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Hi All,

I have captured the logs at both ends for a end-to-end Wi-Fi call using two phones. They have mainly SIP and RTP packets. I am getting jitter between 5-15 ms for reverse and forward directions. I have a local router which isn't connected to LAN.
My question is: Is it possible to analyze jitter and find the possible root cause of why it is occurring?
Please let me know if any detail i am missing.

Any help is greatly appreciated. Thanks!

asked 11 Jul '12, 12:16

vibgyor2nee's gravatar image

accept rate: 0%

In addition to the main question: is it possible to get jitter or other voice quality metrics via command line?

(10 Jan '14, 04:25) Alex Voron

Please try this:

tshark -nr rtp.pcap -q -z rtp,streams

The output will also show jitter values.


(15 Jan '14, 06:14) Kurt Knochner ♦

Wireshark 1.8

Telephony -> RTP -> Show All Streams

This will show all RTP streams and Min/Max Jitter for each stream (scroll to the right).

Then select one stream and click on Analyze (same as Option: Telephony -> RTP -> Stream Analysis). You will get a much more detailed view of that stream.


Any idea what is the permissible value of Jitter for an end-to-end wi-fi call?

Well, it depends ... Some network providers offer SLAs with 0.5 - 2 ms max jitter. That's pretty good. Others (e.g. Avaja, Cisco) say, 10-20 ms is acceptable. Furthermore there are compensations techniques, like jitter buffers. So a jitter value of 15 ms can cause problems (crippled audio) in one environment and no problems at all in another environment (with jitter buffers).

As I mentioned, Jitter can be caused by numerous factors. Even the VoIP devices (especially soft phones) can cause jitter due to process scheduling in the device. So, if there are no signs of other network problems, you should consider the VoIP devices as a possible source.

As you mentioned that the VoIP endpoints are connected via Wifi, I suggest to check that connection first. A Wifi link can also cause jitter, especially if the network is "crowded" or if there are other interfering radio signals. You can test the jitter of the network with xjperf (UDP tests). I'm not sure how iperf calculates the jitter value, so it may not be comparable with the VoIP jitter value! However, it's something to start with.


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answered 12 Jul '12, 00:45

Kurt%20Knochner's gravatar image

Kurt Knochner ♦
accept rate: 15%

edited 12 Jul '12, 20:40

Thanks Kurt

I got that. That's how i know the jitter values. I need to analyze it and find the reason of occurrence. Is there any way we can do so?

(12 Jul '12, 10:08) vibgyor2nee

you can save the the jitter values as CSV, then use Excel to chart the values together with the frame/packet number. If there are any spikes in the jitter values, go to that packet number any check if there are any signs of general network problems within +/- 10 seconds in the capture file. Unfortunately it's hard to give any detailed advice what to look for, as it may be anything or nothing if it's a problem on the VoIP client itself.

(12 Jul '12, 11:46) Kurt Knochner ♦

Any idea what is the permissible value of Jitter for an end-to-end wi-fi call?

(12 Jul '12, 14:32) vibgyor2nee

see UPDATE in my answer.

(12 Jul '12, 20:15) Kurt Knochner ♦
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question asked: 11 Jul '12, 12:16

question was seen: 26,134 times

last updated: 15 Jan '14, 06:17

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