Changelog archive: 2025

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Overview

• daily-js
• react-native-daily-js
• daily-react
• daily-ios
• daily-android
• daily-python

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daily-js

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0.80.0

• Added support for setting permissions for SIP/PSTN users at join time.
• Removed the use of STUN from Xirsys.
• Updated dependencies to resolve security vulnerabilities.

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react-native-daily-js

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0.77.0

• Added support for setting permissions for SIP/PSTN users at join time.
• Fixed a build issue when using React Native’s New Architecture.
• Removed the use of STUN from Xirsys.
• Updated dependencies to resolve security vulnerabilities.

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daily-python

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0.19.3

• Added an option to ignore audio level when creating a custom audio track.

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0.19.2

• Fixed a segmentation fault that could occur on exit when a custom audio source was still writing data.

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Media Services

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Recording

• Fixed a bug where a failed session asset image URL could cause a recording composition to crash.

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Telephony

• Fixed a bug to correctly extract the displayName from custom SIP headers for dial-in participants.

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Overview

• daily-js
• react-native-daily-js
• daily-react
• daily-ios
• daily-android
• daily-python

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daily-js

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0.79.0

• Improved Krisp error handling.
• Removed a client-side limit on max_live_streams.
• Disabled Redux DevTools in production builds.

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react-native-daily-js

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0.76.0

• Removed a client-side limit on max_live_streams.
• Disabled Redux DevTools in production builds.

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daily-android

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0.31.0

• Added support for the canReceive permission to manage which media tracks remote participants can receive.
• Improved connection reliability by retrying signaling channel connections and using an alternate WebSocket URI format.
• Fixed a bug where custom ICE server configurations were not being applied.

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daily-ios

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0.31.0

• Added support for the canReceive permission to manage which media tracks remote participants can receive.
• Improved connection reliability by retrying signaling channel connections and using an alternate WebSocket URI format.

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daily-python

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0.19.1

• Fixed an audio renderer issue that caused garbled audio with remote stereo tracks.
• Added handling for meeting session ID changes that occur mid-call.

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0.19.0

• Introduced CustomAudioTrack for use as an additional custom track or as the main microphone input.
• Added support for displayName, userId, video, codecs, and permissions in start_dialout().
• Added sample_rate and callback_interval_ms arguments to set_audio_renderer() for more control over audio data.
• Updated add_custom_audio_track() and update_custom_audio_track() to accept a CustomAudioTrack instance.
• Switched to using system-level root certificates on all platforms.
• Fixed a bug where blocking virtual microphones could fail to send audio.

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0.18.2

• Fixed an installation issue on macOS versions older than 15.0.
• Corrected type hints for remove_custom_track().

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0.18.1

• Improved connection reliability by retrying signaling channel connections and using an alternate WebSocket URI format.
• Fixed a bug with virtual microphone completion callbacks.

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Media Services

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Recording

• Meeting move aware recording and streaming is now enabled by default. This feature (previously in pre-beta) automatically resumes recordings and live streams when meetings are transferred between servers during maintenance or failovers, eliminating gaps in recorded content.

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Telephony

• Introduced new webhooks to track call transfer status: calltransfer.triggered, calltransfer.initiated, calltransfer.answered, and calltransfer.complete.
• Added support for passing custom SIP headers during pinless dial-in, which are now available in the sipHeaders object in the webhook payload.
• Added canReceive permission controls for dial-in and dial-out participants via the startDialOut() API and room properties.
• On-demand SIP dial-out endpoints are now the default behavior, improving resource efficiency.

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Transcription

• Transcripts are now saved to storage every two minutes by default, improving data resilience.
• The /transcript and /transcripts API endpoints now include roomName, participantMinutes, and isVttAvailable for more detailed information.
• Fixed a race condition that could cause duplicate transcription messages when participant settings were updated rapidly.

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Overview

• daily-js
• react-native-daily-js
• daily-react
• daily-ios
• daily-android
• daily-python

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daily-js

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0.78.0

• Fixed ejection logic to prevent incorrect ejections or joins when a user’s local clock is out of sync. ⚠️ Note: Participants can no longer join if eject_after_elapsed is set to a negative value.
• Fixed an OverconstrainedError on Chrome that occurred when providing an invalid audio deviceId.
• Fixed an issue where unhandled errors during Krisp initialization could cause problems when joining a call.
• Reverted a change that disabled background effects on Windows when software WebGL is in use. Instead, a video-processor-warning is now logged, as this scenario can significantly reduce video frame rates.
• Improved networkState calculations by removing recvPacketLoss, which could be misleading.
• Added instanceId to webhooks for raw tracks recordings.
• Added performance logging for enumerateDevices() to help identify slow join times, especially on Firefox when the tab is not in focus.
• Removed the beta feature warning from testCallQuality().

