The AudioHook protocol requires every message to carry a `parameters`
object. `_create_message` conditionally included it only when parameters
were truthy, so pong responses and closed responses without
outputVariables were sent without the field.
Clients that validate message structure (including the Genesys reference
implementation) rejected these messages, which broke server sequence
tracking and prevented outputVariables from reaching the Architect flow.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Align with the OpenTelemetry GenAI semantic convention
gen_ai.system_instructions for system prompts. The old "system"
attribute name was unrelated to gen_ai.system (which is for
provider name).
Replace adapter-based extraction in traced_llm with direct reads from
_settings.system_instruction (priority) and context messages (fallback).
The old approach had three bugs: signature mismatch with Anthropic
adapter, key name inconsistency, and unnecessary overhead from full
message/tools conversion.
Also deduplicate the system instruction in spans -- it was appearing as
both "system" and "param.system_instruction".
Adds a FrameOrder enum with ARRIVAL (default, existing behavior) and
PIPELINE (pushes frames in pipeline definition order). This lets callers
guarantee output ordering between parallel pipelines — e.g. ensuring
image frames precede audio frames — without needing a separate reordering
processor downstream.
Updates the 05-sync-speech-and-image example to use FrameOrder.PIPELINE,
removing the ImageBeforeAudioReorderer class entirely.
- Add WakePhraseUserTurnStartStrategy for gating interaction behind wake
phrase detection, with timeout and single_activation modes
- Add default_user_turn_start_strategies() and
default_user_turn_stop_strategies() helper functions
- Deprecate WakeCheckFilter in favor of the new strategy
- Extend ProcessFrameResult to stop strategies for short-circuit evaluation
- Fix MinWordsUserTurnStartStrategy including filtered text in output
* Fix empty user transcription causing spurious interruption in Nova Sonic
Skip _report_user_transcription_ended() when _user_text_buffer is empty,
which happens when the initial prompt is text-only. Previously, an empty
TranscriptionFrame was pushed upstream, triggering a chain reaction:
on_user_turn_stopped → UserStartedSpeakingFrame → interruption →
premature BotStoppedSpeaking → multiple response start/stop cycles.
* Improve TextFrame and assistant end of turn logic
Now, SPECULATIVE text results are used to push the LLMTextFrame,
AggregatedTextFrame, and TTSTextFrame. Additionally, the TTSTextFrames
are push at the end of the corresponding audio segment.
* Remove BotStoppedSpeakingFrame fallback from Nova Sonic
Now that assistant response end is detected directly from Nova Sonic
contentEnd events (END_TURN and INTERRUPTED), the BotStoppedSpeakingFrame
handler is no longer needed. Inline the cleanup logic in reset_conversation.
* Remove duplicate reconnection logic from Gradium STT
The _receive_messages method had its own while-True reconnect loop,
duplicating the reconnection handling already provided by
WebsocketService._receive_task_handler (exponential backoff, max
retries, error reporting). Flatten to just the inner message loop
and let the base class handle reconnection.
* Align Gradium STT VAD handling with base class patterns
Replace the process_frame override with a _handle_vad_user_stopped_speaking
override, which is the proper hook provided by STTService. Move
start_processing_metrics() into run_stt (matching Gladia's pattern).
Remove unused FrameDirection and VADUserStartedSpeakingFrame imports.
* Add transcript aggregation delay after flushed to capture trailing tokens
Gradium flushed response can arrive before all text tokens have been
delivered. Instead of finalizing immediately on flushed, start a short
timer (100ms) that allows trailing tokens to accumulate before pushing
the final TranscriptionFrame.
* Add changelog for PR #4066
* Change default encoding to pcm_16000
* Decouple encoding from sample_rate in Gradium STT
The encoding parameter now takes just the base type (pcm, wav, opus)
and the sample rate is derived from the pipeline audio_in_sample_rate,
assembled dynamically via input_format_from_encoding(). This fixes the
mismatch where SAMPLE_RATE=24000 was passed to the base class while
encoding defaulted to pcm_16000.
List-valued settings like keyterm, keywords, search, redact, and replace
were being converted to strings before being passed to the SDK connect()
method. The SDK expects lists so its encode_query can produce repeated
query params (keyterm=a&keyterm=b).
Expose enable_dialout as a configure() parameter (default False) so
dial-out examples can opt in without needing to build DailyRoomProperties
manually.
Narrow misleading Optional type hints on parameters that never accept
None, extract the duplicated token_exp_duration * 60 * 60 calculation,
remove unnecessary forward-reference quotes on DailyMeetingTokenProperties,
and clarify why enable_dialout is explicitly set to False.
Refactor language_to_soniox_language to use resolve_language + LANGUAGE_MAP
pattern consistent with other services. Fix resolve_language fallback to use
str(language) instead of language.value so plain strings don't crash.
