With TCP you don’t really get any
packets lost eventually, however if your streaming client somehow decides not to
wait for a retransmitted packet, the retransmitted packet could be dropped by
the client itself.
Given TCP is reliable, however, by looking
at the number of retransmissions, you may have some idea about how good/bad the
network is.
If you’d like to have a quick look
at the round-trip time, you may wish to use Statistics->TCP Stream Graph->Round
Trip Time Graph, which gives you some idea about the end-to-end delay (roughly
half the round-trip time).
For better analyzing, it may be good to
grasp some more TCP basics.
Cheers,
Ji
From:
wireshark-users-bounces@xxxxxxxxxxxxx
[mailto:wireshark-users-bounces@xxxxxxxxxxxxx] On Behalf Of Fabiana moreno
Sent: 10 April 2008 17:31
To: Community support list for
Wireshark
Subject: Re: [Wireshark-users]
end-to-end delay calculation using tcp
So every retransmission
is a packet lost?
On 10/04/2008, Hansang Bae
<hbae@xxxxxxxxxx> wrote:
Fabiana moreno
wrote:
> Hello guys,
>
> I am streaming a video using tcp and i was wondering how could i
> calculate certain parameters such as end-to-end delay or frame loss?? is
> it hard to acomplish?
>
You are streaming or downloading the video? if you're streaming, I
guess you use a buffer of some sort?
end to end delay can be somewhat tricky but there are some rules of
thumb you can use to calculate it.
1) Look at SYN and SYN+ACK. The time it takes for syn_ack
to come back
is your round trip delay
2) Look at ACKs coming back after two successive full sized packets.
This too can be used to calculate the round trip time. The reason is
that the receivers tcp stack will immediately ACK after receiving two
full size packets (w/o app intervention).
You can also look for packet loss by using the filter
"tcp.analysis.flags" display filter and looking for retransmissions.
--
Thanks,
Hansang
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