"Pinned" endpoints are deprecated, use "selected" instead.

Make sure screenshare w/ presenter gets the full 2500 Kbps bitrate.

The max bitrate for VP9 is enforced by adding the b=AS:<limit> line in the SDP since there is no way to configure the max. bitrates for the individual SVC streams using RTCRtpSender.setParameters.

* expose meeting-id-set event

* typo

Force the client to reload when the bridge that is handling the media goes down.
This mitigates issues seen on the bridge because of a client re-joining the call with the same endpointId, BWE issues, etc.
This behavior is configurable through 'enableForcedReload' setting in config.js.

On Firefox it throws:
DOMException: AudioContext.createMediaStreamSource: Connecting AudioNodes from AudioContexts with different sample-rate is currently not supported.

RN doesn't support RTCRtpSender yet. Therefore, media is suspended on RN by changing the media direction in the SDP whenever the client receives an ideal height of 0 for sender constraints on the bridge channel.

After ICE failure and subsequent recovery of the media connection to the bridge using session-terminate, the local connection status needs to updated when ICE connection is established on the new peerconnection. The local connectivity status will be shown as Lost otherwise.

* Support the urn:xmpp:extdisco:2 namespace specified by XEP-0125

* squash: Fix lint errors and drop unused code.

Co-authored-by: Jasper Hugo <jasper@jasperhugo.com>

Fixes https://github.com/jitsi/lib-jitsi-meet/issues/1466

* Added useTurnUdp option to API documentation

* Fixed typo

* feat: Detects shard changed when using websockets.

* squash: Simplifies code.

* squash: Rename method.

* squash: Skips checking for shard changed on the initial connection.

Determine the preferred codec for a given endpoint based on the config.js settings and the codecs supported by the endpoint.
The preferred codec is published in presence and then used by the other endpoints in the call during join/leave to determine
if the codec needs to be changed on the fly. Different codecs can be configuered for p2p/jvb connections.
The preferredCodec/disabledCodec settings under videoQuality will have precedence over the older settins like preferH264/disableH264.

I see we have an option to use replace track in conference level which its missing in this API. thanks

The jid received is the main jid in internal representation with [tenant]roomname@domain.com and we cannot compare it directly to roomname@conference.tenant.domain.com.

* fix: Drops unused parameter of join and sendPresence.

* feat: Skips using disco-info for features.

Uses presence to publish features added externally.
Recognizes jigasi participants using a specific presence extension used by jigasi.

* squash: Fixes tests.

* squash: Adds e2ee to the features in presence.

* squash: Fix function name and docs.

* squash: Drops detecting jigasi from initiator.

Using the newly added presence features.

* squash: Drops unused var.

* squash: Fix comments.

* squash: Adds a constant and for E2EEE feature.

which should include the colon in order not to match something like
  a=simulcast-is-cool

in order to not match a=msid-somethingelse, a=ridiculous and similar

Bumps [ini](https://github.com/isaacs/ini) from 1.3.5 to 1.3.7.
- [Release notes](https://github.com/isaacs/ini/releases)
- [Commits](https://github.com/isaacs/ini/compare/v1.3.5...v1.3.7)

Signed-off-by: dependabot[bot] <support@github.com>

There are cases where we start p2p on the first participant received presence, where there are more presences coming. This starts the process of p2p resulting and few iq errors while terminating it quickly after starting it.
On a normal joining as third participant we see around 38 send or received iq messages, in the problem scenario we see 60.

Explicitly check if all the encodings report the same scaleResolutionDownBy value before trying to ensure they match the expected values. This makes Chrome VP9 work again.

Due to no timeout value passed the disconnection was not detected.

Now stores the time of the raw input when ping is received. Before this change we were storing the Date.now and comparing to _getTimeSinceLastServerResponse and we needed small adjustment of values as it may lead to no pings sent.

* fix: Always query the main domain for features.

* squash: Always use the main domain for turn credentials and ping.

* squash: Fix tests.

* feat: Make enableRemb signal client remb support.

* feat: Make enableTcc signal client tcc support.

* ref: Remove the enableOpusRed conference-wide option.

* ref: Remove the "enforcedBridge" option.

* ref: Remove pre-configured jigasi address.

Only one audio or video track is correctly supported by lib-jitsi-meet.
This is an attempt to mitigate an issue reported by the community
where the user's audio is played in the conference even though the UI
shows the user as muted.