The above two pictures are my measurement and simulation setups respectively. The Replay block plays a blf file of length 6 minutes containing total of 2,413,161 CAN frames from two CAN channels.
The above picture explains the bench setup. Canoe reads the blf file and transmits the CAN frames on two CAN channels. Microcontroller (MuC) receives the CAN frames converts them in to Ethernet IPV4 UDP packet and transmit again to the Canoe.
When i run this configuration, i am getting below errors.
1. System - CAN driver: Reception overrun - messages are lost
2. System CAN X : Message with ID = XXX could not be sent. Driver error 11 in TransmitCANFrame, "XL_ERR_QUEUE_IS_FULL"
3. System Warning: replay loading delay(s)
System ReplayBlock 1(blf_file.blf): 15 times, 7347.46 ms total
I assumes this was due to Canoe performance issue or CAN driver issue. So I did the below steps.
1. Modified the CANCaseXL Receive latency->Very fast under Vector hardware Config.
2. Increased the Transmit queue settings->32768 (maximum) under Vector hardware config -> Global settings.
3. I disabled all except one logging block (blf) [As you can see in the measurement setup].
But i still experience the same errors. What could be the problem? Is there any other ways to resolve this?
You need to have termination on both CANs (120 Ohm).
Such error indicates lack of termination.
Related
Let's say I have a car with different sensors: several cameras, LIDAR and so on, the data from this sensors are going to be send to some host over 5G network (omnetpp + inet + simu5g). For video it is like 5000 packets 1400 bytes each, for lidar 7500 packets 1240 bytes and so on. Each flow is encoded in UDP packets.
So in omnetpp module in handleMessage method I have two sentTo calls, each is scheduled "as soon as possible", i.e., with no delay - that corresponds to the idea of multiple parallel streaming. How does omnetpp handle situations, when it needs to send two different packets at the same time from the same module to the same module (some client, which receives sensor data streams)? Does it create some inner buffer on the sender or receiver side, therefore allowing really only one packet sending per handleMessage call or is it wrong? I want to optimize data transmission and play with packet sizes and maybe with sending intervals, so I want to know, how omnetpp handles multiple streaming at the same time, because if it actually buffers, maybe than it makes sense to form a single package from multiple streams, each such package will consist of a certain amount of data from each stream.
There is some confusion here that needs to be clarified first:
OMNeT++ is a discrete event simulator framework. An OMNeT++ model contains modules that communicate with each other, using OMNeT++ API calls like sendTo() and handleMessage(). Any call of the sendTo() method just queues the provided message into the future event queue (an internal, time ordered queue). So if you send more than one packet in a single handleMessage() method, they will be queued in that order. The packets will be delivered one by one to the requested destination modules when the requested simulation time is reached. So you can send as many packets as you wish and those packets will be delivered one by one to the destination's handleMessage() method. But beware! Even if the different packets will be delivered one by one sequentially in the program's logic, they can still be delivered simultaneously considering the simulation time. There are two time concepts here: real-time that describes the execution order of the code and simulation-time which describes the time passes from the point of the simulated system. That's why, while OMNeT++ is a single threaded application that runs each events sequentially it still can simulate infinite number of parallel running systems.
BUT:
You are not modeling directly with OMNeT++ modules, but rather using INET Framework which is a model directly created to simulate internet protocols and networks. INET's core entity is a node which is something that has network interface(s) (and queues belonging to them). Transmission between nodes are properly modeled and only a single packet can travel on an ethernet line at a time. Other packets must queue in the network interface queue and wait for an opportunity to be delivered from there.
This is actually the core of the problem for Time Sensitive Networks: given a lot of pre-defined data streams in a network, how the various packets interfere and affect each other and how they change the delay and jitter statistics of various streams at the destination, Plus, how you can configure the source and network gate scheduling to achieve some desired upper bounds on those statistics.
The INET master branch (to be released as INET 4.4) contains a lot TSN code, so I highly recommend to try to use it if you want to model in vehicle networks.
If you are not interested in the in-vehicle communication, bit rather want to stream some data over 5G, then TSN is not your interest, but you should NOT start to multiplex/demultiplex data streams at application level. The communication layers below your UDP application will fragment/defragment and queue the packets exactly how it is done in the real world. You will not gain anything by doing mux/demux at application layer.
I have a hardware IO task (write/read serial message) that has a strict jitter requirement of less than 200 micro seconds. I need to be able to isolate both a CPU core/s and hardware/interrupt.
I have tried 2 things that have helped but not gotten me all the way there.
Using <termios.h> to configure the tty device. Specifically setting VMIN=packet_size and VTIME=0
Isolcpus kernel argument in /etc/default/grub and running with taskset
I am still seeing upwards of 5 ms (5000 us) of jitter on my serial reads. I tested this same application with pseudo serial devices (created by socat) to eliminate the HW variable but am still seeing high jitter.
My test application right now just opens a serial connection, configures it, then does a while loop of writes/reads.
