Been trying to find the issue here (Unity devs make it seem it's a design choice somehow), but I've been trying to simulate lag to see how my game works over the internet (I've also tried connecting a client from a tethered hotspot and it disconnected as well) and found that any amount of packet loss (even just 1%) will eventually lead to a disconnection within a couple minutes.
My game receives data just fine, as you can see the remote players moving around just fine until the disconnection. I think it has something to do with UNET's heartbeat packets not being resent for some reason, and has soon as it drops the first one you get disconnected.
If this is a design choice, I can't see how Unity would think you could have a rock solid connection when you obviously have some who could be playing off a cellular connection. Anyone know anything about this? I've tried asking on the Unity forums as well, and there's been no replies for over a week now.
Thanks
Related
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.
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.
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
I'm new to driver programming in general and also to USB. However, I managed to write a driver for Windows CE (6.0) and I also had access to an USB-Sniffer to read all traffic between the host and the device.
The problem now occurs on some boards (2 out of the 3 I have):
When the device has no data to send and I issue an Interrupt-In-Transfer the device sends an ACK.
So far this is expected. However, something (I guess either the USB-Controller or WinCE) seems to automatically issue more IN-Transfers (3 on one board, 4 on another) and I get subsequent ACK. This isn't a problem so far either.
But the next IN-Transfer will also result in an ACK, no matter if there is data to send or not, I receive zero bytes in the driver.
Yet, when I look at the USB-Sniffer the proper telegram was send, however 2 more IN-Transfers are automatically issued and are responded with an ACK. So it seems like the data is overwritten by the ACK.
I tried everything that came to my mind so far: Reset the pipe, close and reopen the connection, but nothing seems to work out properly. Resetting the Pipe solves the problem in about half of the cases though. I really ran out of ideas for solving the problem.
Is there a way to tell the USB-Controller (or WinCE or whatever causes this behaviour) to always only issue one single transfer?
EDIT
Turns out it was a threading issue. Unfortunately I wasn't the one who fixed it and I have no access to the working solution, thus I cannot give further details.
one of the our distributed apps are using heart beat to detect the peer's disconnection(e.g. LAN line broken, etc) .
is the heart beating necessary?
Maybe, what will do you if you don't get the heart beat?
If you have no way to recover there is
no point in having a heart beat.
If you are using call-back from the server to the client, you need a way that the client can ask the server to resent all lost call-backs, this is not easy.
Also if you don’t get a heart beat it does not mean a message will not get there later, as there can be all sort of network delays, is it safe to just resent your messages?
The heart beat is the easy bit, the
hard bit is what to do when the heart
does not beat!
Yes. TCP would only show that the physical connection is still alive (ie. the socket was not teared down by routers or by OS). But will tell nothing about the application availability. If the process at the other end of your pipe is in a while(1); loop and is not processing your requests, you aren't really connected to it.
That is quite a good way to know that you are still connected to the other end at the "application level" and applications can still talk. Otherwise you would have to make assumptions that "the other end" has nothing to "say", which would be hard to separate from "the other end actually lost network connectivity 35 seconds ago".