QoS in TCP and UDP protocol

[Armansah Hs <armansah.hs () gmail com>]

Hello,
I want to ask some questions about *e2e delay, jitter, packet loss* in TCP
and UDP protocol.
I already post a question at "ask wireshark" and get one answer but there
is a lack of information I need. (
https://ask.wireshark.org/questions/38705/qos-in-tcp-and-udp-protocol)

------------------------- edited
I have 4 files of captured traffic in wireshark, both client
(192.168.100.6) and server (192.168.100.1) have a captured traffic and time
synchronized.
I want to ask you some questions about how to get the QoS parameters in
wireshark (*end-to-end delay, jitter, throughput, and packet loss*)

I have two scenario

   - Transfer file between source and destination
   - Video streaming using VLC (RTSP protocol)

so far, I've got TCP throughput and RTP Throughput.

*Transfer file*

   - TCP throughput - in the Statistic -> Summary (done)
   - *TCP e2e delay, jitter, and packet loss. *I want to know how to get
   these parameters.


*Video Streaming*

   - RTP throughput  - in the Statistic -> Summary (done)
   - *RTP e2e delay. *I want to know how to get this parameter.

My friend taught me that RTP packet loss and RTP jitter can be found in *RTP
-> Telephony -> Show all streams.*
but when I go in that menu, there is no jitter and there are 2 detected RTP
stream (<5% packet loss and >40% packet loss)

-------------------------
this is what Jaap say in the answer section

   - TCP throughput: that can be derived from the protocol interaction at a
   single endpoint, hence is available.
   - TCP e2e delay, jitter, packet loss: Hard to do based on a single
   capture, apart from the packet loss maybe, as retransmissions would
   indicate as such. Not aware of a ready made analysis function right now.
   - RTP throughput: that can be derived from the protocol interaction at a
   single endpoint, hence is available.
   - RTP packet loss: Be aware, you use an Telephony analysis feature for
   video, that doesn't work. Unfortunately the RTP statistics are not profile
   aware and geared towards telephony only. And even then only the simplest
   cases.
   - RTP e2e delay: Hard to do based on a single capture.

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

last but not least, Related to packet loss, do I really have a packet loss
if the retransmission successfully sent to destination?

how to extract TCP and RTP e2e delay; TCP packet loss; TCP and RTP jitter?

where I can find the RTP packet loss and jitter other than through
telephony analysis?


Thanks,

Armansah Hs

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