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react-native-daily-js

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0.75.0

• Fixed ejection logic to prevent issues caused by mismatched local clock times. ⚠️ Note: Participants can no longer join if eject_after_elapsed is set to a negative value.
• Improved networkState calculation by removing recvPacketLoss from the formula.
• Added instanceId to webhooks for raw tracks recordings.
• Added performance logging for enumerateDevices() to help diagnose slow join times.

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daily-python

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0.18.0

• Introduced support for creating, updating, and removing custom audio tracks. You can now use CallClient.add_custom_audio_track(), CallClient.update_custom_audio_track(), and CallClient.remove_custom_audio_track() for advanced audio manipulation.
• Improved canReceive permission support to ensure track states ("off", offReasons) update correctly when permissions are granted or revoked.
• Audio and video renderer callbacks (set_audio_renderer(), set_video_renderer()) now receive the audio or video source as the last argument.
• Reduced “no subscription for consumer” and “Failed to close consumer” log messages from error to debug, as these often occur during normal participant disconnections.

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Media Services

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Recording

• Instance ID in Webhooks: The recording.ready-to-download and raw-tracks-recording-error webhooks now include an instance_id to uniquely identify raw tracks recording sessions.
• Fixed an issue to ensure the instanceId is correctly propagated for raw tracks recordings from the initial API call through to the final webhook event.

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Telephony

• dialin.ready Webhook: Introduced the dialin.ready webhook event, which notifies you when a room’s SIP endpoint is ready to receive an inbound call.
• origin Field in Dial-Out Webhooks: Dial-out webhook events now include an origin field, specifying the PSTN number or SIP URI that initiated the call.
• Default SIP Codec: Added a domain-level property to set default codecs for SIP/PSTN calls. Opus is now the default codec for all dial-in calls.
• On-Demand SIP Endpoints: SIP endpoints for dial-out are now created on-demand when startDialOut() is called, improving resource efficiency and room creation reliability.
• Error Messaging: Enhanced error messages for dial-in/out configurations during room creation.

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Overview

• daily-js
• react-native-daily-js
• daily-react
• daily-ios
• daily-android
• daily-python

​

daily-js

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0.77.0

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Introduced canReceive Permission

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Improved Network Quality Statistics

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Bug fixes and improvements

• Fixed proxyUrl support in dailyConfig.
• Optimized background effects on Windows by disabling them when hardware acceleration is unavailable.
• Updated internal dependencies to resolve security vulnerabilities.
• Simplified recordings_bucket response in room() method for enhanced security.

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0.76.0

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Bug fixes and performance improvements

• Updated Krisp noise cancellation library to v2.2.1.
• Fixed switching back from custom video track to device track via setInputDevicesAsync().
• Fixed iceTransportPolicy configuration to correctly override domain defaults.
• Added enable_dialout to TypeScript definitions for DailyRoomInfo.

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0.75.2

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Bug fixes

• Fixed unnecessary video track re-acquisition during input settings updates.

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0.75.1

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Bug fixes

• Fixed video constraint preservation when changing devices.
• Removed console error for null deviceId in setInputDevicesAsync.

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0.75.0

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Enhanced Participant Type for Dial-in/out

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Dynamic Input Track Constraints

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Bug fixes and improvements

• Fixed track event reliability for rapid muting/unmuting and custom tracks.
• Fixed track event participant info on leave.
• Improved Krisp initialization error handling.
• Fixed ondevicechange undefined error on certain systems.
• Updated Banuba SDK to eliminate need for 'unsafe-eval' CSP directive with avoidEval: true.
• Updated minimum supported version to 0.64.0.

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react-native-daily-js

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0.74.0

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Introduced canReceive Permission

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Improved Network Quality Statistics

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Bug fixes and improvements

• Added caller ID support for SIP PSTN transfers in sipCallTransfer().
• Fixed proxyUrl support.
• Updated dependencies for security fixes.
• Simplified recordings_bucket response in room() method.

