Is there any tool existing or any shortcut that can be used to extract print job payload from IPP packets(captured via wireshark utility)?
Write now I'm manually inspecting/dissecting the packets byte per byte which is kind of excruciating especially with requests which spans several packet exchanges (HTTP Continue).
Thanks
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I have two apps sending tcp packages, both written in python 2. When client sends tcp packets to server too fast, the packets get concatenated. Is there a way to make python recover only last sent package from socket? I will be sending files with it, so I cannot just use some character as packet terminator, because I don't know the content of the file.
TCP uses packets for transmission, but it is not exposed to the application. Instead, the TCP layer may decide how to break the data into packets, even fragments, and how to deliver them. Often, this happens because of the unterlying network topology.
From an application point of view, you should consider a TCP connection as a stream of octets, i.e. your data unit is the byte, not a packet.
If you want to transmit "packets", use a datagram-oriented protocol such as UDP (but beware, there are size limits for such packets, and with UDP you need to take care of retransmissions yourself), or wrap them manually. For example, you could always send the packet length first, then the payload, over TCP. On the other side, read the size first, then you know how many bytes need to follow (beware, you may need to read more than once to get everything, because of fragmentation). Here, TCP will take care of in-order delivery and retransmission, so this is easier.
TCP is a streaming protocol, which doesn't expose individual packets. While reading from stream and getting packets might work in some configurations, it will break with even minor changes to operating system or networking hardware involved.
To resolve the issue, use a higher-level protocol to mark file boundaries. For example, you can prefix the file with its length in octets (bytes). Or, you can switch to a protocol that already handles this kind of stuff, like http.
First you need to know if the packet is combined before it is sent or after. Use wireshark to check it the sender is sending one packet or two. If it is sending one, then your fix is to call flush() after each write. I do not know the answer if the receiver is combining packets after receiving them.
You could change what you are sending. You could send bytes sent, followed by the bytes. Then the other side would know how many bytes to read.
Normally, TCP_NODELAY prevents that. But there are very few situations where you need to switch that on. One of the few valid ones are telnet style applications.
What you need is a protocol on top of the tcp connection. Think of the TCP connection as a pipe. You put things in one end of the pipe and get them out of the other. You cannot just send a file through this without both ends being coordinated. You have recognised you don't know how big it is and where it ends. This is your problem. Protocols take care of this. You don't have a protocol and so what you're writing is never going to be robust.
You say you don't know the length. Get the length of the file and transmit that in a header, followed by the number of bytes.
For example, if the header is a 64bits which is the length, then when you receive your header at the server end, you read the 64bit number as the length and then keep reading until the end of the file which should be the length.
Of course, this is extremely simplistic but that's the basics of it.
In fact, you don't have to design your own protocol. You could go to the internet and use an existing protocol. Such as HTTP.
I'm running some software that sniffs local packets, then encapsulates the full, raw packet data in a TCP packet and sends it to the server.
What I'd like to do is have the server receive the packet data and put it into a PCAP file for download by anyone who is connected to the server.
Basically what I need is some inspired information regarding the PCAP file format and how to make entries into a PCAP file using raw packet data without using a packet sniffer.
Can this be done and if so how?
Impossible to tell without looking at your data and knowing what tool you are using to capture the packet.
It might be as simple as adding a global header.
The global header contains the magic number, GMT offset, timestamp
precision, the maximum length of captured packets (in octets), and the
data link type. This information is followed by zero or more records
of captured packet data.
If you are using tcpdump then the output might already be pcap compatible. If not, just install tcpdump-libpcap and then use that version.
EXAMPLE PCAP STRUCTURE:
SOURCE: https://www.lesliesikos.com/pcap/
For some example code with PCAP headers, check out this Python program:
https://www.codeproject.com/Tips/612847/Generate-a-quick-and-easy-custom-pcap-file-using-P
I have captured SNMP traps/informs from my network (a mix of V1 and V2c) and I wish to write a camel pipeline to replay them in order to test my trap processing engine. In order to do this, I must resend the traps with the source IP of the original sender (since that is part of the criteria for identification and correct processing of a trap).
I thought I would send the resulting UDP datagrams using Netty (although I'm open to writing a stand-alone component or using Mina or any other approach, but the use of the Camel SNMP component does not immediately seem appropriate). I have implemented similar functionality in Python, and I needed to write to a raw socket. I have looked through the netty component source in camel and was not able to see how I might use it unmodified to use a raw socket.
Does anyone out there have an example of using netty with raw sockets that they could share, bonus points if it includes some reference to camel or better still if it calls out a way I can do this with something more high-level than a raw socket (ie a regular UDP datagram with some kind of modifier to set the source IP).
Many thanks
I am developing a client server application where the client application sends different types of messages to server. One type message consists of file descriptor that is to be passed between processes.
Generally on Posix API pages, not much information is found about sendmsg and recvmsg. My question is if the sent message is too big that cannot be sent atomically, will the attached file descriptor be sent for each pieces of the message, or just first one?
Why this confuses me is that on connected sockets, if messages are sent too quickly, kernel is merging messages to each other, then file descriptors (integer number) must be merged aligned with messages as well.
UNIX domain sockets support passing file descriptors or process credentials to ... The send(2) MSG_MORE flag is not supported by UNIX domain sockets. ... For historical reasons the ancillary message types listed below are specified with ... To pass file descriptors or credentials over a SOCK_STREAM
I know that it is possible that multiple packets would be stacked to the buffer to be read from and that a long packet might require a loop of multiple send attempts to be fully sent. But I have a question about packaging in these cases:
If I call recv (or any alternative (low-level) function) when there are multiple packets awaiting to be read, would it return them all stacked into my buffer or only one of them (or part of the first one if my buffer is insufficient)?
If I send a long packet which requires multiple iterations to be sent fully, does it count as a single packet or multiple packets? It's basically a question whether it marks that the package sent is not full?
These questions came to my mind when I thought about web sockets packaging. Special characters are used to mark the beginning and end of a packet which sorta leads to a conclusion that it's not possible to separate multiple packages.
P.S. All the questions are about TCP/IP but you are welcomed to share information (answers) about UDP as well.
TCP sockets are stream based. The order is guaranteed but the number of bytes you receive with each recv/read could be any chunk of the pending bytes from the sender. You can layer a message based transport on top of TCP by adding framing information to indicate the way that the payload should be chunked into messages. This is what WebSockets does. Each WebSocket message/frame starts with at least 2 bytes of header information which contains the length of the payload to follow. This allows the receiver to wait for and re-assemble complete messages.
For example, libraries/interfaces that implement the standard Websocket API or a similar API (such as a browser), the onmessage event will fire once for each message received and the data attribute of the event will contain the entire message.
Note that in the older Hixie version of WebSockets, each frame was started with '\x00' and terminated with '\xff'. The current standardized IETF 6455 (HyBi) version of the protocol uses the header information that contains the length which allows much easier processing of the frames (but note that both the old and new are still message based and have basically the same API).
TCP connection provides for stream of bytes, so treat it as such. No application message boundaries are preserved - one send can correspond to multiple receives and the other way around. You need loops on both sides.
UDP, on the other hand, is datagram (i.e. message) based. Here one read will always dequeue single datagram (unless you mess with low-level flags on the socket). Event if your application buffer is smaller then the pending datagram and you read only a part of it, the rest of it is lost. The way around it is to limit the size of datagrams you send to something bellow the normal MTU of 1500 (less IP and UDP headers, so actually 1472).