6 Nov
2016
6 Nov
'16
3:29 p.m.
Hi,
I'm just starting with Kamailio, and trying to validate a possible solution
design.
I started offering hosted PBXs (FreePBX or Elastix as customer wish), each
one with a public IP, connected with SIP trunks from Net2Phone provider. RTP
from/to customers calls is going in and out through PBX, and not directly
customer <----> sip trunk provider. Billing is provided by SIP trunk
provider, so with just CDR from freepbx/Elastix is enough.
But, I would like to introduce 2 main changes. First a sip router/proxy in
front of PBXs like kamailio, and doing just a basic domain based routing to
each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs
to internal IPs. Second, to know if would be possible to redirect RTP
from/to customer-net2phone provider directly, not going through PBXs.
Net2phone has public IPs for RTP, customer normally in their homes behind
NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using
Asterisk 12 (could be 11 or 13 too tbe o reach this).
I would like to kamilio be as simple and static as possible, not handling
REGISTER or LOCATION. Just to route all traffic based on each domain to each
corresponding PBX. My idea will be to pre-provide all subdomains to each
internal IP using subdomains like "customer0001.domain.com,
customer0002.domain.com." and customers would login as
100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> ,
101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001
and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> ,
101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002
for example. Just using as it was a firewall doing NAT and Asterisk
(freepbx/elastix) behinds handles everything as they are doing already.
Maybe kamailio could check/filter some malicious packets like I'm doing now
with fail2ban on PBXs.
Can be this possible?
In that case where do I need to setup SIP trunk? On kamailio or on internal
PBXs as I have it now?
For second part, just to know and understand if would be possible customer
"talks" RTP directly to Net2phone SIP trunk provider on doing or receiving
external calls. Not going to my DC to save BW.
How I could do that?
Thanks a lot in advance!
Regards,
elkato