It is not 100% clear to me when it happens, but I think it could happen
in some race condition where a track is unmuted when it's being disposed
or something around those lines.
The fact is that any muted tracks are disposed by replaceLocalTrack(track.jitsiTrack, null) and they should not be used anymore.
Supposedly fixes a crash:
Failed to add local track to conference Track has been already disposed
In version 1.15 the storage backend was rewritten, which hopefully allows us to
fix this crash on Android:
Caused by java.lang.IllegalStateException: attempt to re-open an already-closed object: SQLiteDatabase: /data/user/0/org.jitsi.meet/databases/RKStorage
at android.database.sqlite.SQLiteClosable.acquireReference(SQLiteClosable.java:55)
at android.database.sqlite.SQLiteDatabase.queryWithFactory(SQLiteDatabase.java:1160)
at android.database.sqlite.SQLiteDatabase.query(SQLiteDatabase.java:1036)
at android.database.sqlite.SQLiteDatabase.query(SQLiteDatabase.java:1204)
at com.reactnativecommunity.asyncstorage.AsyncStorageModule$1.doInBackgroundGuarded(AsyncStorageModule.java:159)
at com.reactnativecommunity.asyncstorage.AsyncStorageModule$1.doInBackgroundGuarded(AsyncStorageModule.java:146)
at com.facebook.react.bridge.GuardedAsyncTask.doInBackground(GuardedAsyncTask.java:35)
at com.facebook.react.bridge.GuardedAsyncTask.doInBackground(GuardedAsyncTask.java:19)
at android.os.AsyncTask$2.call(AsyncTask.java:305)
at java.util.concurrent.FutureTask.run(FutureTask.java:237)
at com.reactnativecommunity.asyncstorage.AsyncStorageModule$SerialExecutor$1.run(AsyncStorageModule.java:63)
at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1133)
at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:607)
at java.lang.Thread.run(Thread.java:760)
- Fixed react native community slider to work on both android and ios
- Removed InviteButton from native menus
- Fixed buttons spacing in native OverflowMenu
- Participant context menu details are shown only for remote participants
With react-native-webrtc 1.89.2 the remote SDP is properly updated before
onaddstream is fired so it's no longer needed.
Also, for readability, IPv6 address synthesis has been moved to a standalone
utils file.
THe new version fixed a longstanding problem with RN not updating the JS side
SDP representation properly. This will allow us to remove a hack we currently
have to sidestep this.
- Remove button list from interface_config.js since it has been deprecated for a
while
- Alphabetically sort buttons in config.js and constants.js to make it easier to
add / remove items
- Add missing invite and toggle-camera buttons to default constants
- Remove no longer existing "fodeviceselection" button
Fixes: https://github.com/jitsi/jitsi-meet/issues/9605
* feat(JingleSessionPC): Enable unfied plan by default for chrome p2p.
* fix(JingleSessionPC): Fix startMuted cases for p2p unified plan. Chrome doesn't create a decoder for ssrc in the remote description when there is no local source and the endpoint is offerer. Initiating a renegotiation with the endpoint as a responder fixes this issue. Add a workaround until Chrome fixes this bug.
* fix: Missed SSRCs in Unified Plan with several "ssrc-group:FID" groups. (#1658)
e6648fac96...b815157a22
* fix(TPC): Fix the screenshare issue when user starts video muted on chrome. Munge 3 ssrcs in the SDP for chrome in unified plan always for the simulcast case.
053a26604d...e6648fac96
* fix(JingleSessionPC): Disable unified-plan for p2p chrome. Do not enable unified plan for p2p chrome by default until StartMutedTest is fixed. Fix media direction for case when there are no local and remote sources, should be set to 'inactive' in that case.
3a313a244d...053a26604d
* fix(codec-selection): Fix VP9 codec switching issue in Chrome unified-plan. Munge only the m-line that corresponds to the source that the browser will be sending. Do not select VP9 on Firefox. Detect support for RTCRtpTransceiver#setCodecPreferences correctly.
89a7e2d9cd...3a313a244d
* fix(RTC): In unified-plan mode, disable the low resolution streams for low fps SS. In unified plan impl, it is not possible to enable/disable simulcast during the call since the same sender is re-used for all local video tracks. Therefore, disable the low resolution simulcast streams for low fps screensharing so that the bridge forwards only the highest resolution stream which is important for low fps screensharing.
f974007ca6...89a7e2d9cd
* fix(JingleSessionPC): Do not check if the ssrc already exists in the RD when adding a ssrc-group from source-add.
* feat: Switch to unified plan on chrome by default unless explicitly disabled.
