I am testing websocket communication by sending small messages (180 bytes) per 15 milliseconds. I always get for some time (10 or 21 seconds) big delay (even 250 milliseconds). I checked it in wireshark and i saw that for this specific seconds i got a lot of tcp window update segments. Like on this screen.
It is little strange because Calculated window size is pretty big (about 258944 bytes). Example time for communication:
If i increase message size i dont have this problem and i dont get this segments.
Related
There's a following setup (it's basically a pair of TWS earbuds and a smartphone):
2 audio sink devices (or buds), both are connected to the same source device. One of these devices is primary (and is responsible for handling connection), other is secondary (and simply sniffs data).
Source device transmits a stream of encoded data and sink device need to decode and play it in sync with each other. There problem is that there's a considerable delay between each receiver (~5 ms # 300 kbps, ~10 ms # 600 kbps and # 900 kbps).
It seems that synchronisation mechanism which is already implemented simply doesn't want to work, so it seems that my only option is to implement another one.
It's possible to send messages between buds (but because this uses the same radio interface as sink-to-source communication, only small amount of bytes at relatively big interval could be transferred, i.e. 48 bytes per 300 ms, maybe few times more, but probably not by much) and to control the decoder library.
I tried the following simple algorithm: secondary will send every 50 milliseconds message to primary containing number of decoded packets. Primary would receive it and update state of decoder accordingly. The decoder on primary only decodes if the difference between number of already decoded frame and received one from peer is from 0 to 100 (every frame is 2.(6) ms) and the cycle continues.
This actually only makes things worse: now latency is about 200 ms or even higher.
Is there something that could be done to my synchronization method or I'd be better using something other? If so, what would be the best in such case? Probably fixing already existing implementation would be the best way, but it seems that it's closed-source, so I cannot modify it.
I have read a bunch of posts on SO regarding the computation of end-to-end delay in Veins, but have not found an answer to be fulfilling in explaining why the delay is seemingly too low.
I am using:
Veins 4.7
Sumo 0.32.0
Omnetpp 5.3
Channel switching is turned off.
I have the following code, sending a message from the transmitting node:
if(sendMessage) {
WaveShortMessage* wsm = new WaveShortMessage();
sendDown(wsm);
}
The receiving node computes the delay using the wsm creation time, but I have also tried setting the timestamp on the transmitting side. The result is the same.
simtime_t delay = simTime() - wsm -> getCreationTime();
delayVector.record(delay);
The sample output for the delay vector is as follows:
Item# Event# Time Value
0 165 14.400239402394 2.39402394E-4
1 186 14.500240403299 2.40403299E-4
2 207 14.600241404069 2.41404069E-4
3 228 14.700242404729 2.42404729E-4
Which means that the end-to-end delay (from creation to reception) is equivalent to roughly a quarter of a millisecond, which seems to be quite low - and a fair bit below what is typically reported in the literature. This seems to be consistent with what other people have reported as being an issue (e.g. end to end delay in Veins)
Am I missing something in this computation? I have tried adding load on the network by adding a high number of vehicular nodes (21 nodes within a 1000x50 sandbox on a straight highway, with an average speed of 50 km/h), but the result seems to be the same. The difference is negligible. I have read several research papers that suggest that end-to-end delay should increase dramatically in high vehicular densities.
This end-to-end delay is to be expected. If your application's simulation model does not explicitly model processing delay (e.g., by an application running on a slow general purpose computer), all you would expect to delay a frame is propagation delay (lightspeed, so negligible here) and queueing delay on the MAC (time from inserting frame into TX queue until transmission finishes).
To give an example, for a 2400 bit frame sent at 6 Mbit/s this delay is roughly 0.45 ms. You are likely using slightly shorter frames, so your values appear to be reasonable.
For background information, see F. Klingler, F. Dressler, C. Sommer: "The Impact of Head of Line Blocking in Highly Dynamic WLANs" (DOI 10.1109/TVT.2018.2837157), which also includes a comparison of theory vs. Veins vs. real measurements.
I'm trying to simulate an imergancy breaking application using veins and analyze its performance. Research papers on 802.11p shows that as beacon frequency and number of vehicles increase delay should increase considerably due to mac layer delay of the protocol ( for 50 vehicles at 8Hz - about 300ms average delay).
But when simulating application with veins delay values does not show much different ( it ranges from 1ms-4ms).I've checked the Mac layer functionality and it appears that the channel is idle most of the time. So when a packet reaches Mac layer the channel has already been idle for more than the DIFS so packet gets sent quickly. I tried increasing packet size and reducing bitrate. It increase the previous delay by a certain amount. But drastic increase of delay due to backoff process cannot be seen.
Do you know why this happens ???
As you use 802.11p the default data rate on the control channel is 6Mbits (source: ETSI EN 302 663)
750Mbyte/s = 750.000bytes/s
Your beacons contain 500bytes. So the transmission of a beacon takes about 0.0007 seconds. As you have about 50 cars in your multi lane scenario and for example they are sending beacons with a 10 hertz frequency, it takes about 0.35s from 1 second to transmit your 500 beacons.
In my opinion, this are to less cars to create your mentioned effect, because the channel is idling for about 60% of the time.
I'm using a SAML21 board to accept some data over a serial connection, and at the moment, just mirror it to a serial port on a computer. However this data is 6 bytes at ~250Hz(It was closer to 3KHz before). As far as I can tell I'm tracking the start and end bytes correctly, however my columnar alignment gets out of whack on occasion in realterm.
I have it set up for 6 bytes in single mode. SO all columns should be presenting the same bytes up and down. However, over time as I increase the rate at which I mirror(I am still receiving the data at a fixed rate) the first column's byte tends to float.
I have not used realterm at speeds this high before, so I am not aware of it's limitations.
WINAPI has a function GetMessageTime() that returns the time a message was generated in system time, which has a resolution of 10 to 16 ms. Is there an effective way to get the time an event occurred in interrupt time (100 ns precision), or in some other format with at least 1 ms precision?
Even with Raw Input, the message timing isn't delivered with < 10 ms resolution. (The timing data comes from the WM_INPUT message.) As far as I can tell from the keyboard driver sources, the timing data simply isn't collected with < 10 ms resolution.