I am designing the software controlling several serial ports, operating system is OpenWrt. The device application is running in is single core ARM9 # 450 MHz. Protocol for serial ports is Modbus.
The problem is with Modbus slave implementation. I designed it in real-time manner, looping reading data from serial port (port is open in non-blocking mode with 0 characters to wait for and no timeouts). The sequence/data stream timeout is about 4 milliseconds # 9600/8/N/1 (3.5 characters as advised). Timeout is checked if application does not see anything in the buffer, therefore if application is slower that incoming stream of characters, timeouts mechanism will not take place - until all characters are removed from the buffer and bus is quiet.
But I see that CPU switches between threads and this thread is missed for about 40-60 milliseconds, which is a lot to measure timeouts. While I guess serial port buffer still receives the data (how long this buffer is?), I am unable to assess how much time passed between chars, and may treat next message as continuation of previous and miss Modbus request targeted for my device.
Therefore, I guess, something must be redesigned (for the slave - master is different story). First idea coming to my mind is to forget about timeouts, and just parse the incoming data after being synchronized with the whole stream (by finding initial timeout). However, there're several problems - I must know everything about all types of Modbus messages to parse them correctly and find out their ends and where next message starts, and I do not see the way how to differentiate Modbus request with Modbus response from the device. If developers of the Modbus protocol would put special bit in command field identifying if message is request or response... but it is not the case, and I do not see the right way to identify if message I am getting is request or response without getting following bytes and checking CRC16 at would-be byte counts, it will cost time while I am getting bytes, and may miss window for the response to request targeted for me.
Another idea would be using blocking method with VTIME timeout setting, but this value may be set to tenths of seconds only (therefore minimal is 100 ms), and this is too much given another +50 ms for possible CPU switching between threads, I think something like timeout of 10 ms is needed here. It is a good question if VTIME is hardware time, or software/driver also subject to CPU thread interruptions; and the size of FIFO in the chip/driver how many bytes it can accumulate.
Any help/idea/advice will be greatly appreciated.
Update per #sawdust comments:
non-blocking use has really appeared not a good way because I do not poll hardware, I poll software, and the whole design is again subject to CPU switching and execution scheduling;
"using the built-in event-based blocking mode" would work in blocking mode if I would be able to configure UART timeout (VTIME) in 100 us periods, plus if again I would be sure I am not interrupted during the executing - only then I will be able to have timing properly, and restart reading in time to get next character's timing assessed properly.
Related
I am developping an application on the STM32 SPBTLE-1S module (BLE 4.2). The module connects to a Raspberry Pi.
When the connection quality is low, a disconnection will sometimes occur with error code 0x28 (Reason: Instant Passed) before the connection timeout is reached.
Current connection settings are:
Conn_Interval_Min: 10
Conn_Interval_Max: 20
Slave_latency: 5
Timeout_Multiplier: 3200
Reading more on this type of error, it seems to happen when "an LMP PDU or LL PDU that includes an instant cannot be performed because the instant when this would have occurred has passed." These paquets are typically used for frequency hopping or for connection updates. In my case, they must be frequency hoping paquets.
Any idea on how to prevent these disconnections caused by "Instant Passed" errors? Or are they simply a consequence of the BLE technology?
Your question sounds similar to this one
In a nutshell, there's only two possible link layer requests that can result in this type of disconnect (referred to as LL_CONNECTION_UPDATE_IND & LL_CHANNEL_MAP_IND in the latest v5.2 Bluetooth Core Spec)
If you have access to the low level firmware for the bluetooth stack on the embedded device, what I've done in the past is increase the number of slots in the future the switch "Instant" is at so there's more time for the packet to go through in a noisy environment.
Otherwise the best you can do is try to limit the amount of times you change connection parameters to make that style of disconnect less likely to occur. (The disconnect could still be triggered by channel map change but I haven't seen many BLE stacks expose a lot of configuration over when those take place.)
I currently have a game server with a customizable tick rate, but for this example let's suggest that the server is only going to tick once per second or 1hz. I'm wondering what's the best way to handle incoming packets if the client send rate is faster than the server's as my current setup doesn't seem to work.
I have my udp blocking receive with a timeout inside my tick function, and it works, however if the client tick rate is higher than the server, all of the packets are not received; only the one that is being read at the current time. So essentially the server is missing packets being sent by clients. The image below demonstrates my issue.
So my question is, how is this done correctly? Is there a separate thread where packets are read constantly, queued up and then the queue is processed when the server ticks or is there a better way?
Image was taken from a video https://www.youtube.com/watch?v=KA43TocEAWs&t=7s but demonstrates exactly what I'm explaining
There's a bunch going on with the scenario you describe, but here's my guidance.
If you are running a server at 1hz, having a blocking socket prevent your main logic loop from running is not a great idea. There is a chance you won't be receiving messages at the rate you expect (due to packet loss, network lag spike or client app closing)
A) you certainly could create another thread, continue to make blocking recv/recvfrom calls, and then enqueue them onto a thread safe data structure
B) you could also just use a non-blocking socket, and keep reading packets until it returns -1. The OS will buffer (usually configurable) a certain number of incoming messages, until it starts dropping them if you aren't reading.