* fix(VADAudioAnalyser): NPE error evaluating this._vadEmitter.on (#1652)
* Small fix in tokens doc.
b43a9fa0ee...f974007ca6
* Created desktop reactions menu
Moved raise hand functionality to reactions menu
* Added reactions to chat
* Added animations
* Added reactions to the web mobile version
Redesigned the overflow menu. Added the reactions menu and reactions animations
* Make toolbar visible on animation start
* Bug fix
* Cleanup
* Fixed overflow menu desktop
* Revert mobile menu changes
* Removed unused CSS
* Fixed iOS safari issue
* Fixed overflow issue on mobile
* Added keyboard shortcuts for reactions
* Disabled double tap zoom on reaction buttons
* Refactored actions
* Updated option symbol for keyboard shortcuts
* Actions refactor
* Refactor
* Fixed linting errors
* Updated BottomSheet
* Added reactions on native
* Code cleanup
* Code review refactor
* Color fix
* Hide reactions on one participant
* Removed console log
* Lang fix
* Update schortcuts
* fix(participants): Change from array to Map
* fix(unload): optimise
* feat: Introduces new states for e2ee feature.
Stores everyoneSupportsE2EE and everyoneEnabledE2EE to minimize looping through participants list.
squash: Uses participants map and go over the elements only once.
* feat: Optimizes isEveryoneModerator to do less frequent checks in all participants.
* fix: Drops deep equal from participants pane and uses the map.
* fix(SharedVideo): isVideoPlaying
* fix(participants): Optimise isEveryoneModerator
* fix(e2e): Optimise everyoneEnabledE2EE
* fix: JS errors.
* ref(participants): remove getParticipants
* fix(participants): Prepare for PR.
* fix: Changes participants pane to be component.
The functional component was always rendered:
`prev props: {} !== {} :next props`.
* feat: Optimization to skip participants list on pane closed.
* fix: The participants list shows and the local participant.
* fix: Fix wrong action name for av-moderation.
* fix: Minimizes the number of render calls of av moderation notification.
* fix: Fix iterating over remote participants.
* fix: Fixes lint error.
* fix: Reflects participant updates for av-moderation.
* fix(ParticipantPane): to work with IDs.
* fix(av-moderation): on PARTCIPANT_UPDATE
* fix(ParticipantPane): close delay.
* fix: address code review comments
* fix(API): mute-everyone
* fix: bugs
* fix(Thumbnail): on mobile.
* fix(ParticipantPane): Close context menu on click.
* fix: Handles few error when local participant is undefined.
* feat: Hides AV moderation if not supported.
* fix: Show mute all video.
* fix: Fixes updating participant for av moderation.
Co-authored-by: damencho <damencho@jitsi.org>
* fix(virtual-background): Style adjustments on virtual background dialog on small screens.
* fix(virtual-background): Style adjustments on virtual background dialog on small screens.
Co-authored-by: tudordan7 <tudor.pop@decagon.tech>
* feat(toolbox) allow any toolbox button to be displayed as main
fixes the previous behaviour where only a certain set of buttons were whitelisted for being displayed in the main toolbar
* code review
* code review - fix avatar icon position
* fix(TPC): Do not remove ssrcs from remote desc for p2p. In unified plan, re-use of m-line (i.e., adding an SSRC, removing it and then adding it back) causes the browser to not render the media on Chrome and Safari. The WebRTC spec is not clear as to how browsers have to behave, this doesn't cause any issues on Firefox. As a workaround, only change the media direction and leave the ssrc in the remote desc. This automatically triggers a 'removetrack' event on the associated MediaStream and the track can be removed from the UI.
* Drops old prosody versions from the tokens instructions
0cdfb79c2e...b43a9fa0ee
Adding some more interface translation from main.json, specifically, the participants button, participants pane and virtual background selection dialog window
* squash: Set capScreenShareBitrate flag every time a new pc is created.
* feat(RTC): Add the ability to change desktop share fps. Provide a method for changing the capture fps for desktop tracks during the call. These changes to the lib are needed for making it configurable from the UI.
46ec23fcdc...229015a6f3
* feat: Change the screenshare capture fps from UI.
Add the ability to change the capture frame rate for screenshare from the UI. The fps becomes effective only on screen shares that are started after the setting is changed.
* squash: add missing JSDOCs and translations for frames-per-second.
* fix(virtual-background): Fix resize action and prevent mirror behaviour on desktop share as a virtual background.
Co-authored-by: tudordan7 <tudor.pop@decagon.tech>
* feat: Initial UI part for A/V moderation.
Based on https://github.com/jitsi/jitsi-meet/pull/7779
Co-authored-by: Gabriel Imre <gabriel.lucaci@8x8.com>
* feat: Hides context menu in p2p or only moderators in the meeting.
* feat: Show notifications on enable/disable.
* feat(moderation): Add buttons to participant list & notifications
* fix(moderation): Fix raised hand participant leaving
* feat(moderation): Add support for video moderation
* feat(moderation): Add mute all video to context menu
* feat(moderation): Redo participants list 'More menu'
* fix: Fixes clearing av_moderation table.
* fix: Start moderation context menu
* fix(moderation): Show notification if unapproved participant tries to start CS
Co-authored-by: Gabriel Imre <gabriel.lucaci@8x8.com>
Co-authored-by: Vlad Piersec <vlad.piersec@8x8.com>
* fix(RTC): Do not overwrite other constraints when resolution option is used. When the resolution option was being used, all the other constraints like frameRate and facing mode were being overwritten.
24627e1b95...46ec23fcdc
* feat(vpaas, recording): Show recording link to recording initiator
This applies only for jaas users for now but is easily extensible.
Changed the recording sharing icon according to ui design.
* fix(vpaas, recording): Guard for deployment info
* fix(TPC): Filter ssrcs differently while extracting the SSRC map from SDP. Use 'msid' for plan-b clients and 'cname' for unified-plan clients.
fad985e95a...d5e60583b8
* fix(TPC): fix local resolution/fps stats. Browsers do not generate a 'msid' attribute for ssrcs in unified plan mode, use mediaType as a key for the TrackSSRCInfo map.
* fix(recording): Send participant id when recording starts/stops (#1632)
8057f12a39...2259d44185
* fix(RTC): Adjust the media direction for p2p conn. For p2p connections, the media direction needs to be adjusted after every source-add/source-remove is processed based on the availability of local sources.
* fix(RTC): Use a enum for media direction.
5738c80baf...d9d9b7fc31
If we mute a video in Youtube it is stored in the current browser session and if someone shares a video it will start muted and we don't have the control to unmute it.
* fix(JingleSessionPC): Disable unified-plan for p2p. Disable cross browser p2p using unified plan until all the issues are fixed.
0993c8e93d...5738c80baf
Previously gravatars (external resources) were preloaded even if
disableThirdPartyRequests was set to true in the config, as the
config may be empty at the time of preloading.
Closes: #5670
Signed-off-by: Christoph Settgast <csett86@web.de>
* i18n: zhTW: new translation
* Fixed the proper semantic of toolbar.hangup
* i18n: zhTW: new translation
* Fixed the proper semantic of speaker related strings
* i18n: zhTW: new translation
* Added startupoverlay.genericTitle
* fix(LocalSdpMunger): Fix unit test.
* fix(CodecSelection): Call RTCRtpTransceiver#setCodecPreferences before renegotiation. Call RTCRtpTransceiver#setCodecPreferences with the preferrred codec order before every createOffer/createAnswer. This ensures that the codec preference is enforced even when there is no local description available yet while the preferred codec is being set immediately after media session creation.
* fix(JingleSessionPC): Add a workaround for chrome issue. The 'signalingstatechange' event for 'stable' is fired after the 'iceconnectionstatechange' event for 'completed' is fired on chrome in Unified plan. This prevents the client from switching the media connection to the p2p connection once the ice connection for p2p gets established.
* fix(Logging): Log enhancements. Add a preifx to logs for idenitifying the type of TPC/jingleSessionPC.
* feat(TPC): Enable unified-plan support for Chromium based browsers. This can be controlled through the config.js option 'enableUnifiedOnChrome'.
* fix(TPC): Do not configure encodings on Safari until reneg. Avoid configuring the encodings on Chromium/Safari until simulcast is configured for the newly added track using SDP munging which happens during the renegotiation.
* fix(TPC): Do not configure encodings on chromium immediately after replace track. Avoid configuring the encodings on chromium immediately after replace track since the encoding params are read-only until the renegotation is done.
* fix: send json message (#1180)
be3e2a69f2...3fb44f7695
* fix(SDP): Add missing msid for p2p sources.
* fix(TPC): Don't convert plan-b<->unified-plan SDPs for p2p.
* squash: Implement review comments.
* fix(JingleSessionPC): Do not try to re-use inactive mid for new remote ssrcs. The direction was marked as 'inactive' only on Firefox as Safari had audio issues when an inactive mid is re-used. Chrome (in unified-plan) needs the direction of the mid in remote desc to be set to 'inactive' for a 'removetrack' to be fired on the associated media stream whenever a remote source is removed.
* fix(SDP): Drop SSRCs whenever the transceiver direction is 'inactive' or 'recvonly'. This is needed only for JVB connections. Add unit tests for LocalSdpMunger.
* fix: Ignore startAudioMuted/startVideoMuted for p2p. The tracks will not be added when the call switches from jvb to p2p for an endpoint that joins muted by focus.
