As we all know, in the CAN bus communication protocol, sender know whether the data was successfully sent. I send socketcan data as follows.
ret = write (socket, frame, sizeof (struct can_frame));
However, even if the CAN communication cable is disconnected, the return value of ret is still 16(=sizeof (struct can_frame)).I queried the information and found that the problem was due to the tx_queue of the network stack used by socketcan. When write is called multiple times, the buffer is full and the return value of ret is -1.
But this is not the behavior I expect, I hope that every frame of data sent will immediately get the status of success or failure.
By
echo 0> / sys / class / net / can0 / tx_queue_len
I want to cancel the tx_queue, but it does not work.
What I want to ask is, is there a way to cancel the tx_queue of socketcan, or to get the status of the each sending frame about controller through the API (such as libsocketcan).
Thanks.
You cannot use write() itself to discover whether a CAN frame was successfully put on the bus, because all it does is write the frame to the in-kernel socket buffer. The kernel then moves the frame to the transmit queue of the SocketCAN network interface, followed by the driver moving it to the transmit buffer of the CAN controller, which finally puts the frame on the bus. What you want is a direct write which bypasses all those buffers, but that's not possible with SocketCAN, even if you set the transmit queue length to 0.
However, there is another way to get confirmation. If you enable the CAN_RAW_RECV_OWN_MSGS socket option (see section 4.1.4 and 4.1.7 in the SocketCAN documentation), you will receive frames that were successfully sent. You'll need to use recvmsg() so you get the message flags. msg_flags will have the MSG_CONFIRM bit set for a frames that was successfully sent by the same socket on which it is received. You won't be informed of failures, but you can detect them by using a timeout for the confirmation.
It's not an ideal solution because it mixes the read and write logic in your application. One way to avoid this would be to use two sockets. One for writing and reading MSG_CONFIRM frames, the other for reading all other frames. You could then create a (blocking) write function that does a write() followed by multiple calls to recvmsg() with an appropriate timeout.
Finally, it is useful to enable error frames (through the CAN_RAW_ERR_FILTER socket option). If you send a frame on a socket with a disconnected cable, this will typically result in a bus off state, which will be reported in an error frame.
Related
For c send function(blocking way) it's specified what function returns with size of sent bytes when it's received on destinations. I'm not sure that I understand all nuances, also after writing "demo" app with WSAIoctl and WSARecv on server side.
When send returns with less bytes number than asked in buffer-length parameter?
What is considered as "received on destinations"? My first guess it's when it sit on server's OS buffer and server application is notified. My second one it's when server application recv call have read it fully?
Unless you are using a (somewhat exotic) library, a send on a socket will return the number of bytes passed to the TCP buffer successfully, not the number of bytes received by the peer (see Microsoft´s docs for example).
When you are streaming data via a socket, you need to check the bytes effectively accepted into the TCP send buffer. That´s why usually a send command is inside a loop that will issue several sends if needed.
Errors in send are local: for example if the socket is closed by the peer during a sending operation (making your socket invalid) or if the operation times out (TCP buffer not emptying, i. e. peer not receiving data fast enough or some other trouble).
After all send is completed you have no easy way of knowing if the peer received all the bytes you sent. You´ll usually just issue closesocket and make sure that your socket has a proper linger option set (i. e. only close after timeout or sucessfully finishing the send). Alternatively you wait for a confirmation by the peer (for example via a recv that returns zero bytes, indicating that the connection was gracefully closed).
Edit: typo
I was trying to manipulate SOME/IP messages by falsifying their content(Payload) sent between 2 ECUs at run time.
After setting up the Hardware VN6510A MAC Bypassing and integrating it in the data traffic path between those 2 ECUs to monitor and control all Ethernet data streams.
ECU A ---> eth1 interface --VN6510A-- eth2 interface ---> ECU B
I successfully catch our target SOME/IP messages and I also succefully manipulate their paylod.
But at the end we got 2 SOME/IP messages: the real coming message and the falsified message forwarded at the same time.
How could we bound those 2 SOME/IP messages, the real message and the falsified message together, so that we could have just one falsified SOME/IP message, knowing that I am using the same SOME/IP message handle.
I used the callback function void OnEthPacket(LONG channel, LONG dir, LONG packet) to register a received Ethernet packet.
Probably by setting your VN.... to "Direct" and not "MAC Bypassing"
Well we could not manipulate Messages at run time using the vector box VN6510A Solution because simply their box doesn't support this feature.
Is it possible to implement the equivalent of Socket.Poll in async/await paradigm (or BeginXXX/EndXXX async pattern)?
A method which would act like NetworkStream.ReadAsync or Socket.BeginReceive but:
leave the data in the socket buffer
complete after the specified interval of time if no data arrived (leaving the socket in connected state so that the polling operation can be retried)
I need to implement IMAP IDLE so that the client connects to the mail server and then goes into waiting state where it received data from the server. If the server does not send anything within 10 minutes, the code sends ping to the server (without reconnecting, the connection is never closed), and starts waiting for data again.
In my tests, leaving the data in the buffer seems to be possible if I tell Socket.BeginReceive method to read no more than 0 bytes, e.g.:
sock.BeginReceive(b, 0, 0, SocketFlags.None, null, null)
However, not sure if it indeed will work in all cases, maybe I'm missing something. For instance, if the remote server closes the connection, it may send a zero-byte packet and not sure if Socket.BeginReceive will act identically to Socket.Poll in this case or not.
And the main problem is how to stop socket.BeginReceive without closing the socket.
After studying the "window size" concept, what I understood is that it keeps packet before sending over wire and till acknowledgement come for earliest packet . Once this gets filled up, subsequent packet will be dropped. Somewhere I also have read that TCP is a streaming protocol, and packet is what related to IP protocol at Network layer .