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0.73.0

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Bug fixes

• Fixed switching from custom video track back to device track.
• Fixed iceTransportPolicy configuration override.
• Added enable_dialout to TypeScript definitions.

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0.72.1

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Reverted Erroneous API Changes

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0.72.0

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Enhanced Participant Type for Dial-in/out

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Added Audio-Only Raw Tracks Recording

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Bug fixes

• Fixed track event reliability for muting and custom tracks.
• Updated internal REST APIs to use native utilities.
• Added participant info to track-stopped events on leave.

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0.71.0

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Bug fixes and improvements

• Added sessionId to dial-in events.

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daily-android

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0.30.0

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Ignore Network Devices API

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Bug fixes

• Fixed ICE server configuration application via CallClient.setIceConfig().

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0.29.0

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Automatic Cloud Recording Start

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Bug fixes and improvements

• Fixed reconnection with expired meeting tokens.
• Updated dependencies (ffmpeg, libav, libvpx) for security patches.

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daily-ios

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0.30.0

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Bug fixes

• Fixed ICE server configuration application via setIceConfig().

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0.29.0

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Automatic Cloud Recording Start

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Bug fixes

• Fixed reconnection with expired meeting tokens.

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daily-python

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0.17.0

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Improved Linux Compatibility

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0.16.1

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Bug fixes

• Fixed virtual speaker device selection.

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0.16.0

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Update canReceive Permissions Support

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Bug fixes

• Fixed ICE server configuration application.

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0.15.0

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Automatic Cloud Recording Start

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Bug fixes

• Fixed reconnection with expired meeting tokens.

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Daily Prebuilt

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Transcription Language Defaults

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Media Services

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Recording and Live Streaming

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New Features

• Scale Mode Overrides: Added videoSettings.split.scaleModeOverrides for individual participant scaling in split layout compositions.
• Background Color Support: VCS render pipeline now supports custom background colors via backgroundColor parameter.

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Bug fixes

• Fixed audio-only participant filtering in baseline compositions.
• Fixed instance ID handling for raw tracks recording.

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Telephony

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Enhanced Caller ID Control

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API Improvements

• Domain dial-in configs endpoint now supports filtering by multiple phone numbers.
• Purchased phone numbers endpoint now includes pagination support.
• Room creation API provides more specific telephony provisioning error messages.

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Bug fixes and improvements

• Improved SIP INVITE handling to prevent signaling state errors.
• Enhanced error messages for invalid call transfers.
• Fixed layout updates for disconnected SIP participants.
• Added comprehensive transfer guide documentation.

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Transcription

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Bug fixes

• Fixed race condition when stopping or updating transcription immediately after starting.

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Overview

• daily-js
• react-native-daily-js
• daily-react
• daily-ios
• daily-android
• daily-python

​

daily-js

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0.74.0

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Updated Krisp CSP Requirements

script-src 'self' https://unpkg.com/@daily-co/daily-js blob:;

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Follow-up fix for Safari Audio Processing Detection

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Fixed Call Quality Test Error Handling

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Added Session ID to Dial-in Events

dialinConnected: { sessionId: 'abc-123', sipFrom: '+1234567890', ... }

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Version Support Update

• The oldest supported version of daily-js is now 0.63.0.

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react-native-daily-js

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0.71.0

• Added sessionId to dial-in events

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daily-android

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0.28.0

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Added support for Cloudflare TURN/STUN servers

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Fixed Android Proguard Build Configuration

• Added proper Proguard configuration for optional Banuba SDK dependencies
• Ensured R8 optimization doesn’t fail on missing video processor classes
• Maintained backwards compatibility with existing video processor integrations

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0.27.0

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Fixed crash when creating and deleting VideoTextureView/VideoTrackCapturerToSurface without also creating a CallClient.

• Moved native library initialization to shared context to prevent premature loading
• Added safety checks to prevent cleanup of uninitialized native contexts
• Improved synchronization of client tracking in shared context

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Enhanced Call Reliability during Meeting Moves

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daily-ios

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0.28.0

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Added support for Cloudflare TURN/STUN servers

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0.27.0

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Enhanced Call Reliability during Meeting Moves

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daily-python

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0.14.2

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Fixed type hints for CallClient.send_app_message()

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0.14.1

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Enhanced App Message Serialization

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0.14.0

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Added DTMF Support for Dial-out Calls

• Send numeric tones (0-9), symbols (#, *) and letters (A-D) to control interactive voice systems
• Control automated phone menus and voicemail systems during dial-out calls
• Specify session ID and tone sequences via simple settings object

