Re: SIP-T and Q.Sig

[Dave Siegel <dave@siegelie.com>]

On Mon, Jun 14, 2004 at 02:42:55PM -0400, Daniel Golding reportedly typed:
> Thanks, Dave.
> 
> When connecting different vendor's softswitches together, what is the common
> signaling now? Q.sig? IAX2?

We only use a softswitch from one vendor on the internal network, which
is the SONUS PSX/GSX, and we have it configured for SIP (but of course
it also uses Diameter+, a SONUS proprietary signalling protocol).

We have a session border controller that sits between us and our VoIP
customers, which supports only SIP today but will support H.323 in the
future.

IAX, or Inter-Asterisk eXchange, is only supported on Asterisk IP-PBX's
to my knowledge.  I have played with it a bit on my home Asterisk box,
but it is largley a toy protocol at this point.  I do get a sense that
it might be the makings of an underground Internet VoIP cloud, however.
It'll be interesting to see where that one goes.

Dave

> 
> Or is so little of that going on, that there is no baseline?
> 
> Thanks,
> Dan
> 
> On 6/14/04 2:02 PM, "Dave Siegel" <dave@siegelie.com> wrote:
> 
> > I don't know of anyone actually using SIP-T yet, although many list it
> > on their supported features list.
> > 
> > My understanding of SIP-T is that it is nothing more than an SS7 message
> > encapsulated inside of a SIP message.
> > 
> > This could be particularly useful for VoIP gateways, as the Term Codes used
> > in the Voice world do not map well to SIP error messages.  For example,
> > there are three different term codes that map to a SIP 404 Not Found
> > error, but in the voice world, 2 of those codes should result in a
> > incompleted call, where as the 3rd would result in an alt route.  If
> > the downstream that generated the error message was a VoIP Bridge and it
> > encapsulated the original SS7 message, the originating switch might have
> > a better idea of how to handle the failure scenario.
> > 
> > Again, like I said, I'm not aware of anyone actually using it yet...I
> > wanted to keep quiet for a few days to see if anyone else spoke up first.
> > 
> > Dave
> > 
> > On Thu, Jun 10, 2004 at 02:09:04PM -0400, Daniel Golding reportedly typed:
> >> 
> >> Can anyone shed light on the use of these protocols for inter-call server
> >> signaling? I've heard SIP-T is become prevalent for this, but the RFC seems
> >> to indicate that it was originally intended for SS7 to VoIP interoperation,
> >> rather than voip switch to voip switch signaling.
> >> 
> >> Thanks,
> >> 
> >> -- 
> >> Daniel Golding
> >> Network and Telecommunications Strategies
> >> Burton Group
> >> 
> >> 
> >> 
> >> 
> >> 
> >> --
> >> To unsubscribe send a message to voip-peering-request@psg.com with
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> >> An archive is at <http://psg.com/lists/voip-peering/>.
> 
> -- 
> Daniel Golding
> Network and Telecommunications Strategies
> Burton Group
> 

-- 
Dave Siegel                     http://www.siegelie.com/people/dsiegel/
Oro Valley, AZ
 "Let us be thankful for the fools. But for them, the rest of
  us could not succeed."  -- Mark Twain


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