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.
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
Refactored STUN/TURN fetch to be done only once, when successful, per session and cache it in mem to avoid too many reqs. Current way is a bit dumb, this should increase reliability a bit more. The caching is configurable so folks who want to use very short lived TURN credentials can disable it
Add a fallback STUN config option to be used when the default STUN/TURN fetch fails
Clean the safari/no candidate generation pre flight check from 3rd party STUNs
Fix deadlock in audio join when STUN/TURN fetch failed
