VoIP performance and SIP call quality test report for SIP Tester - RTP jitter, MOS, delays

StarTrinity SIP testerVersion 3.1.5225.41627, compiled at 2014-04-22 19:07 UTC
StarTrinity SIP tester uptime0d 1h 5m 14s
Remote SIP 'User-Agent' header
Remote SIP 'Server' headerStarTrinity.SIP 2014-04-22 19:07 UTC
Measurement started at2014-04-23 17:07:16
Measurement duration0d 1h 5m 14s
Packet analyseroff
Lightweight media processingoff
Actual average incomplete calls count (number of channels)16,85
Received SIP calls count0
Total average received calls per second0,00
Answered received calls count0
Attempted outgoing calls count2169515
Total average attempted calls per second554,26
Session establishment rate100,00% (2169508/2169515)
Failed outgoing calls count (total)0,00% (0/2169515)
    with status = 408 (Request Timeout)0,00% (0/2169515)
    with status = 486 (Busy Here)0,00% (0/2169515)
    with status = 487 (Request Terminated)0,00% (0/2169515)
Answered outgoing calls count2169508
Successfully completed calls2169508
Answered duration (min/avg/max)0,01ms/1,25ms/75,53ms

Stress parameters for outgoing calls

Min interval between calls1,80ms, fixed
Max calls per second555,56
Max number of current calls30,00

Performance indicators

IndicatorNcallsMinAverageMaxPercentile 90%95%98%99%99.5%99.8%99.9%99.95%99.98%99.99%
Max RX RFC3550 jitter (ms)

0

Max RX instant jitter (ms)

0

Max RX delta (ms)

0

Lost RX packets (%)

0

Max TX instant jitter (ms)

0

Answer delay (ms)

2169508

0,000,48194,60

0,00

0,50

9,96

15,10

19,28

24,61

28,30

32,40

43,02

58,42

-24dB delay (ms)

0

Media threads delay (ms)

2169508

0,020,6012,29

0,05

2,30

3,47

4,24

4,98

6,21

8,38

10,20

11,46

11,83

Signalization thread delay (ms)

2169508

0,001,2475,53

1,76

3,28

4,49

5,49

6,61

8,51

10,65

13,38

35,77

56,98

GUI thread delay (ms)

2169508

0,3343,744974,18

16,92

20,01

477,55

1383,33

2174,53

3046,02

3633,55

3941,42

4281,22

4605,86

Scripts

Outgoing CallXML script<callxml>
<call value="sip:123456@devpc:5080" callerId="1111" sendSdpInInitialInvite="false" />
<on event="answer">
<exit />
</on>
</callxml>

Settings

Desired audio codecUnknown
Forced audio codecUnknown
Jitter buffer initial size10ms
Jitter buffer max size10ms
Jitter buffer prefetch max size10ms
Jitter buffer prefetch min size10ms

Report was automatically generated at 2014-04-23 18:12:30