RTP Jitter calculation (Telephony->RTP Stream Analysis)

["Chaswi Przellczyk" <cp70 () gmx de>]

Dear all,

I'm trying to match
---------------------------
http://wiki.wireshark.org/RTP_statistics
---------------------------
How jitter is calculated

Wireshark calculates jitter according to RFC3550 (RTP):

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 SHOULD be 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(i) = J(i-1) + (|D(i-1,i)| - J(i-1))/16 

---------------------------
to the following code from tap-rtp-common.c (rtp_packet_analyse):
---------------------------
                if( !statinfo->first_packet )
                { /* Calculate the current jitter(in ms) */
                        expected_time    = statinfo->time + (nominaltime - statinfo->lastnominaltime);
                        current_diff     = fabs(current_time - expected_time);
                        current_jitter   = (15 * statinfo->jitter + current_diff) / 16;

                        statinfo->delta  = current_time - (statinfo->time);
                        statinfo->jitter = current_jitter;
                        statinfo->diff   = current_diff;
                }
                statinfo->lastnominaltime = nominaltime;

---------------------------

And I can't find (m)any similarities between the two.

Can anyone lend a hand?

Thanks!
CP.

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