Maps WebSocket's 1006 error to BBB's 1002, the same way it was done with old sip.js version
Set user agent's number of reconnection attempts to the same value as older sip.js version
Changed the maximum attempts of the UserAgent reconnection (this should be changed when binding audio's websocket to meteor's connection state).
Added a log to monitor WS reconnect attempts.
When closing/reloading tab with active microphone, audio exits successfully but a wrong log-error (1005) is shown.
We now process closing/reloading tab the same way we do when user hangup the call.
For some reason (still investigating), using turn/coturn on 443/tcp makes firefox's iceGathering process (during echo test) takes 12+ seconds (tested on webrtc's trickle page with multiple instances).
This was found when testing the current default (15s) on production with a private turn/coturn server on port 443/tcp. For default bbb setup (stun only), echo test still runs fast.
To avoid adding extra delay to iceGathering on this scenario (Firefox + turn on 443/tcp), i am just setting the default value back to the 5s (old default).
So , for those who wants to reduce the 1004 occurrences, increasing the iceGatheringTimeout could help (just be aware this adds delay on the mentioned scenario).
Added a default 'MEDIA' option: iceGatheringTimeout. This option allows admin to set a higher ICE gathering timeout, which can help when getting ICE errors during audio negotiation (eg 1004)
Default value set to 15s (current default is 5s).
Sometimes, when user already joined audio session, RTCPeerConnection may
find new ICE candidates, which triggers 'connected' state for peer's
'onconnectionstatechange' event. When this happens we process this
new state the same way when user is not running an audio session, which
makes html5client popup an annoying 'Audio Connected' message.
The audio keeps working fine, but this can make user think that there's a
connection issue, or the audio is reconnecting, while audio is ok.
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
Current sessionDescriptionHandlerModifiers of SIP.js acts before ICE gathering is done. This means we are not able to modify/strip candidates in local SDP.
This modifier acts on local SDP, by allowing user to modify SDP before it is sent on INVITE message.
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
If an ejected user tries to enter in the meeting using the current url
html5 client keep trying to validate that user, but without success
causing a validateAuthToken message spam until the connection times out.
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
