Does Reach-ability class keep sending / receiving data in iOS dev? - objective-c

I have been working on Reachability class for a while and have tried both the one from Apple sample and the one from ddg. I wonder whether the Reachability class keep sending / receiving data after starting the notifier.
As I'm developing an app which connect to different hosts quite often, I decided to write a singleton and attach the reachability classes I need on it. The reacability classes would be initiated and start their notifiers once the app start. I use the singleton approach as I want this singleton class to be portable and can be applied to other apps without much rewriting. I am not sure if it is good idea to implement like this but it worked quite well.
However, someone reported that the battery of his device drain significantly faster after using the app and someone reported more data usage. My app does not send / receive data on background so I start wondering if it is related to the reachability.
I tried profiling the energy usage with Instrument and I notice that there are continuous small data (few hundred bytes in average) coming in via the network interfaces even I put my app in idle. However, there are almost no data sending out.
I know that Reachability requires data usage when initiate (resolving DNS etc) but I am not sure that whether it still keep using data after starting notifier. Does anyone can tell?
I am not familiar with the low-level programming, it would be nice if someone could explain how does the Reachability work.

I use Reachability, and while I haven't monitored the connections, I have browsed the code, and I can't see any reason why it would keep sending ( or receiving).
If you have a ethernet connection to your Mac, it is quite easy to check. Enable sharing over wifi of your ethernet connection. Install little snitch, it will run in demo mode for three hours after every boot. Turn off the data connection on the test device and connect it to your mac over wifi.
This will allow you to see any network access your test device is making.
If this isn't possible, you can also run your app in the simulator as the network side should be the same, so you should be able to check.
There are also a ton of other tools to track network activity, but I think little snitch is the easiest to use.

Related

Can I poll my USB HID device without first sending a command

I was able to make a working HID USB stack on my "StartUSB for PIC" board for the 18F2550 microcontroller. I based it on one of the MLA libraries, which was made for the 18F45K50 (MLA 2018_11_26, hid_custom, picdem_fs_usb_k50.x), but I converted it to work with the 18F2550 (there might have been easier ways, but only learned to work with PIC about 1 month ago). On the host side, I'm using LibUsbDotNet (also here, there might be easier ways - the documentation on this library really sucks) on a Windows 10 machine.
I'm using the HID class, full speed, and all seems to work. Although, I get some random errors on the host PC (see below), but doing one close/re-open cycle on the host side when getting the error is kind of solving it. Dirty, but it works. So I kind of ignore this now.
Win32Error:Win32Error:GetOverlappedResult Ep 0x01
995:The I/O operation has been aborted because of either a thread exit or an application request.
I'm not an expert on USB (yet). But all examples I'm seeing are based on 1) you send first a command to the device and 2) then you retrieve the answer from the device. I did some performance tests, and see that this indeed shows that I can do about 500 cycles/second. I think that is correct, because each cycle, sending command and retrieving answer, each takes 1 msec.
But do I really need to send a command? Can't I just keep reading endlessly, and when the device has somthing to say, it does send the data in an IN transaction, and when not it ignores which creates a timeout on the host side. That would mean that I can poll at 1000 cycles/second? Unfortunately, I have tried it by changing my implementation on the PIC, but I get very weird results. I think I have issues with suspend mode. That brings me to another question - how can I make the device get out of suspend mode (means that not the host, but the device should be triggering this event). I have searched the MLA library for command such as "wakeup", "resume", ... but couldn't find anything.
So, to summarize, 2 questions:
Conceptual: Can I send data from device to host without being requested for it by a command from the host?
For PIC experts: How can I have a device trigger for a wakeup from suspend mode?
And indeed, the answer is Yes on the first question.
In the meantime, I found another link on the web that contains a Visual Studio C# implementation of a USB library including all the source files.
If you're interested, this is the link
This C# host implementation works as a charm. Without sending a command to the device, I get notified immediately if a button is pressed. Great!
It also proofs that my earlier device implementation based on the original MicroChip MLA, is 100% correct. I stress tested the implementation by sending a "toggle LED command" as fast as I could, and I reach 1000 commands/second. Again great!
I think that LibUsbDotNet isn't that perfect after all. As I wrote above, I get rather unstable communication (Win32Error). But with this implementation, I don't get a single error, even after running for half an hour # 1000 commands/second.
So for me, case closed.

