I want to send variable sized packets between 2 linux OSes over an internal network. The packet is variable sized and its length and CRC are indicated in the header which is also sent along with the packet. Something roughly like-
struct hdr {
uint32 crc;
uint32 dataSize;
void *data;
};
I'm using CRC at the application layer to overcome the inherent limitation of TCP checksums
The problem I have is, there is a chance that the dataSize field itself is corrupted, in which case, I dont know where the next packet starts? Cos at the reciver, when I read the socket buffer, I read n such packets next to one another. So dataSize is the only way I can get to the next packet correctly.
Some ideas I have is to-
Restart the connection if a CRC mismatch occurs.
Aggregate X such packets into one big packet of fixed size and discard the big packet if any CRC error is detected. The big packet is to make sure we lose <= sizeof of one packet in case of errors
Any other ideas for these variable sized packets?
Since TCP is stream based, data length is the generally used way to extract one full message for processing at the application. If you believe that the length byte itself is wrong for some reason, there is not much we can do except discard the packet,"flush" the connection and expect that the sender and receiver would re-sync. But the best is to disconnect the line unless, there is a protocol at the application layer to get to re-sync the connection.
Another method other than length bytes would be to use markers. Start-of-Message and End-of-Message. Application when encountering Start-of-Message should start collecting data until End-of-Message byte is received and then further process the message. This requires that the message escapes the markers appropriately.
I think that you are dealing with second order error possibilities, when major risk is somewhere else.
When we used serial line transmissions, errors were frequent (one or two every several kBytes). We used good old Kermit with a CRC and a packet size of about 100 bytes and that was enough: I encountered many times a failed transfer because the line went off, but never a correct transfer with a bad file.
With current networks, unless you have very very poor lines, the hardware level is not that bad, and anyway the level 2 data link layer already has a checksum to control that each packet was not modified between 2 nodes. HDLC is commonly used at that level and it uses normaly a CRC16 or CRC32 checksum which is a very correct checksum.
So the checksum as TCP level is not meant to detect random errors in the byte stream, but simply as a last defense line for unexpected errors, for exemple if a router gets mad because of an electrical shock and sends full garbage. I do not have any statistical data on it, but I am pretty sure that the number of errors reaching the TCP level is already very very low. Said differently, do not worry about that: unless you are dealing with highly sensitive data - and in that case I would prefere to have two different channels, former for data, latter for a global checksum - TCP/IP is enough.
That being said, adding a control at the application level as an ultime defense is perfectly acceptable. It will only process the errors that could have been undetected at data link and TCP level, or more probably errors in the peer application (who wrote it and how was it tested?). So the probability to get an error is low enough to use a very rough recovery procedure:
close the connection
open a new one
restart after last packet correctly exchanged (if it makes sense) or simply continue sending new packets if you can
But the risk is much higher to get a physical disconnection, or a power outage anywhere in the network, not speaking in a flaw in application level implementations...
And do not forget to fully specify the byte order and the size of the crc and datasize...
Related
When will a TCP packet be fragmented at the application layer? When a TCP packet is sent from an application, will the recipient at the application layer ever receive the packet in two or more packets? If so, what conditions cause the packet to be divided. It seems like a packet won't be fragmented until it reaches the Ethernet (at the network layer) limit of 1500 bytes. But, that fragmentation will be transparent to the recipient at the application layer since the network layer will reassemble the fragments before sending the packet up to the next layer, right?
It will be split when it hits a network device with a lower MTU than the packet's size. Most ethernet devices are 1500, but it can often be smaller, like 1492 if that ethernet is going over PPPoE (DSL) because of the extra routing information, even lower if a second layer is added like Windows Internet Connection Sharing. And dialup is normally 576!
In general though you should remember that TCP is not a packet protocol. It uses packets at the lowest level to transmit over IP, but as far as the interface for any TCP stack is concerned, it is a stream protocol and has no requirement to provide you with a 1:1 relationship to the physical packets sent or received (for example most stacks will hold messages until a certain period of time has expired, or there are enough messages to maximize the size of the IP packet for the given MTU)
As an example if you sent two "packets" (call your send function twice), the receiving program might only receive 1 "packet" (the receiving TCP stack might combine them together). If you are implimenting a message type protocol over TCP, you should include a header at the beginning of each message (or some other header/footer mechansim) so that the receiving side can split the TCP stream back into individual messages, either when a message is received in two parts, or when several messages are received as a chunk.