Either way is fine, however for individual game clients, I prefer the second simple approaches when knowing I'm on a thread that is servicing the socket at a reasonable rate (5hz seems pretty low, but may be appropriate for your game). If there's a chance you are stalling the servicing thread (level loading, etc), then go with the first approach, so you don't detect it as a disconnection if you miss sending/receiving a periodic keepalive message.
On the server side, if I'm planning on a large number of clients/data, I go to great lengths to efficiently read from sockets - using IO Completion Ports on Windows, or epoll() on Linux.
Your server could have to have a thread to tick every 5 seconds just like the client to receive all the packets. Anything not received during that tick would be dropped as the server was not listening for it. You can then pass the data over from the thread after 5 ticks to the server as one chunk. The more reliable option though is to set the server to 5hz just like the client and thread every packet that comes in from the client so that it does not lock up the main thread.
For example, if the client update rate is 20, and the server tick rate is 64, the client might as well be playing on a 20 tick server.
The GATT architecture of BLE lends itself to small fixed pieces of data (20 bytes max per characteristic). But in some cases, you end up wanting to “stream” some arbitrary length of data, that is greater than 20 bytes. For example, a firmware upgrade, even if you know its slow.
I’m curious what scheme others have used if any, to “stream” data (even if small and slow) over BLE characteristics.
I’ve used two different schemes to date:
One was to use a control characteristic, where the receiving device notify the sending device how much data it had received, and the sending device then used that to trigger the next write (I did both with_response, and without_response) on a different characteristic.
Another scheme I did recently, was to basically chunk the data into 19 byte segments, where the first byte indicates the number of packets to follow, when it hits 0, that clues the receiver that all of the recent updates can be concatenated and processed as a single packet.
The kind of answer I'm looking for, is an overview of how someone with experience has implemented a decent schema for doing this. And can justify why what they did is the best (or at least better) solution.
After some review of existing protocols, I ended up designing a protocol for over-the-air update of my BLE peripherals.
Design assumptions
we cannot predict stack behavior (protocol will be used with all our products, whatever the chip used and the vendor stack, either on peripheral side or on central side, potentially unknown yet),
use standard GATT service,
avoid L2CAP fragmentation,
assume packets get queued before TX,
assume there may be some dropped packets (even if stacks should not),
avoid unnecessary packet round-trips,
put code complexity on central side,
assume 4.2 enhancements are unavailable.
1 implies 2-5, 6 is a performance requirement, 7 is optimization, 8 is portability.
Overall design
After discovery of service and reading a few read-only characteristics to check compatibility of device with image to be uploaded, all upload takes place between two characteristics:
payload (write only, without response),
status (notifiable).
The whole firmware image is sent in chunks through the payload characteristic.
Payload is a 20-byte characteristic: 4-byte chunk offset, plus 16-byte data chunk.
Status notifications tell whether there is an error condition or not, and next expected payload chunk offset. This way, uploader can tell whether it may go on speculatively, sending its chunks from its own offset, or if it should resume from offset found in status notification.
Status updates are sent for two main reasons:
when all goes well (payloads flying in, in order), at a given rate (like 4Hz, not on every packet),
on error (out of order, after some time without payload received, etc.), with the same given rate (not on every erroneous packet either).
Receiver expects all chunks in order, it does no reordering. If a chunk is out of order, it gets dropped, and an error status notification is pushed.
When a status comes in, it acknowledges all chunks with smaller offsets implicitly.
Lastly, there is a transmit window on the sender side, where many successful acknowledges flying allow sender to enlarge its window (send more chunks ahead of matching acknowledge). Window is reduced if errors happen, dropped chunks probably are because of a queue overflow somewhere.
Discussion
Using "one way" PDUs (write without response and notification) is to avoid 6. above, as ATT protocol explicitly tells acknowledged PDUs (write, indications) must not be pipelined (i.e. you may not send next PDU until you received response).
Status, containing the last received chunk, palliates 5.
To abide 2. and 3., payload is a 20-byte characteristic write. 4+16 has numerous advantages, one being the offset validation with a 16-byte chunk only involves shifts, another is that chunks are always page-aligned in target flash (better for 7.).
To cope with 4., more than one chunk is sent before receiving status update, speculating it will be correctly received.
This protocol has the following features:
it adapts to radio conditions,
it adapts to queues on sender side,
there is no status flooding from target,
queues are kept filled, this allows the whole central stack to use every possible TX opportunity.
Some parameters are out of this protocol:
central should enforce short connection interval (try to enforce it in the updater app);
slave PHY should be well-behaved with slave latency (YMMV, test your vendor's stack);
you should probably compress your payload to reduce transfer time.
Numbers
With:
15% compression,
a device connected with connectionInterval = 10ms,
a master PHY limiting every connection event to 4-5 TX packets,
average radio conditions.
I get 3.8 packets per connection event on average, i.e. ~6 kB/s of useful payload after packet loss, protocol overhead, etc.
