When using a non beaconing Zigbee network, I know that the 802.15.4 spec defines the use of CSMA-CA to control when two devices get access to a channel to make sure no two nodes "step on each others toes" so to speak. My understanding is that very simply, it requires each node to "listen before talking". Is that correct? Is there more information on the Zigbee implementation of this? In other words, where do I go to learn more about how to program a Zigbee chip to implement the same?
Also, if i have 20 end nodes sending data asynchronously to one coordinator, is the channel access mechanism enough to ensure that they do not broadcast at the same time and flood the coordinator? If five nodes (for example) attempt to broadcast at the same time, how will mutual exclusion be ensured? Where can I get some details on that?
Thanks
Rishi
The maximum size of a 802.15.4 packet is 1024 bits of payload. So the maximum duration of the frame (running in standard 250kbps rate on the 2.4GHz band) is about 5ms when you take preamble etc into account. If your end devices are polling at 1 poll/second it should easily manage 20 end nodes I think. If it gets too much the exponential backoff should ease the collision rate.
I'm sure you've seen these when searching, but just in case:
http://www.prismmodelchecker.org/casestudies/zigbee.php
http://www.dagstuhl.de/Materials/Files/07/07101/07101.FruthMatthias.Slides.pdf
http://www-public.it-sudparis.eu/~gauthier/Tools/802_15_4_MAC_PHY_Usage.pdf
Related
Looking at:
OMNET++: How to obtain wireless signal power?
and
https://github.com/inet-framework/inet/blob/master/examples/wireless/scaling/omnetpp.ini
there seem to be no power consumption related settings to packets that are sent in a UnitDiskRadio.
Is there a way of setting packet power consumption in a unit disk radio medium, or, conversely, communication range in ApskScalarRadioMedium?
UnitDiskRadio is a simplified version of a radio, where you are not interested in the transmission, propagation, attenuation etc. details. You just want to have a clear cut transmission distance. Above that, the transmission always fails, below that the transmission always succeed. This is simple, fast and suitable if you want to simulate high level behavior like application level or routing. You really don't care how much your radio draws from a power grid (or battery) in this case.
On the other hand, if you are interested in low level details, the whole radio transmission process should be modeled. In this case, you model the power draw and based on that transmission and there is no clear cut transmission range. Whether a transmission succeeds is a probabilistic outcome depending on power, antenna configuration, encoding, modulation, noise and a lot of other stuff, so you cannot set it as a simple "range".
TLDR: No, you cannot set both of them on the same radio.
PS: and make sure that you do not mix and match various power parameters. The first question you linked is about getting the power of a received packet (i.e. how strong that signal was when it was received). The second link show how to configure the transmission power (that goes out on the antenna), and in the question you are referring to power consumption which is a third thing, meaning how much you draw from a battery to make the transmission. They are NOT the same thing.
Consider IEEE 802.15.4 Protocol superframe structure
(Image Src: Google)
IEEE 802.15.4 Superframe Structure
In this structure Contention Access Period(CAP) is always followed by Contention Free Period(CFP).
So is there any particular reason behind keeping CAP first and then CFP? Could it be other way around?
Thank you.
It can't really be the other way around because that is what is in the standard. Obviously, you are free to implement your own use of the radio but then I guess it wouldn't be 802.15.4!
The designers of the standard probably had good reason to place the CAP before the CFP (and if you are really interested I imagine it will be documented somewhere in the IEEE meeting minutes etc). My guess is that I think it would have these following benefits:
devices have to wake up their receiver to listen for the beacon frame, and thus if they have any ad-hoc comms to perform (like collecting a pending message or negotiating a connection etc) they can do it straight away and then go to sleep for the rest of the superframe
having the CAP first allows any devices that do not have a GTS to power down their radio for as long as possible
having the CAP first provides time for devices to negotiate a GTS before the CFP starts, thus reducing the latency to their first GTS (i.e. it would be possible to hear a beacon, associate, and obtain a GTS prior to the very next CFP)
I'm making an application for a running competition on a fixed track. I'm investigating what systems could be used and tough of using a stick containing a GPS/DGPS module and a Zigbee enabled chip to communicate the location to a server.
