[SIPForum-discussion] Diffrerence betwen SIP_T and SIP
WIGNELL, Clifford (Clifford)
cwignell at alcatel-lucent.com
Fri May 23 01:15:11 UTC 2008
Hello Javier and group,
I have not see a response on this so yet, so I thought I would have a go
at answering it...
Part 1, what are the differences between SIP, SIP-I and SIP-T.
SIP-I and SIP-T are variants or extensions of SIP, which carry PSTN/PLMN
signalling (i.e. ISUP) information in the body of the messages; SIP-I/T
provides for:
* Interworking of ISUP parameters to SIP headers to enable SIP
based routing/features in the IP Core
* Encapsulation of ISUP using MIME header to enable transparent
ISUP through an IP core
SIP-T/I specifies interworking between ISUP/BICC and SIP
The interworking specifications include:
* Architectural context
* Mappings for specific parameters
* Mechanism for encapsulating ISUP messages.
SIP-T is specified in three IETF RFC's 3372, RFC 3398, RFC 3578
SIP-I is the ITU specification for SIP interworking with International
ISUP (Q.761-Q.764) and BICC (Q.1902.1-Q.1902.4) in Q.1912.5 It
specifies actions for three profiles, A, B, and C
North American variations in ANSI T1.PP.679-2004
SIP-I (Q.1912.5) and SIP-T are generally consistent; except for subtle
differences in terms of options, level of detail, and explicit vs.
implicit requirements
SIP-I assumes a suitable "trust domain" exists between gateways
SIP-T assumes it is always necessary to use cryptography verify
identity, establish trust, and to secure communication.
Q.1912.5 has been developed in the ITU-T and therefore has benefited
from extensive input from numerous carriers.
SIP-I is more robust relative to mapping ISUP messages and parameters.
SIP-T has been developed in IETF, and therefore has benefited from a
careful analysis of security considerations and provides more detail in
the context of public IP networks where "trust domains" may not exist.
The following is a SIP-T message, as you can see it is a typical SIP
INVITE with a MIME block added, in this case the ISUP is contained with
in the bock marked -testing and -testing-; the SIP call flow has some
implicates as there are a number of mid call ISUP messages which may be
present, if you look at the RFC's you will see examples of these.
INVITE sip:3228880002 at 10.11.17.3 SIP/2.0
Via: SIP/2.0/UDP 10.11.30.4:5070;branch=z9hG4bK63773d8-15942
Max-Forwards: 70
From: sip:4771110001 at 10.11.30.4:5070;tag=56077021
To: <sip:4772220001 at 10.11.17.3;user=phone>
Call-ID: 9234 at 10.11.30.4
CSeq: 1 INVITE
P-Asserted-Identity: Spectra2 <sip:4772220001 at 10.11.17.3:5070>
Privacy: none
Contact: <sip:4771110001 at 10.11.30.4:5070>
Content-Type: multipart/mixed;boundary="testing"
Content-Length: 299
MIME-Version: 1.0
--testing
Content-Type: application/sdp
v=0
o=- 2890844526 2890844526 IN IP4 10.11.30.4
s=-
c=IN IP4 10.11.30.4
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
--testing
Content-Type: application/isup; base=ansi88; version=ansi
01 00 00 00 00 03 0f 16 0c d0 18 10 98 9e !,` c0 c6 e6 07 03 10 t'"00 10
07 03 10 t17 11 00 10 00
--testing-
Part 2, where do you use SIP-I/T; any time you wish to transit calls
between PSTN/PLMN networks (over IP) where you want to retain the ISUP
information, some ISUP parameters affect the way calls are handled and
billed for so this is important in many cases. The second reason you may
want to use SIP-I/T is in the case you want to signal to the PSTN/PLMN
the need for a particular feature such as Call Completion to Busy
Subscriber (camp on), this requires ISUP signalling to go to the
terminating switch.
I know that is a bit long but I hope it answers your question.
Best regards
Cliff Wignell
Alcatel-Lucent
Director: NSA, CBG, APAC
Phone: +61 3 9664 3847
Cell: +61 419 747 373
-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Javier Balbuena
Gamarra
Sent: Tuesday, 20 May 2008 8:31 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] Diffrerence betwen SIP_T and SIP
Hi,
Could you please explain me the diffrence between Sip-T, Sip-I and Sip.
when i use each one ?.
Thanks
Javier Balbuena Gamarra
Gerencia Planificacion de Red
Tlf : 210-2293, 995061214
E-mail : jbalbuena at tp.com.pe
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