Could use advice on how to bring jitter down to 200 us or less. I am considering moving to a dual boot RTOS/Linux with shared memory, but would rather solve on one OS.
Real Application description:
Receive message from USB serial
Get PTP (precision time protocol) time within 200 us of receiving the first bit
Write packet received along with timestamp to shared memory buffer shared with a python application: <timestamp, packet>.
Loop.
Also on another isolated HW/core:
Read some <timestamp, packet> from a shared memory buffer
Poll PTP time until <timestamp>
Transmit <packet> at within 200 us of <timestamp> over USB serial
Loop
To reduce process latency/jitter:
Isolate some cores
/etc/default/grub... isolcpus=0
Never interrupt RT tasks
set /proc/sys/kernel/sched_rt_runtime_us to -1
Run high priority on isolated core
schedtool -a 0 -F -p 99 -n -20 -e $CMD
To reduce serial latency/jitter
File descriptor options O_SYNC
Ioctl ASYNC_LOW_LATENCY
Termios VMIN = message size and VTIME = 0
Use tcdrain after issuing write commands
I've read many stack overflow questions similar to this, but I don't think any of the answers really satisfied my curiosity. I have an example below which I would like to get some clarification.
Suppose the client is blocking on socket.recv(1024):
socket.recv(1024)
print("Received")
Also, suppose I have a server sending 600 bytes to the client. Let us assume that these 600 bytes are broken into 4 small packets (of 150 bytes each) and sent over the network. Now suppose the packets reach the client at different timings with a difference of 0.0001 seconds (eg. one packet arrives at 12.00.0001pm and another packet arrives at 12.00.0002pm, and so on..).
How does socket.recv(1024) decide when to return execution to the program and allow the print() function to execute? Does it return execution immediately after receiving the 1st packet of 150 bytes? Or does it wait for some arbitrary amount of time (eg. 1 second, for which by then all packets would have arrived)? If so, how long is this "arbitrary amount of time"? Who determines it?
Well, that will depend on many things, including the OS and the speed of the network interface. For a 100 gigabit interface, the 100us is "forever," but for a 10 mbit interface, you can't even transmit the packets that fast. So I won't pay too much attention to the exact timing you specified.
Back in the day when TCP was being designed, networks were slow and CPUs were weak. Among the flags in the TCP header is the "Push" flag to signal that the payload should be immediately delivered to the application. So if we hop into the Waybak
machine the answer would have been something like it depends on whether or not the PSH flag is set in the packets. However, there is generally no user space API to control whether or not the flag is set. Generally what would happen is that for a single write that gets broken into several packets, the final packet would have the PSH flag set. So the answer for a slow network and weakling CPU might be that if it was a single write, the application would likely receive the 600 bytes. You might then think that using four separate writes would result in four separate reads of 150 bytes, but after the introduction of Nagle's algorithm the data from the second to fourth writes might well be sent in a single packet unless Nagle's algorithm was disabled with the TCP_NODELAY socket option, since Nagle's algorithm will wait for the ACK of the first packet before sending anything less than a full frame.
If we return from our trip in the Waybak machine to the modern age where 100 Gigabit interfaces and 24 core machines are common, our problems are very different and you will have a hard time finding an explicit check for the PSH flag being set in the Linux kernel. What is driving the design of the receive side is that networks are getting way faster while the packet size/MTU has been largely fixed and CPU speed is flatlining but cores are abundant. Reducing per packet overhead (including hardware interrupts) and distributing the packets efficiently across multiple cores is imperative. At the same time it is imperative to get the data from that 100+ Gigabit firehose up to the application ASAP. One hundred microseconds of data on such a nic is a considerable amount of data to be holding onto for no reason.
I think one of the reasons that there are so many questions of the form "What the heck does receive do?" is that it can be difficult to wrap your head around what is a thoroughly asynchronous process, wheres the send side has a more familiar control flow where it is much easier to trace the flow of packets to the NIC and where we are in full control of when a packet will be sent. On the receive side packets just arrive when they want to.
Let's assume that a TCP connection has been set up and is idle, there is no missing or unacknowledged data, the reader is blocked on recv, and the reader is running a fresh version of the Linux kernel. And then a writer writes 150 bytes to the socket and the 150 bytes gets transmitted in a single packet. On arrival at the NIC, the packet will be copied by DMA into a ring buffer, and, if interrupts are enabled, it will raise a hardware interrupt to let the driver know there is fresh data in the ring buffer. The driver, which desires to return from the hardware interrupt in as few cycles as possible, disables hardware interrupts, starts a soft IRQ poll loop if necessary, and returns from the interrupt. Incoming data from the NIC will now be processed in the poll loop until there is no more data to be read from the NIC, at which point it will re-enable the hardware interrupt. The general purpose of this design is to reduce the hardware interrupt rate from a high speed NIC.
