Need Advice on aplicability of Asterisk for this setup

[Nimroduk]

I've decided to assign myself the project of moving each of the 25 sites I develop/admin/support to VoIP for interbranch calls.

I am hoping to use to a centralised server to manage setting up the calls, with the actual voice data going Point-to-Point. Any POTS calls should continue as they do now so that there is no need for me to use a SIP Provider. Is this possible with Asterisk?

A little more info:

Each site
* Is independant of the others.
* Has, at minimum, a 1mb ADSL connection.
* Has a DNS reversable address (not all sites have fixed IP)
* Has (at least for now) 1 POTS line that *needs* to be both VoIP and POTS.
* There is no VPN between sites.

My initial thought is to buy a SPA-3102 ATA for each site and configure said ATAs to use the same Asterisk server (internet facing) for call registration. I need to be sure though that the voice data will not end up being relayed via the server but will go PtP; none of the sites have the upload bandwidth to cope with the level of calls.

I have had a read through the Asterisk site but I cannot see anything about using Asterisk as a server for remote, non-VPNed, clients.

Any and all advice welcomed

 

[CBasicAsslember]

Has, at minimum, a 1mb ADSL connection.

Is that minimum the download speed?

 

[Nimroduk]

Sorry, yes its 1mb down w/ upto 512kb up.

 

[CBasicAsslember]

Your limitation is going to be the lowest data rate. Keep in mind that the lowest rate is only an average. You also need to know if the provider supports QoS.

 

[Nimroduk]

The provider does not support QoS.

For an initial "trial" rollout I am quite happy to just provide a single VoIP line to each site; so a maximum of 1 call per DSL connection.

The area I am least clear on is whether the actual call data goes point to point or whether it is proxied via the server ?

i.e.

Site 1 - Asterisk Server
Site 2 - Single VoIP line
Site 3 - Single VoIP line

When Site2 calls Site 3 does the all the data traffic pass through Site1 or is the call setup between Site2 and Site3; with Site1 purely providing the initial call setup information ?
I am struggling to explain this

 

[Nimroduk]

Finally answered my own question

http://upload.wikimedia.org/wikipedia/en/9/9b/SIP_signaling.png

RTP is point to point

 

[altaphon]

Not necessarily. SIP is point to point *if* you set it up that way. If the endpoints can reinvite, they can establish a point to point connection (usually desirable). But you can program the server to stay in the path, and you must if you want to collect information on the call, monitor it, etc.