I am having difficulty about understanding tcp traffic between server working on 8000 port on my pc and my browser. I am trying to display an image in my browser that is hosted on this server. While loading the browser page, I started the wireshark and captured the traffic. My goal is to confirm that the transferred data with tcp is equal to 2621 bytes which is the size of the actual image.
Here is the traffic:
Here, I thought that the packet in which the image is transferred is this boxed packet. But its size is 262 bytes and not equal to actual image size. How can figure out in which packets the image is tranferred and how can I see its size? Thanks for helping.
Use Fiddler Classic to inspect HTTP/HTTPS traffic to and from browsers.
https://www.telerik.com/fiddler/fiddler-classic
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When call is made between two chan_mobile channels, all works fine.
But not when call is established between SIP and chan_mobile (through simple bridge).
SIP -> mobile is clear and fine with controllable packet size in sip.conf
Reverse direction voice going in too small packets causing near 180 kbits/s and loosing packets because of out of order on SIP phone side (if not enabled permissive mode on RTP).
So question is how to increase packets size originated by chan_mobile and going to SIP?
Setting like
allow=ulaw:100
not works in chan_mobile.conf
Chan mobile to chan_sip connection is via sln internal bridge ALWAYS.
You can control only outgoing packets(from your server sip to remote sip) by set
allow=g729:20
Or similar form. Please note, that codec should support length of packet you asking.
You have no any way control other side, so no way control inbound direction.
For more info see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
I have a big PCAP file downloaded from this website. (You can download the original pcap file : 368 MBs)
You can also download a short version that contains only some of the buggy packets.
There is something strange with some packets inside this file. There are 1113 packets labeled with sFlow inside it that it doesn't matters which wireshark-filter you apply to the packets, you always will see them (or part of them) in the window:
To be more clear let see some screen shots:
No filter applied:
Filter to see tcp packets only:
Filter to see udp packets only:
Filter to see packets with ip.addr == 68.64.21.64
What's wrong with these packets?
These packets are of sFlow type. They are used for network sampling, so they contain samples of other network packets within. Display filter seems to be applied not only to sFlow packet itself, but to every inner packet as well. So "tcp" display filter leaves those sFlow packets (they are obviously udp), which contain tcp sample within. Same for address filtering.
You can inspect inner packets as shown on the picture
Not sure if filter behavior is correct, I was amused by output as well. I think, it'd be great to open ticket at Wireshark bug database to hear developers' opinion.
When the client initiates the connection with the SYN bit set, Wireshark (and TCPDump) show the MSS as being 1460. However, when the same packet is delivered to the host, Wireshark (and TCPDump) show the MSS as being 1416.
Can anybody please explain why there's a discrepancy of 44 bytes?
The image below shows the MSS received by the host. Sorry but I don't have a screenshot showing the client's initial SYN 1460 MSS.
During actual data transfer, the 1416 is used as an MSS (1404 for payload and 12 for options such as the TSVal)
My original thought was that it has something to do with Path MTU discovery, and that some space is being reserved for any additional headers that may be added on while the packet is making it's way from the sender to the destination. Am I correct in thinking so? If so, is there a way to find a breakdown of how these are being used?
After consulting the university's network admin, we concluded that that a lower MSS was being imposed by the network for load reasons.
I'm using lwip stack on my embedded platform. I have connected the board to my PC via ethernet. My application running on board, dumps the image data out of ethernet. PC applications waits for header, after header it decodes the data and displays the image.
This is for debug purpose only. My images are 4MBytes and i receive 20 Frames per second. So it will be 80MBytes data per second.
Is is advisable to use TCP or UDP?
I tried using TCP, but my send buffers becomes full and it will wait around 200ms to receive acknowledge. Mean time i loose 5-6 images coming from sensor. Can this be fixed if i use UDP?
Thanks,
Sathya
I suggest you apply some kind of compression to your images before sending them to the network.
That said, if you use UDP, you may get better transferrate, but you do need receiving code that can handle lost packets (discard image or request resend or pad affected area)
I'm trying to reverse engineer an application, and i need help understanding how TCP window size works. My MTU is 1460
My application transfers a file using TCP from point A to B. I know the following:
The file is split into segments of size 8K
Each segment is compressed
Then each segment is sent to point B over TCP. These segment for a text file can be of size 148 Bytes, and for a pdf 6000 Bytes.
For a text file, am i supposed to see the segments of 148 attached to one another to form one large TCP stream? and then it is split according to the Window Size?
Any help is appreciated.
The receiver application should see the data in teh same way, the sender application sent it. TCP uses byte-streaming and so it collects all the bytes in an in-order manner and delivers it to the application. MTU is largely an internal semantics to TCP and does not take into application-layer packet boundaries. If TCP has enough data to send in its send buffer (each TCP socket has its own send buffer, btw), then it will package its next segment worth MTU size and sends it; to be more precise, it deducts TCP and IP header from the MTU size.