Fragmentation should be transparent to a TCP application. Keep in mind that TCP is a stream protocol: you get a stream of data, not packets! If you are building your application based on the idea of complete data packets then you will have problems unless you add an abstraction layer to assemble whole packets from the stream and then pass the packets up to the application.
The question makes an assumption that is not true -- TCP does not deliver packets to its endpoints, rather, it sends a stream of bytes (octets). If an application writes two strings into TCP, it may be delivered as one string on the other end; likewise, one string may be delivered as two (or more) strings on the other end.
RFC 793, Section 1.5:
"The TCP is able to transfer a
continuous stream of octets in each
direction between its users by
packaging some number of octets into
segments for transmission through the
internet system."
The key words being continuous stream of octets (bytes).
RFC 793, Section 2.8:
"There is no necessary relationship
between push functions and segment
boundaries. The data in any particular
segment may be the result of a single
SEND call, in whole or part, or of
multiple SEND calls."
The entirety of section 2.8 is relevant.
At the application layer there are any number of reasons why the whole 1500 bytes may not show up one read. Various factors in the internal operating system and TCP stack may cause the application to get some bytes in one read call, and some in the next. Yes, the TCP stack has to re-assemble the packet before sending it up, but that doesn't mean your app is going to get it all in one shot (it is LIKELY will get it in one read, but it's not GUARANTEED to get it in one read).
TCP tries to guarantee in-order delivery of bytes, with error checking, automatic re-sends, etc happening behind your back. Think of it as a pipe at the app layer and don't get too bogged down in how the stack actually sends it over the network.
This page is a good source of information about some of the issues that others have brought up, namely the need for data encapsulation on an application protocol by application protocol basis Not quite authoritative in the sense you describe but it has examples and is sourced to some pretty big names in network programming.
If a packet exceeds the maximum MTU of a network device it will be broken up into multiple packets. (Note most equipment is set to 1500 bytes, but this is not a necessity.)
The reconstruction of the packet should be entirely transparent to the applications.
Different network segments can have different MTU values. In that case fragmentation can occur. For more information see TCP Maximum segment size
This (de)fragmentation happens in the TCP layer. In the application layer there are no more packets. TCP presents a contiguous data stream to the application.
A the "application layer" a TCP packet (well, segment really; TCP at its own layer doesn't know from packets) is never fragmented, since it doesn't exist. The application layer is where you see the data as a stream of bytes, delivered reliably and in order.
If you're thinking about it otherwise, you're probably approaching something in the wrong way. However, this is not to say that there might not be a layer above this, say, a sequence of messages delivered over this reliable, in-order bytestream.
Correct - the most informative way to see this is using Wireshark, an invaluable tool. Take the time to figure it out - has saved me several times, and gives a good reality check
If a 3000 byte packet enters an Ethernet network with a default MTU size of 1500 (for ethernet), it will be fragmented into two packets of each 1500 bytes in length. That is the only time I can think of.
Wireshark is your best bet for checking this. I have been using it for a while and am totally impressed
I'm writing a c program that sends the output of a bash shell over a tcp connection. To make my program more responsive, I used setsockopt() to enable TCP_NODELAY, which disables Nagle's buffering algorithm. This worked great, except rarely there is a lag in large messages. As in, if the message is more than around 500 bytes (probably 512). The first 500 bytes will go through (quickly in small messages), then there'll be a 1-2 second delay before the rest is received all at once. This only happens once every 10-15 times a large message is received. On the server side, the message is being written to the socket one byte at a time, and all of the bytes are available, so this behavior is unexpected to me.
My best guess is that there's a 512 byte buffer somewhere in the socket that's causing a block? I did some time tests to see where the lag is, and I'm pretty sure it's the socket itself where the lag is occurring. All of the data on the server side is written without blocking, but the client receives the end of the message after a lag. However I used getsockopt() to find the socket's receive and send buffers, and they are well over 512 bytes - 66000 and 130000 respectively. On the client side, I'm using express js to receive the data in a handler (app.on('data', function(){})). But I read that this express function does not buffer data?
Would anyone have a guess why this is happening? Thanks!