* fix(RTC): Do not suppress the source updates on Firefox. If the msid attribute is missing, then remove the ssrc from the transformed description so that a source-remove is signaled to Jicofo. This happens when the direction of the transceiver (or m-line) is set to 'inactive' or 'recvonly' on Firefox. Not signaling these source updates creates issues with remote track handling on the other endpoints in the call.
* fix(RTC): Set transceiver direction after RTCRtpSender#replaceTrack. This fixes the issue where TRACK_REMOVED event is not fired when a remote track is removed from the peerconnection. Fixes https://github.com/jitsi/lib-jitsi-meet/issues/1612 and https://github.com/jitsi/jitsi-meet/issues/8482.
60c5667957...be3e2a69f2
In case limited those connection will be whitelisted and unlimited. Updates existing configurations to make sure prosody update will not break it by limiting too much.
Uses 28c16c93d79a version of the module: https://modules.prosody.im/mod_limits_exception.html
Will be available in prosody 0.12.
* fix(caps): Disable TCC on Firefox. There is a known issue with Firefox where the BWE gets halved on every renegotiation causing the low upload bitrates from the Firefox clients.
* fix: Drops unused config, fixesjitsi/lib-jitsi-meet#1620.
* fix(e2ee): destroys olm session on disabling e2ee
f95a455c08...60c5667957
* Added mod_reservations prosody plugin
* Removed comments re mutex
* Add support for HTTP retries and expose config to tweak retry behaviour
* Removed TODO comment. Feature implemented
* Added multi-tenant support
* renamed config var and default to always including tenant name in name field
* Simplified handling of multi-tenant
* Fixed bug with DELETE not called on reservation expiry
* fix: Fixes destroying room.
Co-authored-by: damencho <damencho@jitsi.org>
* fix(TPC): Return default codec if the local sdp is not available. Get the correct media type when generating the source identifier.
88560a8a5e...9eb4af1e80
* feat(prosody-modules): Moves a function for getting room to util.
* feat: Audio/Video moderation.
* squash: Fix docs.
* squash: Changes a field name in the message for adding jid to whitelist.
* squash: Moves to boolean from boolean string.
* squash: Only moderators get whitelist on join.
* squash: Check whether in room and moderator.
* squash: Send to participants only message about approval.
Skips sending the whole list.
* feat: Separates enable/disable by media type.
Adds actor to the messages to inform who enabled it.
* squash: Fixes reporting disable of the feature.
* squash: Fixes init of av_moderation_actors.
* squash: Fixes av_moderation_actor jid to be room jid.
* squash: Fixes comments.
* squash: Fixes warning about shadowing definition.
* squash: Updates ljm.
* fix: Fixes auto-granting from jicofo.
* squash: Further simplify...
* fix(JingleSession): Move the ssrc identifier generation to LocalSdpMunger.
* fix(logger): Logging enhancements. Get rid of noisy logs related to SDP transformations which are redundant. Fix formatting and add missing information.
7cbd9c8f2a...923aa449c4
The client now listens for changes to lastN, selectedEndpoints and maxReceiverVideoQuality in redux to trigger sending bridge message in the new format. This fixes an issue where the stage view <-> tile view changes prompt two receiver constraints messages to be sent, first with the maxHeight update and then with the selected endpoints update.
* fix(quality-control): Propagate the height constraints to p2p session. If the application is using the new receiver constraints, propagate the height constraint to the p2p session as well.
* build(deps): bump lodash from 4.17.19 to 4.17.21
* chore(deps): bump hosted-git-info from 2.8.8 to 2.8.9
74a90f7035...7cbd9c8f2a
* fix(quality-control): fix constraints sent on channel initialization. Do not send old format constraints if no constraints are set before the channel is initialized.
* chore(deps) run npm audit fix
* chore(deps) update webrtc-adater@8.0.0
86c7a35817...74a90f7035
Unpin the screenshare when the screensharing participant leaves. Switch to tile view if no other participant was pinned before screenshare was auto-pinned, pin the previously pinned participant otherwise.
* Added lobby component translations for bg language
* Update translation of shared youtube video button and label
* Translate security litterals in bg. Update lobby translations
* Add dependency for promise.allSettled. Older chrome versions like M72 do not support Promise.allSettled.
* fix(conference): Enable p2p for unified plan clients.
* fix(TPC): Use addTrack instead of addStream in Unified-plan impl.
* Add missing spaces in debug logs.
ad5692d6aa...e362c89eb6
* fix(SDP): Move all SDP related files to a different dir. SDP utility classes are spread across RTC and XMPP directories now, moving these class files to a 'sdp' directory.
* fix(stats): Return promise for getStats. Switch to returning a Promise for getStats. Reset frame rate stat to 0 when video is suspended as a result of endpoint falling out of last-n.
