VoIP performance and SIP call quality test report for RTT over wireless LAN -  distance is 3 meters - RTP jitter, MOS, delays

StarTrinity SIP testerVersion 3.1.5317.43110, compiled at 2014-07-23 19:57 UTC
StarTrinity SIP tester uptime0d 14h 22m 40s
Remote SIP 'User-Agent' header
Remote SIP 'Server' headerStarTrinity.SIP 2014-07-20 21:06 UTC
Measurement started at24.07.2014 0:50:38
Measurement duration0d 14h 22m 40s
Operation modeActive - generation and receiving SIP calls
Lightweight media processingoff
Memory consumed by SIP Tester562MB
Current calls count (number of channels)min = 0,00; average = 0,99; max = 1,00
Received SIP calls count0
Total average received calls per second0,00
Attempted outgoing calls count8343
Total average attempted calls per second0,16
Session establishment rate100,00% (8343/8343)
Failed outgoing calls count (total)0,00% (0/8343)
    with status = 408 (Request Timeout)0,00% (0/8343)
    with status = 486 (Busy Here)0,00% (0/8343)
    with status = 487 (Request Terminated)0,00% (0/8343)
Answered calls count8343
Successfully completed calls8342
Answered duration (min/avg/max)6063,00ms/6080,07ms/6668,00ms

Stress parameters for outgoing calls

Min interval between calls144,12ms, fixed
Max calls per second6,94
Max number of current calls1,00

Performance indicators

IndicatorNcallsMinAverageMaxPercentile 90%95%98%99%99.5%99.8%99.9%99.95%99.98%99.99%

Caller lost packets (%)

8342

0,000,000,00

0,00

0,00

0,00

0,00

0,00

0,00

0,00

0,00

0,00

0,00

Caller G.107 MOS

8342

4,414,414,41

4,41

4,41

4,41

4,41

4,41

4,41

4,41

4,41

4,41

4,41

Caller G.107 R-factor

8342

93,2093,1993,20

93,20

93,20

93,20

93,20

93,20

93,20

93,20

93,20

93,20

93,20

Caller max delta (ms)

8342

50,1251,27190,46

52,07

52,10

52,19

53,06

55,51

57,37

58,20

58,90

60,20

190,46

Caller max RFC3550 jitter (ms)

8342

0,040,2515,51

0,36

0,40

0,45

0,50

0,67

0,91

0,99

1,08

1,23

15,51

Caller mean RFC3550 jitter (ms)

8342

0,020,072,42

0,09

0,11

0,12

0,13

0,16

0,18

0,21

0,22

0,26

2,42

Called lost packets (%)

8342

0,000,000,00

0,00

0,00

0,00

0,00

0,00

0,00

0,00

0,00

0,00

0,00

Called G.107 MOS

8342

4,414,414,41

4,41

4,41

4,41

4,41

4,41

4,41

4,41

4,41

4,41

4,41

Called G.107 R-factor

8342

93,2093,1993,20

93,20

93,20

93,20

93,20

93,20

93,20

93,20

93,20

93,20

93,20

Called max delta (ms)

8342

20,5426,58823,25

37,09

45,09

58,50

65,62

74,00

136,44

197,90

234,25

529,95

823,25

Called max RFC3550 jitter (ms)

8342

0,120,9650,39

2,28

3,07

4,74

5,29

7,17

13,07

17,43

25,34

49,97

50,39

Called mean RFC3550 jitter (ms)

8342

0,060,217,26

0,36

0,46

0,64

0,76

1,05

1,69

2,64

5,29

6,75

7,26

100 response delay (ms)

8342

1,008,80548,00

4,00

5,00

8,00

502,00

503,00

505,00

506,00

514,00

543,00

548,00

Answer delay (ms)

8342

17,0027,33567,00

23,00

25,00

32,00

520,00

522,00

523,00

527,00

531,00

562,00

567,00

-24dB delay (ms)

8341

121,01148,01683,04

149,01

152,01

160,01

638,04

644,04

648,04

651,04

653,04

681,04

683,04

Media threads delay (ms)

8342

0,592,0077,41

2,82

3,01

3,93

4,74

5,76

8,34

10,09

11,95

41,67

77,41

Signaling thread delay (ms)

8342

9,159,8019,35

9,87

9,91

9,98

10,04

11,02

13,84

15,40

16,61

17,77

19,35

GUI thread delay (ms)

8342

0,018,34493,03

16,66

18,65

19,66

27,10

44,22

59,95

69,73

78,66

80,62

493,03

RX_WG67_PTT2SQU_DELAY

8342

1,0012,2573,00

18,00

20,00

21,00

23,00

56,00

69,00

71,00

71,00

72,00

73,00

Packet analyser statistics

Statuson
Total packets0 dropped, 3688771 detected
Processing delay0,3ms
SIP+RTP packets3603355
RTP packets3552990
SIP packets50365
INVITE8449
INVITE retransmissions106
RE-INVITE0
'100 Trying' responses to INVITE8343
'180 Ringing' responses to INVITE0
'183 Session Progress' responses to INVITE0
Error responses to INVITE0
'200 OK' responses to INVITE8407
BYE8417
BYE retransmissions75
'200 OK' responses to BYE8342
CANCEL0
CANCEL retransmissions0
'200 OK' responses to CANCEL0
ACK8407

Scripts

Outgoing CallXML script<callxml>
<call value="sip:111@192.168.123.1:5090" callerId="1111" />
<on event="answer">
 <setrtpextension wg67_pttType="1" wg67_pttId="16" />
 <playaudio value="music.wav" maxtime="3s" />
 <setrtpextension wg67_pttType="0" wg67_pttId="16" />
 <playaudio value="music.wav" maxtime="3s" />
 <getwg67info ptt2squdelayvar="ptt2squdelay" ptttypevar="ptttype" pttidvar="pttid" squvar="squ" pmvar="pm" pttsvar="ptts" sctvar="sct" />
 <log value="WG67 measurements: ptt2squdelay=$ptt2squdelay;ms, ptttype=$ptttype;, pttid=$pttid;, squ=$squ;, pm=$pm;, ptts=$ptts;, sct=$sct;" />
 <writecdr field="RX_WG67_PTT_ID" value="$pttid;" />
 <writecdr field="RX_WG67_PTT2SQU_DELAY" value="$ptt2squdelay;" numeric="true" qualityIsAscending="false" />
 <!-- save measured delay to CSV CDR file, GUI report and history charts on "reports/statistics" tab -->
 <exit />
</on>

</callxml>

Settings

Desired audio codecUnknown
Forced audio codecUnknown
Jitter buffer initial size10ms
Jitter buffer max size80ms
Jitter buffer prefetch max size20ms
Jitter buffer prefetch min size10ms

Report was automatically generated at 24.07.2014 15:13:19