If the MOH thing is really true, a more "realistic" test would be to run playback(demo-instruct). Since I know that I will eventually cross this bridge in real life/real time, I devised this test on my Asterisk 10.0 box
Dialplan (in default context) exten => 3366,1,answer() exten => 3366,n,playback(demo-instruct,noanswer) exten => 3366,n,playback(demo-instruct,noanswer) exten => 3366,n,playback(vm-goodbye,noanswer) exten => 3366,n,hangup() SIPP command ./sipp -l 399 -d 99000 -m 399 -s 3366 -p 5061 -sn uac 127.0.0.1 -trace_err I was able to do 260 concurrent calls with no issues. The 2 playbacks for demo-instruct were to cover 99 seconds since the file is only 67 seconds long. For the 300/1000 call scenario, you would need to duplicate the line accordingly. The limiting factor for me was my rtp.conf. I set up a range of 10001-10520 which stopped at 260 since each "call" allocates 4 rtp slots (2 in use and 2 for transfer, etc). -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Stefan Schmidt Sent: Thursday, February 09, 2012 10:06 AM To: [email protected] Subject: Re: [asterisk-users] Asterisk V/s FreeSwitch Am 09.02.12 16:45, schrieb Patrick Lists: > Iirc a long time ago there was a discussion about load testing by > playing MoH was not a realistic test. Something about all MoH music > getting streamed synchronized so basically Asterisk only has to stream > one file and sorta multiplex that single output to all the established > calls (legs). this load tests are mostly about sip signal handling and not so much about rtp streaming but this moh class which i use had 100 files and random set to yes, so its atleast not soo bad. > [snip] > >> btw my normal production machines which are just the same virtual >> machines like this test system. i also had 330 concurrent calls, some >> with transcoding, many database lookups, musiconhold, pickup ... and >> the sysload was around 1.0 ;) > > The difference (13500 with MoH versus 330 with a real dialplan) shows > that it makes sense to mimic your dialplan in your test scenario as > much as possible to see how far you can realistically push the box and > still keep things stable and sound quality good. This 330 concurrent calls was only the highest value which i had on a normal production system and its really hard to build a test setup which presents a system with 4000 sip peer doing some calls. but the sound quality was still good even with 10000 calls in my tests. > Regards, > Patrick > best regards stefan -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
