Open Stack

On Thu, Aug 15, 2013 at 8:20 PM, Russell Bryant <rbryant at redhat.com> wrote:
> On 08/15/2013 07:02 PM, Paul Belanger wrote:
>> On Tue, Aug 13, 2013 at 5:54 PM, Russell Bryant <rbryant at redhat.com> wrote:
>>> On 08/13/2013 05:35 PM, James E. Blair wrote:
>>>> Hi,
>>>>
>>>> On Friday, August 16th at 1700 UTC, we'd like to get a bunch of people
>>>> calling into the Asterisk conference bridge in order to help load test
>>>> it.  You can find connection instructions here (for SIP and PSTN):
>>>>
>>>>   https://wiki.openstack.org/wiki/Infrastructure/Conferencing
>>>>
>>>> We'll use conference number 6000.  Please also join #openstack-infra on
>>>> IRC so we can discuss the results and any problems out-of-band.
>>>
>>> Paul Belanger and/or I will set up something to generate extra calls at
>>> the server to create load.  We can pretty easily have calls come in that
>>> randomly inject a sound prompt here and there.
>>>
>> I have something setup for tomorrow if we needed it.
>>
>
> Cool.  I was going to work on something in the morning.
>
> What were you thinking of doing?  I was thinking of originating a bunch
> of calls via SIP with something like this on the call generating machine
> (untested):
>
> ; Say a random number (1-1000) at a random time between 1 and 60 seconds
> exten => foo,1,Answer()
>     same => n,While(1)
>     same => n,Wait(${RAND(1,60)})
>     same => n,SayNumber(${RAND(1,1000)})
>     same => n,EndWhile()
>
> Basically, we want to simulate some people talking, sometimes at the
> same time, but not often all at once.
>
> There's actually a higher load on the conference the more people are
> talking at the same time.  There's an optimization where it doesn't mix
> in a person's audio stream if they are currently silent.  If we wanted
> to load up the worst case (everyone talking over each other constantly),
> we could have all the streams play something constantly (like just
> comment out the Wait above).
>
This works.  I was planning on just originating a few channels at the
same time, playing some demo prompts.  Either way, it should be
straightforward to decide what to use.

I was planning on using the voip.ms number to start, then switching to
SIP if we had time / wanted a 2nd snapshot.

-- 
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger