When getting disconnected with 1001 ("websocket closed unexpectedly" error) we were creating a new SIP session, therefore a new FreeSWITCH channel.
While reconnecting the socket, instead of closing the SIP session, we keep it alive during reconnection (audio should keep working in the meantime). When reconnected we keep using this same session (avoiding the creation of an extra one).
We also better handle WebSocket error codes from SIP.js.
FF immediately closes websocket when unloading page, so we now to stop user agent when 'beforeunload' event is triggered, to avoid leaving open sessions in FreeSWITCH when user leaves page.
Cursor coordinates are calculated using the presentation SVG object
DOMMatrix. When getting this matrix, some browsers (Firefox at least)
responds it as null if the svg object does not have a visible area.
This adds a check before trying to transform the cursor coordinates
using the matrix inverse so we avoid calling a method from a null object.
If there isn't a DOMMatrix to be used as reference, returns a simple out of
bounds SVGPoint (-1, -1)
Users must return their audio to the main room before joining a different one.
Since the audio transfer and the UI state manager doesn't provide a shortcut for
jumping from a breakout room to another, avoid making this opiton available.
When refusing ("thumbs down" button) echo test, user is able to select a different input device. This should work fine for chrome, firefox and safari (once user grants permission when asked by html5client).
For output devices, we depend on setSinkId function, which is enabled by default on current chrome release (2020) but not in Firefox (user needs to enable "setSinkId in about:config page). This implementation is listed as (?) in MDN.
In other words, output device selection should work out of the box for chrome, only.
When selecting an outputDevice, all alert sounds (hangup, screenshare , polling, etc) also goes to the same output device.
This solves #10592
This considerably changes the way we process audio signaling and start audio elements in user's browser.
We now avoid using AudioContext element for both microphone and listenonly calls, once it is unstable for some iOS devices (cracky audio, user stops hearing audio after a while).
Increased default value for listenOnlyCallTimeout: this avoids activating FreeSWITCH's fallback when ICE negotiation takes longer than 15sec (tested on DO).
Increased listenonly logs.
This fixes#8133#10388
The mic mute is done client side via the track`s enabled flag, which generates silent when false. This still tracks the server/freeswitch mute state, so server-side mutes will be reproduced in the client
