VoIP Mailing List Archives :: View topic - [asterisk-users] Question about SS7

Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community

[asterisk-users] Question about SS7

VoIP Mailing List Archives Forum Index -> Asterisk Users

View previous topic :: View next topic

Author

Message

michael_bulk at wildga...
Guest

Posted: Wed May 14, 2008 3:08 pm Post subject: [asterisk-users] Question about SS7

You can think of it like cable TV. You aren't required (at the moment)
to have cable to receive television signals, but if you want extra
features you have to go with cable (or something like it).

With a PSTN line, you can make calls, send caller-id, etc. but SS7
allows you to receive large amounts of data, allows you to grow your
network over 100 T1's, etc.

Asterisk doesn't natively do SS7 signalling, but I know there are ways
of doing it using third party drivers and soforth. Most importantly,
you need a way to connect to a SS7 network (e.g. connection at a
collocation).

This is an overly simplistic explanation, but there's lots more
information out there on the benefits.

Mik

mark morreny wrote:

Quote:

Hi,

I have read about SS7 recently and learnt that it is a signalling
protocol used in PSTN for call management, setup, etc. The thing that I
don't understand is how SS7 plays a role in VOIP. When I make calls
between landline and Asterisk via PSTN, I don't need to do anything with
SS7. Is it because the SS7 signalling is already done by Asterisk
already? From the prespective of implementing Asterisk, what kind of
SS7 support is needed? Is SS7 something needs to be concerned about
when using Asterisk with T1/E1?

I hope someone can help me to clearify these doubts that I am having.

Thanks,
Mark

------------------------------------------------------------------------

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

abalashov at evaristes...
Guest

Posted: Wed May 14, 2008 3:37 pm Post subject: [asterisk-users] Question about SS7

mark morreny wrote:

Quote:

That depends entirely on the application of Asterisk.

In general, as others have commented, SS7 does not play a role in VoIP
and does not concern you as an Asterisk user.

SS7 is an out-of-band signaling system is used inside telcos for
trunking between network elements -- mainly softswitches or traditional
big-iron TDM switches (such as the Lucent 5E, Nortel DMS, etc.) -- and
to provide certain database access features used by that type of
equipment and service provider, such as LNP (Local Number Portability
information), CNAM (directory listings), certain billing processes, etc.
SS7 operates on a carrier level and at global dimensions.

In other words, the phone company's switches talk to each other -- and
to other phone companies' switches -- using SS7. That is why it is
irrelevant to your PSTN experience as an end-user. You never see any SS7.

SS7 is _not_ used at the network edge, and is _not_ commonly used in
handoff to users, even ones who order very large amounts of access
circuits from a telco. These are typically done as PRI and ISDN PRI
signaling is used to control them; that is a completely different
animal than SS7 signaling. You can still aggregate PRIs into trunk
groups, couple T1s full of B channels into one logical signaling group
controlled by a single D channel using NFAS, etc. There is no need to
do SS7 outside of an inter-carrier environment. So, while I don't know
exactly what sort of service you are envisioning when you say "with
T1/E1," if it's available to you, it probably does not use SS7.

Where SS7 can become somewhat relevant is if you are a CLEC trying to
get away with using Asterisk in some switching capacity, and are wanting
to interconnect with a carrier in the full sense of the term (as in,
actual interconnection, not access circuits). That's typically done
over SS7. But Asterisk is completely the wrong tool for that job anyway.

In other words, for all practical intents and purposes, you don't need
SS7 and do not need to know about it.

--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

abalashov at evaristes...
Guest

Posted: Wed May 14, 2008 3:47 pm Post subject: [asterisk-users] Question about SS7

Alex Balashov wrote:

Quote:

SS7 is _not_ used at the network edge, and is _not_ commonly used in
handoff to users, even ones who order very large amounts of access
circuits from a telco.

Which is not to say that private SS7 links do not find their way into
such situations on occasion, for various specialised applications. But
if the question is, how is this brought to bear directly on VoIP... it's
not.

--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

michael_bulk at wildga...
Guest

Posted: Wed May 14, 2008 3:51 pm Post subject: [asterisk-users] Question about SS7

How VOIP relates to SS7 is fairly simple; you aren't going to connect to
an SS7 network with VOIP unless you have some kind of bridge between the
two.

As for what SS7 is able to send in terms of data, there is a huge list.
This includes data sent before a channel is aquired, data sent during,
as well as after a call is placed.

In all, what others have said is true; you only need to worry about SS7
if you're required to.

mark morreny wrote:

Quote:

Hi Mik

Could you elaborate on what you mean by " SS7 allows you to receive
large amounts of data, allows you to grow your
network over 100 T1's, etc."? Why is SS7 a compusory requirement for a
large network instead of "all" network? I have read quite a bit about
SS7 from Google and the part that is not discussed about is the
relationship of SS7 with VOIP. When is SS7 needed in VOIP and what kind
of benefits SS7 can deliver and how?

Thanks alot for your info.

Regards,
Mark

------------------------------------------------------------------------

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

jra at baylink.com
Guest

Posted: Wed May 14, 2008 3:53 pm Post subject: [asterisk-users] Question about SS7

On Wed, May 14, 2008 at 04:37:06PM -0400, Alex Balashov wrote:

Quote:

SS7 is an out-of-band signaling system is used inside telcos for
trunking between network elements -- mainly softswitches or traditional
big-iron TDM switches (such as the Lucent 5E, Nortel DMS, etc.) -- and
to provide certain database access features used by that type of
equipment and service provider, such as LNP (Local Number Portability
information), CNAM (directory listings), certain billing processes, etc.
SS7 operates on a carrier level and at global dimensions.

Listen when Alex talks. He clearly knows whereof.

Cheers,
-- jra
--
Jay R. Ashworth Baylink jra at baylink.com
Designer The Things I Think RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274

Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Joseph Stalin)

dinesh at alphaque.com
Guest

Posted: Thu May 15, 2008 5:07 am Post subject: [asterisk-users] Question about SS7

On Wed, 14 May 2008 17:06:54 -0400, Alexander Lopez wrote:

Quote:

SS7 helps carriers maximize the use of the circuits that interconnect
them with others. Instead of using a channel and having it open for 30
seconds as the call is setup, user gets signaling (busy, ringing, not in
service), and call is torn down. It can get the result in a split
second with out using any of my channels, all out of band and digital
rather than analog, (see 2600 signaling)

simplistically, ss7 is like sip which sets up the call, and the circuit
itself is the rtp streams which are then built when the call is connected.
likewise, you can have the sip exchange go through one path/route and rtp
through another.

--
Regards, /\_/\ "All dogs go to heaven."
dinesh at alphaque.com (0 0) http://www.openmalaysiablog.com/
+==========================----oOO--(_)--OOo----==========================+
| for a in past present future; do |
| for b in clients employers associates relatives neighbours pets; do |
| echo "The opinions here in no way reflect the opinions of my $a $b." |
| done; done |
+=========================================================================+

Display posts from previous:

	VoIP Mailing List Archives Forum Index -> Asterisk Users	All times are GMT - 5 Hours
Page 1 of 1

You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum