Three LTE small cells located in a room approximately same distance between each other which are having some interference. All the cells are same having bandwidth (15MHz) and same frequency configured.
Each cell is connected with a three separate UE(Phone) and calculated downlink and uplink throughput,
Each phone is receiving ~ ( 25 to 38 Mbps).
Later i changed the bandwidth of 1 cell to 10 MHz and other two cells to 5 Mhz and appropriate frequency and calculated downlink and uplink throughput,
Each phone is higher throughput compared to previous test.~( 29 to 46 Mbps).
My question here is , as the bandwidth is reduced significantly i assume the through also should drop as well, but the throughput is higher than before.
All the cells are same having bandwidth (15MHz) and same frequency configured.
That's the key moment, because they do interfere with each other. What is the reason of using the same frequency band for three cells?
Each phone is higher throughput compared to previous test.~( 29 to 46 Mbps).
My question here is , as the bandwidth is reduced significantly i assume the through also should drop as well, but the throughput is higher than before.
Most likely, reducing the bandwidth reduced the level of total interference.
Related
What would be the best practice for writing to the flash on a regular basis.
Considering the hardware I am working on is supposed to have 10 to 20 years longevity, what would be your recommandation? For example, is it ok I write some state variables every 15 minutes thru Preferences?
That depends on
number of erase cycles your Flash supports,
size of the NVS partition where you store data and
size and structure of the data that you store.
Erase cycles mean how many times a single sector of Flash can be erased before it's no longer guaranteed to work. The number is found in the datasheet of the Flash chip that you use. It's usually 10K or 100K.
Preferences library uses the ESP-IDF NVS library. This requires an NVS partition to store data, the size of which determines how many Flash sectors get reserved for this purpose. Every time you store a value, NVS writes the data together with its own overhead (total of 32 bytes for primitive data types like ints and floats, more for strings and blobs) into the current Flash sector. When the current sector is full, it erases the next sector and proceeds to write there; thereby using up sectors in a round robin fashion as write requests come in.
If we assume that your Flash has 100K erase cycles, size your NVS partition is 128 KiB and you store a set of 8 primitive values (any int or float) every 15 minutes:
Each store operation uses 8 * 32 = 256 bytes (32 B per data value).
You can repeat that operation 131072 / 256 = 512 times before you've written to every sector of your 128KiB NVS partition (i.e. erased every sector once)
You can repeat that cycle 100K times so you can do 512 * 100000 = 51200000 or roughly 5.1M store operations before you've erased every sector its permitted maximum number of times.
Considering the interval of 15 minutes creates 365 * 24 * 4 = 35040 operations per year, you'd have 51200000 / 35040 = 1461 years until Flash is dead.
Obviously, if your Flash chip is rated at 10K erase cycles, it drops to only 146 years.
There's probably some NVS overhead in there somewhere that I didn't account for, and the Flash erase cycle ratings are not 100% reliable so I'd cut it in half for good measure - I would expect 700 or 70 years in real life.
If you store non-primitive values (strings, blobs) then the estimate changes based on the length of that data. I don't know to calculate the exact Flash space used by those but I'd guess 32B plus length of your data multiplied by 10% of NVS overhead. Plug in the numbers, see for yourself.
I have setup a sample Kafka cluster on AWS and am trying to identify maximum throughput possible with the given configurations. I am currently following post provided here for this analysis.
https://engineering.linkedin.com/kafka/benchmarking-apache-kafka-2-million-writes-second-three-cheap-machines
I would appreciate it if you could clarify the following issues.
I observed a throughput of 40MB/s for messages of size 512 bytes ( single producer - single consumer ) with given hardware. Assume I need to achieve a throughput of 80MB/s.
As I understand one way to do this to increase the number of partitions per topic and increase the number of threads in producer and consumer. ( Assuming I do not change the default values for batch size, compression ratio etc. )
How to find the maximum throughput possible with given hardware? The point after which we are required to improve our hardware resources if we are to further improve the throughput?
( In other words how to make the decision "With X GB RAM and Y GB disk space this is the maximum throughput I can achieve. If I need to further improve the throughput I have to upgrade RAM to XX GB and disk space to YY GB" )
2.Should we scale the cluster vertically or horizontally? What is the recommended approach?
Thank you.
If we define throughput as the volume of data transmitted over the network per second, the maximum throughput should not exceed #machine number * bandwidth. Given a single machine whose NIC is configured with 1Gbps, the max TPS on single machine cannot be larger than 1Gbps. In your case, TPS is 40MB/s, namely 320Mbps，which is quite less than 1Gbps, meaning there is still room for improvement. However, if your target is far larger than 1Gbps, you definitely need more machines.