Continuous device and connection issues with routed Tokbox session

We’ve been using the Tokbox platform for several months now with a Javascript web-client as well as an Android phone client, where sessions and connections are managed by a Python server. While integration and bring-up went well on both ends (client and server), we continue to encounter problems with the in-session audio and video experience.
Sessions are always routed and always between two participants only, with much use of a collaborative editor.
The in-session experience is like a coin toss: we never know how it’s going to go, and that’s becoming a business threat.
Web-Client: A/V Resources
The most common problem is the acquisition of audio and/or video: at the beginning of a session, one or the other participants may have problems hearing or seeing the other. Allocating a new connection to establish new streams does not fix that, nor does restarting the browser.
Question: What’s the recommended way to detect possible resource locks (e.g. does another application hog the camera/microphone)?
Web-Client: Network
Bandwidth and packet loss are a challenge, for example this inspector graph:
Audio and video of both participants is all over the place, and while we can not control the network connections the web-client should be able to reliably give useful information.
Question: Other than continuous connection monitoring with getStats() and maybe the experimental navigator.connection property, how can the web-client monitor network connectivity?
Pre-Call Test
We recommend to customers to run a pre-call test and have implemented it on our site as well. However, results of that test often times do not reflect the in-session connectivity. Worse, a pre-call test may detect a low (no video) bandwidth while Skype works just fine.
Question: How can that be?
I'm a member of the TokBox development team. I remember you reported an issue with the Python SDK, thanks for that!
Web-Client: A/V Resources
Most acquisition issues are detected by the JS SDK and if they aren't then we'd really like to hear about it! Please report reproduction steps or affected session IDs to TokBox support (referencing this StackOverflow question): https://support.tokbox.com/hc/en-us/requests/new
Most acquisition errors appear as OT_HARDWARE_UNAVAILABLE or OT_MEDIA_ERR_ABORTED errors. Are you detecting and surfacing these errors to your users? There is also the special OT_CHROME_MICROPHONE_ACQUISITION_ERROR error which is due to a known issue with Chrome that has been mostly fixed since Chrome 63 (see https://bugs.chromium.org/p/webrtc/issues/detail?id=4799).
Web-Client: Network
Unfortunately this is one of the more difficult issues to troubleshoot. Yes, Subscriber#getStats() is the best tool we have at our disposal and is a wrapper around the native RTCPeerConnection#getStats() function. Unfortunately we don't have much control over the values returned by the native function and if you think our SDK is returning incorrect values when compared with values from RTCPeerConnection#getStats() then please let us know!
It would be worthwhile confirming whether the issue is reproducible in all browsers or only a particular one. If you have detailed data regarding the inaccuracy of the native RTCPeerConnection#getStats() function then we could work together to report it to the browser vendor(s).
Fortunately we have just released the new Publisher#getStats() function which lets you get the publisher side of the stats. This should help you narrow down the cause of a connectivity issue to either a publisher or subscriber side. Please let us know if this helps with tracking down these issues.
Pre-Call Test
Again, these tests are based on Subscriber#getStats() which in turn are based on RTCPeerConnection#getStats(), the accuracy of which is out of our hands, but we'd love any reproduction steps to either fix a bug in our client SDK or report a bug to the browser vendors.
Just to confirm though, when you say you've implemented a pre-call test in your site, did you use the official JavaScript network test module? https://github.com/opentok/opentok-network-test-js This is actually what's used by the TokBox pre-call test.
#Aiham, thanks for responding, I've been looking at the the new Publisher#getStats() you linked to (thank you!), so we too can give our users some way of visibly seeing the network conditions that might be affected the quality of their call (and who's causing it). However, it seems as though bytes / packets sent goes up sharply as the number of subscribers increases, even though we're in a routed session.
Am I wrong to expect the Publisher#getStats() statistics to stay fairly stable regardless of the number of subscribers then receiving that stream in a routed session? I expected the nature of a routed call to mean it's sent once to the OpenTok Media Servers, and the statistics would end there.

Why sending and receiving JSON data from the server in IPhone is so slow?

