I have used kamaillo with rtpproxy at certain sites of mine to handle media
anchoring where NAT or network routing issues were a huge problem.. the
kamaillo box acts as a mini-SBC, and also as failover routing for sites that
have cloud backup of certain endpoints.
my customers mostly wont allow having the phone system also be their internet
gateway or even have a static public IP attached to it.. (some converged
network models require PC's plugged into the phones.. and without special
considerations this breaks PCI DSS compliance if the PBX has a public IP on it..
while I enjoy astlinux being an all-in-one appliance. (I use several of them at
home.. we chose not to use astlinux for our product), i think there comes a
time when it can end up with too much on it.
when I get to the point of needing kamaillo or rtpproxy thats when I also am at
the point that I need an additional box on site to handle the advanced or
special situation.
christopher
From: Michael Knill <michael.kn...@ipcsolutions.com.au>
To: AstLinux Users Mailing List <astlinux-users@lists.sourceforge.net>
Sent: Monday, November 10, 2014 5:24 AM
Subject: [Astlinux-users] Kamailio use cases
Hi group
I am very interested in the inclusion of Kamailio into Astlinux.Just wondering
if anyone has any use cases to describe where they are currently using it.
I have a multisite customer (and more to come Im sure), all with their own
local Astlinux system and local trunk. This setup presents real problems
transferring calls as the media is not released (please tell me if this should
not be the case) when transferring between the systems. This is especially ugly
when an incoming call is answered by a central console and then transferred
back to an extension at the original site. Yes I realise that you can have a
console that is registered to both systems and it rings on different lines but
its not really seamless e.g. can’t do a Call Pickup etc from local phones when
this line is ringing. And even if you could, you would end up with the same
hairpin problem if you transferred it back.
My understanding is that Kamailio will solve this problem. Asterisk remains the
routing engine however all phones register to Kamailio and all RTP ends up
going between the endpoints. If this is the case, then there should never be
any hair pinning and only ever a single hop. Obviously you would need to remove
all NAT on the network through VPN.
Will this work as described? Is it a reasonable use case?
The Kamailio config does look pretty scary.
RegardsMichael Knill
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