VoIP performance and SIP call quality test report for 3CX Phone System 12.0.33463.0 - RTP jitter, MOS, delays

SIP Server software 3CX Phone System 12.0.33463.0
SIP Server hardware Lenovo 3000 G530 laptop, Pentium Dual-Core T3100 1.9GHz 2 cores
Network 100Mbps ethernet, PC to PC
SIP Tester hardware Lenovo G700 laptop, Pentium 2020M 2.4GHz 2 cores
StarTrinity SIP tester Version 3.1.5137.41327, compiled at 2014-01-24 18:57 UTC
StarTrinity SIP tester uptime 2d 17h 37m 40s
Measurement started at 2014-01-25 1:28:29
Measurement duration 2d 15h 52m 17s
Actual average incomplete calls count (number of channels) 0,58
Received SIP calls count 3954
Total average received calls per second 0,02
Answered received calls count 3954
Attempted outgoing calls count 4751
Total average attempted calls per second 0,02
Session establishment rate 100,00% (4751/4751)
Failed outgoing calls count (total) 0,00% (0/4751)
    with status = 408 (Request Timeout) 0,00% (0/4751)
    with status = 486 (Busy Here) 0,00% (0/4751)
    with status = 487 (Request Terminated) 0,00% (0/4751)
Answered outgoing calls count 4751
Successfully completed calls 8706
Answered duration (min/avg/max) 12951,29ms/13209,90ms/15120,19ms

Stress parameters for outgoing calls

Min interval between calls 25864,16ms, fixed
Max calls per second 0,04
Max number of simultaneous calls 3,97

Performance indicators

Indicator Ncalls Min Average Max Percentile 90% 95% 98% 99% 99.5% 99.8% 99.9% 99.95% 99.98% 99.99%
Max call jitter (ms)

8424

0,00 6,39 154,00

9,00

20,00

28,00

39,00

58,00

69,00

79,00

87,00

142,00

154,00

Answer delay (ms)

4751

143,37 379,27 7111,50

406,45

412,62

417,57

463,58

905,52

3705,98

4949,06

5648,23

7111,50

7111,50

-24dB delay (ms)

8706

230,01 483,79 7261,42

525,03

1006,06

1022,06

1030,06

1042,06

2803,16

3955,23

5075,29

6466,37

7261,42

Media threads delay (ms)

8706

0,15 2,49 25,98

3,94

4,35

5,82

7,01

8,30

11,17

13,20

24,08

25,70

25,98

Signalization thread delay (ms)

8706

0,00 0,02 5,97

0,01

0,01

0,01

0,26

0,42

2,82

4,03

5,52

5,90

5,97

Scripts

Outgoing CallXML script <callxml>
<call maxtime="19530ms" value="$random_least_busy_uac_registration();" />
<on event="answer">
<playaudio value="ivr2" repeat="10000" maxtime="15000ms" />
<exit />
</on>
</callxml>
Incoming CallXML script <callxml>
<accept />
<playaudio value="ivr3" />
<exit />
</callxml>

Settings

Desired audio codec Unknown
Forced audio codec PCMA
Jitter buffer initial size 10ms
Jitter buffer max size 80ms
Jitter buffer prefetch max size 20ms
Jitter buffer prefetch min size 10ms

Comments

We have reduced number of channels from 4 to 2, and jitter has improved. However, answer delay is still not good - 7.1 seconds. From SIP trace we see that some SIP INVITE packets get ignored - see image. We will continue investigation with a different hardware.

Report was automatically generated at 2014-01-27 17:20:46