[asterisk-users] Asterisk V/s FreeSwitch
Danny Nicholas
danny at debsinc.com
Thu Feb 9 13:17:20 CST 2012
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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stefan Schmidt
Sent: Thursday, February 09, 2012 10:06 AM
To: asterisk-users at lists.digium.com
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
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