The test environment was an Intel quad core machine with 8 GB and FreeSWITCH Version 1.0 installed. The load was generated by SIPP running on a different machine.
On this standard server, and just a little bit of tweaking, with 2000 calls in 50 different conference rooms with each conference being recorded to a file - the call quality was still ok and cpu was 50% idle.
Here are the SIPP result screens
Each call lasted about 4 and a half minutes with more or less 10% of that time prerecorded alaw sound was streamed by SIPP at different intervals throughout the call. 2002 calls were connected at peak, with 2000 generted by SIPP and 2 calls were placed manually to test the call quality.
I would say the results are quite impressive, what do you think ?
1 comment:
this test I complete with Asterisk
By edit
extensions.conf and add the following lines :
[sipp]
exten => 1001,1,Answer
exten => 1001,2,SetMusicOnHold(default)
exten => 1001,3,WaitMusicOnHold(93)
exten => 1001,4,Hangup
and
edit sip.conf and add the following lines :
[sipp]
type=friend
context=sipp
host=dynamic
port = 6000
user=sipp
canreinvite = no
disallow=all
allow=ulaw
but for Freeswitch I don't know how to test it.
please help me
Best Regards
julia
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