VoIP performance and SIP call quality test report for ASUS USB-N53 Wireless USB Adapter, 10 G711 channels - RTP jitter, MOS, delays

StarTrinity SIP tester Version 3.1.5179.4411, compiled at 2014-03-06 22:27 UTC
StarTrinity SIP tester uptime 0d 4h 15m 26s
Measurement started at 2014-03-07 21:52:45
Measurement duration 0d 3h 13m 37s
Packet analyser off
Actual average incomplete calls count (number of channels) 9,40
Received SIP calls count 8341
Total average received calls per second 0,72
Answered received calls count 5720
Attempted outgoing calls count 0
Total average attempted calls per second 0,00
Session establishment rate
Failed outgoing calls count (total) 0
    with status = 408 (Request Timeout) 0
    with status = 486 (Busy Here) 0
    with status = 487 (Request Terminated) 0
Answered outgoing calls count 0
Successfully completed calls 8351
Answered duration (min/avg/max) 10613,29ms/13040,52ms/13135,87ms

Performance indicators

Indicator Ncalls Min Average Max Percentile 90% 95% 98% 99% 99.5% 99.8% 99.9% 99.95% 99.98% 99.99%

-24dB delay (ms)

8351

165,01 205,23 2669,15

199,01

213,01

226,01

240,01

345,02

1493,09

1596,09

1783,10

2641,15

2669,15

Media threads delay (ms)

8351

1,06 5,43 62,44

7,56

12,94

15,36

17,41

25,44

31,42

50,66

54,61

56,49

62,44

Signalization thread delay (ms)

8351

0,00 1,23 40,19

0,01

10,69

12,76

15,27

18,57

23,56

24,71

27,38

36,15

40,19

GUI thread delay (ms)

8351

0,01 8,66 43,15

16,70

18,36

19,88

20,87

24,08

28,12

30,01

31,84

42,48

43,15

Scripts

Incoming CallXML script <callxml>
<accept/>
<playaudio value="ivr3" />
<exit />
</callxml>

Settings

Desired audio codec Unknown
Forced audio codec G711A
Jitter buffer initial size 10ms
Jitter buffer max size 80ms
Jitter buffer prefetch max size 20ms
Jitter buffer prefetch min size 10ms

Report was automatically generated at 2014-03-08 1:06:23