[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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