Hi

New to Kamailio and FreeSwitch, loosely familiar with SIP mechanics, and not a 
complete network idiot... but please be gentle. :)

We're trying to get Kamailio set up in front of a FreeSwitch-based SIP 
application server, to do some simple policy controls and, using one of the RTP 
proxy modules, to help handle the media side of things. Our SIP trunks come 
from our provider on a VLAN addressed as 10.x, and they provide an upstream 
proxy/media server (10.y address). The link also carries Internet traffic so 
we're running it through a Cisco ASA which breaks out the VLANs and static NATs 
the SIP stuff to our DMZ (10.z address). This is where we want to put the 
Kamailio box, where it should receive the calls and route them back through the 
ASA to the internal address (10.w range - all 10.w/x/y/z are separate, 
non-overlapping networks).

Outbound calls from the application server (there are very few) should pass 
back up to Kamailio which will validate the numbers against an approved list 
(the app should only dial a subset of numbers) and pass them to the upstream 
proxy if necessary. We need to ensure that inbound calls from outside trombone 
through our application server so that the caller doesn't get billed for any 
calls, but this should be the default I think as that side of things is handled 
by FS (the app stands up a new call and bridges it to the incoming). This 
topology should allow us to use Kamailio to hide details of the FS app from the 
outside world, and make life easier for the app developer who should just 
send/receive to the Kamailio box with no further thought or complexity.

If we put the FreeSwitch box in the DMZ where we want to put Kamailio, and then 
turn on SIP packet inspection in the ASA, calls flow but quality is poor for 
some callers. If we turn off SIP packet inspection, we get no audio - I can't 
find any clear Cisco documentation for this but I think the SIP inspection 
stuff in the ASA seems to handle RTP NAT fixups and the like too. But with no 
docs it may as well be magic, and I want to remove it if possible. With 
Kamailio in the DMZ and FS internal, we loosely followed 
http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc and hacked 
out the voicemail and Lua stuff, but couldn't get calls to flow, whether or not 
we used the SIP inspection on the ASA. We were confused why this example isn't 
define as WITH_NAT and doesn't use a local RTP proxy. Surely that would be 
needed?

Questions:

1.    Should the proposed topology, with Kamailio + an RTP proxy behind a 
firewall, relaying to FS on an inside interface, work? (Can't see why not)

2.    Does it need a local RTP proxy on the Kamailio box, particularly if we 
turn off the ASA SIP inspect stuff?

3.    Can you recommend which RTP proxy to use? There seem to be at least 3 
that work with Kamailio. The box is CentOS 6.5, and it would be nice to use 
known-to-work packages rather than compile from source. (But eh, if I haveta).

4.    Can anyone point me to some docs to explain what ports need to be open 
between the Kamailio box and my upstream proxy/media server? I can be more 
liberal between inside and DMZ I guess.

5.    Is static NAT in this environment going to bite me, or should it be OK?

6.    Is there any better documentation that we should be using to make this 
easier, or should I just man up and try harder?

TIA
Sean
Volunteer for our Red Puppy street appeal to help puppies
like Gordy become guide dogs.
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