* Add Inworld Realtime LLM service
Adds a WebSocket-based realtime service for Inworld's cascade
STT/LLM/TTS API with semantic VAD, function calling, and streaming
transcription support.
New files:
- src/pipecat/services/inworld/realtime/ (service, events)
- src/pipecat/adapters/services/inworld_realtime_adapter.py
- examples/foundational/19zb-inworld-realtime.py
Also includes:
- websockets dependency for inworld extra in pyproject.toml
- Adapter and settings tests matching OpenAI/Grok realtime patterns
- Fix for double-response when server-side VAD is enabled
* Prefer init-provided system instruction in Inworld Realtime
Adopt _resolve_system_instruction() from BaseLLMAdapter, matching the
pattern applied to OpenAI Realtime, Grok Realtime, Gemini Live, and
Nova Sonic in the pk/realtime-services-init-v-context-system-instructions-cleanup
branch.
* Update changelog entry with PR number
* Fix changelog format to use bullet point
* Polish PR: default model, example cleanup, changelog update
- Change default model from gpt-4.1-nano to gpt-4.1-mini
- Add function calling demo to example
- Remove demo-testing artifact from system instruction
- Mention Router support in changelog
* Address PR review feedback for Inworld Realtime
- Move example to examples/realtime/realtime-inworld.py
- Change initial context role from "user" to "developer"
- Remove explicit sample rates from example; sync them in
_ensure_audio_config so Inworld gets the transport's actual rates
- Add audio race condition guard in _handle_evt_audio_delta (matches
OpenAI realtime pattern)
- Convert remaining "system"/"developer" messages to "user" in adapter
- Add clarifying comment for local-VAD vs server-VAD metrics paths
* Simplify example, add provider tracking, remove local VAD path
- Remove function calling from example, switch model to xai/grok-4-1-fast-non-reasoning
- Add pipecat-realtime session key prefix and provider_data metadata
for Inworld traffic attribution
- Remove local VAD code path (Inworld only supports server-side VAD)
- Use typed InputAudioBufferAppendEvent for audio sends
* Default TTS model to inworld-tts-1.5-max
* Remove dead shimmed tools code, set STT/VAD defaults
- Remove non-functional AdapterType.SHIM custom tools code from adapter
- Default STT model to assemblyai/u3-rt-pro
- Default VAD eagerness to low
Remove the deprecation proxy infrastructure that allowed old-style flat
imports (e.g. `from pipecat.services.openai import OpenAILLMService`).
Users must now import from specific submodules
(`from pipecat.services.openai.llm import OpenAILLMService`), which is
already the established pattern across all internal code and 179+ examples.
- Strip 32 proxy `__init__.py` files to empty
- Strip 3 non-proxy files with bare star imports (minimax, sambanova, sarvam)
- Strip google/gemini_live `__init__.py` re-exports
- Remove DeprecatedModuleProxy class and helpers from services/__init__.py
- Remove ruff per-file ignore for services/__init__.py
- Fix 2 examples using old-style imports
Napoleon's Attributes section creates class-level attribute docs that
duplicate the __init__ parameter docs when napoleon_include_init_with_doc
is enabled. Using Parameters avoids the duplication.
Remove the deprecated text_aggregator parameter from TTSService,
CartesiaTTSService, and RimeTTSService, and the deprecated text_filter
parameter from TTSService. Users should use LLMTextProcessor before
the TTS service instead. Update the voice-switching example to use
LLMTextProcessor with PatternPairAggregator.
Add SmallestSTTService using the Pulse WebSocket API for real-time
transcription. Includes SmallestSTTSettings dataclass, 32-language
support with resolve_language fallback, VAD-driven finalize signal,
and SMALLEST_TTFS_P99 latency constant.
Also adds X-Source and X-Pipecat-Version headers to Smallest STT
and TTS WebSocket connections.
Example files like openai.py shadow installed packages when Python adds the
script directory to sys.path. Prepend the parent folder name to each example
file (e.g. openai.py -> function-calling-openai.py). Also split
thinking-and-mcp/ into separate mcp/ and thinking/ directories.
Replace the nested services/speech/ and services/function-calling/ with
top-level voice/ and function-calling/ directories. Update eval script
paths and README to match.
Move 304 examples from a flat numbered directory into 14 descriptive
subfolders: getting-started, services (speech + function-calling),
transcription, vision, realtime, persistent-context,
context-summarization, update-settings (stt/tts/llm), turn-management,
thinking-and-mcp, transports, video-avatar, video-processing, and
features.
Strip numbered prefixes from filenames (e.g. 07c-interruptible-deepgram.py
becomes services/speech/deepgram.py) since the folder context makes them
redundant. Keep numbered prefixes only in getting-started/ where ordering
matters.
Update eval script paths and README to match the new structure.
Add detailed trace-level logging to _apply_previous_response_optimization
showing why the optimization was applied or fell back to full context,
including the relevant data for debugging.
Use append_to_context=False for the filler TTSSpeakFrame in the
function-calling example to avoid altering the conversation history
and breaking the previous_response_id prefix match.
Introduce a WebSocket variant of the OpenAI Responses API service that
maintains a persistent connection to wss://api.openai.com/v1/responses
for lower-latency inference. The WebSocket variant automatically uses
previous_response_id to send only incremental context when possible,
falling back to full context on reconnection or cache miss.
The WebSocket variant becomes the new default OpenAIResponsesLLMService,
and the HTTP variant is renamed to OpenAIResponsesHttpLLMService. Both
share a private base class with common settings, parameter building,
and run_inference (always HTTP) logic.
OpenPipe was acquired by CoreWeave in September 2025. The Python package
hasn't been updated since June 2025 and the repo since 2024. The openpipe
package caps openai<=1.97.1, creating dependency conflicts with other
extras. Remove the dead integration to clean up the codebase.
- Add Nebius LLM service wrapping OpenAI-compatible Token Factory API
- Set supports_developer_role = False (Nebius rejects developer role)
- Default to openai/gpt-oss-120b model (supports function calling)
- Add Nebius function-calling example and env.example entry
- Fix Sarvam developer role support
- Update examples to use developer role for intro messages
Gemini 3.1 Flash Live won't reliably report ending its turn until
after it says something following a tool call. Restructure the system
instruction so the model says goodbye *after* calling
end_conversation, and add a comment explaining the deferred EndFrame
behavior that makes this work.
All recent Gemini Live models (including the default
gemini-2.5-flash-native-audio-preview-12-2025, and going at least as
far back as gemini-2.5-flash-native-audio-preview-09-2025) only
support AUDIO as a response modality. We considered using
`modalities=TEXT` as a Pipecat-level signal to suppress audio output
frames (so developers could pair Gemini Live with an external TTS),
but the output transcription from the API arrives too late relative
to the audio to be useful for driving an external TTS service.
For now, just log a warning when a TEXT modality is configured
(at init or via set_model_modalities) and proceed as normal. The 26d
text-modality example is removed since it no longer represents a
viable configuration.