Since TCP_NODELAY means send every piece of data as a packet as soon as possible without combining data together, it sounds like you are sending tons of packets. Since you are writing one byte at a time it could send packets with just one byte of payload and a much bigger frame. This would work fine most of the time but as soon as the first packet drops for whatever reason the receiver would need to go into error-correction mode on the TCP socket to ask for retransmission of the dropped packet. That would incur at least one round-trip latency and perhaps several. It sounds like you are getting lucky for the first several hundred packets (500 bytes worth) and then typically hitting your first packet drop and slowing way down due to error correction. One simple solution might be to write in larger chunks, say 10 bytes at a time, instead of 1 byte so that the chance of hitting a dropped packet is much less. Then you would expect to see this problem as often as you do only for messages around 5000 bytes or so. In general setting TCP_NODELAY will cause things to go faster at first but wind up hitting the first dropped packet sooner simply because TCP_NODELAY will not decrease the number of packets you send per amount of data. So it increases or leaves the number of packets the same which means your chance of hitting a dropped packet within a certain amount of data will go up. There is a tradeoff here between interactive feel and first hiccup. By avoiding TCP_NODELAY you can delay the typical amount of data that will be sent before the first error retransmission is hit on average.
Get a network capture using tcpdump or wire-shark. Review the packet transmission time line, this will help distinguish network problems from software implementation issues. If you see retransmissions you may have a network issue, if you see slow acks you might find it better to NOT use 'No Delay' since Ack delay can stall a 'No Delay' connection.
I have developed a single Server/multiple Clients udp application, where Server can handle x number of clients at a time. The Server has x number of threads each thread dedicated to one Client.
The code works perfectly fine. Now I want to check my application for all possible scenarios i.e. validate my application. For this purpose, I need to design a test best.
Initial Design:
The test bed I initially designed has following functionalities:
The Server GUI has a button on it. When the button is clicked, the
each thread in the Server reads a text file, picks up few bytes of
the text file, and sends those chunks to its respective clients. The
thread then picks next chunk of bytes from the text file, sends those
chunks to the client and so on until EOF is found.
The Client on the other side keep receiving these chunks of bytes,
creates a text file, and keeps storing these chunks of bytes in its
text file.
When EOF from Server is received, the Client starts sending the
completely received text file back to the Server over its Socket.
When the file is completely received back (echoed), the Server then
compares the two text files, the Sent file and the echoed one. If
both files are same, the communication process has occurred without
any fault and the communication protocol is validated.
The above mentioned validation technique (sending the text file, receiving the echoed file and then comparing both) checks the following things:
The number of bytes sent = number of bytes receieved.
No data is corrupted.
The data is receieved in proper order.
If any of the above mentioned three conditions is not fulfilled, that means that there is some error in communication.
Now I have been asked to make changes to this test bead and add more functionlities to it. Does the procedure that I am using actually can check above mentioned 3 conditions in all scenarios?
Are there some other conditions that must be checked besides above mentioned 3 conditions.
What could be other methods of checking communication protocol except the one I desgined i.e. Sending a text file and getting it echoed and then comparing.
I have to implement more functionlities to his test bed for making validation system more efficient or completely replece the above test bed with some better option.
Please help me with your suggestions.
Thanks in advane :)
The first two of your conditions are guaranteed by UDP. Picking "a few bytes", i.e. anything less than 65535 bytes (64kiB isn't really a "few" bytes) will result in a single datagram being sent, and anything larger than that will fail. Though you will not want to max out the largest possible datagram size, as it will incur IP fragmentation (staying below 1280 bytes is a good idea).
You will be able to receive exactly the amount you sent or nothing at all, never more or less. UDP does not guarantee that any datagram that is sent out arrives (it cannot guarantee that, since IP does not), but it does guarantee that the entire datagram arrives as-is -- or nothing. Never anything in between.
It further guarantees that the data inside the datagram matches its checksum (the underlying protocols including IP/ethernet/ATM further do their own checksumming) and thus arrives in the same binary representation as it was sent. In other words, data arrives in order (inside the datagram) and is not corrupted.
It is of course in theory possible that a bit error passes all 3 layers of checksums, but this is extremely unlikely and will not happen in practice. Unless you need to guard against someone maliciously tampering with packets, you do not need to worry. The kinds of bit errors that happen accidentially are reliably picked up by the checksums used in the protocols.