What I assumed till was that I have declared a Buffer (inside code) which I fill with some data and send this Buffer using socket. I declared a buffer of 10000 bytes and send it repeatedly using socket over 10 Gbps link .
I have following assumptions and questions. Please verify and help
If I want to send a packet of 64,256,512 etc. bytes, declared buffer inside code of that much space and send over socket. Each execution of send() command will send one packet of that much size .
So if I want to study the packet size variation effect on throughput, what do I have to do? Do I need to vary buffer size in code?
What are the socket buffer which we set using SO_SNDBUF and SO_RECVBUF? Google says it's buffer space for socket. Is it same as TCP window size or something different? Which parameter is more suitable to vary or to increase throughput?
Also there are three parameter in socket buffer: Min, Default and Max. Which one should I vary to my experiment and to get more relevance?
If I want to send a packet of 64,256,512 etc. bytes , Declared buffer inside code of that much space and send over socket .Each execution of send() command will send one packet of that much size.
Only if you disable the Nagle algorithm and the size is less than the path MTU. You mustn't rely on this.
So if I want to Study the Packet size variation effect on throughput, What I have to do , vary buffer space in Code?
No. Vary SO_RCVBUF at the receiver. This is the single biggest determinant of throughput, as it determines the maximum receive window.
what are the socket buffer which we set using SO_SNDBUF and SO_RCVBUF
Send buffer size at the sender, and receive buffer size at the receiver. In the kernel.
It's Same as TCP Window size
See above.
or else different ? Which parameter is more suitable to vary to increase throughput ?
See above.
Also there are three parameter in Socket Buffer min Default and Max . Which one should I vary for My experiment to get more relevance
None of them. These are the system-wide parameters. Just play with SO_SNDBUF and SO_RCVBUF for the specific sockets in your application.
TCP does not directly expose a way to control the way packets are sent since it is a stream protocol. But you can make the TCP stack send packets by disabling the Nagle algorithm. That way all data that you send will be sent out immediately instead of being buffered. Data will be split into packets of MTU size which is like ~1400 bytes. Depends on the link.
To answer (2): Disable nagling and invoke send with buffers of < 1400 bytes. Use Wireshark to make sure you got what you wanted.
The buffer settings have nothing to do with any of this. I know of no valid reason to touch them.
In general this question is probably moot since you seem to want to send a lot of data. Just leave Nagling enabled and send big buffers (such as 64KB).
I do some experience on Windows 10:
code from https://docs.python.org/3/library/socketserver.html#asynchronous-mixins,
RawCap for loopback capture,
WireShark for watching result.
The primary client code is:
def client(ip, port, message):
sock = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
sock.setsockopt(socket.SOL_SOCKET,socket.SO_RCVBUF, 100000)
sock.connect((ip, port))
sock.sendall(bytes(message, 'ascii'))
response = str(sock.recv(1024), 'ascii')
print("Received: {}".format(response))
Here is the result(the server port is 11111):
you can see, the tcp recive window size is the same as SO_RCVBUF, may it is platform indepent, you can verify it on other platform.
on https://msdn.microsoft.com/en-us/library/windows/hardware/ff570832(v=vs.85).aspx
The SO_RCVBUF socket option determines the size of a socket's receive buffer that is used by the underlying transport.
verified this.
Also, when I set SO_SNDBUF = 100000, it have no affects on the tcp transmission between client and server, as server just can discard data if client send much data one time.
So, if you want to change SO_RCVBUF to max Throughput, you can refer http://packetbomb.com/understanding-throughput-and-tcp-windows/, the os may offer func to detect ideal send backlog (ISB).
I am writing the callout driver for Hyper-V 2012 where I need to filter the packets sent from virtual machines.
I added filter at FWPM_LAYER_EGRESS_VSWITCH_TRANSPORT_V4 layer in WFP. Callout function receive packet buffer which I am typecasting it to NET_BUFFER_LIST. I am doing following to get the data pointer
pNetBuffer = NET_BUFFER_LIST_FIRST_NB((NET_BUFFER_LIST*)pClassifyData->pPacket);
pContiguousData = NdisGetDataBuffer(pNetBuffer, NET_BUFFER_DATA_LENGTH(pNetBuffer), 0, 1, 0);
I have simple client-server application to test the packet data. Client is on VM and server is another machine. As I observed, data sent from client to server is truncated and some garbage value is added at the end. There is no issue for sending message from server to client. If I dont add this layer filter client-server works without any issue.
Callback function receives the metadata which incldues ipHeaderSize and transportHeaderSize. Both these values are zero. Are these correct values or should those be non-zero??
Can somebody help me to extract the data from packet in callout function and forward it safely to further layers?
Thank You.
These are the TCP packets. I looked into size and offset information. It seems the problem is consistent across packets.
I checked below values in (NET_BUFFER_LIST*)pClassifyData->pPacket.
NET_BUFFER_LIST->NetBUfferListHeader->NetBUfferListData->FirstNetBuffer->NetBuffe rHeader->NetBufferData->CurrentMdl->MappedSystemVa
First 24 bytes are only sent correctly and remaining are garbage.
For example total size of the packet is 0x36 + 0x18 = 0x4E I don't know what is there in first 0x36 bytes which is constant for all the packets. Is it a TCP/IP header? Second part 0x18 is the actual data which i sent.
I even tried with API NdisQueryMdl() to retrieve from MDL list.
So on the receiver side I get only 24 bytes correct and remaining is the garbage. How to read the full buffer from NET_BUFFER_LIST?