The Google Cast extension would not work on my Chromebook Pixel which I had pinned to the beta channel of ChromeOS. Once I did a recovery and reverted to the stable channel on my Chromebook, the Google Cast extension works.
I 100% get that beta channel will (almost certainly) not be officially supported, but I'm hoping as part of being forward-looking the extension will eventually/soon get to "seems to work" status on the beta channel -- because today's beta channel leads to tomorrow's stable channel.
Related
In the last couple of weeks I've been developing a boot loader that performs a firmware update on a certain device. The setup is as follows:
The firmware binary and its respective SHA1 hash are stored in a web server;
The device is composed of an ESP8266 and a STM32 microcontroller (STM32F401 or STM32F030, there two hardware versions, but the one I'm using is the F401). The ESP is used only with AT+ commands, i.e., I did not built it's firmware, just used the latest version from Espressif.
The idea is that, the STM32 bootloader should use the ESP to download the firmware hash and binary from the webserver and then boot the firmware if the hash is OK. The download is made using the ESP in passive mode, i.e. the STM has to manually request X bytes to read from the ESP buffer, currently I'm using 1 MTU (1460 bytes).
At first, the connection to the webserver was made using HTTP and everything worked perfectly, however, I had to change it to HTTPS, and that's where the problem starts. After the STM has received around 100kB of the firmware (which has 110kB), the ESP only provides 30 bytes per request (which should be around 1 MTU), thus, making the download time extremely large.
I've already did some digging trying to find out if this is related to the ESP, but didn't find anything. Also, the point where this 30 byte download rate starts to happen isn't always at the 100kB mark, I've tested with a 170kB firmware and It started to happen at 160kB ish, so, it looks like it's always the last 10kB.
I've also added some delays in the firmware when the packet size becomes smaller than 1 MTU, to give more time for the ESP to process the packet, since the SSL decryption stuff takes longer to process; but it did not help.
My question is: is there some characteristic in the HTTPS/SSL protocols that reduces the packet length? What could be the causes of what is happening here?
Lately numerous network requests with Alamofire made from our iOS device fail with the following error:
Error Domain=NSPOSIXErrorDomain Code=28 "No space left on device"
UserInfo={_NSURLErrorFailingURLSessionTaskErrorKey=LocalDataTask
.<3>,
_kCFStreamErrorDomainKey=1, _NSURLErrorRelatedURLSessionTaskErrorKey=( "LocalDataTask .<3>" ),
_kCFStreamErrorCodeKey=28}
Our app has a mechanism to send a network request if the user has moved +- 10 meters. This is checked every 5 seconds, so in theory every five seconds a call can be made. The network request fails occasionally with this message, returning no status code and the above error.
The message implies the error has to do with available disk/memory space on the device. However, after checking both there is no link to be found since there is plenty of space available. Also, the error occurs on multiple devices, all running iOS 14.4 or higher.
Is there information available regarding error code 28 and what could be the culprit on iOS devices? Even better; how can this error be prevented?
To answer the occurrence of the error itself:
NSPOSIXErrorDomain Code=28 "No space left on device"
With logs in the Xcode terminal:
2021-05-07 15:56:50.873428+0200 MYAPP[21757:7406020] [] nw_path_evaluator_create_flow_inner NECP_CLIENT_ACTION_ADD_FLOW 05CD829A-810D-412F-B86E-7524369359E8 [28: No space left on device]
2021-05-07 15:56:50.877243+0200 MYAPP[21757:7400322] Task <5504BCDF-7DFE-4045-BD4B-E75054636D5B>.<1> finished with error [28] Error Domain=NSPOSIXErrorDomain Code=28 "No space left on device" UserInfo={_NSURLErrorFailingURLSessionTaskErrorKey=LocalUploadTask <5504BCDF-7DFE-4045-BD4B-E75054636D5B>.<1>, _kCFStreamErrorDomainKey=1, _NSURLErrorRelatedURLSessionTaskErrorKey=(
"LocalUploadTask <5504BCDF-7DFE-4045-BD4B-E75054636D5B>.<1>"
), _kCFStreamErrorCodeKey=28}
It appears to get called when there are too many NSURLSessions created, reaching a limit of (in our tests) 600-700 sessions, which are not maintained or closed properly. The error started to get thrown since iOS 14, so it is interesting to see if there was a limit introduced.
