Using this larger value helps reducing ocurrences 1005/1010 errors for Chrome, avoiding an openssl's error which interrupts the dtls handshake (Chrome triggers "DTLS timeout expired" error)
This key size is also the default value used by freeswitch on switch_core_cert.c
Fixed two occurrences where the tryGenerateIceCandidates workaround rejected without an error, which borked the callers error handling
Also put it behind a config flag. This workaround used to be important when Kurento didnt infer prflx candidates properly, but that`s no longer the case. With the flag, we can disable the workaround to see if there`s any visible regression and hopefully remove it down the road
This adds the possibility to configure the SIP Via header to plain WS to allow reverse proxying from WSS to WS, internally, to work around a bug in freeswitch where the WSS stack would get deadlocked due to a still unidentified bug in there that has to do with SSL termination
Although Chrome's default is now unified plan, Chrome <-> FreeSWITCH ICE connection fails for some Chrome installations (specially those running on Windows).
FS ICE fails when Chromes's SDP has "a=mid:<index>" (instead of "a=mid:audio").
This fixes Error 1010 and situations where echo test takes too long.
This fixes#6414 regression, once we do the same older version of SIP.js used to do.
We now use both peer's connectionstatechange and iceconnectionstatechange to monitor ICE state for audio sessions.
The same way we did with old sip.js version, we leave iceconnectionstate trigger audio actions , such as connect, disconnect, reconnect.
We still listen for 'failed' state for connectionstatechange event, because chrome triggers this (tested on 86+).
This should reduce the audio error 1010 ocurrences, once some browsers (specially Chrome/Android) don't trigger connectionstatechangeevent.
This might reduce problems reported in #10708, which still needs more investigation though.
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.
