[Asterisk-Users] SIP calls-per-second performance test tool

[Juan J. Sierralta P.]

On Mon, 2004-05-10 at 16:16, John Todd wrote:
> http://sipp.sourceforge.net/
> 
> Anyone care to throw this at Asterisk to see what happens?   I would, 
> but I am having significant temporal shortfalls recently due to the 
> apparent warping of the space/time continuum when I answer the phone 
> with clients/associates.  It seems that entire days pass by before I 
> hang up... very odd, and very counter-productive to getting good 
> Asterisk work done.

	Ok. Test report:

	I set up an UAC which was generating 10cps of 10s duration and the
corresponding UAS which received this calls. The command used to
generate the calls which were GSM was:

	sipp 192.168.65.100 -s 700 -sf uac.xml -d 10000 -r 10

	The command to receive the calls on another box was:

	sipp -sf uas.xml

	I´m using my own uac.xml and uas.xml just to talk GSM, I monitored
using my 7960 agains a MusicOnHold.
	On my Xeon 2.4Ghz no call were dropped and no audio problems. Note that
I use nat=yes and canreinvite=no for UAC/UAS on sip.conf. It seems that
SIPP doesn´t support authentication for now.
	For 40cps of 10sec duration (which means 400 concurrent calls) it works
just fine for me. At 50cps of 10sec duration no call are dropped but I
start seeing some SIP packets retransmitions.
	At 60cps lots of call gets dropped but the funny thing is that the
audio through the 7960 isn´t much affected.

	Real nice tool.

-- 
Juanjo sin .sig