I'm developing a protocol on top of TCP. I'm wondering what is the best way to detect control messages in this case.
Example:
Client sends the control message START.
Server sends data ABCD, then sends the control message END.
Client detects END and closes connection
The issue is that "END" can be fragmented. So the client might end up reading ABCDEN and then D.
What would be the optimal way to detect the control message "END" in this case?
Note: The server does not know how many bytes in total it will be sending!
TCP is a byte stream, it has no concept of message boundaries, so you must delimit your messages in your protocol. The receiver of a message can then look for the delimiter to know where one message ends and the next begins. For a text-based protocol like you showed, CRLF (a carriage return followed by a line feed) is a commonly used delimiter in many text-based Internet protocols, including HTTP, FTP, POP3, SMTP, etc. In the case of variable-length data, you still need an end-of-data delimiter if you can't send the data size ahead of time. In your example, sending "\r\nEND\r\n" after the data would act as that delimiter.
Related
To generate error on CAN, I've done change data field. But It seems just change numerical things.
I want to know how to modify ACK or CRC field to inject error.
Can I change that field with software?
No, you cannot change that from software, since that part of the message is always constructed on CAN Communication Controller level and down (Physical Layer).
Basically, the ACK field is not set in SW. It is "completed" by other nodes while the message is sent, and the bitstream arrived to the ACK bit slot.
The CRC is constructed on Communication Controler level, upon the payload the application wishes to send.
So in order to inject such faults in a CAN message, you need a special HIL (Hardware in the Loop) device, which will forcefully overwrite fields of your choosing.
One such device is a CANSTress from Vector, but there are many others.
Regarding NACK error, you can simulate that without HIL, if you have a simulation environment.Or, simply do not turn on the other nodes on the cluster, ensuring there is no other node to ACK the message. Beware, disconnecting CANH and CANL cables will result in a different error type.
After that I send different runicast messages with the function runicast_send, how can I understand which message was acknowledged when the callback sent_runicast is triggered?
The runicast.h file states:
The runicast primitive adds two packet attributes: the single-hop
packet type and the single-hop packet ID. The runicast primitive
uses the packet ID attribute as a sequence number for matching
acknowledgement packets to the corresponding data packets.
but I didn't understand how to do it in practice. Can somebody provide an example?
One way would be to look at the field sndnxt of struct runicast_conn *c before you send the packet, and then compare that value of the packetbuf_attr(PACKETBUF_ATTR_PACKET_ID) in the "sent" callback of your code.
However note that by default the runicast packet ID is just 2 bits long. Enough to demultiplex the ACK in most cases, but may be insufficient for your purposes. (The packet ID size in bits can be changed by redefining RUNICAST_PACKET_ID_BITS.)
Also Rime is obsolete. Don't use it in your code, especially production code unless you know what you're doing. runicast was never one of the highlights of Rime, I doubt there are no better alternatives (e.g. the uIPv6 stack) for what you want to do.
I am writing two programs (server.vi) and (client.vi). that communicate with each other over a TCP connection.
After the client opens a TCP connection with the server, the server responds with a packet of type "A". The client sends another packet of type "A" back to the server as an acknowledgement. At this point the server starts sending a continous stream of packets of type "B" to the client. And the client starts sending a continous stream of packets with type "C".
This means the sending an receiving of packets with types B and C will be in parallel.
How should I implement something like this in labview?
Here is one idea i have and I was hoping someone could either comment or provide a better suggestion.
the server has two while loops
a. first while loop consists of a TCP read function that receives packets of type "C".
b. second while loop consists of a TCP write function that sends packets of type "B"
the client has two while loops
a. first while loop consists of a TCP write function that sends packets of type "C"
b. second while loop consists of a TCP read function that receives packets of type "B".
This way we are sending and receiving packets of type "B" and "C" in parallel.
All the while loops are independent of each other and are essentially infinite unless both client and server programs are stopped.
Does this make any sense? Is there a more clever / better aproach to doing this?
That sounds like the appropriate way to have two processes run in parallel in LabVIEW, yes.
Have a look at the examples that come with LabVIEW - in LV 2012 there's a 'TCP Communicator - Active.vi' (Help->Find Examples->Networking->TCP & UDP) that looks like it does something similar to what you're describing.
You need to figure out when and how to stop each loop - the example above uses a local variable but you could also do it with a notifier, for example.
I tried using PARSE on a PORT! and it does not work:
>> parse open %test-data.r [to end]
** Script error: parse does not allow port! for its input argument
Of course, it works if you read the data in:
>> parse read open %test-data.r [to end]
== true
...but it seems it would be useful to be able to use PARSE on large files without first loading them into memory.
Is there a reason why PARSE couldn't work on a PORT! ... or is it merely not implemented yet?
the easy answer is no we can't...
The way parse works, it may need to roll-back to a prior part of the input string, which might in fact be the head of the complete input, when it meets the last character of the stream.
ports copy their data to a string buffer as they get their input from a port, so in fact, there is never any "prior" string for parse to roll-back to. its like quantum physics... just looking at it, its not there anymore.
But as you know in rebol... no isn't an answer. ;-)
This being said, there is a way to parse data from a port as its being grabbed, but its a bit more work.
what you do is use a buffer, and
APPEND buffer COPY/part connection amount
Depending on your data, amount could be 1 byte or 1kb, use what makes sense.
Once the new input is added to your buffer, parse it and add logic to know if you matched part of that buffer.
If something positively matched, you remove/part what matched from the buffer, and continue parsing until nothing parses.
you then repeat above until you reach the end of input.