* Fix: sysMessageHandler not deleted (#1590)
* task(e2ee): switch back to GCM
463e213b3f...7667117117
* fix(quality-control): Send the new constraint on join. Fixes the case where the old format height constraint is sent on join for a jvb media session.
7dedb59b9c...463e213b3f
* fix(quality-control): Switch to new receiver constraints by default. Use the new receiver constraints unless it is explicitly disabled through config.js.
3c9913ed61...7dedb59b9c
* feat: Exposes a hook to mod_external_services data.
The hook can be used to get turn servers and credentials from another module.
* feat: JiConOp2 pushes a message with some info to clients.
* feat: JiConOp adds config for shard name feature.
* squash: Changes message type to service-info.
* squash: Drops the event in external_services.
* fix(authentication) login dialog now closes when connection is established
* fix(authentication) fixed shibboleth auth
* fix(authentication) renamed authenticateExternal func to authenticate and updated its logic
* fix(authentication)removed logindialog.js and created actions.any
* fix(authentication) removed focus from externalauthwindow
* fix(authentication) removed private sign from some actions and added openLoginDialog to actions.any
* fix(authentication) exported all from actions.any
* fix(authentication) reverted change regarding externalAuth
* fix(authentication) fixed indentation
* fix(JingleSession): Increase the ICE candidate gathering timeout to 150ms. This will reduce the numbers of transport-info IQs sent by the client.
* fix(TPC): Fix error handling for getStats.
ca325f5ef9...0dc1540a44
* fix(stats): Use promise-based getStats on all browsers. Get rid of the browser specific keys and use the standard spec-compliant fields for stats. Get the resolution/fps for remote streams from 'inbound-rtp' stats. Use the 'track' stats for the local resolution/fps since these take the active simulcast streams into account.
8b3dc59374...ca325f5ef9
* fix(SS): Implement a 2500Kbps limit for VP9 SS.
* fix(RTC): Remove stream effect before disposing the track. Remove the effect instead of stopping it so that the original stream is restored on both the local track and on the peerconnection. Fixes issues when a stream with effect applied is replaced on the pc after it is muted, also fixes https://github.com/jitsi/lib-jitsi-meet/issues/1537.
* fix: Drops unused config.
1f3f85978d...baa78aca40
* language update: main.json and main-de.json
* language update: main-de.json
* language update: main-de.json
* revert changes in main.json and delete same entries in main-de.json
Co-authored-by: qwertiko <gross@qwertiko>
- added a few missing lines
- changed some fragmented phrases, so that they sound fluent, once reunited
- gave coherence to the usage of the persons (I, or you) in some mismatching title and dialog boxes
In https://github.com/jitsi/jitsi-meet/pull/8673 we inadvertently removed the
backwards compatibility code which would show the security button when the
"info" button is configured in interface_config. The security button replaced
the info button.
* lang:New translation Hindi(hi)
Work in progress. I will update this on the way. I also want to quickly test this out. Thanks
* add new lang Hindi(hi)
* add HIndi(hi)
* Update main-hi.json
* Get rid of stats debug message, fix typo with codec type.
* fix(receiveVideoController): Do a deep copy of constraints for comparsion.
* fix(codec-selection): Fix codec selection for unified plan browsers.
93af5ada95...2e598a4bda
* fix(receiveVideoController): Do not send redundant video constraints to the bridge.
* feat(stats): Add a new bridge message "EndpointStats" for stats. Use the new Colibri message "EndpointStats" for broadcasting the local stats. The bridge then will be able to filter the endpoint stats and send them only to the interested parties instead of broadcasting it to all the endpoints in the call.
* Test RTCRtpReceiver.getCapabilities before using
2b94da12e8...93af5ada95
Set higher preference for screenshare over dominant speaker when trying to elect a participant for large-video. This prevents the dominant speaker from taking over the stage when a user toggles tile view on and off while a screenshare is in progress.
Promise.allSettled is supported from RN 0.63 onwards and is not supported on the current version, use a polyfill for that shims Promise.allSettled if its unavailable or noncompliant.
Co-authored-by: Saúl Ibarra Corretgé <saghul@jitsi.org>
* fix(TPC): get ssrc info per ssrc and not per mline.
* feat: Consider absence of A/V muted from presence as muted.
* Feature: Moderator can revoke moderator role to others and himself (#1532)
4191198233...0e180efdfa
* fix(JingleSession): Avoid renegotiation when user with no sources leaves the call.
* feat: participant kick reason add
* ref(RTC): remove legacy pc constraints. Stop using the legacy pc constraints that are no longer wired up to WebRTC.
* fix(deps) update webrtc-adapter to v7.7.1
087a8e19eb...4191198233
* fix(load-test): Fixes unmuting loadtest client.
Fixes the case where audio track was not added due to jicofo muting clients.
* squash(load-test): Drop noAutoLocalAudio and change add track logic.
