[Freeswitch-users] Load test - performance not even matching Asterisk

[Ken Rice]

Try using something like SIPP for load testing you can load test to much
higher numbers 
http://www.freeswitch.org/eg/load_test.tgz  is what we use for testing so
you can duplicate the results...

Also, look at the configuration you are doing and determine if you really
need all the features that are there... Things like presence tracking,
certain CDR loggers, and a few other things under high CPS loads can cause
more problems then you think...

Hint... Mount freeswitch/db as a ram drive in linux this is a big
performance booster (since it takes the load of sqlite of the hdd), also
turn off presence tracking on all sip profiles that don't need it...

Something doesn't sound right on the 500k of rtp... Also remember that
asterisk RTP stack doesn't handle async rtp.. It depends on receiving a
packet to transmit a packet


> From: Jon Bruel <jbr at consiglia.dk>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 4 Oct 2008 11:03:40 +0200
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Load test - performance not even matching
> Asterisk
> 
> Hi all
> An update on the performance measurements:
> The measurements I have referred to earlier all involved an Asterisk as
> the call generator. Somehow this setup leads to extensive rtp bandwidth
> usage. Each channel used around 500 kbps. If a phone is entered into the
> loop, this is reduced to the expected 64 kbps. I have not found any
> reason for this, but it certainly fouls up the test, and I have changed
> the test setup.
> Further, and since the earlier tests, the network has been updated to a
> Gbits network.
> I have now made two new test:
> 1) Using WinSIP from Touchstone as a call generator.
> 2) Using the Asterisk as one component, and setting up a chain of calls
> which goes forth and back from the Asterisk and the FS. All call are
> started from a real phone, and after 100 loops, where the calls are
> answered and sent on by the dial plan, the calls are terminated by an
> tone (<action application="gentones" data="%(500000,0,400)"/>) in the
> FS.
> The two test show similar top-figures at similar loads.
> The first test would be my preferable, but it is limited to 50 calls due
> to the trial licence limitations. Using an external non-FS and
> non-Asterisk device will eliminate some uncertainties, that's why it
> would be preferred.
> The other test has been done with 600, 400 and 200 channels (300, 200
> and 100 calls), and the results of the top command are:
> cpu sy ni id wa hi si total
> * 600 10 30 0 33 0 2 25 100
> FS600 22 33 0 30 0 0 15 100
> 0
> * 400 7 18 0 67 0 1 7 100
> FS400 14 17 0 62 0 0 7 100
> 0
> * 200 3 10 0 84 1 0 2 100
> FS200 7 8 0 82 1 0 2 100
> The results do not show significant differences between the capacity
> behaviour of the Asterisk (*) and the FS. The also show an expected
> interrupt load (si) proportional to the square of the call load.
> Still the FS does not really outperform the Asterisk - which I find
> disappointing. Any comments are welcome.
> 
> 
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