I've researched the subject (on the internet) but I was wondering if anyone has some practical advice/experience with using a Zigbee mesh/star topology in a dynamic environment wich could apply to this use case. I'm also very interested in using a mesh topology where the data is transmitted (hopping) trough the different Zigbee modules to the server.
Runners are holding a stick; run around the track and than pass the stick on to the next team member.
I am not particularly clear about your goal. But I'd like to say a few things.
First, using GPS/DGPS to measure which team reaches the finish line is inaccurate. Raw GPS data is horrible in accuracy (varying in 1 - 10 meters, well, around that), also the sampling rate of a GPS module is low (say once a second?) How do you determine exactly which team reaches the finish line first?
Second, to use a mobile ZigBee chip to communicate in real-time is hard. I assume your stick has a ZigBee end device. When it is moving (which in your case is pretty fast), it must dynamically find and associate with new parent routers, which takes time and depending on the wireless environment, it might involve several retries. So you will imagine a packet is only successfully delivered to the other end after 100ms or even a second. This might not be a problem if your stick records the exact time when a team reaches the finish line. Since you have a GPS module in the stick so there is no problem in getting very accurate time.
I wish I could play music or video on one computer, and have a second computer playing the same media, synchronized. As in, I can hear both computers' speakers at the same time, and it doesn't sound funny.
I want to do this over Wi-Fi, which is slightly unreliable.
Algorithmically, what's the best approach to this problem?
EDIT 1
Whether both computers "play" the same media, or one "plays" the media and streams it to the other, doesn't matter to me.
I am certain this is a tractable problem because I once saw a demo of Wi-Fi speakers. That was 5+ years ago, so I'm figure the technology should make it easier today.
(I myself was looking for an application which did this, hoping I wouldn't have to write one myself, when I stumbled upon this question.)
overview
You introduce a bit of buffer latency and use a network time-synchronization protocol to align the streams. That is, you split the stream up into packets, and timestamp each packet with "play later at time T", where T is for example 50-100ms in the future (or more if the network is glitchy). You send (or multicast) the packets on the local network, to all computers in the chorus. The computers will all play the sound at the same time because the application clock is synced.
Note that there may be other factors like OS/driver/soundcard latency which may have to be factored into the time-synchronization protocol. If you are not too discerning, the synchronization protocol may be as simple as one computer beeping every second -- plus you hitting a key on the other computer in beat. This has the advantage of accounting for any other source of lag at the OS/driver/soundcard layers, but has the disadvantage that manual intervention is needed if the clocks become desynchronized.
hybrid manual-network sync
One way to account for other sources of latency, without constant manual intervention, is to combine this approach with a standard network-clock synchronization protocol; the first time you run the protocol on new machines:
synchronize the machines with manual beat-style intervention
synchronize the machines with a network-clock sync protocol
for each machine in the chorus, take the difference of the two synchronizations; this is the OS/driver/soundcard latency of each machine, which they each keep track of
Now whenever the network backbone changes, all one needs to do is resync using the network-clock sync protocol (#2), and subtract out the OS/driver/soundcard latencies, obviating the need for manual intervention (unless you change the OS/drivers/soundcards).
nature-mimicking firefly sync
If you are doing this in a quiet room and all machines have microphones, you do not even need manual intervention (#1), because you can have them all follow a "firefly-style" synchronizing algorithm. Many species of fireflies in nature will all blink in unison. http://tinkerlog.com/2007/05/11/synchronizing-fireflies/ describes the algorithm these fireflies use: "If a firefly receives a flash of a neighbour firefly, it flashes slightly earlier." Flashes correspond to beeps or buzzes (through the soundcard, not the mobo piezo buzzer!), and seeing corresponds to listening through the microphone.
This may be a bit awkward over very large room distances due to the speed of sound, but I doubt it'll be an issue (if so, decrease rate of beeping).