Trying to mimic jitsi-meet.
* squash(load-test): Fix adding video.
Check whether context is that of an Activity before launching the Jitsi Conference Activity. If context is not an activity context, apply flag FLAG_ACTIVITY_NEW_TASK to the Jitsi Activity Intent to ensure activity can launch without error.
This scenario would manifest when a user attempts to launch the Jitsi Actvity from a Widget... for example.
https://developer.android.com/about/versions/pie/android-9.0-changes-all#fant-required
* squash: Use different function syntax.
* squash: Fix lint errors.
* Process stats immediately before setting the interval.
* feat(ReceiveVideoController): Add the ability to send constraints in the new format. Add the ability to send the bridge messages for the receiver video constraints in the new format directly.
676c7a9105...5796d83bb1
The majority of the code is in the WASM file, the JS is just 9KB.
It's so little, in fact, that the performance hint for the main bundle didn't
have to be adjusted.
The majority of the code is in the WASM file and models, this is just a few KB.
It's so little, in fact, that the performance hint for the main bundle didn't
have to be adjusted.
For the video to play on Safari mobile browser, the playsInline attribute needs to be set to true. Set the mute attribute as well which was accidentally removed in code refactor.
Fixes this Apple Store Connect warning:
~~~
ITMS-90473: CFBundleShortVersionString Mismatch - The CFBundleShortVersionString
value '1.0' of extension 'jitsi-meet.app/PlugIns/JitsiMeetBroadcast
Extension.appex' does not match the CFBundleShortVersionString value '21.0.0' of
its containing iOS application 'jitsi-meet.app'.
~~~
We will filter the initial presence where participant is announced as `participant` and shortly after that we send a second presence with the new `moderator` role.
* feat(browser-support): Add support for WKWebview based browsers. Apple added getUserMedia support for WkWebview based browsers like chrome and Firefox on iOS 14.3. These browsers behave as Safari does on iOS. Therefore, extend the Safari checks to these webkit based browsers as well.
08ce96d881...e60f09b189
* squash: Always get lastN value from JitsiConference instance.
* fix(lastN): Return the correct lastN value for the conference.
* Use unified plan for mobile browsers on iOS
d31b5a2d5e...08ce96d881
* fix(conference): Do not signal muted tracks on join. Do not add the muted audio/video tracks to the peerconnection on join. The tracks will be added when the user unmutes for the first time. This reduces the number of remote sources that will be added when a participant joins a large call where everyone joins muted (startAudioMuted/startVideoMuted setting).
e83fb93d2d...d31b5a2d5e
Do not add the muted audio tracks to peerconnection until the user unmutes the first time. This applies to startSilent, startWithAudioMuted and startAudioMuted/startVideoMuted config.js settings.
* Update dateUtil.js
* version up moment
* exclude unnecessary languages in Moment.js from webpack
* add Occitan of Moment.js
* Fixed auto-formatting
* add require missing by mistake
* Only show more numbers link if multiple numbers are available
* Fixed some linter errors
* Try to make flow happy
* Fixed another linter error
* Another try to make eslint happy
* Silence eslint
* Added new config to enable individual sharing features
* make config values url friendly
* Add new setting to whitelist
* Fixed some linter issues
* Fixed more linter issues
* Fixed merge error
* Check if interfaceConfig is defined
* Only show more numbers link if there is more than one number
* fix(RTC) fix device selection not being available
* fix(TPCUtils): undefined is not an object (evaluating 'this.tpcUtils.replaceTrack(e,t).then')
4c668023b3...e6ef4e7ae9
* fix(TPC): Remove the existing track instead of overwriting. When a second remote track of the same mediatype is received for an endpoint, remove the existing track before creating the new remote track.
9beb47fe5f...4c668023b3
* fix(e2ee) fix disabling E2EE
* fix(e2ee) fix key index after ratchetting
* fix: Drop caps handling (#1495)
* fix(SendVideoController): Apply the sender constraint only when it changes. There were cases where the bridge was sending the same constraint multiple times causing redundant calls to getParameters/setParameters on the RTCRtpSender.
* feat: Use the new bridge signaling format.
* fix(gum) update permissions prompt detection
c534f74884...6a7b16c33e
* Add buttons to send messages/set nickname.
* Redesign message/nickname inputs.
* Pin messages to the input.
* Add keyboard avoider for Safari.
* Make chat content scrollable on mobile.
* fix(SendVideoController): Apply the sender constraint only when it changes. There were cases where the bridge was sending the same constraint multiple times causing redundant calls to getParameters/setParameters on the RTCRtpSender.
* fix(gum) update permissions prompt detection
beaff3dd02...7f919faacc
* feat(load-test): Senders unmute themselves if muted by policy.
* feat(load-test): Adds option to skip creating local audio track.