This way, upload of a 60 kB image is done in less than 10 seconds, the whole process (connection, discovery, transfer, image verification, decompression, flashing, reboot) under 20 seconds.
It depends a bit on what kind of central device you have.
Generally, Write Without Response is the way to stream data over BLE.
Packets being received out-of-order should not happen since BLE's link layer never sends the next packet before it the previous one has been acknowledged.
For Android it's very easy: just use Write Without Response to send all packets, one after another. Once you get the onCharacteristicWrite you send the next packet. That way Android automatically queues up the packets and it also has its own mechanism for flow control. When all its buffers are filled up, the onCharacteristicWrite will be called when there is space again.
iOS is not that smart however. If you send a lot of Write Without Response packets and the internal buffers are full, iOS will silently drop new packets. There are two ways around this, either implement some (maybe complex) protocol for the peripheral notifying the status of the transmission, like Nipos answer. An easier way however is to send each 10th packet or so as a Write With Response, the rest as Write Without Response. That way iOS will queue up all packets for you and not drop the Write Without Response packets. The only downside is that the Write With Response packets require one round-trip. This scheme should nevertheless give you high throughput.
In talking to a MODBUS device, is there an upper bound on how long a device can take to respond before it's considered a timeout? I'm trying to work out what to set my read timeout to. Answers for both MODBUS RTU and TCP would be great.
In MODBUS over serial line specification and implementation guide V1.0 section 2.5.2.1 MODBUS Message ASCII Framing There are suggestions that inter-character delays of up to 5 seconds are reasonable in slow WAN configurations.
2.6 Error Checking Methods indicates that the timeouts are configured without specifying any values.
The current Modicon Modbus Protocol Reference Guide PI–MBUS–300 Rev. J also provides no quantitative suggestions for these settings.
Your application time-sensitivity, along with the constraints that your network enforces, will largely determine your choices.
If you identify the worst-case delays you can tolerate, take half that time to allow a single retransmission to fail, subtract reasonable transmission times for a message of maximal length, then you should have a good candidate for a timeout. This will allow you to recover from a single error, while not reporting errors unnecessarily often.
Of course, the real problem is, what to do when the error occurs. Is it likely to be a transient problem, or is it the result of a permanent fault that requires attention?
Alexandre Vinçon's comment about the ACKNOWLEDGEMENTs is also relevant. It may be your device does not implement this, and extended delays may be intended.
The specification does not mention a particular value for the timeout, because it is not possible to normalize a timeout value for a wide range of MODBUS slaves.
However, it is a good assumption that you should receive a reply within a few hundreds of milliseconds.
I usually define my timeouts to 1 second with RTU and 500 ms with TCP.
Also, if the device takes a long time to reply, it is supposed to return an ACKNOWLEDGE message to prevent the expiration of the timeout.
I am designing and testing a client server program based on TCP sockets(Internet domain). Currently , I am testing it on my local machine and not able to understand the following about SIGPIPE.
*. SIGPIPE appears quite randomly. Can it be deterministic?
The first tests involved single small(25 characters) send operation from client and corresponding receive at server. The same code, on the same machine runs successfully or not(SIGPIPE) totally out of my control. The failure rate is about 45% of times(quite high). So, can I tune the machine in any way to minimize this.
**. The second round of testing was to send 40000 small(25 characters) messages from the client to the server(1MB of total data) and then the server responding with the total size of data it actually received. The client sends data in a tight loop and there is a SINGLE receive call at the server. It works only for a maximum of 1200 bytes of total data sent and again, there are these non deterministic SIGPIPEs, about 70% times now(really bad).
Can some one suggest some improvement in my design(probably it will be at the server). The requirement is that the client shall be able to send over medium to very high amount of data (again about 25 characters each message) after a single socket connection has been made to the server.
I have a feeling that multiple sends against a single receive will always be lossy and very inefficient. Shall we be combining the messages and sending in one send() operation only. Is that the only way to go?
SIGPIPE is sent when you try to write to an unconnected pipe/socket. Installing a handler for the signal will make send() return an error instead.
signal(SIGPIPE, SIG_IGN);
Alternatively, you can disable SIGPIPE for a socket:
int n = 1;
setsockopt(thesocket, SOL_SOCKET, SO_NOSIGPIPE, &n, sizeof(n));
Also, the data amounts you're mentioning are not very high. Likely there's a bug somewhere that causes your connection to close unexpectedly, giving a SIGPIPE.
SIGPIPE is raised because you are attempting to write to a socket that has been closed. This does indicate a probable bug so check your application as to why it is occurring and attempt to fix that first.
Attempting to just mask SIGPIPE is not a good idea because you don't really know where the signal is coming from and you may mask other sources of this error. In multi-threaded environments, signals are a horrible solution.
In the rare cases were you cannot avoid this, you can mask the signal on send. If you set the MSG_NOSIGNAL flag on send()/sendto(), it will prevent SIGPIPE being raised. If you do trigger this error, send() returns -1 and errno will be set to EPIPE. Clean and easy. See man send for details.