I'm sorry but it seems pretty far-fetched to try and blame SIP shortcomings on "Bell-heads". Since when have they (we) been that influential?
My point to Henry was just that this "problem" is not specific to PSTN gateways; so it wouldn't be appropriate to confine its solution to such a device. The need for media interaction prior to call completion (which as Martin notes can be 2-way) arises from the intermediated service model; which is as applicable to VoIP as to circuit switched networks. Tim > -----Original Message----- > From: Dean Willis [mailto:[EMAIL PROTECTED] > Sent: Thursday, August 21, 2008 2:54 PM > To: Dwight, Timothy M (Tim) > Cc: Henry Sinnreich; Dan Wing; [email protected]; [EMAIL PROTECTED] > Subject: Re: [Sip] Early media as an endpoint application > > > 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 > > > _______________________________________________ Sip mailing list https://www.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use [EMAIL PROTECTED] for questions on current sip Use [EMAIL PROTECTED] for new developments on the application of sip
