As per the definition of the 'Bit error' by protocol developer organization - Bosch : A bit error is detected at the bit time when the bit value monitored by the trasmitter is not the same as the bit actually transmitted by it.
For example, consider a practical scenario on the CAN bus : There are 2 CAN nodes A and B with an Identifier each to transmit on the bus. These 2 nodes start to transmit their respective CAN IDs on the bus and the arbitration mechanism begins. After the Arbitarion is done with, the node with the HIGH priority CAN ID will get the CAN bus access to continue transmission of remaining bits of its CAN frame. The other node [or any other nodes which might be present] on the bus become the receivers of that CAN frame and do not attempt to transmit anything during this time.
Question : If during this time , there is only 1 node which is transmitting and all other nodes are in receive mode, how can a bit error occur ?
1] Could a bit error occur because of a disturbance / EMI effect on the bus line ?
2] Could the sampling and iterpretation of bit sent by the node become faulty at the chip level, leading the CAN chip itself to detect it as a bit error ?
3] Any other reason leading to this ?
Bit error occurs when Transmitted data is != to Rx Data. Though all the other nodes have now gone to receiving mode, due to problems in the senders transceiver/noise effects, a bit error can occur. Every bit error possibility in the CAN Frame(upto EOM) will be checked by the sender
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As far as I unterstand BLE uses two 1-bit fields for applying sequence numbers to packets (SN, NESN). From my (admittedly basic) knowledge about (wireless) communication a 1-bit sequence number is perfectly fine as long as a sender does not continue sending data until the last message is acknowledged by the receiver.
Because of that it is trivial to understand how BLE works with a one-packet-per-interval scheme. However BLE allows multiple packets in a single connection interval. So far I couldn't find any information on how this scenario is handled without allocating more bits for larger sequence numbers.
Any pointers in the right direction on where I'm going wrong or where I can read up on this would be appreciated.
Is there a limit on maximum number of packets (LE_DATA) that could be send by either slave or master during one connection interval?
If this limit exists, are there any specific conditions for this limit (e.g. only x number of ATT data packets)?
Are master/slave required or allowed to impose such a limit by specification?
(I hope I'm not reviving a dead post. But I think the section 4.5.1 is better suited to answer this than 4.5.6.)
The spec doesn't define a limit of packets. It just states the following:
4.5.1 Connection Events - BLUETOOTH SPECIFICATION Version 4.2 [Vol 6, Part B]
(...)
The start of a connection event is called an anchor point. At the anchor point, a master shall start to transmit a Data Channel PDU to the slave. The start of connection events are spaced regularly with an interval of connInterval and shall not overlap. The master shall ensure that a connection event closes at least T_IFS before the anchor point of the next connection event. The slave listens for the packet sent by its master at the anchor point.
T_IFS is the "Inter Frame Space" time and shall be 150 μs. Simply put it's the job of the master to solve this problem. As far as I know iOS limits the packet number to 4 per connection event for instance. Android may have other hard coded limits depending on the OS version.
There is max data rate that can be achieved both on BT and BLE. You can tweak this data rate by changing MTU (maximum transmission unit - packet size) up to max MTU both ends of transmission can handle. But AFAIK there is no straight constraint on number of packets, besides physical ones imposed by the data rate.
You can find more in the spec
I could find the following in Bluetooth Spec v4.2:
4.5.6 Closing Connection Events
The MD bit of the Header of the Data Channel PDU is used to indicate
that the device has more data to send. If neither device has set the
MD bit in their packets, the packet from the slave closes the
connection event. If either or both of the devices have set the MD
bit, the master may continue the connection event by sending another
packet, and the slave should listen after sending its packet. If a
packet is not received from the slave by the master, the master will
close the connection event. If a packet is not received from the
master by the slave, the slave will close the connection event.
Two consecutive packets received with an invalid CRC match within a
connection event shall close the event.
This means both slave and masters have self-imposed limit on number of packets they want to transmit during a CI. When either party doesn't wish to send more data, they just set this bit to 0 and other one understands. This should usually be driven by number of pending packets on either side.
Since I was looking for logical limits due to spec or protocol, this probably answers my question.
Physical limits to number packets per CI would be governed by data rate, and as #morynicz mentioned, on MTU etc.
From my understanding, the limit is: min{max master event length, max slave event length, connection interval}.
