I found this about UDP, could you please explain me its meaning (with examples possibly)
"Packets have definite boundaries which are honored upon receipt, meaning a read operation at the receiver socket will yield an entire message as it was originally sent."
It means that UDP Datagrams are received entire and intact or not at all. They may arrive more than once, or out of order.
You can't describe that as 'streaming'.
Each call to sendto() or sendmsg()sends a single datagram.
Related
Like this figure,enter image description here
I can find many openflow1.3 which are multipart requests in this packet, but I don't know why happened here?
Actually, isn't it only one openflow1.3 here ?
It is related to openflowjava do serialize, wireshark, NIC, tcp nagle's algorithim?
Thanks!
TCP is a byte stream, i.e. packet has no semantic from the perspective of the application layer (i.e. OpenFlow). At the transport level there can be multiple application layer "messages" in a single TCP packet, messages cross packet boundaries etc - it does not matter for the application. While often TCP packet boundaries are also message boundaries because of timing in the application and maybe a disabling of NAGLE algorithm, the assumption that TCP packet boundaries are always message boundaries is wrong and any reliance on this will often cause sporadic and hard to reproduce problems.
And based on this what you see is also not a "multipart request". These are just multiple OpenFlow messages (application level) send at the same time or shortly after each other and they are put together into the same transport level entity (packet) since this way it is less overhead to transport each message.
I'm attempting to make a low-level packet reader, sort of like wireshark, for learning purposes and fun.
Going into http parsing, I quickly found that there are these TCP streams, and that data doesn't all come in one packet.
EDIT: What I consider to be the stream I've been told is not the right term. David Schwartz referred to it as a Query and EJP as a Sequence of Segments.
So I did a little reading on TCP, and I've read everywhere about the sequence and acknowledgment numbers, how they increase and wrap around etc..
Connection establishment and termination I already had knowledge of.
To be honest I've never been too good at interpretation or using google, but I've been at this for a bit, and I just can't seem to understand how the server/client ends a TCP stream.
I'm asking this because, after sending the totality of the http request, the server immediately replies with the page, so, as I've been going back and forward through packets in wireshark and I can't seem to find what delimits this stream (anything indicating length in the first packet or difference between the penultimate and last packets).
Is there no actual delimiter in TCP, and you just get the total length from the http headers's Content-Length?
As seen here, the first 3 packets are connection establishment.
The 3 after those are the sending of the http request (with first being ACK (the selected one), and the latest 2 being PSH & ACK).
The two after those are ACKs from the server, followed by the entirety of the response.
The last one is my ACK and the connection is never closed for the time I "recorded".
Thank you.
The end of a TCP stream is marked by a segment whose FIN bit is sent.
However that isn't the answer to your real question, which is about the end of an HTTP request or response, which is determined by:
The Content-length, if present, or
The final chunk in a chunked transfer, if used, or
The FIN
if there is a body, otherwise the blank line after the headers as pointed out by #Barmar below.
I've heard it said that providing exactly-once delivery is almost impossible. At the same time, TCP is said to provide guaranteed delivery. If TCP does not provide exactly-once guaranteed delivery then does it provide at-most-once or at-least-once
We could say that TCP provides at-least-once delivery and exactly-one processing, with regards to the following definitions:
At-least-once delivery: a TCP message will be delivered at least once to the destination. More specifically, it will keep re-transmitting with specific timeouts if no ACK(knowledgement) is received, so that it will eventually be delivered. However, if some of these re-transmissions were not lost (but just delayed), then more than one copies of the message will be delivered.
Exactly-once processing: each TCP message will be processed by the destination node exactly once. More specifically, the destination will watch out for duplicate messages (checking the IDs of each received message). So, even if a message is delivered twice, the destination node will only process it (pass it to the application level) once and ignore the duplicates received later.
Exactly once is clearly impossible. What if the network connection is severed and never recovers?
I'm trying to make a simple server/application in Erlang.
My server initialize a socket with gen_tcp:listen(Port, [list, {active, false}, {keepalive, true}, {nodelay, true}]) and the clients connect with gen_tcp:connect(Server, Port, [list, {active, true}, {keepalive, true}, {nodelay, true}]).
Messages received from the server are tested by guards such as {tcp, _, [115, 58 | Data]}.
Problem is, packets sometimes get concatenated when sent or received and thus cause unexpected behaviors as the guards consider the next packet as part of the variable.
Is there a way to make sure every packet is sent as a single message to the receiving process?
Plain TCP is a streaming protocol with no concept of packet boundaries (like Alnitak said).
Usually, you send messages in either UDP (which has limited per-packet size and can be received out of order) or TCP using a framed protocol.
Framed meaning you prefix each message with a size header (usualy 4 bytes) that indicates how long the message is.
In erlang, you can add {packet,4} to your socket options to get framed packet behavior on top of TCP.
assuming both sides (client/server) use {packet,4} then you will only get whole messages.
note: you won't see the size header, erlang will remove it from the message you see. So your example match at the top should still work just fine
You're probably seeing the effects of Nagle's algorithm, which is designed to increase throughput by coalescing small packets into a single larger packet.
You need the Erlang equivalent of enabling the TCP_NODELAY socket option on the sending socket.
EDIT ah, I see you already set that. Hmm. TCP doesn't actually expose packet boundaries to the application layer - by definition it's a stream protocol.
If packet boundaries are important you should consider using UDP instead, or make sure that each packet you send is delimited in some manner. For example, in the TCP version of DNS each message is prefixed by a 2 byte length header, which tells the other end how much data to expect in the next chunk.
You need to implement a delimiter for your packets.
One solution is to use a special character ; or something similar.
The other solution is to send the size of the packet first.
PacketSizeInBytes:Body
Then read the provided amount of bytes from your message. When you're at the end you got your whole packet.
Nobody mentions that TCP may also split your message into multiple pieces (split your packet into two messages).
So the second solution is the best of all. But a little hard. While the first one is still good but limits your ability to send packets with special characters. But the easiest to implement. Ofc theres a workaround for all of this. I hope it helps.
I'm writing a program using Java non-blocking socket and TCP. I understand that TCP is a stream protocol but the underlayer IP protocol uses packets. When I call SocketChannel.read(ByteBuffer dst), will I always get the whole content of IP packets? or it may end at any position in the middle of a packet?
This matters because I'm trying to send individual messages through the channel, each messages are small enough to be sent within a single IP packet without being fragmented. It would be cool if I can always get a whole message by calling read() on the receiver side, otherwise I have to implement some method to re-assembly the messages.
Edit: assume that, on the sender side, messages are sent with a long interval(like 1 second), so they aren't going to group together in one IP packet. On the receiver side, the buffer used to call read(ByteBuffer dst) is big enough to hold any message.
TCP is a stream of bytes. Each read will receive between 1 and the maximum of the buffer size that you supplied and the number of bytes that are available to read at that time.
TCP knows nothing of your concept of messages. Each send by client can result in 0 or more reads being required at the other end. Zero or more because you might get a single read that returns more than one of your 'messages'.
You should ALWAYS write your read code such that it can deal with your message framing and either reassemble partial messages or split multiple ones.
You may find that if you don't bother with this complexity then your code will seem to 'work' most of the time, don't rely on that. As soon as you are running on a busy network or across the internet, or as soon as you increase the size of your messages you WILL be bitten by your broken code.
I talk about TCP message framing some more here: http://www.serverframework.com/asynchronousevents/2010/10/message-framing-a-length-prefixed-packet-echo-server.html and here: http://www.serverframework.com/asynchronousevents/2010/10/more-complex-message-framing.html though it's in terms of a C++ implementation so it may or may not be of interest to you.
The socket API makes no guarantee that send() and recv() calls correlate to datagrams for TCP sockets. On the sending side, things may get regrouped already, e.g. the system may defer sending one datagram to see whether the application has more data; on the receiving side, a read call may retrieve data from multiple datagrams, or a partial datagram if the size specified by the caller is requires breaking packet.
IOW, the TCP socket API assumes you have a stream of bytes, not a sequence of packets. You need make sure you keep calling read() until you have enough bytes for a request.
From the SocketChannel documentation:
A socket channel in non-blocking mode, for example, cannot read
any more bytes than are immediately available from the socket's input buffer;
So if your destination buffer is large enough, you are supposed to be able to consume the whole data in the socket's input buffer.