How to sync dual-channel transcripts via OpenAI Whisper (VAD silence stripping destroys absolute timestamps)

Well honestly your split-channel approach is actually pretty solid already the timestamps/speaker reconstruction logic looks mostly correct to me.

The bigger issue feels more like Whisper struggling with

• narrowband 8kHz telephony audio

• overlapping speech/interruption handling

• and context reconstruction across independently transcribed channels.

A few things stand out from your output tho

• “Policy Test” instead of “caller side test”

• numbers normalized weirdly

• sentence continuation split oddly across timestamps

• interruption timing drift around the “blue” overlap

So that usually looks more like ASR inference limitations than AGI/Asterisk sync problems.

One thing I’d seriously test

• upsample audio to 16kHz before transcription (sox or ffmpeg)

• even though no new information is created, Whisper tends to behave noticeably better on resampled telephony audio.

Also maybe try

• adding small silence padding at start of both legs before transcription

• forcing shorter segments/VAD chunking

• aligning merged segments by midpoint timestamps instead of raw start times only.

Hmm.. Your actual synchronization pipeline honestly seems cleaner than most PBX transcription setups I’ve ever seen