If, on the other hand, you do need to guard against malicious modification of your data, you must add a MAC (or a checksum and encrypt the entire packet -- adding a checksum alone is useless).
To ensure that data spanning several datagrams arrives in order, you must add sequence numbers to your packets (in the same manner TCP does). And with that, you can as well use TCP, which is likely more efficient and less error-prone. One of the main reasons why one would want to use UDP is normally because in-order delivery and reliability are not needed, or sometimes reliability is needed, but not in-order delivery.
In-order delivery is the main cause of TCP's latency during packet loss (in absence of packet loss, TCP is exactly as "fast" as UDP), so if this is needed, there is no sane reason not to use TCP in the first place. It is a protocol that has been fine-tuned and worked reliably for literally billions of people for 4 decades.
Also, using one socket and one thread per client is possibly not the best approach. The disk won't read any faster, and the network card won't send any faster either. UDP doesn't need a socket per client either. When using TCP, you'll have no other choice but to use one socket per client, but still multiplexing using a readiness notification system will give you much better performance and fewer opportunities for threading errors.
Also, sending back a checksum such as one of the SHA family (or a MAC, if it needs to be secure) may be more efficient than echoing back the whole lot of data. The likelihood that the checksum matches and the data accidentially doesn't is neglegible.
Entire revision control systems that manage millions of lines of code for millions of people (such as git) rely on the fact that this just doesn't happen to identify files (well, it does happen of course, you just won't live to see it).
I have a question here ? Why UDP why not TCP? especially when you are worried for packet order and data corruption. According to me(I may be wrong), UDP is good only when the data is timesensitive like video stream.
Secondly, yes there are other methods of checking integrity of transmitted data. Simplest may be checking the MD5 and SHA1 checksum.
Does the procedure that I am using actually can check above mentioned 3 conditions in all scenarios?
yes
What could be other methods of checking communication protocol except the one I desgined i.e. Sending a text file and getting it echoed and then comparing.
It doesn't have to be a file, but it has to be something you can check once you get the response. You could just generate some random data and hold on to it until you get the response.
You'd have to tell us what you really want to test. If you are trying to make sure that UDP doesn't give you bad data or out of order data, you're using the wrong protocol. You're not testing anything by seeing if you get the exact data in the exact order you send it over UDP except for the networking infrastructure you have in place.
You say you want to test your application for "all possible scenarios", but that doesn't even mean anything. You're testing to see if a behavior that is part of the UDP specification exists and trying to see that it doesn't? Well, it does. Even if you never see it.
I'm writing my own drivers for LPC2148 and a question came to mind.
How do I receive a message of unspecified size in UART?
The only 2 things that come to mind are: 1 - Configure a watchdog and end the receiving when the time runs out. 2- make it so that whenever a meswsage is sent to it there must be an end of message character.
The first choice seems better in my opinion, but I'd like to know if anybody has a better answer, and I know there must be.
Thank you very much
Just give the caller whatever bytes you have received so far. The UART driver shouldn't try to implement the application protocol, the application should do that.
It looks like a wrong use for a watchdog. I ended up with three solutions for this problem:
Use fixed-size packets and DMA; so, you receive one packet per transaction. Apparently, it is not possible in your case.
Receive message char-by-char until the end-of-message character is received. Kind of error-prone, since the EOM char may appear in the data, probably.
Use a fixed-size header before every packet. In the header, store payload size and/or message type ID.
The third approach is probably the best one. You may combine it with the first one, i.e. use DMA to receive header and then data (in the second transaction, after the data size is known from the header). It is also one of the most flexible approaches.
One more thing to worry about is to keep bytestream in sync. There may be rubbish laying in the UART input buffers, which may get read as data, or you can get only a part of a packet after your MCU is powered (i.e. the beginning of the packet had already been sent by that time). To avoid that, you can add magic bytes in your packet header, and probably CRC.
EDIT
OK, one more option :) Just store everything you receive in a growing buffer for later use. That is basically what PC drivers do.
Real embedded uart drivers usually use a ring buffer. Bytes are stored in order and the clients promise to read from the buffer before it's full.
A state machine can then process the message in multiple passes with no need for a watchdog to tell it reception is over
better to go for option 2) append end of transmission character to the transmission string.
but i suggest to add start of transmission also to validate that you are receiving actual transmission.
Watchdog timer is used to reset system when there is a unexpected behavior of device. I think it is better to use a buffer which can store size of data that your application requires.
I am sending data from a linux application through serial port to an embedded device.
In the current implementation a byte circular buffer is used in the firmware. (Nothing but an array with a read and write pointer)
As the bytes come in, it is written to the circular bufffer.
Now the PC application appears to be sending the data too fast for the firmware to handle. Bytes are missed resulting in the firmware returning WRONG_INPUT too mant times.
I think baud rate (115200) is not the issue. A more efficient data structure at the firmware side might help. Any suggestions on choice of data structure?
A circular buffer is the best answer. It is the easiest way to model a hardware FIFO in pure software.
The real issue is likely to be either the way you are collecting bytes from the UART to put in the buffer, or overflow of that buffer.
At 115200 baud with the usual 1 start bit, 1 stop bit and 8 data bits, you can see as many as 11520 bytes per second arrive at that port. That gives you an average of just about 86.8 µs per byte to work with. In a PC, that will seem like a lot of time, but in a small microprocessor, it might not be all that many total instructions or in some cases very many I/O register accesses. If you overfill your buffer because bytes are arriving on average faster than you can consume them, then you will have errors.
Some general advice:
Don't do polled I/O.
Do use a Rx Ready interrupt.
Enable the receive FIFO, if available.
Empty the FIFO completely in the interrupt handler.
Make the ring buffer large enough.
Consider flow control.
Sizing your ring buffer large enough to hold a complete message is important. If your protocol has known limits on the message size, then you can use the higher levels of your protocol to do flow control and survive without the pains of getting XON/XOFF flow to work right in all of the edge cases, or RTS/CTS to work as expected in both ends of the wire which can be nearly as hairy.
If you can't make the ring buffer that large, then you will need some kind of flow control.
There is nothing better than a circular buffer.
You could use a slower baud rate or speed up the application in the firmware so that it can handle data coming at full speed.
If the output of the PC is in bursts it may help to make the buffer big enough to handle one burst.
The last option is to implement some form of flow control.
What do you mean by embedded device ? I think most of current DSP and processor can easily handle this kind of load. The problem is not with the circular buffer, but how do you collect bytes from the serial port.
Does your UART have a hardware fifo ? If yes, then you should enable it. If you have an interrupt per byte, you can quickly get into trouble, especially if you are working with an OS or with virtual memory, where the IRQ cost can be quit high.
If your receiving firmware is very simple (no multitasking), and you don't have an hardware fifo, polled mode can be a better solution than interrupt driven, because then your processor is doing only UART data reception, and you have no interrupt overhead.
Another problem might be with the transfer protocol. For example if you have long packet of data that you have to checksum, and you do the whole checksum at the end of the packet, then all the processing time of the packet is at the end of it, and that is why you may miss the beginning of the next packet.
So circular buffer is fine and you have to way to improve :
- The way you interact with the hardware
- The protocol (packet length, acknoledgment etc ...)
Before trying to solve the problem, first you need to establish what the problem really is. Otherwise you might waste time trying to fix something that isn't actually broken.
Without knowing more about your set-up it's hard to give more specific advice. But you should investigate further to establish what exactly the hardware and software is currently doing when the bytes come in, and then what is the weak point where they're going missing.
A circular buffer with Interrupt driven IO will work on the smallest and slowest of embedded targets.
First try it at the lowest baud rate and only then try at high speeds.
Using a circular buffer in conjunction with IRQ is an excellent suggestion. If your processor generates an interrupt each time a byte is received take that byte and store it in the buffer. How you decide to empty that buffer depends on if you are processing a stream of data or data packets. If you are processing a stream simply have your background process remove the bytes from the buffer and process them first-in-first-out. If you are processing packets then just keep filing the buffer until you have a complete packet. I've used the packet method successfully many times in the past. I would implement some type of flow control as well to signal to the PC if something went wrong like a full buffer or if packet-processing time is long to indicate to the PC when it is ready for the next packet.
You could implement something like IP datagram which contains data length, id, and checksum.
Edit:
Then you could hard-code some fixed length for the packets, for example 1024 byte or whatever that makes sense for the device. PC side would then check if the queue is full at the device every time it writes in a packet. Firmware side would run checksum to see if all data is valid, and read up till the data length.