Now here is where things get a little weird, especially if you have been looking at nice clean diagrams of the OSI model where higher levels of the stack fit cleanly on top of each other. Oh no, my friend, the real world is far more complicated than that. That NIC that you might have been thinking of as a straightforward layer 2 device, for example, knows how to direct packets from the same TCP flow to the same CPU/ring buffer. It also knows how to coalesce adjacent TCP packets into larger packets (although this capability is not used by Linux and is instead done in software). If you have ever looked at a network capture and seen a jumbo frame and scratched your head because you sure thought the MTU was 1500, this is because this processing is at such a low level it occurs before netfilter can get its hands on the packet. This packet coalescing is part of a capability known as receive offloading, and in particular lets assume that your NIC/driver has generic receive offload (GRO) enabled (which is not the only possible flavor of receive offloading), the purpose of which is to reduce the per packet overhead from your firehose NIC by reducing the number of packets that flow through the system.
So what happens next is that the poll loop keeps pulling packets off of the ring buffer (as long as more data is coming in) and handing it off to GRO to consolidate if it can, and then it gets handed off to the protocol layer. As best I know, the Linux TCP/IP stack is just trying to get the data up to the application as quickly as it can, so I think your question boils down to "Will GRO do any consolidation on my 4 packets, and are there any knobs I can turn that affect this?"
Well, the first thing you can do is disable any form of receive offloading (e.g. via ethtool), which I think should get you 4 reads of 150 bytes for 4 packets arriving like this in order, but I'm prepared to be told I have overlooked another reason why the Linux TCP/IP stack won't send such data straight to the application if the application is blocked on a read as in your example.
The other knob you have if GRO is enabled is GRO_FLUSH_TIMEOUT which is a per NIC timeout in nanoseconds which can be (and I think defaults to) 0. If it is 0, I think your packets may get consolidated (there are many details here including the value of MAX_GRO_SKBS) if they arrive while the soft IRQ poll loop for the NIC is still active, which in turn depends on many things unrelated to your four packets in your TCP flow. If non-zero, they may get consolidated if they arrive within GRO_FLUSH_TIMEOUT nanoseconds, though to be honest I don't know if this interval could span more than one instantiation of a poll loop for the NIC.
There is a nice writeup on the Linux kernel receive side here which can help guide you through the implementation.
A normal blocking receive on a TCP connection returns as soon as there is at least one byte to return to the caller. If the caller would like to receive more bytes, they can simply call the receive function again.
In Ethernet networks, the MAC layer is the first layer to detect the destination address of the received message.
my questions: is that means that the transceiver shall take a copy of each message on the bus and forward it to the MAC layer who will decide to accept that message or discard it? If so, this means that the MAC layer must have a very large buffers to save all that intended and non intended message. am I correct ?
The MAC layer does not typically have much buffering. It may not even be able to store a full packet. Packets instead stream through the MAC.
Packets enter and exit the MAC one flit at a time. It may take hundreds of cycles for a full packet to pass into a MAC depending on the size of the packet and the width of the interface. For example, a MAC with an 8-byte interface (8-byte flit size) will take 1000 cycles to receive an 8kB packet.
The MAC may only have 800 bytes of buffering. In that case, the packet will start coming out the other end after 100 cycles when only 10% of the packet has entered. In fact, many MACs have a latency well below 100 cycles.
Packets which are rejected on the basis of destination address stream in one side but nothing comes out the other side. The frames are simply forgotten/dropped as they arrive.
I'm running a client server configuration over Ethernet and measuring packet latency at both ends. The client (windows) is sending packets every 5 ms (confirmed with wire shark) as it should. Yet, the server (embedded linux) only receives packets at 5 ms intervals for a few seconds, at which point it stops for 300 ms. After this break the latency is only 20 us. After another period of about a few seconds it takes another break for 300 ms. This repeats indefinitely (300ms break, 20 us packet latency burst). It seems as if the server program is being optimized mid-execution to read IO in shorter bursts. Why is this happening?
Disclaimer: I haven't posted the code, as the client and server are small subsets of more complex applications, however, I am willing to factor it out if an obvious answer doesn't present itself.
This is UDP so there is no handshake or any flow control mechanism. Those 300 ms must be because of work the server is doing in the processing of the UDP messages received. During those 300 ms the server has surely lost around 60 messages that were not read from client.
You might probably want to check the server does not take more than 5 ms in processing each message if it uses one thread to process. If the server uses multi-threading to process the messages and the processing takes some time, even if it takes 1 ms, you might be in a situation where at some point all threads are competing for resources and they don't finish in time to read the next message. For the problem you are describing I would bet the server is multithreaded and you have that problem. I cannot assure that 100% for lack of info though. But in any case, you want to check the time it takes to process messages because you might be dealing with real-time requirements.
I spaced out the measurements to 1 in every 1000 packets and now it is behaving itself. I was using printf every 5ms which must've eventually filled the printf tx queue entirely. This then delayed the execution for 300ms. Once printf caught its breath, the program had a queue full of incoming packets and thus was seemingly receiving packets every 20 us.