[Asterisk-Users] SER Interaction: Agents and Extensions

[Matthew Boehm]

> brett-asterisk at worldcall.net wrote:
>
> >
> > I think you might be missing the point here. SER is a raw SIP processor.
> > So for a second throw everything you know about Asterisk + SIP out the
> > window and go back to vanilla SIP. Getting used to a B2BUA in the call
> > path kinda beats some of the raw power of SIP up. Think of how a SIP URI
> > is formed. That domain portion is kinda like a context, right?
> > furthermore, SER can "do stuff" with that.
> >
> > I'm doing my own eval with SER for a very large deployment. But I'm just
> > getting started. I had SER running about a year ago, but it's been about
> > that long since I really toyed with it.
> >
> > One of the call flows I'm about to try is:
> > PSTN GW -> SER -> Asterisk "Transfer"/re-invite -> SER -> Phone
> >
> > The idea is that SER manages my PSTN gateway. I can always just stack
> > more Asterisk servers on, SER I'll never really need to expand (there is
> > a redundant SER Server, removing the need for clustering).  Then the
> > call gets "sent" to asterisk for smart call processing, however actual
> > setup of the media gets resent back to SER. I'm not sure if I'll be able
> > to do this, but  I may be able to do it with re-invites. Any thoughts?
> > -Brett
>
> SER newbie here.  Why do you need Asterisk for Sip->SIP setup?  And if
> there is a reinvite, is that for the RTP stream only or for the SIP
> transactions as well?  Will you lose the BYE transaction if there is a
> reinvite?
>
> Also, how many SIP registrations do you expect to maintain on each SER
box?

I was hoping to maintain somewhere around 1000 registrations per SER box.
I'm pretty sure Asterisk would get sluggish maintaining that many SIP
registrations.

My biggest concern is this: bandwidth. If we get a customer who has 20
stations (aka UAs) but only wants to be able to have 4 inc/outgoing PSTN
conversations, then we only need <G729 bandwidth> * 4 between our main
asterisk server and this customer. If any office-mates want to 4-digit dial
eachother, those conversations should not traverse the main bandwidth.

It saves my company time, money and effort because we could sign up 14 more
customers like the one above and still only need 1 T1 between the asterisk
box and the internet. But if all inter-office communications traversed the
asterisk box, we would need alot more bandwidth.

Our first solution to this was to put asterisk boxes at any customer
location that needed more than 10 UAs. But then we run into a billing
problem cause those 10 UAs would register to the local asterisk box and not
the main server and I would not get account code information from each UA
cause the main server's cdr's would show the call comming from same location
no matter which UA made the call. (can you say "run-on sentence"?)

If I can get re-invites working great, then I should have no worries about
inter-office communication. SER should be able to connect 2 office-mates to
eachother even if they are both behind the same NAT, or behind different
NATs.

-Matthe