We currently create local audio track even when starting audio muted. Adding the option to control that can load test that for clients or signalling.
* ref(QualityController): Split send and receive video constraints handling.
* fix: Save guards _features to be always empty and nver undefined. (#1493)
d1f0ab4d5a...c534f74884
* fix(load-test): Always create local audio track.
When audio mutes will mute the track. Also fixes previous change where we do not add any of the tracks to the room.
* squash: Fix lint errors.
* fix(GUM-permissions): cache permissions on init.
* feat: Reuse billingId from localstorage as jitsiMeetId.
* fix(example) simplify
* feat(docs) mvoe API documentatrion to the handbook
84357ce1a8...d1f0ab4d5a
* feat(conference): Enable forced reload of client on bridge failure.
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.
The client skips the pre-join page when the page reloads.
* squash: refactor the restart logic.
* squash: fix description
* squash: dispatch conferenceWillLeave action before reload.
Add the ability to configure different max bitrates for VP8 and VP9.
Set max bitrate for presenter to 2500 Kbps irrespective of the configured max bitrates for video.
479dd98...77978f0.
* Enforce fixed column number at various width breakpoints.
* Bring back the filmstrip at small sizes but hide it.
* Change default maximum columns to 7.
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.
LJM update - 3570339360...be18ff34be.
When an endpoint that doesn't support the preferred codec (VP9) joins a conference, all the other endpoints fallback to VP8 until the endpoint leaves the call.
Set version to 1.0.0 with a very large version code so it's automatically kept
around when pushing new versions.
Additionally drop some no longer needed icon assets (bubblewrap did this).
* Fix toolbox buttons not displaying properly when chat is open.
* Open chat in fullscreen dialog past custom thresholds when mobile/desktop toolbox would become unusable due to chat
* Remove mobile chat check when displaying toolbox
* feat: Add mod_client_proxy and mod_roster_command.
Taken from prosody-modules 4317:456b9f608fcf with the
mod_roster_command patch applied.
* feat: Use mod_client_proxy to proxy to jicofo.
It appears that at the time of this writing, creating audio tracks blocks
the browser's main thread for a long time on safari. Wasn't able to confirm
which part of track creation does the blocking exactly, but not creating
the tracks seems to help and makes the UI much more responsive.
* Show recording started notification to the initiator
* Translate 'recording.on' language key for English and Turkish
Translate 'liveStreaming.on' language key for English and Turkish
* fix(jitsi-meet-web-config.postinst) allow cert and key pre-selection
* fix(jitsi-meet-web-config.postinst) jvb-hostname gets value from db_go instead of db_get
Co-authored-by: Jakob Pfeiffer <pgp-jkp@pfeiffer.ws>
Due to how the filmstrip size if computed I don't think there is a good way to
animate the change in size, so just ignore the toolbar, it will be hidden soon
enough.
* Updating and uniforming italian translation
- translate uniformly «meeting» to «conferenza», «chat» to «conversazione», ellipsys to «...», verbs in -ing with «in corso»
- correct a few typos
- update a message with old and unused placeholder
- translate some English messages
* typo
- add missing double quotes
* Fixed translation for "meeting" to "riunione"
Swift has a longstanding bug where a framework and a type cannot be named the
same. We have somehow managed to not run into this, but it now seems to be
hitting us.
Since this is a breaking change, this starts the road for SDK 3.0.
* fix(main-ko.json) Update overall korean spelling & words
* fix(_welcome_page.css) update .insecure-room-name-warning_margin-top from 5px to 15px
* fix(_welcome_page.css) initialize .insecure-room-name-warning_margin-top from 15px to 5px
* fix(main-ko.json) add keyboardShortcuts videoQuality
* fix(main-ko.json) Update overall korean spelling & words
* fix(_welcome_page.css) update .insecure-room-name-warning_margin-top from 5px to 15px
* fix(_welcome_page.css) initialize .insecure-room-name-warning_margin-top from 15px to 5px
* Add option to open Etherpad on join
For sites that focus on collaborative editing during meetings, add
an option which, when set, will automatically open etherpad when a
participant joins.
* Add openSharedDocumentOnJoin to config whitelist
This also adds some config file doc comments about the option,
including a note about the choice not to honor it in the mobile app.
...caused by bad state as a result of timing issue around the prejoin flow.
If get user media call is delayed for a while and if user joins
the conference, when it hasn't completed then confrence.js will not assign
'localAudio' and 'localVideo' variables and will create additional media
tracks on unmute operation and add them to JitsiConference via replaceTracks
operation.
