RTP packets have sequence numbers. If packets 1, 2, and 5 are in the
trace, then packets 3 and 4 are considered lost. Wireshark flags
sequence arrival errors. However, packets played out depend on the exact
characteristics of the dejitter buffer in the playout device. Wireshark
does not model dejitter buffers, so it can't tell you what would be
dropped due to jitter or sequence errors.
Similarly, GSM 06.10 decoder software will not tell you exactly what the
listener hears unless you model the dejitter buffer correctly.
PacketScan (http://gl.com/packetscan.html#protocols) will do what you
want, but it's not cheap.
Harvey S. Cohen
-----Original Message-----
From: wireshark-users-bounces@xxxxxxxxxxxxx
[mailto:wireshark-users-bounces@xxxxxxxxxxxxx] On Behalf Of Adam Harding
Sent: Tuesday, March 27, 2007 10:49 AM
To: Wireshark-users@xxxxxxxxxxxxx
Subject: [Wireshark-users] Please help with difference between
packets"lost" and "dropped"
Please could someone help me to understand RTP Packet Loss in Wireshark!
Does the Packet Loss percentage in RTP analysis mean packets lost in in
the network before they reach Wireshark? If so how does Wireshark know
they got lost then?
How can I differentiate between packets lost due to the network and
packets dropped by the application due to jitter at the recieving end?
Does the sequence errors help show me any of this at all?
Many thanks,
Adam Harding
ps. Does anyone know of a plug in or some way I can get RTP player to
support the GSM 06.10 codec? Cheers.
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