So far SIP-T occurred sporadically on this mailing list. I simply try
identify the relevance of it in this context do not take personally mu
comments.
On Feb 19, 2009, at 10:39 PM, Alex Balashov wrote:
The problem is that outside of the VoIP cottage industry, this stuff
isn't "legacy" by any stretch of the imagination, in any way, shape,
or form. We're just used to fancifully imagining that it is.
Adrian Georgescu wrote:
Hm,
It is very hard to judge the benefits of performing all the nice to
have feature at a higher level protocol while still having to
support legacy expensive infrastructure underneath.
Now, last time I heard about SIP-T was by an ECMA standard a few
years ago. ECMA is a sort of inverse pyramid European standards
body that nobody listens to. Basically, they are sponsored by
vendors to endorse 'standards' because they posses an EU stamp. The
word here in Europe goes that if something went to the extent of
geting an ECMA official endorsement, one knows that it is a
standard with no future and no company invests in it anymore.
Maybe I am wrong and this has much more sense in the US.
Adrian
On Feb 19, 2009, at 8:43 PM, Alex Balashov wrote:
To expand on this just a little bit:
While here in the VoIP cottage industry we mostly deal with SIP to
begin with, in that we use ISDN gateways for connecting to
carriers, get SIP trunking from our carriers/ITSPs, and so on, the
reality is that most stuff in the PSTN carrier space is still done
with big-iron TDM equipment, at least here in the US. If you want
to be a competitive carrier, you *must* interconnect with the
incumbent telco using SS7; no ands, buts, ors.
That doesn't mean there aren't a lot of opportunities to deploy
SIP internally inside the service delivery core. The main benefit
SIP provides there is that it is so high-level and easy to
manipulate. As a result, a lot of mediation, logging, billing,
analysis, translation, LCR can be done easily and inexpensively.
Before SIP and H.323 came along, doing this kind of stuff required
a box that did all that and spoke SS7 or, at the very least ISDN Q.
931, and that is much more expensive, inflexible, and difficult to
manipulate.
Promoting this traffic to a higher-level protocol stack that has
more applications and tools to deal with it allows the development
of solutions for doing sophisticated telco-world stuff using
commodity hardware and open methodologies, open-source style.
That has triggered a wave of new products and paradigms in the
telco space in a way that is analogous to how Asterisk et al have
revolutionised the PBX space.
One example of this is TransNexus' NexOSS/NexSRS product (www.transnexus.com
<http://www.transnexus.com>). They use the OSP (Open Settlement
Protocol) module for OpenSER and/or for Asterisk (depending on
whether a B2BUA is required) internally inside their product to
perform a lot of neat AAA and routing functions (e.g. the NexSRS
route server). Their ability to do this benefits precisely from
the fact that the traffic can be moved onto a higher-level
protocol plane and away from proprietary, expensive, closed and
inflexible stuff that has been a defining feature of the telco
world. If you can turn the traffic into SIP or H.323, they can
deal with it, but if it's SS7 or PRI, they can't. The world is
going more "soft[ware]."
At the same time, the telco space is not a SIP world right now;
the network edges are still SS7, and the market really hasn't
settled on a good private SIP interconnection/peering strategy and
implementation for intercarrier settlement. So, for the most part
SIP trunking is used for customer access only. The SS7
information must be conserved in this type of setup, and that's
one of the reasons the sort of thing that SIP-T is exists.
Alex Balashov wrote:
Adrian Georgescu wrote:
Why should SIP-T still exist? Is it cheaper than having a
gateway? What is the practical use case for investing in such
technology?
I am eager to learn
We've used it extensively in work with CLECs that operate TDM
switches such as the Metaswitch, Lucent LCS/Telica, etc.
When a carrier operates more than one switch, SS7 interconnection
between them is generally required so, for the same basic reasons
an internal iBGP mesh or partial mesh (confederation) between two
border routers is required for IP. One switch must be aware of
numbers routed or ported into the other switch, and so on.
The reason for its existence is that if both network elements
support SIP-T, it allows you to replace an SS7 IMT (inter-machine
trunk) with an IP-based mechanism for this interconnection. This
allows you to move the traffic over a data network and get all
the benefits that this brings; economies of scale through
decreased facilities, oversubscription, etc. The main benefit is
the elimination of TDM trunk exhaust; SS7 IMTs are physically
bundles (trunk groups/TCICs) of DS0s, usually consisting of one
or more T1s, and sometimes DS3s or more. That means that when a
large volume of calls is running between the two switches, you
could burn up all your SS7 trunks. Running the calls as SIP-T
allows you to use something like a gigabit network core to make
that problem go away somewhat -- a key benefit of VoIP in most
other scenarios with which you are familiar with.
At the same time, the switches still need ISUP attributes carried
in SS7 IAMs and ACMs for billing, because that's just the
information they operate on internally. SIP-T provides an IP-
based way to encapsulate that information.
SIGTRAN (essentially, SS7-over-IP) is another way to do this.
However, SIP-T is lightweight and easier to deploy. It also
allows you to use existing SIP network elements (proxies, session
border controllers, etc.) to route and manage the traffic. For
example, if you were using OpenSIPS + ACC + FreeRADIUS as a CDR
catcher, you could run the "SS7" calls between two switches and
log the appropriate information as custom attributes. There are
no good open-source implementations for SIGTRAN - nothing as turn-
key as Kamailio or OpenSIPS. SIP is high-level and much easier
to deal with and manipulate using a far wider range of tools.
SIP-T is also becoming an attractive external interconnect option.
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
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