This was added as an option (websocketKeepAliveInterval), which is the interval to send keep alive messages.
Setting websocketKeepAliveInterval to 0 disables the keep alive, producing the same old behavior.
This helps avoid websocket disconnection due to socket inactivity, preventing it to unnecessarily reconnect.
Also, sometimes reconnect fails and error 1005 is triggered.
Fixes problems reported in #10985.
Also reduces occurrences of error 1005.
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
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).
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
This adds support for screensharing on Safari 12.1+ or whichever has getDisplayMedia enabled
To be conservative and backwards compatible, I`ve maintained the current gUM-based code available as a fallback for those still using older versions of FF or Chrome
Made screenshare constraints configurable. The constraints config will be piped directly to gDM, so it`s just a regular gDM constraint dictionary
We need to increase the length of the voice conference. If we have lots of meetings running,
there is a high chance of collision.
Need corresponding changes to FreeSWITCH dialplan.
In bbb_echo_test.xml, change to `expression="^echo(\d{5,11})$"`.
In bbb_conference.xml, change to `expression="^(\d{5,11})$"`.