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Added SIP REFER Support

• Transfer ongoing SIP dial-in calls to external SIP endpoints
• Control transfers using session ID and destination endpoint
• Support for standard SIP REFER mechanism

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Added Daily-to-Daily SIP Call Transfer Support

• Transfer SIP dial-in calls between Daily rooms
• Transfer dial-out calls to other SIP/PSTN endpoints
• Control transfers using session ID and destination endpoint

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Changed field name in Dial Events

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Enhanced Call Reliability during Meeting Moves

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Daily Prebuilt

• Enhanced accessibility across Prebuilt with ARIA attributes, improved keyboard navigation, semantic HTML, updated labels and roles, and better screen reader support to meet WCAG standards.
• Fixed issue where GIFs with empty titles were not rendered in chat.
• Fixed a case where flickering in chat input height occurred while typing.

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Media Services

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Recording and Live Streaming

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Support Multiple Recording Types with Multi-Instance Recording

// Start a cloud recording
await call.startRecording({
  instanceId: '123e4567-e89b-12d3-a456-426614174000',
  type: 'cloud',
  layout: {
    preset: 'default',
  },
});

// Start raw-tracks recording in parallel
await call.startRecording({
  instanceId: '987fdef0-c65b-11d1-b2e2-526614174000',
  type: 'raw-tracks',
});

• New type parameter in startRecording() (docs) API accepts 'cloud', 'raw-tracks', or 'local'
• Recording types must be enabled via enable_recording property on room/domain
• Each recording instance requires a unique UUID via instanceId parameter

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Added Audio-Only Support for Raw Tracks Recording

// Start a raw-tracks recording with audio only
await call.startRecording({
  instanceId: '123e4567-e89b-12d3-a456-426614174000',
  type: 'raw-tracks',
  layout: {
    preset: 'raw-tracks-audio-only',
  },
});

• New raw-tracks-audio-only preset captures only audio tracks, reducing storage requirements and simplifying post-processing workflows
• Preset must be specified at recording start - cannot be changed via updateRecording()
• Compatible with existing raw-tracks processing tools for audio extraction and analysis

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Added Raw Tracks Processing Tools

• New raw-tracks-tools CLI provides tools for analyzing, converting and compositing WebM files from raw-tracks recordings
• Supports track alignment, synchronization, and layout composition using Daily’s VCS engine
• Complements existing GUI tools like CloudConvert and Handbrake with more precise control over WebM processing

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Added Audio Recording Loss Detection

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Improved Recording Reliability with Network Optimizations

• Increased UDP buffer sizes to prevent packet loss between servers
• Added server-to-server retransmission between SFU and media workers
• Reduces recording freezes and dropped tracks in environments with network congestion

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Fixed Missing Tracks in Recordings

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Telephony

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Renamed Dialout Timeout Configuration Field

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Added Participant Type Support in participants() for Phone Participants

• New participant types: sip-dial-in, sip-dial-out, pstn-dial-in, pstn-dial-out
• Automatically set for all phone-based participants, including umbrella participants

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Fixed Dialout ICE Connection Reliability

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Added dialout_config to Room Configuration

• dialout_config.allow_room_start: Allow dialouts to start an empty room
• dialout_config.max_idle_timeout_sec: Control how long a dialout participant can remain in a room alone
• dialout_config.dialout_geo: Specify region for dialout sessions

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Transcription

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Unified Transcription Configuration Across Live and Batch Processing

// Live transcription
await call.startTranscription({
  language: 'fr-CA',
  model: 'nova-2-general',
});

• All configuration options from live transcription (language, model, punctuate, etc.) now supported in batch processing
• Previous undocumented params field maintained for backward compatibility
• Added new extra field for additional Deepgram parameters in both APIs

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Added Room Name in Transcription API

// List transcripts with room name filter
GET /api/transcripts?room_name=my-room*

// Transcript info now includes room name
{
 "transcriptId": "123",
 "roomId": "456",
 "roomName": "my-room",
 // ...other fields
}

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Webhooks

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New transcription webhook events

• transcript.started indicates transcription has begun, including metadata like room ID, instance ID and initial status
• transcript.ready-to-download fires when transcription completes, providing S3 location details for downloading the transcript
• transcript.error provides detailed error information and context if transcription fails during any stage

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Other

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Added support for Cloudflare TURN/STUN servers