I found that, in Linux, there are many loadable modules for congestion control algorithms of TCP (cubic, new-reno, veno, vegas,...). But there are two modules that make me confused, one is "HTCP" and the other one is "highspeed". Doesn't HTCP stands for highspeed TCP? So what is the differences between "HTCP" and "highspeed" module here?
Thanks in advance for pointing out the differences.
Doesn't HTCP stands for highspeed TCP?
No. It stands for TCP for high-speed and long-distance networks and is described in this document from the Hamilton Institute. HighSpeed TCP is published in RFC 3649.
So what is the differences between "HTCP" and "highspeed" module here?
The common point first, is that they want to turn high bandwidth long distance networks more efficient. The main difference, is that HighSpeed TCP relies on the packet drop rate while H-TCP relies on time elapsed since the last packet drop. As a result, H-TCP seems faster to have its Window Size to recover after a congestion event, which will then give a higher throughput.
Related
I want to make sure the latency between my app and the bluetooth headphones is accounted for, but I have absolutely no idea how I can get this value. The closest thing I found was:
BluetoothLEPreferredConnectionParameters.ConnectionLatency which is only available on Windows 11... Otherwise there isn't much to go on.
Any help would be appreciated.
Thanks,
Peter
It's very difficult to get the exact latency because it is affected by many parameters - but you're on the right track by guessing that the connection parameters are a factor of this equation. I don't have much knowledge on UWP, but I can give you the general parameters that affect the speed/latency, and then you can check their availability in the API or even contact Windows technical team to see if these are supported.
When you make a connection with a remote device, the following factors impact the speed/latency of the connection:-
Connection Interval: this specifies the interval at which the packets are sent during a connection. The lower the value, the higher the speed. The minimum value as per the Bluetooth spec is 7.5ms.
Slave Latency: this is the value you originally mentioned - it specifies the number of packets that can be missed before a connection is considered lost. A value of 0 means that you have the fastest most robust connection.
Connection PHY: this is the modulation on which the packets are sent. If both devices support 2MPHY, then the connection should be quicker.
Data Length/MTU Extension: these are two separate features but I am looping them together becuase the effect is the same - more bytes are sent per packet, which results in a higher throughput. The maximum value is 251 bytes per packet.
You can find more information about these parameters here:-
A Practical Guide to BLE Throughput
Maximizing BLE Throughput: Everything You Need to Know
Bluetooth 5 Speed - How to Achieve Maximum Throughput
And below are some other links that might help you understand what is supported on UWP:-
Bluetooth Developer FAQ
BluetoothLEConnectionParameters.OptimizedProperty
Bluetooth LE Preferred Connection Parameter Class
Bluetooth LE Connection PHY class
How to Change MTU Size and PHY on Windows UWP C++
I'm new to BLE, and bluetooth in general, but I'm on a project that includes communication via BT 5.
As the BLE communication has to transmit around 2 bytes, to 1 MB at a time, I'm looking for a way to optimize the transmission time.
I know the pros n cons for the lower transmission freq (125 kbps), and for the highest transmission freq (2 Mbps), and for the DLE of 251 PDU bytes, but what I see from different forums and articles, the throughput mostly depend on the connection parameters as the connection interval and the packets per connection event. But where does the transmission frequency come in?
I've tried searching this forum for an answer, and several others, and even the BT core specification, but I haven't been able to find a solution for my problem.
If you read my answer at Why is BLE 4.2 faster than BLE 4.1, you can see that there are many components affecting the overall transfer speed.
You first have the radio transmission rate itself, which sets the upper limit.
You then have the overhead between all packets that becomes less apparant as longer packets you have.
The connection interval and length of each connection event can be important if you want the throughout to be high. If there is only one connection and the Bluetooth chip is not too stupid, the connection event length will fill the connection interval and therefore the connection interval doesn't really matter. However, if there are other conflicting radio events scheduled in a way that the connection event must be closed, the transmission cannot continue until the next connection event. So in this case, the throuhput will be higher if you lower the connection interval. So as a summary it highly depends on which Bluetooth stack the chip runs, how it's configured by the host and how many active connections you have.
The transmission rate controls your underlying bitrate but on top of that sits different layers of the BLE protocol which slow down the realizable throughput. This article has general derivation of how the different layers impact throughput in case that's useful!
I am trying to figure out what the maximum throughput of a Bluetooth 2.1 SPP connection is.
I found 2 publications concerned with the topic (1, 2) and they both show diagrams, which show the throughput as a function of the Signal to noise ratio (that I can assume to be perfect for my concideration) and the kind of ACL package used. My problem is, I have no Idea which ACL packets are used. How is this decision made? Is it made on the fly, like "what's needed to transfer the current data is used"?
