[asterisk-users] any valid up-to-date info about Kamailio-Asterisk integration ?
Kirill Marchuk
62mkv at mail.ru
Thu Jan 29 02:43:02 CST 2015
Hi all
Have recently watched Matt Jordan's session on Kamailio World 2014
On slides 26-29 of his presentation
(http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf)
he speaks about a (completely new, for me at least) approach to build
scalable telephony systems, using N instances of Kamailio and N
instances of Asterisk
Are there any whitepapers, howtos, "implementation experience reports",
whatever, available, that would describe such an approach in details and
help some not-so-advanced admins to at least understand "if is it what
they need, or not exactly, or not at all" ?
We are planning to look closer at Kamailio (or any other proxy, like
OpenSip) as a way to do both load-balancing and failover solutions, so
that refusal of any Asterisk instance should have minimal possible
effect on the overall system availability.
A lot of questions howevere arise, like: what if one SIP user got
REGISTERed at Server 1, and the other on Server 3, so how can they call
one another ?
Also, outbound registrations can be done from one instance at a time,
say it's done from Server1 for Trunk1, so how can users, that got
authenticated at Server2, call thru that registration (Trunk1) ?
Also, Kamailio itself has to be protected from failing, and probably
even from overload...
Would be great to read something in-depth about that
Thanks!!
Kirill Marchuk
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