I've used this in a real-time EDI tcp server which has an "always on" tcp port in order to break up a (potentially) continuous stream of input data, which actually piggy-backs messages end to end.
details
The best way to setup this system is to use /no-wait and loop until the port closes (you receive none instead of "").
Also make sure you have a way of checking for data integrity problems (like a skipped byte, or erroneous message) when you are parsing, otherwise, you will never reach the end.
In my system, when the buffer was beyond a specific size, I tried an alternate rule which skipped bytes until a pattern might be found further down the stream. If one was found, an error was logged, the partial message stored and a alert raised for sysadmin to sort out the message.
HTH !
I think that Maxim's answer is good enough. At this moment the parse on port is not implemented. I don't think it's impossible to implement it later, but we must solve other issues first.
Also as Maxim says, you can do it even now, but it very depends what exactly you want to do.
You can parse large files without need to read them completely to the memory, for sure. It's always good to know, what you expect to parse. For example all large files, like files for music and video, are divided into chunks, so you can just use copy|seek to get these chunks and parse them.
Or if you want to get just titles of multiple web pages, you can just read, let's say, first 1024 bytes and look for the title tag here, if it fails, read more bytes and try it again...
That's exactly what must be done to allow parse on port natively anyway.
And feel free to add a WISH in the CureCode database: http://curecode.org/rebol3/
I am about to write a message protocol going over a TCP stream. The receiver needs to know where the message boundaries are.
I can either send 1) fixed length messages, 2) size fields so the receiver knows how big the message is, or 3) a unique message terminator (I guess this can't be used anywhere else in the message).
I won't use #1 for efficiency reasons.
I like #2 but is it possible for the stream to get out of sync?
I don't like idea #3 because it means receiver can't know the size of the message ahead of time and also requires that the terminator doesn't appear elsewhere in the message.
With #2, if it's possible to get out of sync, can I add a terminator or am I guaranteed to never get out of sync as long as the sender program is correct in what it sends? Is it necessary to do #2 AND #3?
Please let me know.
Thanks,
jbu
You are using TCP, the packet delivery is reliable. So the connection either drops, timeouts or you will read the whole message.
So option #2 is ok.
I agree with sigjuice.
If you have a size field, it's not necessary to add and end-of-message delimiter --
however, it's a good idea.
Having both makes things much more robust and easier to debug.
Consider using the standard netstring format, which includes both a size field and also a end-of-string character.
Because it has a size field, it's OK for the end-of-string character to be used inside the message.
If you are developing both the transmit and receive code from scratch, it wouldn't hurt to use both length headers and delimiters. This would provide robustness and error detection. Consider the case where you just use #2. If you write a length field of N to the TCP stream, but end up sending a message which is of a size different from N, the receiving end wouldn't know any better and end up confused.
If you use both #2 and #3, while not foolproof, the receiver can have a greater degree of confidence that it received the message correctly if it encounters the delimiter after consuming N bytes from the TCP stream. You can also safely use the delimiter inside your message.
Take a look at HTTP Chunked Transfer Coding for a real world example of using both #2 and #3.
Depending on the level at which you're working, #2 may actually not have an issues with going out of sync (TCP has sequence numbering in the packets, and does reassemble the stream in correct order for you if it arrives out of order).
Thus, #2 is probably your best bet. In addition, knowing the message size early on in the transmission will make it easier to allocate memory on the receiving end.
Interesting there is no clear answer here. #2 is generally safe over TCP, and is done "in the real world" quite often. This is because TCP guarantees that all data arrives both uncorrupted* and in the order that it was sent.
*Unless corrupted in such a way that the TCP checksum still passes.
Answering to old message since there is stuff to correnct:
Unlike many answers here claim, TCP does not guarantee data to arrive uncorrupted. Not even practically.
TCP protocol has a 2-byte crc-checksum that obviously has a 1:65536 chance of collision if more than one bit flips. This is such a small chance it will never be encountered in tests, but if you are developing something that either transmits large amounts of data and/or is used by very many end users, that dice gets thrown trillions of times (not kidding, youtube throws it about 30 times a second per user.)
Option 2: size field is the only practical option for the reasons you yourself listed. Fixed length messages would be wasteful, and delimiter marks necessitate running the entire payload through some sort of encoding-decoding stage to replace at least three different symbols: start-symbol, end-symbol, and the replacement-symbol that signals replacement has occurred.
In addition to this one will most likely want to use some sort of error checking with a serious checksum. Probably implemented in tandem with the encryption protocol as a message validity check.
As to the possibility of getting out of sync:
This is possible per message, but has a remedy.
A useful scheme is to start each message with a header. This header can be quite short (<30 bytes) and contain the message payload length, eventual correct checksum of the payload, and a checksum for that first portion of the header itself. Messages will also have a maximum length. Such a short header can also be delimited with known symbols.
Now the receiving end will always be in one of two states:
Waiting for new message header to arrive
Receiving more data to an ongoing message, whose length and checksum are known.
This way the receiver will in any situation get out of sync for at most the maximum length of one message. (Assuming there was a corrupted header with corruption in message length field)
With this scheme all messages arrive as discrete payloads, the receiver cannot get stuck forever even with maliciously corrupted data in between, the length of arriving payloads is know in advance, and a successfully transmitted payload has been verified by an additional longer checksum, and that checksum itself has been verified. The overhead for all this can be a mere 26 byte header containing three 64-bit fields, and two delimiting symbols.
(The header does not require replacement-encoding since it is expected only in a state whout ongoing message, and the entire 26 bytes can be processed at once)
There is a fourth alternative: a self-describing protocol such as XML.