To clarify, both the master and slave devices (specifically, the BLE stack thereof) typically have event length or "GAP event length" times. This time limit may be used to allow a central and/or advertiser and/or broadcaster to schedule the "phase offset" of more than one BLE radio activity, and/or limit the CPU usage of the BLE stack for application processing needs. E.g. a Nordic SoftDevice stack may have a default event length of 3.75ms that is indefinitely extendable (up to the connection interval) based on other demands on the SoftDevice's scheduler. In Android and iOS BLE implementations, this value may be opaque or not specified (e.g. Android may set this value to "0", which leaves the decision up to the controller implementation associated with the BLE chip of that device).
Note also that either the master or the slave may effectively "drop out" of a connection event earlier than these times if their TX/RX buffers are filled (e.g. Nordic SoftDevice stack may have a buffer size of 6 packets/frames). This may be implemented by not setting the MD bit (if TX buffer is exhausted) and/or "nacking" with the NESN bit (if RX buffer is full). However, while the master device can actually "drop out" by ending the connection event (not sending any more packets), the slave device must listen for master packets as long as at least one of master and slave have the MD bit set and the master continues to transmit packets (e.g. the slave could keep telling the master that it has no more data and also keep NACKing master packets because it has no more buffer space for the current connection event, but the master may keep trying to send for as long as it wants during the connection interval; not sure how/if the controller stack implements any "smarts" regarding this).
If there are no limits from either device in terms of stack-specified event length or buffer size, then presumably packets could be transmitted back and forth the entire connection interval (assuming at least one side had data to send and therefore set the MD bit). Just note for throughput calculation purposes that there is a T_IFS spacing (currently specified at 150us) between each packet and before the end of the connection interval.
I've never worked with bluetooth before. I have to sends data via BLE and I've found the limit of 20 bytes per chunk.
The sender is an Arduino and the receiver could be both an Android or a Node.js app on a pc.
I have to send 9 values, stored in float values, so 4 bytes * 9 = 36 bytes. I need 2 chunks for all my data via BLE. The receiving part needs both chunks to process them. If some data are lost, I don't care.
I'm not expert in network protocols and I think I have to give each message an incremental timestamp so that the receiver can glue the two chunks with the same timestamp or discard the last one if the new timestamp is higher. But I'm not sure how to do a checksum, if I really need it or not, if I really have to care about it, or if - for a simple beta version of my system - I can ignore all those problems..
Does anyone can give me some advice? Like examples of similar situations handled with BLE communication?
You can get around the size limitation using the "Read Blob Request" of ATT. It allows you to read an attribute and also give an offset. So, you can use it to read the attribute with an offset of 0, if there's more than ATT_MTU bytes than you can request again with the offset at ATT_MTU*1, if there's still more ATT_MTU*2, etc... (You can read it in 3.4.4.5 of the Bluetooth v4.1 specifications; it's in the 4.0 spec too but I don't have that in front of me right now)
If the value changes between request, I'm not sure how you could go about detecting such a change. You could have the attribute send notifications when there's a change to interrupt the process in case the value changes in the middle of reading it.
I have created a small wireless network using a few PIC microcontrollers and nRF24L01 wireless RF modules. One of the PICs is PIC18F46K22 and it is used as the main controller which sends commands to all other PICs. All other (slave) microcontrollers are PIC16F1454, there are 5 of them so far. These slave controllers are attached to various devices (mostly lights). The main microcontroller is used to transmit commands to those devices, such as turn lights on or off. These devices also report the status of the attached devices back to the main controller witch then displays it on an LCD screen. This whole setup is working perfectly fine.
The problem is that anybody who has these cheap nRF24L01 modules could simply listen to the commands which are being sent by the main controller and then repeat them to control the devices.
Encrypting the commands wouldn’t be helpful as these are simple instructions and if encrypted they will always look the same, and one does not need to decrypt it to be able to retransmit the message.
So how would I implement a level of security in this system?
What you're trying to do is to prevent replay attacks. The general solution to this involves two things:
Include a timestamp and/or a running message number in all your messages. Reject messages that are too old or that arrive out of order.
Include a cryptographic message authentication code in each message. Reject any messages that don't have the correct MAC.
