Switches from using JitsiConference.isJvbConnectionInterrupted
flag to peerconnection's ICE state. The interrupted
flag is not set if the connection fails initially.
fix(safari): Use standard audio constraints on non-chromium browsers
Standard audio constraints need to be passed to gUM for audio input device selection
to work properly on Safari.
Safari doesn't support permissions query, add a workaround for that.
Audio fails on Safari when we try to re-use audio m-lines whose direction has been set to 'inactive'
when the remote source left the conference.
Disable noisy mic detection on Safari as this causes audio input to fail on Safari on iPadOS
A new method which not only has Promise interface,
but also allows to queue IQs regardless of the current
connection's status.
Ping and send ICE failed notification requests are changed
to use this method which makes it more reliable on XMPP
Websocket connections on mobile(which in contrary to BOSH
dies much faster there).
feat(video-quality): control the sender resolution based on video quality settings (#1119)
* feat(video-quality): control the sender resolution based on video quality settings
* fix(video-quality): Apply the settings on newly created p2p/jvb jingle sessions
If a p2p/jvb session is not present when setSenderVideoConstraint is called,
make sure the settings are applied when they are created
e2ee: don't pass plain frames to decoder if we are encrypting (#1116)
If we are encrypting and the key index from a (potentially unencrypted)
frame exceeds our key ring size (1 currently) drop the frame.
This is a heuristic. We currently don't have the signaling for whether a
remote end does encrypt its frames or not.
Reverts #1098, that strategy does not work. When switching from garbage to non-garbage the decoder
would decode subsequent packets based on the wrong keyframe.
e2ee: decode static black frame for decryption errors (#1098)
similar to #1095 use a black keyframe instead of displaying a potentially
unencrypted stream. This was again captured with a modified version of
the webrtc sample, additional logging and a resolution of 320x180
fix(strophe.jingle.js): exception on ICE config modification
If ICE config object gets modified while there's any native
PeerConnection instance holding a reference, it will cause
an exception about final object being modified.
It happens when trying to reconnect XMPP Websocket on RN and
is caused by the fact that new TURN config is fetched after
the connection is re-established.
e2ee: generate silence in case of audio decryption errors
This handles decryption errors for audio differently than the
current version. Instead of forwarding horrible noise to the decoder,
replace the bytes with magic opus bytes for silence:
0xd8fffe
Those bytes were captured on a modified version of
https://webrtc.github.io/samples/src/content/peerconnection/endtoend-encryption/
with more dumping and after disabling the track.
fix(xmpp): clear any last strophe errors on reconnect
If user experiences any failed reconnect during the entire
session, even if followed by a success, then
the "connectionFailed" flag will trigger
JitsiConnectionEvents.CONNECTION_FAILED on hang up while
processing the disconnected transition.
When modifying the room protection (setting or unsetting a password), the muc#roomconfig_passwordprotectedroom should always be defined. This fixes issue #512.
fix(unified-plan): use RTCRtpSender.replaceTrack for mute/unmute/replace operations
Trigger renegotation only when negotiationneeded event is fired on the browser.
Do not disable simulcast for screensharing in unified plan.
Fix the order of the simulcast streams for Firefox
Alfredo Irarrazaval
paweldomas
Hristo Terezov
Jaya Allamsetty
Philipp Hancke
Saúl Ibarra Corretgé
Jaya Allamsetty
damencho
Alexander Shitikov
Guus der Kinderen
Andrei Gavrilescu