* Update the Czech translation of `addPeople`
* Update the Czech translation of `calendarSync`
* Update the Czech translation of `chat`
* Add the Czech translation of `chromeExtensionBanner`
* Update the Czech translation of `connectingOverlay` and `connection`
* Update the Czech translation of `connectionindicator`
* Update the Czech translation of `deepLinking`
* Update various strings in the Czech translation
* Update various strings in the Czech translation
* Fix a trailing comma that broke JSON
* Sort keys in main-cs.json to deduplicate the translation and make diffs from main.json more readable
* Add several missing strings in the Czech translation
* Add several missing strings in the Czech translation, mainly lobby mode
* Add several missing strings in the Czech translation, mainly `prejoin`
* Add the missing Czech translation of `recording` and `security`
* Update various strings in the Czech translation
* Add the missing Czech translation of `accessibilityLabel`
* Add the missing Czech translation of `toolbar`
* Update various strings in the Czech translation
* Update various strings in the Czech translation
* Various edits of the Czech translation and a spell check
* Add missing language names in the Czech translation
* feat: Exposes a method for checking is remote track received and played.
Used for some tests in torture.
* squash: Drop not matching string.
Duplicate translation key with not matching content.
* squash: Moves torture specific functions to features/base/testing.
Listens for media events from the video tag of the large video and stores them in redux.
* squash: Fix comments.
* feat: Listens for media events from the video tag of the remote videos and stores them in redux.
* squash: Fix undefined videoTrack if between switches.
Looks like audio devices must be re-set after focus was lost and regained.
Otherwise some devices (tested on a Samsung Galaxy S9) are in a weird state
where the second microphone is not used when speakerphone is on.
While the base URL remains configurable, this patch reverts back to using
Gravatar.
We noticed high latency with libravatar and contacted them. They are in the
process of migrarting to a better infrastructure (it's a single personal server
at the moment) so we'll re-evaluate once that has happened.
As for why not leave the default and change it on the meet.jit.si installation,
we don't want to kill their server :-)
As per @fremzy, the "Save Logs" feature generates a json
file with a bevy of technical information about the
meeting. This log contains the server name, server IP
address, participant's IP addresses (only in p2p sessions)
e.t.c. While this may be a useful feature for the
admin-like 'moderator', it creates unnecessary exposure
when made readily available to all users in the meeting.
This commit fixes#8036 by a config.js option to enable
the link (disabled by default), thus giving the owner of
the deployment the choice of enabling it or not.
When trying to auto pin screenshare, always select the endpoint even though it happens to be the large video participant in redux. The auto pin screenshare logic kicks in after the track is added. If the screenshare endpoint is not among the forwarded endpoints from the bridge, it needs to be selected again.
@@ -152,3 +152,20 @@ this model but new features should follow this new layout.
When working on an old feature, adding the necessary changes to migrate to the new
model is encouraged.
### Avoiding bundle bloat
When adding a new feature it's possible that it triggers a build failure due to the increased bundle size. We have safeguards inplace to avoid bundles growing disproportionatelly. While there are legit reasons for increasing the limits, please analyze the bundle first to make sure no unintended dependencies have been included, causing the increase in size.
First, make a production build with bundle-analysis enabled:
Jitsi Meet is an open-source (Apache) WebRTC JavaScript application that uses [Jitsi Videobridge](https://jitsi.org/videobridge) to provide high quality, [secure](https://jitsi.org/security) and scalable video conferences. Jitsi Meet in action can be seen at [here at the session #482 of the VoIP Users Conference](http://youtu.be/7vFUVClsNh0).
The Jitsi Meet client runs in your browser, without installing anything else on your computer. You can try it out at https://meet.jit.si.
The Jitsi Meet client runs in your browser, without installing anything on your computer. You can try it out at https://meet.jit.si.
Jitsi Meet allows very efficient collaboration. Users can stream their desktop or only some windows. It also supports shared document editing with Etherpad.
Jitsi Meet allows for very efficient collaboration. Users can stream their desktop or only some windows. It also supports shared document editing with Etherpad.
**NOTE:** If you are looking for Jitsi as a Service (JaaS) please start [here](https://jaas.8x8.vc).
sed -i "s/Component \"focus.$JVB_HOSTNAME\"/Component \"focus.$JVB_HOSTNAME\" \"client_proxy\"\n target_address = \"$JICOFO_AUTH_USER@$JICOFO_AUTH_DOMAIN\"/g" $PROSODY_HOST_CONFIG
PROSODY_CONFIG_PRESENT="false"
fi
# Old versions of jitsi-meet-prosody come with the extra plugin path commented out (https://github.com/jitsi/jitsi-meet/commit/e11d4d3101e5228bf956a69a9e8da73d0aee7949)
# Make sure it is uncommented, as it contains required modules.
if [ ! -f /var/lib/prosody/$JVB_HOSTNAME.crt ]; then
# prosodyctl takes care for the permissions
# echo for using all default values
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