asterisk users - May 2004 - SIP calls-per-second performance test tool

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.

JT

If you can only get it to compile! :P

bkw
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] On Behalf Of John Todd
> Sent: Monday, May 10, 2004 3:16 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] SIP calls-per-second performance test tool
>
>
> 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.
>
> JT
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

> It seems that entire days pass by before I 
> hang up... very odd, and very counter-productive to getting good 
> Asterisk work done.
Telephony is evil.

P.

No h.323 is evil... PURE evil! :P

bkw
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] On Behalf Of Philipp von Klitzing
> Sent: Monday, May 10, 2004 4:09 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] SIP calls-per-second performance test tool
>
> > It seems that entire days pass by before I
> > hang up... very odd, and very counter-productive to getting good
> > Asterisk work done.
>
> Telephony is evil.
>
> P.
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

JT,

I ran this against my home office asterisk box (4 analog lines, about 20 
sip UA's, 2.6G P4, 512MB system).  I just ran the basic test, routing the 
request to Playback(invalid) then Hangup.

During the test I had two UA's (a cisco 7960 and an analog phone connected 
to an ATA 186) dialed into MoH.

Asterisk was running in background with no options to the command line, and 
one remote CLI connection.

The system was able to handle 20 calls per second without any call 
failures.  Beyond 20 calls per second I began to see call failure.  The 
quality of the two MoH calls was perfect the entire time.

I then proceeded to crank up the call volume and right about 200 calls per 
second, all call attempts became failures, and no new calls succeeded).  At 
this point I got some interesting errors on the CLI related to maximum file 
descriptors (which I didn't worry too much about at the time), however, 
when I cranked the call volume back down to under 20 cps, all calls still 
failed.  I had to shut down asterisk and restart to restore the 
system.  However on an interesting note, at no time during any of the tests 
did the MoH calls lose quality or suffer any artifacts.

Interesting program, and I'll set up a much more scientific test system and 
post some results on multiple systems (1G Pentium, 2.6G Pentium, and a Dual 
AMD system on 2.4 and 2.6 kernels) sometime soon.

-Chris


On 01:16 PM 5/10/2004, 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.
 >
 >JT
 >_______________________________________________
 >Asterisk-Users mailing list
 >Asterisk-Users@lists.digium.com
 >http://lists.digium.com/mailman/listinfo/asterisk-users
 >To UNSUBSCRIBE or update options visit:
 >   http://lists.digium.com/mailman/listinfo/asterisk-users
 >
 >

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.
	I?m playing with it.
	Remember to increase the max number of file descriptors and RTP ports.
	I have reached 1500 concurrent SIP/GSM calls against the Echo
application at 50cps of 10s duration on a Xeon 2.4Ghz single cpu. Anyway
I?m trying to make a more real scenario since I believe Echo application
doesn?t process audio too much just copies it.
	When testing I monitor Asterisk against a 7960 calling MusicOnHold at
about 500-600 concurrent calls I start to have audio dropouts.


-- 
Juanjo sin .sig

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