When participants panel is open and we approve a participant to unmute, the notification was not hidden as we were not correctly updating the state. We were expecting a participant object, but an id of the participant was used.
* fix(ConnectionQuality): Do not show red/yellow GSM bars on join. When the user first unmutes their video, the connection quality is shown as poor until the local stats are available. Calculate the connection quality only after the stats are available, i.e., assume 100% until pcStatsInterval has elapsed.
* feat(non-participant-messages) Add a new JitiConferenceEvent for messages ignored by ENDPOINT_MESSAGE_RECEIVED
* fix(precall) respect custom callstats script url for precall test
9e632a77c5...6a3df11ffa
* feat(BridgeChannel): Signal a new videoType for high fps screenshare. This lets the bridge adjust the bitrate allocation for this source so that layers with higher fps are prioritized over layers with higher resolution. As a result, endpoints with restricted downlink will receive a high fps low resolution share as opposed to a high resolution low fps screenshare.
* fix(log) lower severity of overly verbose logs (2)
fa834c2923...9e632a77c5
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)
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.
* 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
* 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
* 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
* 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
* 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
* fix(JingleSessionPC): Disable unified-plan for p2p. Disable cross browser p2p using unified plan until all the issues are fixed.
0993c8e93d...5738c80baf
* 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
* 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
* 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
* 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
Andrei Gavrilescu
Дамян Минков
Jaya Allamsetty
Saúl Ibarra Corretgé
Avram Tudor
Calin Chitu
Hristo Terezov
hmuresan
Steffen Kolmer
Tudor-Ovidiu Avram
dependabot[bot]