[asterisk-biz] Dedicated Virtual Asterisk Boxes

[trixter aka Bret McDanel]

On Sun, 2006-02-05 at 12:12 -0800, Dovid Bender wrote:
> Hello List,
> I am in talks now with a Data center that currently
> offers VPS services. I tried to install asterisk on a
> box and it didnt work. It is because you have to have
> the zapata files compiled on the VPS host. If this is
> done than asterisk will work fine on the VPS. The way
> thier service currently works is as follows:
> You are given an IP address for your virtual machine.
> You are also given a web portal from which you can set
> up/configure and control the virtual machine. You can

sounds like unixhost.com.  A friend got that and it was not stable in
terms of bandwidth or cpu, so if it is them (or someone that has the
same capacity issues) you may find yourself having weird problems that
are hard to repeat.

Generally no you dont need all the zapata stuff, unless you need a
timing source or want hardware support.  Sounds like hardware is out of
the question so it boils down to music on hold nad meetme conferences
which both are popular and require a timing source (app_conference
doesnt require a timing source though).

If the build failed, perhaps more information about that, would be
helpful, and perhaps to the asterisk-users list as that type of problem
is better suited there than here.

As for the service he is planning on rolling out, some people have had
success using things like Xen to load a system running asterisk others
havent, it boils down to system capacity.  It may beta test well with
one user, but toss a bunch of these systems on one box and load test
each system and see.

It is also somewhat difficult to provision virtual systems like that,
some people may say they need only 10 channels and in reality use 2,
others may say 10 and use 50.  As such he will need to work on some
monitoring processes to ensure that boxes dont become overloaded.  Even
knowing which channels do what may not be enough, codec selection can
dramatically impact cpu load.  

I think however a bigger problem may be timing for VoIP, where there is
just enough drift to cause problems.  Figure on average for every
channel asterisk sends a packet every 20ms.  Without a jitter buffer
(which sip doesnt have one) there can be a bit of swing in that number
and cause poor quality audio.

I am not saying dont do it, I am just saying what I think some of the
major problems will be so they can be thought about and a solution
figured out before customers complain.

-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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