| 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 3.1.5135.2568, compiled at 2014-01-21 21:25 UTC |
| StarTrinity SIP tester uptime | 1d 1h 32m 57s |
| Measurement started at | 2014-01-22 1:35:29 |
| Measurement duration | 1d 1h 32m 57s |
| CDR files | download |
| Actual average incomplete calls count (number of channels) | 2,25 |
| Received calls count | 4603 |
| Total average received calls per second | 0,05 |
| Answered received calls count | 4603 |
| Attempted outgoing calls count | 9928 |
| Total average attempted calls per second | 0,11 |
| Session establishment rate | 100,00% (9928/9928) |
| Failed outgoing calls count (total) | 0,00% (0/9928) |
| with status = 408 (Request Timeout) | 0,00% (0/9928) |
| with status = 486 (Busy Here) | 0,00% (0/9928) |
| with status = 487 (Request Terminated) | 0,00% (0/9928) |
| Answered outgoing calls count | 9928 |
| Successfully completed calls | 14527 |
| Answered duration (min/avg/max) | 3662,99ms/10638,51ms/15185,26ms |
| Min interval between calls | 3105,01ms, fixed |
| Max calls per second | 0,32 |
| Max number of simultaneous calls | 3,84 |
| Indicator | Min | Average | Max | Percentile 90% | 95% | 98% | 99% | 99.5% | 99.8% | 99.9% | 99.95% | 99.98% | 99.99% |
|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
| Max call jitter (ms) | 0,00 | 12,18 | 523,00 |
22,00 |
41,00 |
66,00 |
92,00 |
120,00 |
156,00 |
183,00 |
230,00 |
264,00 |
296,00 |
| Answer delay (ms) | 0,00 | 402,40 | 7116,59 |
415,28 |
427,07 |
496,23 |
552,03 |
2428,27 |
5737,29 |
6418,27 |
6811,20 |
7069,98 |
7116,59 |
|
-24dB delay (ms) |
229,01 | 717,94 | 7719,44 |
1028,06 |
1041,06 |
1063,06 |
1090,06 |
1128,06 |
2233,13 |
5516,32 |
7128,41 |
7679,44 |
7719,44 |
|
Media threads delay (ms) |
0,23 | 4,61 | 127,44 |
8,41 |
14,66 |
25,10 |
31,38 |
43,67 |
68,35 |
89,52 |
93,58 |
123,17 |
124,66 |
|
Signalization thread delay (ms) |
0,00 | 3,23 | 159,86 |
9,76 |
9,78 |
9,82 |
10,02 |
12,39 |
17,77 |
19,96 |
26,23 |
118,23 |
132,31 |
| MOS | 1,25 | 4,10 | 4,55 |
3,88 |
3,82 |
3,77 |
3,72 |
3,66 |
3,55 |
3,44 |
3,18 |
1,25 |
1,25 |
| Outgoing CallXML script |
<callxml> <call maxtime="19530ms" value="$random_least_busy_uac_registration();" /> <on event="answer"> <playaudio value="ivr2" repeat="10000" maxtime="15000ms" /> <exit /> </on> </callxml> |
| Incoming CallXML script |
<callxml> <accept /> <verifyaudio reference="ivr2" recognizedreferencevar="ref" maxtime="10000ms" mosvar="mos" /> <writecdr header="REF" value="$ref;" /> <writecdr header="MOS" value="$mos;" numeric="true" qualityIsAscending="true" /> <log value="mos=$mos;, ref=$ref;" /> <exit /> </callxml> |
| 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 |
Although session establishment rate is 100%, some of calls did not pass through 3CX PBX. Reason could be in 3CX license limit of 4 G.711 channels. This is not clear because SIP Tester did not generate more than 4 simultaneous calls. For calls which don't pass PBX, it plays this audio.
Max post dial delay is 7.7 seconds, which is not good. CPU power should be enough to handle 4 G.711 channels.
Audio quality (MOS) is satisfactory for 99.99% of calls. Worst MOS (1.25) is observed because of large post dial delay - download recording. Recording of another call with MOS = 1.76 contains missing word. Reason could be in large jitter - up to 523ms.
We will make more tests of 3CX PBX in order to get better understanding of problem. Also, we will improve accuracy of jitter buffer measurements by capturing RTP packets directly from network driver.
Report is automatically generated at 2014-01-23 3:08:26