I'm making a mobile client for a web site now. And information exchange between my app and server is in JSON (searching users and data on server,sending messages, conversation threading, etc.) But all these features work too slow. I click on the button "send" and then wait for some second before the message will be sent, the same thing with searching, authorization, etc. So I have such a questions:
1. Why it's such a performance overhead?
2. Can it be troubles with the server side or it's JSON parser troubles or may be something else?
3. How can i fix/optimize this? All solutions, advices etc. will be helpful!
I would use Xcode to debug the app to see whether the majority of time is spent loading the data from the server or parsing the JSON once the data is received.
If it is the first, try loading the data from a PC over the same wireless connection and see if it is slow on that too. If so, clearly your server side code needs optimising.
If it is the second and the parsing is slow, you may want to look into using JSONKit instead of the native JSON parser as testing shows it is faster. You may also want to review the structure of your JSON.
One thing I have noticed however is that connections are slower on my iPad than on other machines. I've noticed this when comparing apps I've developed in the simulator to on the device on the same network and when conducting speedtests. As for why this happens, I am not sure - some form of additional overhead in iOS perhaps.
I can save you some time - it has nothing to do with JSON. It has to do with how the your app handles requests in general. It obviously needs optimization on the server.
EDIT:
I suppose it could also be that you might be experiencing high-latency on your phone, but again, that has nothing to do with your app.
Debug it using a regular browser and chrome dev tools (in the network tab) - you'll see that the requests take long even on a desktop at which point you'll have to start fishing around in the server-side code to see what's making it go slow (hint: unoptimized database queries are a big bottleneck....but then again, so is crappy hardware).
Sorry that I couldn't be of more help, but without seeing the entire setup of the server and the code that's going slow (not the client requests, but the server code), that's the best I can do.
Best of luck.

Is the GameKit's communication reliable with GKMatchSendDataReliable?

I'm working with GameKit.framework and I'm trying to create a reliable communication between two iPhones.
I'm sending packages with the GKMatchSendDataReliable mode.
The documentation says:
GKMatchSendDataReliable
The data is sent continuously until it is successfully received by the intended recipients or the connection times out.
Reliable transmissions are delivered in the order they were sent. Use this when you need to guarantee delivery.
Available in iOS 4.1 and later. Declared in GKMatch.h.
I have experienced some problems on a bad WiFi connection. The GameKit does not declare the connection lost, but some packages never arrive.
Can I count on a 100% reliable communication when using GKMatchSendDataReliable or is Apple just using fancy names for something they didn't implement?
My users also complain that some data may be accidentally lost during the game. I wrote a test app and figured out that GKMatchSendDataReliable is not really reliable. On weak internet connection (e.g. EDGE) some packets are regularly lost without any error from the Game Center API.
So the only option is to add an extra transport layer for truly reliable delivery.
I wrote a simple lib for this purpose: RoUTP. It saves all sent messages until acknowledgement for each received, resends lost and buffers received messages in case of broken sequence.
In my tests combination "RoUTP + GKMatchSendDataUnreliable" works even beter than "RoUTP + GKMatchSendDataReliable" (and of course better than pure GKMatchSendDataReliable which is not really reliable).
It nearly 100% reliable but maybe not what you need sometimes… For example you dropped out of network all the stuff that you send via GKMatchSendDataReliable will be sent in the order you've send them.
This is brilliant for turn-based games for example, but if fast reaction is necessary a dropout of the network would not just forget the missed packages he would get all the now late packages till he gets to realtime again.
The case GKMatchSendDataReliable doesn't send the data is a connection time out.
I think this would be also the case when you close the app

WiFi communication to embedded display

I'm trying to create an embedded outdoor display of bus arrival times at my university. I'd like the device to utilize my school's secured WiFi network to show arrival time updates determined from a server script I have running.
I was hoping to get some advice on the high-level operation of this thing -- would it be better for the display board to poll a hosted database via the WiFi network or should I have a script try to communicate with the board directly over 802.11? (Push or Pull?)
I was planning to use a Wifly or WIZnet ethernet board in combination with a wireless access hub. Mostly inspired by this project: http://www.circuitcellar.com/Wiznet/winners/001166.html Would anyone recommend something else over one of the WIZnet boards? I saw SPI/UART options and thought these boards could work with an AVR platform.
And out of curiosity -- if you were to 'cold start' this device (ie, request a bus arrival time by pushing the display's on button) you might expect it to take 10-20 seconds to get assigned an IP and successfully connect to the database, does that sound right?
I'd go pull. In fact, I'd have outdoor display make http or https requests of the server. That way the server could tell it how long to show a given set of data before polling for a new one using standard http page expiration.
I think pull would make it easier to have multiple displays, and to test your server as well. I've also got a gut feeling that this would make your display more secure. Someone would have to hack your server to hijack your display.
There's a very cool looking Arduino-targetted product called the WiShield. Seems super easy to use and he provides some source code. It uses SPI for host communication. If you're not interested in going the Arduino route, I'm sure the source code wouldn't be too hard to port to something like avr-gcc. Check it out, might save you some time and headaches for $55. Worth checking out anyway.