[Asterisk-Users] help / advice needed on a project

[Patrick Zwahlen]

Hi all,
 
I need some help on a potential project. I am an open-source advocate,
and have been working with Linux for a very long time. I am not a VoIP
expert, even though I have been installing a few C*sco call managers. I
have been doing some development in the past, but I am now better at
"gluing" things together, than at doing hardcore devel ;-)

Project summary:
================
I work for a public administration. Our telephony system is 100% VoIP
based (several C*sco call managers, and a few thousand SCCP phones).
SCCP/H323 are used in a flat, dedicated network.

Our project is to open our VoIP infrastructure and to provide basic
services to smaller administrations. There is no need for billing
(service will be free), and no need for advanced services (no
conferencing, for instance). Moreover, each "client" will keep its local
PSTN connection. We will only handle the following cases:

- Our network -> "client's" network (originating from our network)
- "client's" network -> our network (termination in our network)
- "client's" network -> our network -> other "client's" network
(transit)

Following are some other points:
- We have security concerns around these interconnections.
- Smaller administrations may be using SIP or H323.
- They will have up to 10 simultaneous calls
- We need to keep dialplans as simple as possible.

Initial thoughts:
=================
Our idea is to create a second flat network (as a transit network), and
to use border or edge devices. We would interconnect that new network to
our existing network. "Clients" would be connected to that new network
via dedicated devices. These devices would be used for both signalling
and RTP streams.

The recommandations we've had so far were to look at solutions like
Kagoor (http://www.kagoor.com/), and their session border controllers.
Useless to say that the price is far too high (this is targetted at
ISPs, whereas we won't be making any money with these interconnections).

I then brought the idea of creating a POC/pilot using Asterisk. I am
also playing with Soekris boards (http://www.soekris.com/net4801.htm)
during my spare time, and I have recently found the AstLinux
distribution
(http://www.kriscompanies.com/modules.php?name=Content&pa=showpage&pid=3
). I have the feeling that these two would make good and reliable edge
devices.

Using asterisk at the border of the transit network, we could have
IAX-only trunks, that would probably help around NATting...

Questions:
==========
I can be considered as an asterisk newbie, so I obviously need your
recommandations before going further:

- Do we have a chance to solve this problem with Asterisk ?
- Does the described solution make some sense ?
- Do we need any advanced functions (gatekeeper, ...) ?
- Are there ressources/examples of such designs/configurations available
? (voip-info.org is just HUGE...)
- Can an edge device have 2 IAX trunks in failover to our existing
infrastructure ?
- Can a box with 128MB of RAM and a 266MHz CPU (the Soekris) handle 10
simultaneous calls ?
- Probably many other questions to come... ;-)

As you can see, I need your input in order to take a go/no-go decision.

Thanks in advance for your much appreciated help. - Patrick -