Difference between revisions of "Asterisk Background performance test"
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[[Performance tests]] | |||
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<big>'''Goal - to run at least 700 calls on this server.'''</big> | <big>'''Goal - to run at least 700 calls on this server.'''</big> | ||
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== Testing == | == Testing == | ||
Starting from 500 calls we | 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 72: | Line 74: | ||
[[Image:sipp_background_stats.png]] | [[Image: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 | |||
Latest revision as of 11:57, 28 April 2010
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.
504 calls
605 calls
703 calls
806 calls
909 calls
1002 calls
1143 calls
Results
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
