diff --git a/src/langsmith/trace-with-pipecat.mdx b/src/langsmith/trace-with-pipecat.mdx
index 0058a1d421..f50af9572e 100644
--- a/src/langsmith/trace-with-pipecat.mdx
+++ b/src/langsmith/trace-with-pipecat.mdx
@@ -14,7 +14,7 @@ Trace your [Pipecat](https://pipecat.ai/) voice agents to LangSmith with the Lan
 The Pipecat integration requires `langsmith[pipecat]>=0.9.7`.
 </Note>
 
-The integration hooks into the spans Pipecat already emits and maps them onto LangSmith's tracing format, so each conversation becomes a single LangSmith trace, with a span per pipeline stage (STT, LLM, TTS).
+The integration hooks into the spans Pipecat already emits and maps them onto LangSmith's tracing format, so each conversation becomes a single LangSmith trace, with a span per pipeline stage (STT, LLM, TTS). This covers both the STT/LLM/TTS cascade and speech-to-speech (realtime) models. Realtime models (for example, `OpenAIRealtimeLLMService`) need one extra call to capture the user's transcript. See [When using Pipecat with a realtime model](#when-using-pipecat-with-a-realtime-model).
 
 ## Install
 
@@ -95,6 +95,59 @@ configure_pipecat()
 set_thread_id(conversation_id)
 ```
 
+## When using Pipecat with a realtime model
+
+With a speech-to-speech (realtime) model there is no separate speech-to-text stage, so the user's transcript is never emitted as an OTel span. Instead it arrives through the user context aggregator's `on_user_turn_message_added` callback, which Pipecat fires once it has the finalized user text. Without wiring it up, the trace shows only the assistant side.
+
+Call `instrument_user_aggregator` once, right after building the context aggregator, so the SDK subscribes to that callback for you and pairs each transcript with its turn. It correlates by the id you pass to `set_thread_id`, so set that first and pass the same id:
+
+<Note>
+`instrument_user_aggregator` requires `langsmith[pipecat]>=0.10.6`.
+</Note>
+
+```python
+from langsmith.integrations.pipecat import configure_pipecat, set_thread_id
+from pipecat.processors.aggregators.llm_context import LLMContext
+from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
+from pipecat.services.openai.realtime.events import (
+    AudioConfiguration,
+    AudioInput,
+    AudioOutput,
+    InputAudioTranscription,
+    SessionProperties,
+)
+from pipecat.services.openai.realtime.llm import OpenAIRealtimeLLMService
+
+conversation_id = "..."  # any id that identifies the conversation
+span_processor = configure_pipecat()
+set_thread_id(conversation_id)
+
+llm = OpenAIRealtimeLLMService(
+    api_key=openai_api_key,
+    settings=OpenAIRealtimeLLMService.Settings(
+        model="gpt-realtime",
+        session_properties=SessionProperties(
+            # Enable input-audio transcription: OpenAI Realtime sends the model
+            # raw audio and, by default, produces no user-side text. Without this
+            # the context aggregator never fires, so the user's turns never reach
+            # the trace.
+            audio=AudioConfiguration(
+                input=AudioInput(transcription=InputAudioTranscription()),
+                output=AudioOutput(voice="marin"),
+            ),
+        ),
+    ),
+)
+
+context = LLMContext(messages=[{"role": "system", "content": "..."}])
+context_aggregator = LLMContextAggregatorPair(context, realtime_service_mode=True)
+span_processor.instrument_user_aggregator(context_aggregator, conversation_id)  # capture the user transcript
+```
+
+For OpenAI Realtime, also enable input-audio transcription (`InputAudioTranscription`) on the session; otherwise the model receives raw audio and produces no user-side text, so the aggregator never fires. Other realtime services that surface the user's text through the aggregator themselves (for example, Gemini Live) need only the `instrument_user_aggregator` call, with no extra session configuration.
+
+Only call `instrument_user_aggregator` for realtime models. In the STT/LLM/TTS cascade the transcript is already captured (from the speech-to-text stage), so calling it there would record the user's turns a second time.
+
 ## Record the conversation audio
 
 Attach the conversation audio to the trace using Pipecat's [`AudioBufferProcessor`](https://docs.pipecat.ai/server/utilities/audio/audio-recording). Place it after `transport.output()` so it captures what was actually played (after any barge-in truncation), hand it to the integration, and start it once the session is running: