If sendto fails according to the manpage
"On success, these calls return the number of characters sent. On error, -1 is returned, and errno is set appropriately."
I know that with TCP that is definately the case and you should really attempt to send the remaining data as pointed out in Beej's guide to network programming.
However, partially sending a UDP packet makes no sense to me, and this comment seems to imply it.
If the message is too long to pass atomically through the underlying protocol, the error EMSGSIZE is returned, and the message
is not transmitted.
Can someone confirm for me that if I call sendto (or send) with a UDP packet that if it actually doesn't fit in the outbound buffer then I'll get -1 returned with errno set to EMSGSIZE and no partial send as with a stream (TCP) socket?
There is no hidden meaning, the function just returns the count of bytes sent. It is a standard pattern for Unix APIs. Datagrams are all or nothing delivery, receipt is more complicated if the network caused fragmentation to occur but generally the stack hides all the details and presents each complete packet as it is reconstructed.
EMSGSIZE indicates that "the socket requires that the message be sent atomically, but the size of the message to be sent makes this impossible" (see man sendto).
However, the outbound buffer being full isn't necessarily the reason - Linux (for instance) apparently won't fragment UDP packets by default (see man udp).
Related
I am writing a program for receiving udp multicast packet. I came across short read. Is that applicable to udp? How do I ensure that I read one packet at a time? Is that possible?
My packet has a fixed size header followed by a variable length body.
https://www.boost.org/doc/libs/1_60_0/doc/html/boost_asio/overview/core/streams.html#:~:text=When%20a%20short%20read%20or,write()%20and%20async_write()%20.
"I came across short read"
What does that mean? You mean you "heard about it"? Or received the error?
Is that applicable to udp?
No. UDP is connectionless and DATAGRAM.
How do I ensure that I read one packet at a time?
That's how UDP works.
Is that possible?
Something else is not possible, although you can set flags that allow an incomplete datagram when the supplied buffer is too small to receive the complete datagram.
My packet has a fixed size header followed by a variable length body.
SUMMARY
The title of the page you linked was Streams, Short Reads and Short Writes. Your sockets are UDP, that is, not streams.
I ran the client.java only when I filled the form and pressed send button, it was jammed and I could not do anything.
Is there any explanation for this?
enter image description here
TLDR; the User Datagram Protocol (UDP) is "fire-and-forget".
Unreliable – When a UDP message is sent, it cannot be known if it will reach its destination; it could get lost along the way. There is no concept of acknowledgment, retransmission, or timeout.
So if a UDP message is sent and nobody listens then the packet is just dropped. (UDP packets can also be silently dropped due to other network issues/congestion.)
While there could be a prior-error such as resolving the IP for the server (eg. an invalid hostname) or attempting to use an invalid IP, once the UDP packet is out the door, it's out the door and is considered "successfully sent".
Now, if a program is waiting on a response that never comes (ie. the server is down or packet was "otherwise lost") then that could be .. problematic.
That is, this code which requires a UDP response message to continue would "hang":
sendUDPToServerThatNeverResponds();
// There is no guarantee the server will get the UDP message,
// much less that it will send a reply or the reply will get back
// to the client..
waitForUDPReplyFromServerThatWillNeverCome();
Since UDP has no reliability guarantee or retry mechanism, this must be handled in code. For example, in the above maybe the code would wait for 1 second and retry sending a packet, and after 5 seconds of no responses it would report an error to the client.
sendUDPToServerThatMayOrMayNotRespond();
while (i++ < 5) {
reply = waitForUDPReplyForOneSecond();
if (reply)
break;
}
if (reply)
doSomethingAwesome();
else
showErrorToUser();
Of course, "just using TCP" can often make these sorts of tasks simpler due to the stream and reliability characteristics that the Transmission Control Protoocol (TCP) provides. For example, the pseudo-code above is not very robust as the client must also be prepared to handle latent/slow UDP packet arrival from previous requests.
(Also, given the current "screenshot", the code might be as flawed as while(true) {} - make sure to provide an SSCCE and relevant code with questions.)
My understanding is that TCP is considered "reliable" because the receiver acknowledges packet receipt and requests a resend if there is any problem. My file transfer program currently sends files in 32767 byte packets, though I have experimented with all sizes. Sending a 10 meg file that requires 340 packets consistently results in three or four packets on the receiver being significantly smaller than what was sent. I always end up with a file that is very slightly different from the original.
As an example, my log records the size of all packets received:
TCP packet received (32767 bytes)
TCP packet received (32767 bytes)
TCP packet received (14600 bytes)
TCP packet received (32767 bytes)
My sending thread reads the file in 32767 byte chunks and calls a sending sub:
MyFile.Read(Buffer, 0, BufferSize)
SendTCPData(Address, Buffer)
My TCP code is very simple:
Shared Sub SendTCPData(Address As String, ByVal Data As Byte())
Dim Client As New TcpClient(Address, PortNumber)
Dim Stream As NetworkStream = Client.GetStream()
Stream.Write(Data, 0, Data.Length)
Stream.Close()
Client.Close()
End Sub
Can anyone help?
(The post "TCP Client to Server communication" does not deal with with how to handle received packet sizes, which is my question.)
TCP doesn't provide a packet interface to applications, it provides a byte stream interface. TCP does not "glue" bytes together into messages -- it's not a message protocol. If you want code that reads and writes messages using TCP, you'll have to actually write it or use someone else's code that does that.
If you know the sender is sending exactly 32,767 bytes, just keep calling TCP receive functions until you get 32,767 bytes. If you don't know exactly how many bytes the sender is going to send or can't identify the end of the data with some kind of marker, then it's impossible to know when you've gotten all of them.
For the future, before you write any networking code, it's worth taking the effort to document the protocol you're going to use. Take a look at some specifications for existing protocols that layer over TCP (such as SMTP, NNTP, FTP, or HTTP) to see what you need to decide on and document.
If you're sending files over TCP, look closely at some standard for sending files over TCP (such as FTP) and either implement that standard or choose a rational subset of it. At a minimum, reading the standard will give you an idea of the types of decisions you need to make to wind up with a protocol that works. Also, it's essential for debugging -- if the program doesn't work, without a standard it's difficult to determine if it's the server or the client that's at fault because there is no reference to compare them to.
It seems that a UDP packet can be sent without a payload.
The only thing I can think of that doesn't need a payload is for NAT hole punching.
What else could this be used for?
This relates to my previous question Under Linux, can recv ever return 0 on UDP?
I suppose more to the point is that if it's been specified as part of some standard, then it's been thought to be useful somewhere right?
Anything! The UDP packet isn't empty -- it comes with the sender's identity. Therefore, such a packet could be used as a primitive kind of signal: maybe a hello, a goodbye, or a keep-alive.
With interfaces like sendmsg, an empty packet might be used in order to send auxiliary data, like a cmsg structure (which can be used for things like transferring file descriptors between two processes on Linux).
EDIT: One more use: NAT traversal algorithms such as STUN or UDP hole punching.
To answer the question of "why would a protocol do this": the old Daytime protocol just uses the arrival of a UDP packet to send back a reply packet. Similarly, it replies with time value as soon as a TCP connection happens regardless of any actual data that the TCP connection contains.
a UDP packet without payload may be sent to detect if a UDP port is closed. if closed, an ICMP-unreach is replied.
I have an application that transmits some data in a loop.
Underlying protocol is UDP on WinSock. If I don't add sleep(1ms) after each transmit operation most of the data is not sent (or wireshark can not capture it) Have you experienced such a behavour that UDP does not handle repetitive sending in a loop ?
Regards
Tugrul
First thing you should check the return values when you send data to check if data is successfully sent or not.
Second thing, This can happen internal buffer of UDP cannot accommodate more data because previous data is yet not transmitted. So the simplest solution is that each time before send the data you should check if your UDP socket is writable or not. You can do it by calling "select" or "poll" on that UDP socket.