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.
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I am working on an application that intercepts various kinds of traffic. Recently I have been receiving out-of-order segments. This traffic is over TCP. The SIP header is among multiple segments. I am trying to understand a protocol to be followed to reassemble packets that arrive out of order to be able to display them in my application. To clarify the data is segmented by TCP. By receiving out of order, I mean:
SIP INVITE header first half received later, second-half earlier.
TCP seq and ack are such that the segment received later is expected to be received first.
I would greatly appreciate any leads towards established protocols to implement this.
I suspect you may need to look deeper into your architecture as TCP is designed to deliver packets in order.
One thing in particular to check is whether you are using multiple TCP connections in some way, maybe to boost bandwidth - this could allow out of order delivery if different packets could take different TCP connections, but within a TCP connection delivery should still be in order.
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.
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 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 "
While talking about UDP I saw/heard congestion come up a few times. What does that mean?
congestion is when you are trying to send too much data over a limited bandwidth, it cannot send the data faster than the incoming amount so additional packets are dropped.
When congestion occurs, you can see these effects:
Delay due to the queue at one end of the connection being too big, so it takes time for your packet to be transmitted.
Packet loss when new packets are simply dropped, forcing connection resets (and often causing more congestion).
Lower quality of service, protocols like TCP will do a cutback on the transmission rate, so your throughput will be lowered.
Blocking, certain networks have protocol priorities, so your UDP packets may be dropped in favor of allowing TCP traffic through.
Its like a traffic jam, imagine right after a sports game where a parking lot full of cars is trying to empty out into a small side street.
It means that network-connected devices are attempting to send more data across the network than it can handle, e.g. 20 Mbps of data across a 10 Mbps link.
In context of UDP, it's your main source of lost datagrams under ordinary circumstances.
Most LANs use some sort of a collission detection/avoidance system. A congestion typically means that the amount of data which is being transmiited on the medium is causing enough collissions to deteriorate the quality of service defined for that medium.
You may want to read up CSMA/CD at wikipedia.
As UDP packets can often be broadcasted, congestion can occur more often.
Kind regards,
For instance, Ethernet is a broadband protocol. Once a message is sent, every node receives it but ignores if the packet are not sent to them. What happens when two nodes send a packet at the same time? It will cause a collision and data loss.
So, both of the nodes will have to resend the message. To avoid more collisions, nodes are designed to wait a random number of milliseconds. Otherwise they keep going on sending messages simultaneously and packages will collide forever.