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dana-the-stream-gatekeeper's Issues

Video Issues

  • Fix the join/leave issues (I know you're working on this already)
  • Add option to limit video resolution to various common values... 480p, 720p, 1080p, etc.
    This has to be done via pjsip call options.mediaConstraints.
  • Add video element titlebar with display name to video element. Could also have talking indicator, mute status, etc.
  • Add option to set video element display size. Could be handy but not required.

Transcript: Move Sidebar

The sidebar should be moved to the right side of the UI so it doesn't conflict with the chat sidebar.

Chat: Sidebar Persistence

The chat sidebar should persist until the user clicks the toggle button again and shouldn't dim the rest of the UI

No Devices Found

Cannot seem to find my microphone, check and its working with no obvious windows permission issue.

The console shows...

image

Transcript: Relay Transcript to Other Participants

An option (defaulting to off) could be created to send the transcript live to all participants via sendMessage(). You could create a header so the incoming message is redirected to the transcript sidebar instead of the chat sidebar.

Low Priority.

Chat Issues

  • The chat sidebar should persist until the user clicks the toggle button again and shouldn't dim the rest of the UI.
  • The toggle button should have a tooltip.
  • The "ENTER" key should be bound to the send button.
  • The display should be more compact.
  • The time should be displayed with each message (maybe with an option to turn it on/off)

Support multi stream

Asterisk can now support multiple tracks in a webrtc streams

Support putting the screen share into the existing PeerConnection rather than making a new one with silent audio

Extra audio mute/video mute buttons

When I'm the only person in a conference, waiting for others to join, I see a random audio mute/video mute button in the middle of the screen, but only when I mouse-over a particular region of the screen.

Please see the attached screencast.

No error checking before joining a room

Join should make sure that a wssServerUri, extension and password before it allows you to join a room and if you dont have those setup, force you over to the setup page

Handle the Remote Streams Grid better

Currently it'll give you two rows and place everything inside them. if you have the wrong window size things aren't going to work well. We can do better - just need a bit of time.

If im sending a screenshare, don't show it in remote streams

When I'm sending a screenshare I don't want to show it in remote streams. This happens because I have to make a separate session for the screenshare due to Asterisk's limitations

I know im sending a screenshare, so compare the ID of the mediastream to the remote and make sure I don't show it in remote streams

Make it a PWA

  • Enable Service Worker
  • Fix values in the manifest file
  • Handle no network in the app well

Enhance the video component

To be able to pass in a name etc

Add video element titlebar with display name to video element. Could also have talking indicator, mute status, etc.

Undefined _sip causes type errors

Dana doesn't seem to work anymore, due to the following JavaScript error happening quite a few times when trying to join a conference.

The application seems to think all is well, but the connection to the Asterisk server doesn't happen and errors of this nature are in the JS console:

Uncaught (in promise) TypeError: Cannot read properties of undefined (reading 'on')
    at VideoRoom._callee5$ (VideoRoom.js:470:19)
    at tryCatch (runtime.js:64:1)
    at Generator.invoke [as _invoke] (runtime.js:281:1)
    at Generator.next (runtime.js:117:1)
    at asyncGeneratorStep (asyncToGenerator.js:3:1)
    at _next (asyncToGenerator.js:25:1)
VideoRoom.js:410 Uncaught (in promise) TypeError: Cannot read properties of undefined (reading 'on')
    at VideoRoom._callee4$ (VideoRoom.js:410:23)
    at tryCatch (runtime.js:64:1)
    at Generator.invoke [as _invoke] (runtime.js:281:1)
    at Generator.next (runtime.js:117:1)
    at asyncGeneratorStep (asyncToGenerator.js:3:1)
    at _next (asyncToGenerator.js:25:1)
    at asyncToGenerator.js:32:1
    at new Promise (<anonymous>)
    at VideoRoom.<anonymous> (asyncToGenerator.js:21:1)
    at VideoRoom._call (main.8f8d9f05.chunk.js:360:297)

Occurs in both Chromium 70 and Chromium 101.

Similar equivalent errors in Firefox:

Uncaught (in promise) TypeError: this._sip is undefined
    _callee4$ VideoRoom.js:410
    Babel 8
    React 12
    unstable_runWithPriority scheduler.production.min.js:309
    React 3
Uncaught (in promise) TypeError: this._sip is undefined
    _callee5$ VideoRoom.js:470
    tryCatch runtime.js:64
    invoke runtime.js:281
    defineIteratorMethods runtime.js:117
    Babel 2

Transcript: Sidebar Persistence

The transcript sidebar should persist until the user clicks the toggle button again and shouldn't dim the rest of the UI.

Refactor VideoRoom

It became a little bit of a mess during development, refactor it and split it out a bit

Transcript Issues

  • The transcript sidebar should persist until the user clicks the toggle button again and shouldn't dim the rest of the UI.
  • The toggle button should have a tooltip.
  • The sidebar should be moved to the right side of the UI so it doesn't conflict with the chat sidebar.
  • There should be an option to save the contents of the transcript sidebar to a file.

Bonus:

  • An option (defaulting to off) could be created to send the transcript live to all participants via sendMessage(). You could create a header so the incoming message is redirected to the transcript sidebar instead of the chat sidebar.

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