First time here? Check out the FAQ!
THIS IS A TEST INSTANCE. Feel free to ask and answer questions, but take care to avoid triggering too many notifications.
0

Wireshark RTP stream analysis jitter calculation always zero?

  • retag add tags

it seems wireshark RTP stream analysis jitter calculation incorrect, in example below note the inter RTP packet delay variation ( i.e jitter ), but represented as 0ms

any ideas why jitter is always represented as 0ms ? given the apparent inter RTP packet delay variation....

Hazza06's avatar
1
Hazza06
asked 2020-01-11 06:05:40 +0000
Jaap's avatar
13.7k
Jaap
updated 2020-01-12 09:53:04 +0000
edit flag offensive 0 remove flag close merge delete

Comments

I just tried Wireshark 3.2.0 on one of my VoIP pcaps and it shows Jitter correctly. Can you tell us which Wireshark version you are using? Are you able to share the pcap file on a public fileshare? There might be something inside your traffic that triggers this behavior.

SYN-bit's avatar SYN-bit (2020-01-11 10:06:54 +0000) edit

i'm already running wireshark 3.2.0, i've shared the pcap to link below;

https://drive.google.com/file/d/1PlCG...

Hazza06's avatar Hazza06 (2020-01-12 02:19:02 +0000) edit
add a comment see more comments

1 Answer

1

https://wiki.wireshark.org/RTP_statis...

RTP timestamp: RTP timestamp is based on the sampling frequency of the codec, 8000 in most audio codecs and 90000 in most video codecs. As the sampling frequency must be known to correctly calculate jitter it is problematic to do jitter calculations for dynamic payload types as the codec and it's sampling frequency must be known which implies that the setup information for the session must be in the trace and the codec used must be known to the program(with the current implementation).
Chuckc's avatar
3k
Chuckc
answered 2020-01-12 02:54:27 +0000, updated 2020-01-12 03:14:46 +0000
edit flag offensive 0 remove flag delete link

Comments

ok thanks, that makes a lot of sense, as 8x8 do SIP over TLS, and i don't have the X.509 certificate, i'm unable to dissect the SIP part of the call. I am pushing 8x8 to move away from codec / Payload type = 121, to either G.722 or G.711, but they are struggling to achieve this...only getting thier useless scripted support so far.. Thanks for the explanation though.

Hazza06's avatar Hazza06 (2020-01-12 03:17:00 +0000) edit

https://code.wireshark.org/review/git...
* Ignore jitter calculation for clockrate = 0

The analysis can't determine the clockrate for the payload type DynamicRTP-Type-121 so skips jitter calculation.

Chuckc's avatar Chuckc (2020-01-12 03:21:31 +0000) edit

I think that the display in such cases should be amended to display "N/A" or similar instead of 0 for the jitter. An item for someone interested to raise at the Wireshark Bugzilla.

grahamb's avatar grahamb (2020-01-12 12:25:43 +0000) edit

+1 for N/A in the Jitter/Skew columns when they can't be calculated

SYN-bit's avatar SYN-bit (2020-01-12 12:30:59 +0000) edit
add a comment see more comments

Your Answer

Please start posting anonymously - your entry will be published after you log in or create a new account. This space is reserved only for answers. If you would like to engage in a discussion, please instead post a comment under the question or an answer that you would like to discuss.

Add Answer