v0.30.0

Added

• Added automatic fallback to dailywebrtc.com and dailywebrtc.net when daily.co authoritative nameservers are unreachable, improving connection resilience.

Fixed

• Fixed a use-after-free that could cause a segmentation fault in the audio playout thread (e.g. daily-speaker-p) if a CallClient was released (or garbage collected) while still in a call (i.e. without leaving first) and participant audio renderers were registered.

• Fixed a segmentation fault that could occur when a CallClient was released (or garbage collected) while still in a call, especially right before the application exited: the internal media transport teardown now completes before release() returns, and WebRTC internals are no longer torn down at process exit while SDK threads may still be running.

  Note that the recommended way to terminate a call is still to leave() (and wait for its completion) before calling release().

v0.29.1

Fixed

• Fixed an issue where a room configured with enable_recording set to cloud-audio-only was ignored when starting a recording, causing the recording to also capture video.

• Fixed an issue where StartRecordingProperties none properties were serialized to null.

v0.29.0

Added

• Added support for screenAudio input. It is now possible to send a custom screen-share audio track (alongside or independently of screenVideo) with:

{
  "inputs": {
      "screenAudio": {
          "isEnabled": True,
          "settings": {
             "customTrack": {
                 "id": "CUSTOM_TRACK_ID"
             }
          }
      }
  }
}

v0.28.1

Fixed

• Fixed a segmentation fault that could occur after virtual microphone or speaker inputs or publishing updates.

v0.28.0

Added

• Added support for screenVideo custom tracks. It is now possible to send a custom screen share track with:

{
  "inputs": {
      "screenVideo": {
          "isEnabled": True,
          "settings": {
             "customTrack": {
                 "id": "CUSTOM_TRACK_ID"
             }
          }
      }
  }
}

Changed

• Adaptive Bitrate (ABR) is now enabled by default for camera tracks. It can still be disabled by setting allow_adaptive_layers to False in the camera send_settings.

Fixed

• Fixed panics in signalling reconnect paths when room lookup returns no worker.

• Fixed support for cloud-audio-only recording type.

Performance

• Added per-phase connection timing metrics for WebSocket signalling (DNS lookup, TCP connect, TLS handshake, and WebSocket upgrade), complementing the existing HTTP connection timings.

• Replaced ureq HTTP client with hyper + rustls (ring backend). Includes connection timeouts, TLS configuration reuse, and per-request timing metrics for DNS lookup, TCP connect, and TLS handshake.

v0.27.0

Added

• Added auto_silence parameter to CustomAudioSource. When set to True (default), silence is automatically sent when no audio frames are being written. When set to False, no audio is sent while the source is idle.

v0.26.0

Added

• Added debug logs to HTTP requests.

v0.25.0

Added

• Added support for dtmf-event.

• Added provider field to dialout properties. Currently only supports "daily".

• Added extension and waitBeforeExtensionDialSec fields to dialout properties.

• Added optional method field to send DTMF properties. Supported values are "sip-info", "telephone-event", and "auto".

• Added optional digitDurationMs field to send DTMF properties.

v0.24.0

Added

• Added a new CustomVideoTrack. This new track can be used as an additional custom video track (i.e. with custom names) or as the main camera track.

video_source = CustomVideoSource(width=1280, height=720, color_format="RGBA")

video_track = CustomVideoTrack(video_source)

client.join("YOUR_DAILY_ROOM", client_settings={
    "inputs": {
        "camera": {
            "isEnabled": True,
            "settings": {
               "customTrack": {
                   "id": video_track.id
               }
            }
        }
    }
})

# Write video frames
frame = bytes([255, 0, 0, 255] * 1280 * 720)  # RGBA
video_source.write_frame(frame)

Changed

• Updated to PyO3 0.28.2.

Fixed

• Avoid unnecessary signalling reconnection attempts when leave() is called during a network failure.

v0.23.0

Added

• Added a trackType field to the transcription-message event to indicate which track the transcription originated from.

Changed

• Updated to PyO3 0.27.2.

• Updated to maturin 1.10.2.

• Updated libc to 0.2.178.