Linked is a github issue raised stating the same issues on the ktor microservices framework by JetBrains, pointing in the same direction, mentioning the invalidation of sessions to prevent this issue:
https://github.com/ktorio/ktor/issues/1341
In our own project the origin of the problem turned out to be our implementation of the StarScream websocket library. This might not be relevant for the issues others are having, but explained anyways to create a complete picture of the problem. It is the cause and fix of our specific situation.
At first we assumed it had something to do with the URLSession created by Alamofire (networking library used) since POST requests started to get cancelled, and a kill of the app seemed the only solution to do requests again.
However, we also make use of websocket connections using the StarScream library, which attempts to connect to an socket, and if failed retry to connect every two seconds for a max time of two hours. This would mean for two hours, every two seconds, we connect to the socket -> receive a failure to connect -> disconnect the socket -> connect again. Using a singleton of the socket it was thought there was no possibility of creating multiple URLSessions, since the socket was only initiated once. However calling the connect to the socket again would create a new nw_connection object every single time, since the library did not handle the disconnect properly.
image of NWConcrete_nw_connection objects generated in socket connection
The way this was validated was using the instruments app to check for the creation of new nw_connection objects. Logged as a "memory leak" there, the creation of the nw_connection objects was getting logged and the solution was to make sure we disconnect the socket (invalidate the session) properly before connecting again.
I hope to answer a big part of the issue here, and I will mark my own question answered since this was the solution to the problem at hand. I think Apple should consider giving accurate reports on the number of objects created being limited, instead of giving an error "No space left on device".
Just wanted to chime in with more info, since we're experiencing the same issue.
Based on our analytics, this issue only started happening since iOS 14. We've verified it happening on 14.2, 14.4 and 14.5. Naturally the most straightforward cause for this error would be low memory or disk storage. We've excluded this option with additional logging, as you seem to have done as well.
A possibly related SO post has attributed the issue to a network inspecting framework that was enabled in their release build. It's worth checking if you use a similar tool.
Another report of this issue, this time on the Github of AFNetworking (predecessor to the Alamofire library you use), says they were able to fix it by limiting the creation of URLSession objects.
For us personally, neither of these did the trick. We created a support ticket with Apple, but this hasn't lead to a solution. They requested a small sample project that reproduces the issue, but the error only manifested after 7 days of continuous use in our app. If you have a faster way to reproduce this, it may be worth it to submit your own support ticket.
Hopefully this helps you find a solution, if you do please add this to your post to help others!
201 [Deprecation] RTP data channels are no longer supported. The "RtpDataChannels" constraint is currently ignored, and may cause an error at a later date.
But what is the migration path? I did a search and can only find info telling us to migrate to SCTP, without any details.
Remove WebRTC RTP data channels (removed)
I am thinking if we shall use the WebSocket channel (the one we used to establish the WebRTC Connection) to replace RtpDataChnanels.
If you're doing a peer-to-peer application: Use the standard WebRTC data channels; if in Javascript, just remove the constraint saying "I want to use RTP data channels", and most or all of your applications will work just like before, only better.
Why would anyone want to use that old relic?
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.
When I am testing RTMFP serverless connection, I noticed that there is often a very long delay between the NetGroup.Connect.Success and NetGroup.Neighbor.Connected events (where I can start an actual P2P communication).
Sometimes it connects immediately and but sometimes it takes several minutes on the same network. I was not able to find any logic in this.
After I have finally received the Neighbor.Connect, the posting is quick and stable.
I have tested it on ADL <-> Android and ADL <-> ADL (Windows 10), but on all browser the rtmfp flash examples behave the same.
So my question is - Is this a common issue and is there a way to minimize this delay in the connection (without any messing with hardware)?
Thanks!