Instead of injecting research into every LLM call (which bloated prompts
and caused timeouts), do one quick SearXNG search when a caller is picked
up and add a relevant headline to their background. 3s timeout — if search
is slow, caller just doesn't reference news. Zero impact on live conversation.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Research results were bloating the system prompt, making LLM calls
slower and hitting the 20s timeout. Callers don't need news awareness
to have good conversations.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Use local SearXNG at localhost:8888 instead of Google News RSS
- No more 302 redirects or blocked requests — local is fast and reliable
- 5s timeout on all SearXNG requests
- Removed async locks (no contention needed for local service)
- Re-enabled research and headlines
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Google News RSS returns 302: add follow_redirects and User-Agent header
- Cache failed headline fetches for 5min so they don't retry every call
- Add 8s timeout on background research tasks
- Fix keyword extraction: skip short texts, require 2+ proper nouns (not names),
increase min word length to 6, add radio show filler to stop words
- Stops garbage searches like "Megan welcome" and "sounds thats youre"
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Add 30s timeout to all frontend fetch calls (safeFetch)
- Add 20s asyncio.timeout around lock+LLM in chat, ai-respond, auto-respond
- Reduce OpenRouter timeout from 60s to 25s
- Reduce Inworld TTS timeout from 60s to 25s
- Return graceful fallback responses on timeout instead of hanging
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Add ads playback system with backend endpoints and frontend UI
- Diversify AI callers: randomize voices per session, expand jobs/problems/interests/quirks/locations
- Update website tagline and descriptions to "biologically questionable organisms"
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
When off air, callers hear a message and get disconnected. When on
air, calls route normally. Toggle button added to frontend header
with pulsing red ON AIR indicator.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Add streamSid tracking and per-caller send locks for SignalWire
- Improve TTS streaming with real-time pacing and detailed logging
- Block host audio to caller during TTS playback
- Randomize caller names between sessions from name pools
- Update page title and show phone number in UI
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Replace aggressive sentence-count limiting with ensure_complete_thought()
which only trims if the LLM was actually cut off mid-sentence
- Softer prompt guidance for natural brevity instead of rigid sentence count
- max_tokens at 100 as natural length cap
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- max_tokens back to 150 so LLM can finish thoughts
- New limit_sentences() keeps only first 2 complete sentences
- Never cuts mid-sentence — always ends at punctuation
- Applied to both chat and auto-respond paths
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Reduce max_tokens from 100 to 75 for shorter output
- Add truncate_to_complete_sentence() to trim at last punctuation
- Applied to both chat and auto-respond paths
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Much stronger prompt language: "no more than 2 sentences EVER"
- Added "DO NOT ramble" instruction
- Reduced max_tokens back to 100 as hard limit
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Increase max_tokens from 100 to 150 to avoid mid-sentence truncation
- Tighten prompt to 1-2 short sentences with emphasis on completing them
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Mute host mic forwarding while TTS is streaming to prevent interleaving
both audio sources into the same playback buffer
- Replace nearest-neighbor downsampling with box-filter averaging on both
server (host mic) and browser (caller mic) for anti-aliased resampling
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
TTS audio was sent as a single huge WebSocket frame that overflowed the
browser's 3s ring buffer. Now streams in 60ms chunks at real-time rate.
Also increased browser ring buffer from 3s to 10s as safety net.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
The auto-respond function played AI TTS to the local Loopback channel
but didn't send it over WebSocket to the live caller in the browser.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Replace per-callback async task spawning with persistent queue-based sender
- Buffer host mic to 60ms chunks (was 21ms) to reduce WebSocket frame rate
- Reduce server ring buffer prebuffer from 150ms to 80ms
- Reduce browser playback jitter buffer from 150ms to 100ms
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Add CallRecord dataclass for tracking call history
- Extend Session with call_history, active_real_caller, ai_respond_mode, auto_followup
- Add get_show_history() for AI caller context about earlier calls
- Update get_conversation_summary() to handle host/real_caller/ai_caller roles
- Update reset() to clear new fields
- Add tests for all new functionality
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
