Releases: daily-co/react-native-daily-js
0.84.1
0.84.0
Features
- Drastically reduced initial call-machine bundle download size (~2MB smaller), dynamically loading larger portions as-needed.
Other Improvements
- New
digitDurationMsoption insendDTMF()to allow configurable digit duration. - Updated dependencies to remove vulnerabilities.
- Added dashboard logging for dialin/dialout events
Bugfixes
- Removed client-side 3 hour limit for raw-tracks recordings. Limits are handled by the server.
- Improve system idleness check to avoid signaling reconnect retries after 5 minutes of being idle, resulting in bad sigauthz errors. Calls should leave cleanly in this scenario.
Breaking Change
- If you currently host the call-machine-bundle yourself using the advanced
callObjectBundleUrlOverrideconfig, please reach out to help@daily.co for upgrade instructions. You will need a new, second bundle for Krisp noise cancellation to work.
0.83.0
react-native-daily-js 0.83.0
Features
-
Introduced a new option to
startRecording()Β calledΒdataOutputsto enable recording of events that occur during the meeting in addition to the actual recording media. This configuration option takes an array of auxiliary output types. For each type specified, an extra file will be provided with the recording download link (accessible via the REST API). When using a custom S3 bucket, the data output files are written to the same bucket as the AV media recording. These data outputs are timecode-aligned with the recording media, so they can be used in post-processing workflows.Currently three types are supported:Β
event-json,transcript-webvtt,chat-webvttevent-jsonβ produces a JSON that describes all relevant events during the recording session, e.g. when a recording media file started, track updates, etc.transcript-webvttβ produces a WebVTT file with all transcription events. The timecode of this file starts with the recording media (the mp4 or m4a file).- A live transcription needs to be active for the meeting to get these events.
chat-webvttβ produces a WebVTT file with all chat events. The timecode of this file starts with the recording media (the mp4 or m4a file).
The user can pass any combination of options to to capture all events, live transcription, and/or chat messages in separate files during a cloud recording.
Note that these outputs are capturing events, not starting services. If there's no active transcription in the room, there won't be any transcription written to the file. (In other words, passingΒ
dataOutputs: ['transcript-webvtt']Β doesn't start the transcription, it just configures the recording session to also capture transcription events.) -
Added support for new
cloud-audio-onlyrecording type.- This produces m4a files with the
audio/mp4MIME content type on S3. - Can be configured together with other existing recording types.
- Doesnβt share an instance id with RTMP/HLS streaming like regular cloud recording does (because streaming plain audio isnβt typically useful, as many streaming platforms require a video track even if itβs black).
- This produces m4a files with the
Bugfixes
- Fixed issue where sessions will reconnect and try, often successfully, to resume after long periods of idleness when the laptop lid is closed. Now, after 5 minutes of inactivity is detected (on lid open), the session will automatically disconnect.
- Fixed uncaught errors surrounding
join()andstartCamera()when no room url is provided, or the url is invalid. These will now throw and error andjoin()will leave themeetingState()in"left-meeting"to allow trying again.
Other improvements
- Improved the cpu-load-change event by requiring multiple consecutive high CPU readings before reporting a high load and by refining the global decode time calculation to exclude videos with empty frame metrics.
- Added
max_app_message_sizetoDailyRoomInfotypes - Added missing type definitions for streaming and recordings
0.82.0
Bugfixes
- Fixed an issue where the initial send settings were not being respected for cam video.
0.81.0
Features
- Added support for Android 16KB page size.
- Added support for calling phone extensions during dialout operations, allowing users to specify an extension number and wait time before dialing the extension after the main call connects.
Other improvements
- Improved reliability of handling auto-start recording/transcription options when joining room.
0.80.0
Improvements
- Updated mediasoup-client dependency to latest
- Removed deprecated and unused support for
rtp-tracksrecording
0.79.0
Improvements
- Updated dependencies to resolve security vulnerabilities
0.78.0
Features
- Added a new
dialin-errortype:start-failed.
Bugfixes
- Preventing potential null pointer exception in
DailyNativeUtilswhen the module is destroyed before being initialized. For example, if JS never accessed it.
Other improvements
- Updated
enumerateDevices()to report errors only when significant, and added improved logging to trackenumerateDevicestiming.
0.77.0
Features
- Added support for join time permissions for the SIP/PSTN user.
Bugfixes
- Fixed an issue that prevented building
react-native-daily-jswhen the React Native New Architecture wasenabled.
Other improvements
- Removed the use of STUN from Xirsys.
- Updated packages to address security vulnerabilities.
0.76.0
Improvements
- Removed a client-side limit on max_live_streams
- Disabled redux devtools