describing, a PSTN gateway handling a
call from SIP to PSTN would send an initial 200/sendonly on receipt of
the ACM (or equivalent). This would give us a SIP session for the
"early media" phase. Then the gateway could either reinvite or refer/
replaces on the ANS.
The problem with this is that, if you're billing on the INVITE/200
transaction, a naive system starts billing on the first INVITE/200
(the early media). So one needs a smarter billing system that 1)
knows that the call is to a gateway, so it should start billing when
the gateway initiates the bidirectional session, and 2) knows to bill
the user, rather than the gateway. This could be helped by a SIP
exetension of some sort that identifies the early 200 OK as "early"
Now this isn't perfect, as if one forks a call, and one fork goes to a
PSTN gateway, that's the fork you're going to get connected to every
time, as it sends back the quickest 200 OK. But hey, if that's where
you're getting media from, that's where you should be listening,
right? Forking is broken. If it hurts, don't do it. Forking to a PSTN
gateway really requires a B2BUA that understands the "early media
session" model proposed above, and is smart enough to drop the early-
media session if it gets a non-early 200 OK from another leg.
On a pure VoIP system, localized ringback is probably better handled
with extensions to "alert-info".
--
Dean
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