Re: Comments on draft-camarillo-mmusic-sip-isup-bcp-00.txt

[[e-mail]]

I think this will work. In order to help you with the ISUP, I just checked Q763.
The SAM cannot contain any additional information except additional digit(s).
This means that it is not neccessary to send a SAM for each digit, but one SAM
can be sent with as many additional digits as are available, so

"The egress gateway computes the SAMs based on differentials. So, lets say it
receives the INVITE with 1555, then the INVITE with 155512. It then
sends two SAMs, the first with 1, the next with 2."

can be modified with

"It then sends one SAM, with both digits 1 and  2."

Steve Carr
Unisphere Solutions, Inc.





Jonathan Rosenberg <[e-mail]> on 11/18/99 10:57:31 AM

To:   Gonzalo Camarillo <[e-mail]>
cc:   [e-mail], [e-mail],
      [e-mail], [e-mail] (bcc:
      Steve Carr/Internet/ICN)
Subject:  Re: Comments on draft-camarillo-mmusic-sip-isup-bcp-00.txt




What if, rather than encapsulating subsequent digits into INFO, they are
added to the request URI in an INVITE? A local proxy for the gateway
will respond with 484 to all those that don't have enough digits until
it gets one that does have enough. So:






Ingress GW            P             Egress GW

INV tel:1 ------------>
CSeq: 1

INV tel:12 ----------->
CSeq: 2


<---------484 --------
          CSeq: 1

INV tel:121 --------->
CSeq: 3

<---------484 -------
          CSeq: 2

INV tel:1212 --------> ---------------------->   IAM 1212---------->
CSeq: 4

<--------------484----
               CSeq: 3

INV tel:12125---------> ----------------------> SAM 5 ------------->
CSeq: 5

INV tel:121255 -------> ----------------------> SAM 5 ------------->
CSeq: 6

INV tel:[phone] ------> -----------------------> SAM 5 ------------>
CSeq: 7

.......(same thing for digits 1, 2, 1)

INV tel:[phone]----> ---------------------> SAM 2 -------------->
CSeq: 11                                        <---------ACM-------

                         <------484-----------
                                CSeq: 5
                         <------484-----------
                                CSeq: 6
                               .......
                         <------484
                                CSeq: 10
                         <---183--------------
                                CSeq: 11


Ok, so let me explain. Proxies do routing just on longest prefix matches
for INVITE's. So, proxy P can be stateless actually, as it just forwards
INVITEs that match 1212. The ingress gateway sends a re-INVITE (possibly
without waiting for a response to the previous INVITE) every time a new
digit is received from the PSTN. Once the proxy has enough digits to
route, it won't send 484 anymore, and forwards it to the egress gateway.
The egress gateway is smart, and creates an IAM from the first INVITE.
Subsequent re-INVITEs cause SAMs to be sent, containing the difference
in the set of digits from the previous INVITE. Once the egress gateway
receives an ACM (address complete, right?), it responds with a 484 to
all the INVITEs but the last, and the final INVITE with a 183. Things
progress normally from there.

Implications:

1. proxies don't have to know anything about INFO or ISUP. They just
have to know longest prefix match routing for phone numbers. They can
even be stateless.

2. egress gateways that know ISUP effectively re-generate the SAMs

3. if the egress point is not an ISUP gateway, but a PC terminal or
something that happens to have a phone number as its name, it will have
to know enough to reject each INVITE until it gets one with a request
URI/To field that matches its own number. This, BTW, is normal behavior.
A UAS rejects an INVITE if the URI is for an address thats not its own.

4. This works *even* if there is misordering of re-INVITEs. The egress
gateway computes the SAMs based on differentials. So, lets say it
receives the INVITE with 1555, then the INVITE with 155512. It then
sends two SAMs, the first with 1, the next with 2. Lets say, then, that
the INVITE with 15551 comes. Based on CSeq, the egress gateway knows its
old and rejects it.

5. The ingress gateway doesn't depend on special messages or a
completion character, like #, to send an INVITE.




Its very possible I've completely missed something, as I'm hardly an
ISUP expert. Perhaps there is additional data in SAM such that it MUST
be sent across, rather than being regenerated. But, the above solution
seems nice, as it doesn't break anything, works with non-ISUP
terminating points, doesn't require users to dial a # at the end of the
digit string, etc.

Comments?

-Jonathan R.


Gonzalo Camarillo wrote:
>
> It is nicer to send complete URIs in the "to" header. If the i-gateway
> sends the INVITE before receiveing all the digits, the "to" header is
> not complete. It would contain just a few digits of the whole telephone
> number (this assumes that with just some digits the i-gateway figures
> out where to send the INVITE).
>
> Gonzalo
>
> [e-mail] wrote:
> >
> > >The use of a timer to force enbloc signaling is not acceptable. In the case
> > >of SIP bridging, subscribers originating calls from the PSTN will not
accept
> > >an artificial post-dial delay (even if it is just a few seconds long).
> > >Overlap signaling must be supported. It is essential that the overlap
> > >process takes place within the context of a single session - using the
> > >suggested INVITE/CANCEL/INVITE approach would result in multiple abortive
> > >calls being delivered to the terminating agent, which would not be
> > >acceptable.
> > >
> > >Why not send an INVITE message when the ISUP IAM is received and then send
> > >subsequent INFO messages for each ISUP SAM? I understand that "pure" SIP
> > >agents will expect the INVITE to contain the complete address of the called
> > >party. This could be handled by including support for overlap in the
> > >"Require" extensions (draft-roach-mmusic-sip-pstn-require-header-00.txt).
If
> > >the terminator did not support the PSTN extensions, the originator could
> > >then switch to enbloc and wait for the full address to be collected.
> >
> > There are a variety of problems here. The first is determining how
> > intervening proxies will react. If, for example, a proxy is making
> > decisions about where to forward messages based on a called party URI,
> > your suggestion would require it to understand ISUP (for interpretation
> > of SAM message). It also requires it to have number analysis code and
> > dialing plan information (potentially for the entire globe).
> >
> > The problem gets even stranger for redirect servers, which expect to
> > be able to send an immediate response to an INVITE -- not sit around
> > collecting digits.
> >
> > Basically, your suggestion forces overlapped dialing deep into each
> > and every SIP node in the network that is expected to receive calls
> > originating from the PSTN network.
> >
> > I agree that this is a problem that needs to be solved if we intend
> > to use SIP between two PSTN phones; but we need to find a better
> > solution than tunelling SAMs all over the place.
> >
> > We shouldn't overlook human factors: it could be that customers are
> > willing to, for example, be trained to press "#" at the end of a
> > phone number to skip the post-dial-delay.
> >
> > --
> > Adam Roach, Ericsson Inc. |  Ph: [phone] | 1010 E. Arapaho, MS L-04
> > [e-mail]   | Fax: [phone] | Richardson, TX 75081 USA
>
> --
> Gonzalo Camarillo         Phone :  [phone]
> Oy L M Ericsson Ab        Mobile:  [phone]
> Telecom R&D               Fax   :  [phone]
> FIN-02420 Jorvas          Email :  [e-mail]
> Finland

--
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