Furthermore, in the Serial Port Profile specification (chapter 2.3) I found this sentence:
This profile requires support for one-slot packets only. This means that this profile
ensures that data rates up to 128 kbps can be used. Support for higher rates is optional.
The last sentence realy confuses me. How do I find out whether this "option" applies in my case?
This means that in SPP mode, all bluetooth modules should work up to 128kbps, and some modules may work even faster.
Under SPP is RFCOMM, which tries to deliver the packets as quickly as possible. If only one packet is sent in one timeslot, you get the 128kbps. The firmware of the bluetooth module, or the HCI driver however can do things differently.
There are SPP speeds up to 480kbps reported - however this requires that both SPP modules are from the same vendor (e.g. BlueGiga iWrap modules can do this speed).
On the other end, if you're connecting to an unknown device, for example a BT112, or an RN41 module to an Android device, the actual usable SPP bandwidth can be much lower than 128 kbps (I have measurements as low as 10kbps).
In case of some old generation iPhones, the usable SPP bandwidth is around 8 kbps.
It is a wise idea to treat "standards" and "datasheets" very conservative, and do your own measurements if actual net data bandwidth is critical.
Even though BT, BT+EDR has theoretical on-the-air bitrates of 3Mbps, the actual usable net data bandwidth is a way smaller.
On a Linux router I wrote a C-program which uses pcap to get the IP header, and length of the packet. In that way I am able to gather statistics and measure bandwidth based on IP. Pretty neat. :-)
Now the traffic and number of users has grown, and the old program starts to struggle. That is, the router struggles to cope with the massive amount of packets. It's over 50000 packets per second all in all in "prime time".
The program itself is pretty optimized. I don't want to show off, but I believe it's as good as it can get. It reads the IP header, and the packet length. It then converts the IP to a index (just a simple subtract), and the length of the packet is stored (accumulated) in an array. Every now and then (actually a SIGALRM) it stores the array in a MySQL database.
My question is: Is there any other way to tap into an ethernet device to get the bit-stream "cheaper" than pcap?
I can of course modify the ethernet driver to include single IP statistics gathering, but that seems a little overkill.
Basically my program is a 'tcpdump' on a busy eth0 and that will eventually kill my router.
Have you considered PF_RING? It's still the pcap-like world, but on steroids - thanks to the zero-copy mechanism:
As you see, there is a kernel module that provides low-level packet copying into the PF_RING buffer, and there is the userland part that allows to access this buffer.
Who needs PF_RING?
Basically everyone who has to handle many packets per second. The term ‘many’ changes according to the hardware you use for traffic analysis. It can range from 80k pkt/sec on a 1,2GHz ARM to 14M pkt/sec and above on a low-end 2,5GHz Xeon. PF_RING not only enables you to capture packets faster, it also captures packets more efficiently preserving CPU cycles....
I highly recommend you to use PF_RING ZC. It could be found under /userland/examples_zc. it is part of pf_ring.
you can handle and capture tens of Gbps traffics in line rate by pf_ring zc.
I'm trying to understand the correct way to increase the socket buffer size on Linux for our streaming network application. The application receives variable bitrate data streamed to it on a number of UDP sockets. The volume of data is substantially higher at the start of the stream and I've used:
# sar -n UDP 1 200
to show that the UDP stack is discarding packets and
# ss -un -pa
to show that each socket Recv-Q length grows to the nearly the limit (124928. from sysctl net.core.rmem_default) before packets are discarded. This implies that the application simply can't keep up with the start of the stream. After discarding enough initial packets the data rate slows down and the application catches up. Recv-Q trends towards 0 and remains there for the duration.
I'm able to address the packet loss by substantially increasing the rmem_default value which increases the socket buffer size and gives the application time to recover from the large initial bursts. My understanding is that this changes the default allocation for all sockets on the system. I'd rather just increase the allocation for the specific UDP sockets and not modify the global default.
My initial strategy was to modify rmem_max and to use setsockopt(SO_RCVBUF) on each individual socket. However, this question makes me concerned about disabling Linux autotuning for all sockets and not just UDP.
udp(7) describes the udp_mem setting but I'm confused how these values interact with the rmem_default and rmem_max values. The language it uses is "all sockets", so my suspicion is that these settings apply to the complete UDP stack and not individual UDP sockets.
Is udp_rmem_min the setting I'm looking for? It seems to apply to individual sockets but global to all UDP sockets on the system.
Is there a way to safely increase the socket buffer length for the specific UDP ports used in my application without modifying any global settings?
Thanks.
Jim Gettys is armed and coming for you. Don't go to sleep.
The solution to network packet floods is almost never to increase buffering. Why is your protocol's queueing strategy not backing off? Why can't you just use TCP if you're trying to send so much data in a stream (which is what TCP was designed for).