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

[Juan J. Sierralta P.]

On Wed, 2004-05-12 at 18:42, Juan J. Sierralta P. wrote:

> 
> 	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.

	Update. I'm not seeing any RTP traffic generated from SIPP UAC which
means that * won't generate any traffic at all since it recover the
clock from the received stream AFAIK. So maybe the tests aren't
measuring the whole * capacity just the SIP stack. One solution is to
end the calls on a MusicOnHold extension but passing through an
intermediate * which is the measured one.
	So as I saw my test posted to the Wiki I prefer to have a more
realistic tests before going to the Wiki. BTW I'm known as Juanjo not
Juan Jo ;)