Hi Mr. Lars, thanks for answering here too. I have 2 more questions, is there a specific jitter that I should consider for QoS purposes?, can I choose to use as my parameter the mean jitter shown in the RTP streams window? there is no special consideration or rule I must follow to measure the jitter in my QoS analysis?
I was reading the RFC-3550 regarding "delay since last SR (DLSR): 32 bits", I know with Wireshark I can filter RTCP so I would be able to read this information, but how is the delay managed in a Quality of Service Review?. I mean, what has to be less than 150ms? the delay since last SR for every packet for every call? :0S How do you clasify a call to be one with an acceptable delay? I´m confused... I thought I need *one* value of the delay for a whole communication (or call), is that ok??
Once more thanks for your time,
Best Regards,
Andreina
On 10/19/06, Lars Ruoff <Lars.Ruoff@xxxxxxxxxx> wrote:
Andreina, i replied to your private mail, but i also reply to this for
archiving purposes...
Andreina Toro wrote:
> Hi everyone, I have a question regarding the calculation of
> interarrival jitter and the information provided by Wireshark in the
> "RTP Stream Analysis Wndow" for each call.
>
> I can see that Wireshark gives me in the 4th Row of the RTP Stream
> Analysis Wndow the Jitter for each packet of each call.
>
> In the other hand I´ve read that:
>
> "If Si is the RTP timestamp from packet i, and Ri is the time of arrival
> in RTP timestamp units for packet i, then for two packets i and j, D may
> be expressed as
>
> D(i,j)=(Rj-Ri)-(Sj-Si)=(Rj-Sj)-(Ri-Si)
> The interarrival jitter is calculated continuously as each data packet i
> is received from source SSRC_n, using this difference D for that packet
> and the previous packet i-1 in order of arrival (not necessarily in
> sequence), according to the formula
> J=J+(|D(i-1,i)|-J)/16
> Whenever a reception report is issued, the current value of J is sampled."
>
> What I don´t have clear is what this Jitter in the 4th Row represents in
> the interarrival jitter calculation?
Well, it represents just that!
The value in 4th column *is* the value of J(i) according to the above
formula (ref. RFC 3550), starting with J(0):=0 and Ri:=frame.time(i) and
Si:=rtp.timestamp(i) in appropriate units (for conversion between units,
the clock sample rate is used - for details see the code in rtp_analysis.c).
> Can I calculate the jitter J, defined to be the mean deviation, with
> that data? I mean, can I use the values of the jitters of each
> packet given in that RTP Stream Analysis in every call and calculate the
> difference D??
>
> D_m = \frac{1}{n} \sum_{i=1}^n \left| x_i -
> \overline{x} \right|
What do you call "the jitter J"?
As said, the Jitter J(i) on a packet-by-packet basis is defined as above
and viewed in Wireshark RTP analysis in the 4th column.
If you want to have *one* value of J for a whole communication, feel
free the take the (arithmetic) mean over all J(i) (this is done and
shown on the RTP streams window by stream btw.) or use some other
mean/average.
I cannot tell you if one is more representative/common than another though.
best regards,
Lars Ruoff
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