On Thu, Aug 15, 2013 at 8:20 PM, Russell Bryant <[email protected]> wrote: > On 08/15/2013 07:02 PM, Paul Belanger wrote: >> On Tue, Aug 13, 2013 at 5:54 PM, Russell Bryant <[email protected]> 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: [email protected] | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger _______________________________________________ OpenStack-Infra mailing list [email protected] http://lists.openstack.org/cgi-bin/mailman/listinfo/openstack-infra