The synchronization is relative to the position of the listener relative to each speaker. I don't think the reliability of the network would have as much to do with this synchronization as it would the content of the audio stream. In order to synchronize you need to find the distance between each speaker and the listener. Find the difference between each of those values and the value for the farthest speaker. For each 1.1 feet of difference, delay each of the close speakers by 1ms. This will ensure that the audio stream reaches the listener at the same time. This all assumes an open area, as any in proximity to your scenario will generate reflections of the audio waves and create destructive interference. Objects within the area may also transmit sound at a slower speed resulting in delayed sound of their own.
I'm assigned to a project where my code is supposed to perform uploads and downloads of some files on the same FTP or HTTP server simultaneously. The speed is measured and some conclusions are being made out of this.
Now, the problem is that on high-speed connections we're getting pretty much expected results in terms of throughput, but on slow connections (think ideal CDMA 1xRTT link) either download or upload wins at the expense of the opposite direction. I have a "higher body" who's convinced that CDMA 1xRTT connection is symmetric and thus we should be able to perform data transfer with equivalent speeds (~100 kbps in each direction) on this link.
My measurements show that without heavy tweaking the code in terms of buffer sizes and data link throttling it's not possible to have same speeds in forementioned conditions. I tried both my multithreaded code and also created a simple batch file that automates Windows' ftp.exe to perform data transfer -- same result.
So, the question is: is it really possible to perform data transfer on a slow symmetrical link with equivalent speeds? Is a "higher body" right in their expectations? If yes, do you have any suggestions on what should I do with my code in order to achieve such throughput?
PS.
I completely re-wrote the question, so it would be obvious it belongs to this site.
CDMA 1x consists of up to 15 channels of 9.6kbps traffic. This results in a total throughput of 144kbps.
Two channels are used for command and control signals (talking to base stations, associating/disassociating, SMS traffic, ring signals, etc).
That leaves you with up to 124.8kbps.
--> Each channel is one way. <--
They are dynamically switched and allocated depending on the need.
Generally you'll get more download than upload because that's the typical cell phone modem usage. But you'll never get more than 120kbps total aggregate bandwidth.
In practise, due to overhead of 1xRTT encoding, error correction, resends, etc, you'll typically experience between 60kbps and 90kbps even if you have all the channels possible.
This means that you can probably only get 30kbps-60kbps of upload and download simultaneously.
Further, due to switching the channels dynamically (and the fact that the base station controls this more than your modem - they need to manage base station channels carefully to keep channels free for voice calls) you'll lose time when it switches channels - it's not an instantaneous process.
So - 1xRTT can, in theory, give you 124kbps one way, but due to overhead, switching times, base station capacity, or the phone company simply limiting such connections for other reasons, you can't depend on a symmetrical link.
NOTE:
This will vary to some degree based on the provider and the modem. For instance, some modems have 16 channels, and some providers support 16 channels. In some cases those modems and providers work well together and can provide a full 144kbps aggregate raw bandwidth to the application, with only one dedicated channel (which has to work pretty hard) to deal with control, switching, and other issues. Even then, though, with the overhead of the modem communications, then the overhead of PPP, then the overhead of IP, then the overhead of TCP, you're still looking at maybe 100-120kbps total bandwidth, both up and down.
Lastly, no provider yet supports transparent transfer of IP traffic. In other words if you're modem is moving, the modem will switch to a new base station, but you'll completely drop the PPP session and have to restart it, as well as all the TCP sessions and such. You typically won't get the same IP address, and so your TCP sessions will not recover gracefully.
The "fun" aspect to this twist is that this can happen even if you aren't moving. If one base station gets loaded down, you may be transferred to another base station if you are close enough - there are other things that may make your modem transfer even without you moving. So make sure you take this into account, since you seem to be keen on maintaining a full duplex, symmetric channel open. It's tough to write stuff that will recover gracefully, nevermind anticipate it and do it quickly. You would do well to work very closely with a modem manufacturer (such as Kyocera) on this - otherwise you won't get the documentation on how to control the modem chipset at the low level that you need.
-Adam
I think the whole drama with high equal speeds on both directions is because my higher body thinks that they have 144 kbps on uplink AND 144 kbps on DOWNLINK (== TWO pipes). Whereas in reality we have 144 kbps of ONE pipe which is switching directions when I transfer files.
Comment me if I right or wrong, please.