I want to calculate the Round Trip timing for the TCP packets.
But in wireshark, I don't see any particular field for the RTT timing for a TCP packet like its there for the RTP packet.
Wireshark do calculates the RTT graph but i am not finding as how it has been calculated.
Can someone help me out in finding the formula used for the same?
There is nothing inside TCP that gives the round-trip time. It's estimated by the kernel based on how long it takes to receive an ACK to data that was sent. It records the timestamp of when a given sequence number went out and compares it to the timestamp of the corresponding ACK. The initial 3-way handshake gives a decent starting value for this.
However, this is only an estimate as the receiver is free to delay ACKs for a short period if it feels it can respond to multiple incoming packets with a single reply.
RTT frequently changes over the duration of the session due to changing network conditions. The effect is (obviously) more pronounced the further away the endpoints.
If you want to get the values of the RTT calculated by wireshark/tshark, the following did the trick for me to print them on stdout:
tshark -r myfile.pcap -Y 'ip.addr == AA.BB.CC.DD' -T fields -e tcp.analysis.ack_rtt
(where I used the display filter after -Y to restrict the analysis to only one remote host)
If you are using wireshark , it show the iRtt =initial Round Trip and the Rtt of each sent packet , just look at "show packet in new window /seq/ack analyses "
Related
Suppose we have 2 nodes, A and B, directly connected by Internet (we can ignore the underlyng network eg, routers, ISP etc).
We know RTT between nodes (80ms)
We know packet loss (0.1)
We know jitter (1ms)
We know bandwith, A=100/10mbps B=50/5mbps (first value is download, second is upload)
A sends a 1GB file to B by using the TCP protocol (with 64KB segment size).
How many times they need to exchange the file?
How many times it takes to do the same thing by using the UDP
protocol?
EDIT:
i guess the main difference in the calculation between UDP and TCP is that in TCP we need to wait for every packet to be sent before sending the next one. Or, in other words, we have to add in the delay calculation one RTT for every packet. Moreover, packetloss is not considered at all in UDP. I am not sure of what I'm sayng in this edit, so let me know if I'm wrong.
I should find a combination of the lenght of a packet, bandwidth and the link lenght and then find out if there is a combination for which transmitting time of a packet finishes before the first bit of the packet has reached the receiver. Is this even possible?
TCP or UDP?
TCP will require to receive a response from the destination before it actually starts sending the packet thus it won't be possible here.
UDP has no concept of knowing whether or not the packet got received, which means that as soon as the packet has left the sender there is no further communication between the two.
Your question is worded ambiguously though: how can you talk about 'transmission time' (which implies the time between sending and receiving) while comparing that to the 'receiving time' (which is already part of the transmission time).?
We are doing some load testing on our servers and I'm using tshark to capture some data to a pcap file then using the wireshark GUI to see what errors or warnings are showing up by going to Analyze -> expert Info with my pcap loaded in..
I'm seeing various things that I'm not sure or do not completely understand yet..
Under Warnings I have:
779 Warnings for TCP: ACKed segment that wasn't captured (common at capture start)
446 TCP: Previous segment not captured (common at capture start)
An example is :
40292 0.000 xxx xxx TCP 90 [TCP ACKed unseen segment] [TCP Previous segment not captured] 11210 > 37586 [PSH, ACK] Seq=3812 Ack=28611 Win=768 Len=24 TSval=199317872 TSecr=4506547
We also ran the pcap file though a nice command that creates a command line column of data
command
tshark -i 1 -w file.pcap -c 500000
basically just saw a few things in the tcp.analysis.lost_segment column but not many..\
Anyone enlighten what might be going on? tshark not able to keep up with writing data, some other issue? False positive?
That very well may be a false positive. Like the warning message says, it is common for a capture to start in the middle of a tcp session. In those cases it does not have that information. If you are really missing acks then it is time to start looking upstream from your host for where they are disappearing. It is possible that tshark can not keep up with the data and so it is dropping some metrics. At the end of your capture it will tell you if the "kernel dropped packet" and how many. By default tshark disables dns lookup, tcpdump does not. If you use tcpdump you need to pass in the "-n" switch. If you are having a disk IO issue then you can do something like write to memory /dev/shm. BUT be careful because if your captures get very large then you can cause your machine to start swapping.
My bet is that you have some very long running tcp sessions and when you start your capture you are simply missing some parts of the tcp session due to that. Having said that, here are some of the things that I have seen cause duplicate/missing acks.
Switches - (very unlikely but sometimes they get in a sick state)
Routers - more likely than switches, but not much
Firewall - More likely than routers. Things to look for here are resource exhaustion (license, cpu, etc)
Client side filtering software - antivirus, malware detection etc.
Another cause of "TCP ACKed Unseen" is the number of packets that may get dropped in a capture. If I run an unfiltered capture for all traffic on a busy interface, I will sometimes see a large number of 'dropped' packets after stopping tshark.
On the last capture I did when I saw this, I had 2893204 packets captured, but once I hit Ctrl-C, I got a 87581 packets dropped message. Thats a 3% loss, so when wireshark opens the capture, its likely to be missing packets and report "unseen" packets.
