Difference between revisions of "Asterisk Background performance test"

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== Testing ==
== Testing ==


Starting from 500 calls we were increasing calls in 20 cps rate (calls per second).
Starting from 500 calls, we increased calls at a rate of 20 calls per second (cps).


Stats can be seen on following screenshots. Accumulated stats are at the end.
Stats can be seen on following screenshots. Accumulated stats are at the end.
Line 76: Line 76:




At 1143 network limit was reached and calls started to drop.
At 1143 calls, the network limit was reached and calls started to drop.
 
We were able to reach our goal - e.g. 700 sim. calls on this server.


We were able to reach our goal - that is, 700 sim. calls on this server.


== Future ==
== Future ==


In future I see no problem to reach 2000 calls, changes which should be made:
In future I see no problem in reaching 2000 calls. Changes that should be made:


* smaller codec should be used, e.g. GSM
* A smaller codec should be used, e.g. GSM
* 1 Gbit network instead of 100 Mbit we used
* 1 Gbit network instead of the 100 Mbit we used
* more RAM, second CPU, sound files stored into RAM instead of HDD
* More RAM, a second CPU, sound files stored into RAM instead of HDD

Latest revision as of 11:57, 28 April 2010

Performance tests



Goal - to run at least 700 calls on this server.

Hardware

  • Local LAN with 100mb cheap DLink switch
  • Laptop to initiate calls over sipp
  • Tested server:
HP Proliant DL160 G5 E5405
1 x Quad Core Xeon 2Ghz
2 Gb RAM
Asterisk 1.4.18.1
Centos 5.2


Asterisk configuration

For test standard /var/lib/asterisk/sounds/tt-monkeys.gsm from HDD (not from RAM) was used.

sip.conf

[general]
context=playback_test
;rtptimeout=60  
;rtpholdtimeout=300
[1000]
type = friend
host = dynamic
username = 1000
secret = 1000
nat = no
context = playback_test
disallow = all
allow = alaw


extensions.conf

[playback_test]
exten => _X.,1,Answer
exten => _X.,2,Background(tt-monkeys)
exten => _X.,3,Goto(2)


sipp

./sipp -inf users.csv -sf uac_gsm.xml -i 192.168.0.130 -s 123 192.168.0.191
File:Uac gsm.xml <-- uac_gsm.xml file


Testing

Starting from 500 calls, we increased calls at a rate of 20 calls per second (cps).

Stats can be seen on following screenshots. Accumulated stats are at the end.

At each step we dialed additionally with X-Pro to test how it sounds.

  • Sipp background 504 calls.png 504 calls
  • Sipp background 605 calls.png 605 calls
  • Sipp background 703 calls.png 703 calls
  • Sipp background 806 calls.png 806 calls
  • Sipp background 909 calls.png 909 calls
  • Sipp background 1002 calls.png 1002 calls
  • Sipp background 1143 calls.png 1143 calls

Results

Sipp background stats.png


At 1143 calls, the network limit was reached and calls started to drop.

We were able to reach our goal - that is, 700 sim. calls on this server.

Future

In future I see no problem in reaching 2000 calls. Changes that should be made:

  • A smaller codec should be used, e.g. GSM
  • 1 Gbit network instead of the 100 Mbit we used
  • More RAM, a second CPU, sound files stored into RAM instead of HDD