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