The MAC should be at least 64 bits long to prevent brute force forgery attempts. Yes, I know, that's a lot of bits for small messages, but try to resist the temptation to skimp on it. 48 bits might be tolerable, but 32 bits is definitely getting into risky territory, at least unless you implement some kind of rate limiting on incoming messages.
If you're also encrypting your messages, you may be able to save a few bytes by using an authenticated encryption mode such as SIV that combines the MAC with the initialization vector for the encryption. SIV is a pretty nice choice for encrypting small messages anyway, since it's designed to be quite "foolproof". If you don't need encryption, CMAC is a good choice for a MAC algorithm, and is also the MAC used internally by SIV.
Most MACs, including CMAC, are based on block ciphers such as AES, so you'll need to find an implementation of such a cipher for your microcontroller. A quick Google search turned up this question on electronics.SE about AES implementations for microcontrollers, as well as this blog post titled "Fast AES Implementation on PIC18F4550". There are also small block ciphers specifically designed for microcontrollers, but such ciphers tend to be less thoroughly analyzed than AES, and may harbor security weaknesses; if you can use AES, I would. Note that many MAC algorithms (as well as SIV mode) only use the block cipher in one direction; the decryption half of the block cipher is never used, and so need not be implemented.
The timestamp or message number should be long enough to keep it from wrapping around. However, there's a trick that can be used to avoid transmitting the entire number with each message: basically, you only send the lowest one or two bytes of the number, but you also include the higher bytes of the number in the MAC calculation (as associated data, if using SIV). When you receive a message, you reconstruct the higher bytes based on the transmitted value and the current time / last accepted message number and then verify the MAC to check that your reconstruction is correct and the message isn't stale.
If you do this, it's a good idea to have the devices regularly send synchronization messages that contain the full timestamp / message number. This allows them to recover e.g. from prolonged periods of message loss causing the truncated counter to wrap around. For schemes based on sequential message numbering, a typical synchronization message would include both the highest message number sent by the device so far as well as the lowest number they'll accept in return.
To guard against unexpected power loss, the message numbers should be regularly written to permanent storage, such as flash memory. Since you probably don't want to do this after every message, a common solution is to only save the number every, say, 1000 messages, and to add a safety margin of 1000 to the saved value (for the outgoing messages). You should also design your data storage patterns to avoid directly overwriting old data, both to minimize wear on the memory and to avoid data corruption if power is lost during a write. The details of this, however, are a bit outside the scope of this answer.
Ps. Of course, the MAC calculation should also always include the identities of the sender and the intended recipient, so that an attacker can't trick the devices by e.g. echoing a message back to its sender.
I seem to have an issue with IP queue.
I have a linux machine that I am using to run some experiments.
The linux machine is configured to be a router, having two NICs, connecting two other computers, and managing their network traffic.
All incoming packages are captured, using iptables, and analyzed by a C application.
The application analyzing the packets has a built-in delay, as part of the experiment.
So I have one very fast computer sending packets through my linux-router and a (relatively) slow linux-router that analyses and deals with the packets, one by one.
This situation leads to the fact that when I fire up a sender application on one of the computers connected to the linux-router, my IP queue on the linux-router gets filled up (almost) instantaneously.
The IP queue's max length is currently set to 1024, and if it overflows, the packets are dropped. This is expected and i'm OK with it.
But, (and this is where it gets interesting), every now and then I get the following error:
"Failed to receive netlink message: No buffer space available"
At start, I thought this was due to the IP queue overflow, but after some analysis i found that sometimes I get the error even if the IP queue buffer did not overflow, and sometime I DON'T get the message even though the buffer DID overflow.
When I run > cat /proc/net/ip_queue, I get the following table (also used to monitor the IP queue overflow):
Peer PID : 27389
Copy mode : 2
Copy range : 65535
Queue length : 0
Queue max. length : 1024
Queue dropped : 1166875
Netlink dropped : 2916
Looking at the last two values, Queue dropped seems to refer to packets that did not manage to get into the IP queue because the buffer was full. I can see this value rise as I bombard the linux-router. Netlink dropped ( as it's name implies :) ) seems to have to do with the error i'm getting.
I did my best to search for material on this error, but wasn't able to find anything that seemed to point me in the required direction.
Bottom line: Why am I getting this error and what can I do to avoid it?
try increasing the receive buffer space on your netlink receive socket using setsockopt .. this should remove the error !