Rtc | Arduino Library for RTC Ds1302 Ds1307 Ds3231

ImageData isn't part of the test environment used by Jest. You can create a mock definition for ImageData yourself. You can use the setupTestFrameworkScriptFile setting to point to a file that executes before all of your tests run as described in this answer.

For example, if you need window.ImageData to be available on the window add this to your package.json:

Can you try with same id in pending intent and notify.? Notification id in createNotification() method is random id.

fixed by cloning the rtcclient repo, and adding my own custom function getUserEmail

Check first if this was working with 0.6.0 (depending on your RTC version 6.x?; 7.x).

dixudx/rtcclient issue 136 mentions:

v0.7.0 has introduced a change on authentication, refer to #133 for details (and issue 112), which may be not compatible with older Rational Team Concert. From your issue, it seems to be true. Sorry for this.

My suggestion is just sticking with older rtcclient version, such as pip install rtcclient==0.6.0.

The enableLogUpload method is used to send logs to the Agora Server to debug potential problems, if you want to maintain the application logs for yourself you can use a different service like datadog

(ice)connectionstatechange is the right event in general.

If you want more granularity you'll need to poll getStats and looks for stats like framesReceived. But there is no guaranteed frame rate sent from the other side (e.g. in screensharing you go below 1/s).

While the actual ICE statistics like requestsSent seem more useful they happen much less frequently, only once per second and you can loose a packet or it comes late.

In general this is a question of how reliable the detection of the network failure is. If it is too aggressive you end up with a poor UX showing a warning too often. You might not end up that is significantly better than at the cost of introducing complexity that you need to maintain.

looking to have each bucket shown as a % of the total

Combining COUNT(...) with windowed SUM() OVER() to get the total across all groups:

The problem occurs when the repository contains symbolic links and RTC does not have sufficient permission to create the required links on the local Windows file system. The simple workaround for this is to run Visual Studio as administrator.

I've forked and modified your project on GitHub. Working code and setup instructions can be found here.

Here is how it looks with 4 tabs open. You can see local session id on top right and corresponding sessions ids on each remote videos. I used VCam for debugging. That's why the trial text:

There are some issues to fix:

1. Major issue is the comparison if (y != "complete"). You are comparing event object with "complete" string. You end up sending offer SDP without any ICE candidate. Using if (self.peerConn.iceGatheringState === "complete")ref fixes part of the problem. With this change Receiver sees the Sender's video.
2. From Receiver side you create answer and send back SDP immediately. ICE candidate collections starts after you call createAnswer() or createOffer(). So you are sending SDP without any ICE candidates.
   After you fix this other issue is that on Sender side there are no onAddStream and streamEventsChangedHandler handlers. Thus Sender never gets to set received stream to corresponding element.
3. RTCPeerConnection is capable of both sending and receiving video/audio streams with same connection object. You can set setRemoteDescription() on Sender as well. You don't have to maintain separate Sender and Receiver connection objects. Major improvement here could be you combine Sender and Receiver classes in to one Connection class. This way you create only n-1 connections, for n participants, at each participant side. And on server you need to track only n connections and not n*(n-1).
4. RTCPeerConnection.onaddstream is deprecated. Use RTCPeerConnection.ontrack property
5. Serve web pages over secure channel(HTTPS). Because browsers won't let your web page access media hardware over insecure(HTTP).ref. Localhost(127.0.0.1) is exception.
   You can configure tomcat7-maven-plugin in pom.xml to serve pages on HTTPs. Steps to setup HTTPs are in the README.md file.
6. You need to have more timeout delay. 2 seconds is very less. Clients may have very low latency streaming among them, due this direct P2P connection, and high latency connecting to your server. So if they can't connect to server doesn't mean they stopped communicating. Once you establish connection between clients the server's role is over until they logoff explicitly.
7. Due to Chrome autoplay policy you can't start playing other's video directly. So had to add an overly over the page to force user interaction. This won't be an issue once you have login implemented.
8. Acquire local media streams before you create Room object, preferably immediately after page load and before user get past overlay. If user delays providing device access permissions, your Room object will have already completed all connection procedures with other peers without sharing any local audio/video streams. If you handle it later you'll have to add tracks/streams to all connections again.
9. Did little bit of restructuring. On server side I moved public inner classes from MeetSessions.java to separate files. It was getting hard to understand the code. Something with the index.html.
   
   I've tested the code in LAN/Wi-Fi using Windows PC, Android phone, and tablet. If you host your server in public domain you may need TURN server as well for these reasons.