As I mentioned, I captured a really busy interface with no capture filter, so tshark had to sort all packets, when I use a capture filter to remove some of the noise, I no longer get the error.
Acked Unseen sampleHi guys! Just some observations from what I just found in my capture:
On many occasions, the packet capture reports “ACKed segment that wasn't captured” on the client side, which alerts of the condition that the client PC has sent a data packet, the server acknowledges receipt of that packet, but the packet capture made on the client does not include the packet sent by the client
Initially, I thought it indicates a failure of the PC to record into the capture a packet it sends because “e.g., machine which is running Wireshark is slow” (https://osqa-ask.wireshark.org/questions/25593/tcp-previous-segment-not-captured-is-that-a-connectivity-issue)
However, then I noticed every time I see this “ACKed segment that wasn't captured” alert I can see a record of an “invalid” packet sent by the client PC
In the capture example above, frame 67795 sends an ACK for 10384
Even though wireshark reports Bogus IP length (0), frame 67795 is
reported to have length 13194
Frame 67800 sends an ACK for 23524
10384+13194 = 23578
23578 – 23524 = 54
54 is in fact length of the
Ethernet / IP / TCP headers (14 for Ethernt, 20 for IP, 20 for TCP)
So in fact, the frame 67796 does represent a large TCP packets (13194
bytes) which operating system tried to put on the wore
NIC driver will fragment it into smaller 1500 bytes pieces in order to transmit over the network
But Wireshark running on my PC fails to understand it is a valid packet and parse it. I believe Wireshark running on 2012 Windows
server reads these captures correctly
So after all, these “Bogus IP
length” and “ACKed segment that wasn't captured” alerts were in fact
false positives in my case
I just came across this and would like to share my observation of TCP ACKed unseen segment. In my case the client was trying to initiate connection on same source port and destination port it used previously and thus the server was confused and replied with old TCP SEQ number saying I havent seen this new packet
How to find UDP packet's round trip time from Wireshark tool??i am getting lot of upd packets at a time.So i want to find for each packet,how long it take to send a resposne.Is any other tool for accomplishing this??
Wireshark can not help in this, since it only records time it sees UDP packet sent.
Try ping remote_host_IP_addr - it gives statistics regarding RTT (round trip time)
I've been researching TCP congestion control recently, however one question plagues me...
If I understand everything correctly, TCP will not send NEW data unless allowed by the cwnd (congestion window) and rwnd (the receiving side's window). In other words:
if(flightSize < MIN(cwnd, rwnd))
{
// Send some new data (if possible)
// Taking into account other details that we don't need
// to get into such as Nagle's algorithm, etc.
}
Where flightSize is the amount of data that has been sent but not yet acknowledged.
Let us assume that TCP is going along, sending data, and increasing cwnd as appropriate. Let's say cwnd = [10 full packets], and the flightSize == cwnd. Then packet loss occurs in the network, and the sender's retransmission timer goes off. How/When does New Reno retransmit the unacknowledged data?
Here's my current understanding/misunderstanding:
When the timer goes off, the cwnd will be reset to [1 full packet], the oldest sent but unacknowledged packet will be resent, the rto will be doubled, and the retransmission timer will be reset. So if we say the rto was 1 second when the timer went off, it will get updated to 2 seconds, and the retransmission timer will get started again with a wait time of 2 seconds.
Here is why I'm confused:
In the above situation, TCP will resend only a single packet. Even if that packet gets ACKed right away, TCP cannot send any NEW data because cwnd is still less than the flightSize. So what does it do? Sit around and wait until the 2 second retransmission timer goes off again before it resends another packet? Does it force a resend of the old data since it can't send new data? Does it reset the flightSize, and reconsider all previously sent data to be unsent?
I've read all the RFC's I could find, and all kinds of guides and explanations of TCP. I must have missed something somewhere...
Clarification:
I was considering multiple losses, where TCP is not using SACK.
If duplicate acks are received, TCP will resend the oldest ack on the 3rd duplicate ack (fast retrasmit) and will send new data on and after the 4th duplicate ack (fast recovery). My question concerns what happens if the TCP sender gets less than 3 dup acks?
I found the answer in the book "TCP/IP Illustrated, Volume 2", section 25.11, pages 842-844:
[On a retransmission timeout] the next
send sequence number (snd_nxt) is set
to the oldest unacknowledged sequence
number (snd_una). ... By moving
snd_nxt back, [TCP can begin to
retransmit all unacknowledged data].
In other words, the flightSize will get reset, so data can continue to be sent (in slow start mode). It's just that some of this data may be data that has already been sent before. A cumulative ack might come along that prevents all data from being resent though.
Request for clarification: are you considering a single packet loss? Or multiple losses within a window?
In a single loss case, there will be duplicate acknowledgements received because of packets received after the lost one. I believe New Reno will transmit subsequent packets ("NEW data") in response to the duplicate acks. This then resets the timeout timer.