I'm trying to emulate slow net link with command tc. I use netem to emulate delay and packet loss and htb to emulate narrow bandwidth, but I find there is a limit option in netem, what does this option do? will it affect the final bandwidth?
I googled it and find something in http://manpages.ubuntu.com/manpages/raring/man8/tc-netem.8.html
which says:
limits the effect of selected options to the indicated number of next packets.
But I still can not understand what it does.
I don't know exactly what netem is doing, but I've found that if you don't set "limit" to a higher value, netem doesn't work correctly - i.e. it discards packets at higher speeds and possibly has other problems, essentially not accurately emulating a real network.
From the mailing list mentioned by CarlH, Stephen Hemminger said:
The limit value is in packets at least when using the default qdisc
inside netem (tfifo). You can also use pfifo and configure it for
packet limit, or bfifo same only bytes. The value 1000 is low, you
want about 50% more than the max packet rate * delay, unless you are
trying to emulate a router with a small queue.
So for a 1 Gbps link, 1 Gbps / 1500 bytes MTU * 100 ms * 1.5 = 12500.
Command:
sudo tc qdisc add dev eth1 root netem limit 12500 delay 100ms loss 1%
I've been using limit 100000, which seems to work fine, but it seems a lower value may be fine.
From https://lists.linuxfoundation.org/pipermail/netem/2007-March/001091.html
The "limit" parameter refers to the number of buffers allocated in the
netem module.
The limit must be adjusted to support the number of frames delayed
(500ms for e.g.) at a given data rate.
Yours sincerely,
Laurent MARIE
The updated documentation says:
limit packets
maximum number of packets the qdisc may hold queued at a time.
Related
I'm experimenting on an LTE connection for checking the maximum rate of bandwidth can be achieved in the uplink.While creating iperf sessions i observed that i'm not able to go beyond 100Kbps in the uplink when the UDP packet size is set as 1400.Apparently when i increased the packet size to 50000 i was able to achieve 2 Mbps in the same link.
Can someone guide me why this performance difference is observed ?When i tried this in a wired channel there i was able to achieve 10Mbps with UDP packet size set as 1400 itself.
What could be the reason for this?
Will trying TCP/IP instead of UDP increase the data throughput?
It probably matters where fragmentation is done - application or IP stack. Your observations show you that IP stack is more efficient.
TCP will be slower. TCP's built-in congestion control will not allow you to send packets until some of already sent have been ACK-ed. That adds round-trip time to performance considerations.
UDP has no such restrictions. It can (ab)use the network to its full potential.
a little info on what i'm trying to achieve here first. I'm using a Texas Instrument board EVM6678LE, and what i am trying to do is to increase the UDP transfer rate between the board and my PC.
I've increased the MTU on my PC through netsh>interface>ipv4 to 15,000. But when i ping the board from my PC i am only able to ping up to "ping 192.168.2.100 -l 10194", if i ping with 195bytes onwards i'll receive a request timeout. Is this a limitation of my PC or something?
Does anyone have any idea what could be the possible cause of this? Any advice or suggestions at all would be welcome. As the only way to increase the transfer rate i could think of it increasing the per packet size which reduces overhead. And at 10k i have a rate of around 9.1MB/s, and i'm trying to attain 25MB/s.
Thanks!
Increasing the MTU on your PC may not prevent fragmentation. I don't know exactly what is controlling this, but your network card or driver can fragment the packet even when MTU is not reached. Use a sniffer like Wireshark to see how the packets are sent.
About the timeout, it is possible that your board rejects fragmented pings (because of Ping of Death protection). There is also a possibility that its packet buffer is 10kB (10240) bytes long, and can't receive larger packets. Also, make sure that the receiving endpoint have a matching MTU.
Anyway, if you are trying to increase transfer rate, you are on the wrong track. The overhead for UDP is 8 bytes, IP 20 bytes, Ethernet 18 bytes, which make a total of 46 bytes (oh, coincidence, 46+10194 is exactly 10240). 46 bytes overhead for 1024 MTU is 95.5%. 46 bytes for 4096 is 98.9%, 46 bytes for 16384 is 99.7%. That means you gain +3.5% transfer rate from 1024 to 4096 MTU, and another +0.8% from 4096 to 16384. The gain is just ridiculous, and you should just let the MTU to the common default 1500.
Anyway, going from 9.1MB/s to 25MB/s just by changing the MTU is IMPOSSIBLE (if it was, why the PC default is not higher ?). Here I guess you are using Fast Ethernet (100BASE-T), and you are already transferring near full bandwidth. To get higher rates, you would need to have Gigabit Ethernet (1000BASE‑T). That means you need both hardware endpoints to support 1000BASE-T.
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'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)
I intend to write an application where I will be needing to calculate the network bandwidth along with latency and packet loss rate. One of the constraints is to passively measure the bandwidth (using the application data itself).
What I have read online and understood from a few existing applications is that almost all of them use active probing techniques (that is, generating a flow of probe packets) and use the time difference between arrival of the first and last packets to calculate the bandwidth.
The main problems with such a technique are that it floods the network with probe packets, which runs longer and is not scalable (since we need to run the application at both ends).
One of the suggestions was to calculate the RTT of a packet by echoing it back to the sender and calculate the bandwidth using the following equation:
Bandwidth <= (Receive Buffer size)/RTT.
I am not sure how accurate this could be as the receiver may not always echo back the packet on time to get the correct RTT. Use of ICMP alone may not always work as many servers disable it.
My main application runs over a TCP connection so I am interested in using the TCP connection to measure the actual bandwidth offered over a particular period of time. I would really appreciate if anybody could suggest a simple technique (reliable formula) to measure the bandwidth for a TCP connection.
It is only possible to know the available bandwidth by probing the network. This is due to that a 80% utilized link will still send echo-packets without delay, i.e. it will appear to be 0% occupied.
If you instead just wish to measure the bandwidth your application is using, it is much easier. E.g. keep a record of the amount of data you have transferred in the last second divided into 10ms intervals.
Active probing technique and its variants are bandwidth estimation algorithm. You dont want to use these algorithm to measure bandwidth. Note the difference between 'measure' and 'estimate'.
If you want to use tcp to measure bandwidth, you should be aware that tcp bandwidth is influenced by latency.
The easiest way to measure bandwidth using tcp is by sending tcp packets and measure the transferred bandwidth. It will flood the network. None of the non flooding algorithm reliable in high speed network. Plus, non flooding algorithm assume the channel is clear from traffic. If there is other traffic inside the channel, the result would be skewed. Im not suprised if the result would not make sense.