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)
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).?
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 "
I want to get the inter-arrival times of the network packets. I want to use these arrival times for predicting the arrival time of future packet (probably by using Bayesian classifier). Can someone suggest how can I get the inter-arrival times of incoming packets? I don't see any such option in wireshark. Any help will be appreciated.
The pcap (packet capture) API should allow you to get this information.
Here is some example code: link.
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.