hello Adam,
So basically the endpoints have to decode the
information out of the encapsulated information. Won't
this cause some extra processing?
For example on a multiple redirect call:
OCN - received by normal sip signaling. There is
no way to find out (other than looking at the
encapsulated information) if the call is directed
multiple times. So a terminating endpoint (vm
application) has to look at the sip headers as well as
the encapsulated information for *every* call.
Or the other solution is for the originating endpoint
to do some translation. You have to put logic on this
system to map RDN to OCN for multiple redirected
calls. The same applies for the redirect reason (not
translated now).
thank you
alex
--- Adam Roach <adam(_at_)dynamicsoft(_dot_)com> wrote:
These cannot be mapped. There are no equivalents in
the SIP protocol.
/a
-----Original Message-----
From: a c [mailto:lra101(_at_)yahoo(_dot_)com]
Sent: Wednesday, July 10, 2002 20:44
To: iesg(_at_)ietf(_dot_)org
Cc: ietf(_at_)ietf(_dot_)org
Subject: [Sipping] additional optional parameters
for
translation {ISUP
to SIP}
hello,
Could you please include the following the
following
parameters for translation as they are essential
for
voice mail:
- Redirecting Number {for multiple redirected
calls}
IE 00001011
- Redirection Information {reason for forward -
Original reason and redirect reason}- IE 00010011
- Redirection Counter - number of redirections
undergone - IE 00010011
thank you..alex
"There are also quite a few optional parameters
that
can appear in an
IAM message; Q.763 [15] lists 29 in all.
However,
each of these
parameters need not to be translated in order
to
achieve the goals of
SIP-ISUP mapping. As is stated above,
translation
allows SIP network
elements to understand the PSTN context of the
session if they are
not capable of deciphering any encapsulated
ISUP.
Parameters that
are only meaningful to the PSTN will be carried
through PSTN-SIP-
PSTN networks via encapsulation - translation
is
not necessary for
these parameters. Of the aforementioned 29
optional parameters, only
the following are immediately useful for
translation: the Calling
Camarillo, et al. Expires December 30, 2002
[Page 20]
Internet-Draft ISUP to SIP Mapping
July 2002
Party's Number (CIN, which is commonly
present),
Transit Network
Selection (TNS), Carrier Identification
Parameter
(CIP, present in
ANSI networks), Original Called Number (OCN),
and
the Generic Digits
(known in some variants as the Generic Address
Parameter (GAP)).
"
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Last Call: ISUP to SIP Mapping to Proposed
Standard
--------------------------------------------------------------
------------------
To: IETF-Announce: ;
SUBJECT: Last Call: ISUP to SIP Mapping to
Proposed
Standard
From: The IESG <iesg-secretary(_at_)ietf(_dot_)org>
Date: Mon, 08 Jul 2002 17:45:01 -0400
Cc: sipping(_at_)ietf(_dot_)org
Reply-to: iesg(_at_)ietf(_dot_)org
Sender: jhargest(_at_)cnri(_dot_)reston(_dot_)va(_dot_)us
--------------------------------------------------------------
------------------
The IESG has received a request from the Session
Initiation Proposal
Investigation Working Group to consider publishing
each of the
following internet drafts as as Proposed
Standards.
o 'ISUP to SIP Mapping'
<draft-ietf-sipping-isup-03.txt>
o Mapping of ISUP Overlap Signalling to the
Session
Initiation
Protocol
<draft-ietf-sipping-overlap-01.txt>
The IESG plans to make a decision in the next few
weeks, and
solicits final comments on this action. Please
send
any comments to
the iesg(_at_)ietf(_dot_)org or ietf(_at_)ietf(_dot_)org mailing lists
by
August 05, 2002.
Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sipping-isup-03.txt
http://www.ietf.org/internet-drafts/draft-ietf-sipping-overlap-01.txt
--------------------------------------------------------------
------------------
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