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

[Chris A. Icide]

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
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