AFAIK, bandwidth is the most likely cause for the system bottleneck. Unlike CPU and RAM, it's not easy to scale vertically, so a horizontally scaling might be an option.
You could do some maths before scaling. Say the throughput target is "produce 2 billion of records with 512Bytes in 1 hour". That's to say, the TPS has to achieve 2,000,000,000 * 8 * 512 / 3600 / 1024 / 1024 = 2170mbps. Assuming available bandwidth for single machine is 700mbps(Over 70% usage normally brings 'packet loss'), at least 4 machines should be planned for the producer application.
I have a little network with a client and a server, and I'm testing the FrameRate, changing the dimension of the packet. Particulary, I have an image, changing threshold, I extract keypoints and descriptors and then I send a fixed number of packets (with different dimension with different threshold). Problems happen when udp packets are under MTU dimension, reception rate decrease and frame rate tend to be constant. I verify with wireshark that my reception times are correct, so isn't a server code problem.
this is the graph with the same image sends 30 times for threshold with a 10 step from 40 to 170.
i can't post the image so this is the link
Thanks for the responces
I think that none will interest this answer, but we arrived to the conclusion that the problem is a problem in wifi dongle's drivers.
The trasmission window does not go under a determined time's threshold. So under a determined amount of data while time remains constant, decreases.
I want to know how can one calculate bandwidth requirements based upon flows and viceversa.
Meaning if I had to achieve total of 50,000 netflows what is the bandwidth requirement to produce this number? Is there a formula for this. I'm using this to size up flow analyzer appliance. If its license says supports 50,000 flows what does this means. How more bandwidth if i increase I would lose the license coverage?
Most applications and appliances list flow volume per second, so you are asking what the bandwidth requirement is to transport 50k netflow updates per second:
For NetFlow v5 each record is 48 bytes each with each packet carrying 20 or so records, with 24 bytes overhead per packet. This means you'd use about 20Mbps to carry the flow packets. If you used NetFlow v9 which uses a template based format it might be a bit more, or less, depending on what is included in the template.
However, if you are asking how much bandwidth you can monitor with 50k netflow updates per second, the answer becomes more complex. In our experience monitoring regular user networks (using google, facebook, etc) an average flow update encompasses roughly 25kbytes of transferred data. Meaning 50,000 flow updates per second would equate to monitoring 10Gbps of traffic. Keep in mind that these are very rough estimates.
This answer may vary, however, if you are monitoring a server-centric network in a datacenter, where each flow may be much larger or smaller, depending on content served. The bigger each individual traffic session is, the higher the bandwidth will be you can monitor with 50,000 flow updates per second.
Finally, if you enable sampling in your NetFlow exporters you will be able to monitor much more bandwidth. Sampling will only use 1 in every N packets, thus skipping many flows. This lowers the total number of flow updates per second needed to monitor high-volume networks.
I've been tinkering around with the BLE (Bluetooth Low Energy) connectivity classes quiet a bit lately and haven't been able to make it transfer data any faster than 1KB / 5 seconds. I believe, in the documentation, it says the max speed is 60 bytes per 20 milliseconds. With data transfer and counting the Ack transfer after each set of packets, I believe we should be able to go as fast as 1.5KB per second. So my code is around 7-8 times slower than it should be.
I'm just wondering if anyone has been able to do data transfer in BLE as fast as the documentation says it should be able to do. What sort of speed are you getting if faster than mine?
Thanks a lot
see at the guidlines of apple and you will see that a connection update request is required to speed up your connection.
https://developer.apple.com/hardwaredrivers/BluetoothDesignGuidelines.pdf
I have min=20ms max 40 ms
I hope I could help
Roman
If you are able to use higher MTU size (negotiated by the iOS) then you would be able to increase the bandwidth even more, because there is a 4 byte L2CAP header and a 3 byte ATT header that wouldn't be transmitted more than in one packet.
If you are able to transmit 6 packets pr connection interval, then you would be able to put in 35 byte extra per connection interval (the 7 byte header would still be there for the first packet) The MTU size could also be split over several connection intervals, increasing the throughput with 7 more bytes pr connection interval. (Just takes longer time to assemble the packet again.) The max MTU size allowed by ATT is 515 bytes (Max size of att is 512 bytes + 3 byte header for opcode + handle)