I am writing a java applet that will print what a telnet client sends to the connection. Unfortunately, the client splits at 1448 characters.
The code that is proving to be a problem:
char[] l = new char[5000];
Reader r = new BufferedReader(new InputStreamReader(s.getInputStream(), "US-ASCII"));
int i = r.read(line);
I cannot change the source of what the telnet client reads from, so I am hoping it is an issue with the above three lines.
You're expecting to get telnet protocol data units from the TCP layer. It just doesn't work that way. You can only extract telnet protocol data units from the code that implements the telnet protocol. The segmentation of bytes of data at the TCP layer is arbitrary and it's the responsibility of higher layers to reconstruct the protocol data units.
The behavior you are seeing is normal, and unless you're diagnosing a performance issue, you should completely ignore the way the data is split at the TCP level.
The reason you're only getting 1448 bytes at a time is that the underlying protocols divide the transmission into packets. Frequently, this size is around 1500, and there are some bytes used for bookkeeping, so you're left with a chunk of 1448 bytes. The protocols don't guarantee that if you send X bytes in a 'single shot', that the client will receive X bytes in a single shot (e.g. a single call to the receive method).
As has been noted already in the comments above, its up to the receiving program to re-assemble these packets in a way that is meaningful to the client. In generally, you perform receives and append the data you receive to some buffer until you find an agreed-upon 'end of the block of data' marker (such as an end-of-line, new-line, carriage return, some symbol that won't appear in the data, etc.).
If the server is genuinely a telnet server--its output might be line-based (e.g. a single block of data is terminated with a 'end of line': carriage return and linefeed characters). RFC 854 may be helpful--it details the Telnet protocol as originally specified.
Related
I was debugging a webrtc trickle ice exchange the other day and realized I never paid much attention to the candidate messages (generated by calling RTCIceCandidate.toJson() ) that look like this:
{"candidate":"candidate:394300051 1 tcp 1518214911 192.168.1.12 9 typ host tcptype active generation 0 ufrag rfBJ network-id 1",
"sdpMid":"0","sdpMLineIndex":0}
In the above json message exactly what do sdpMid and sdpMLineIndex represent? They always appear to have the same values (either 0/"0" or 1/"1").
Is it correct to say:
That sdpMid corresponds to the a=mid line for a stream in the intial SDP? That is, if the line for the audio stream was declared as a=mid:audio, then the candidate's sdpMid value would have been "audio" as well.
That sdpMLineIndex is the index number of the stream as it appeared in the SDP? That is if audio was first in the SDP, then this value is 0 and if video was second it would be 1?
In other words, sdpMid is a string name for the stream and sdpMLineIndex is an index value. But the standard convention used by most implementations is to just have these values be the same.
Is this correct?
sdpMid and sdpMLineIndex are equivalent for the currently browser-generated offers for simple cases. They are not equivalent for cases like stopping a transceiver (using .stop()) and then generating a new offer. This new offer usually has a new mid which can be an incrementally generated number whereas the sdpMLineIndex may not increment if a previously unused m= line gets recycled.
Effectively they are artifacts from very early versions of the specifications and implementations lagging behind (here for Firefox).
I'm writing a very specific application protocol to enable communication between 2 nodes. Node 1 is an embedded platform (a microcontroller), while node 2 is a common computer.
Such protocol defines messages of variable length. This means that sometimes node 1 sends a message of 100 bytes to node 2, while another time it sends a message of 452 bytes.
Such protocol shall be independent on how the messages are transmitted. For instance, the same message can be sent over USB, Bluetooth, etc.
Let's assume that a protocol message is defined as:
| Length (4 bytes) | ...Payload (variable length)... |
I'm struggling about how the receiver can recognise how long is the incoming message. So far, I have thought about 2 approaches.
1st approach
The sender sends the length first (4 bytes, always fixed size), and the message afterwards.
For instance, the sender does something like this:
// assuming that the parameters of send() are: data, length of data
send(msg_length, 4)
send(msg, msg_length - 4)
While the receiver side does:
msg_length = receive(4)
msg = receive(msg_length)
This may be ok with some "physical protocols" (e.g. UART), but with more complex ones (e.g. USB) transmitting the length with a separate packet may introduce some overhead. The reason being that an additional USB packet (with control data, ACK packets as well) is required to be transmitted for only 4 bytes.
However, with this approach the receiver side is pretty simple.
2nd approach
The alternative would be that the receiver keeps receiving data into a buffer, and at some point tries to find a valid message. Valid means: finding the length of the message first, and then its payload.
Most likely this approach requires adding some "start message" byte(s) at the beginning of the message, such that the receiver can use them to identify where a message is starting.
Usually Win32 API can tell what is the length of output buffer required. One need just pass 0 as buffer length and API returns error BUFFER_TOO_SMALL and number of bytes required.
But it is not the same with SslEncryptPacket. It just returns error about small buffer and that's all.
There is also SslLookupCipherLengths which I suppose should be used for that, but documentation gives no clue about how to calculate output buffer having that info.
Maybe you can tell ? Usually I would reserve + kilobyte , but in my situation I need to know exactly.
You probably already know that in order to go through the TLS/SSL handshake, you repeatedly call SSPI->InitializeSecurityContext (on the client side) or SSPI->AcceptSecurityContext (on the server side).
Once the function returns SEC_E_OK, you should call SSPI->QueryContextAttributes with SECPKG_ATTR_STREAM_SIZES to determine the sizes of the header and trailer. It also tells you the number of SecBuffers to use for the SSPI->EncryptMessage function, and it tells you the maximum size of the message that you can pass to EncryptMessage.
As I understand, the values that are returned may vary depending on the type of encryption that the OS chooses for the connection. I'm not intimately familiar with TLS/SSL but I think it uses 5 bytes for the header, 36 for the footer and 16384 for the maximum message length. You mileage may vary, so that's why you should call QueryContextAttribute(... SECPKG_ATTR_STREAM_SIZES ...).
I want both the client and server to write and read resp. at a constant rate (which can be configured on the GUI of the client) to the buffer.
Say,
I am able to send from the client at 150 bytes per packet
Now, I should be able to read also at 150 bytes per packet on the server too
Since, both are connected through a socket, can we retrieve the socket params (set on the client size, like 150 here) from the server end, using the tcpServer object.
Or is it must to send an initial setup packet which tells about these client params and so accordingly the server can continue?
It's kinda usual to sort message sizes out at the application level. You could indeed send a 'setup message' as the first data after a successful connect. You should send this setup message in a form that will not be misunderstood due to endianness or the number of bytes received per read call. Perhaps a fixed-size messge in ASCII, maybe five bytes:
'00150'
The server can then read five bytes only, convert to integer, save it in the server-client socket object so that the server always knows how many bytes to send and then issue a read call for that number of bytes.
Alternatively, you could use a simple protocol that embeds the size into each message, eg:
SOH
"0"
"0"
"1"
"5"
"0"
[150 bytes of data]
EOT
Rgds,
Martin
I want to read specific number of bytes from udp socket. In tcp socket I can use socket.read where I can specify the amount of data to receive. I don't find similar function for UDP socket. I am using receive_from() where I can specify the amount of data to read, but if there is more data then no data is read and I get following error.
"A message sent on a datagram socket was larger than the internal message buffer or some other network limit, or the buffer used to receive a datagram into was smaller than the datagram itself" std::basic_string<char,std::char_traits<char>,std::allocator<char> >
I am not able to find what value do I need to give for message_flags (3rd arg to receive_from) so that it will read the number of bytes specified. Currenly I am using the following code to read data but it either reads all data or no data.
size_t size=socket.receive_from(boost::asio::buffer((const void*)&discRsp,sizeof(DataStructure)),remote_endpoint,0,errors);
Try this:
socket.set_option(boost::asio::socket_base::receive_buffer_size(65536));