Fix live caller audio latency and choppiness

- Reduce capture chunk from 4096 to 640 samples (256ms → 40ms)
- Replace BufferSource scheduling with AudioWorklet playback ring buffer
- Add 80ms jitter buffer with linear interpolation upsampling
- Reduce host mic and live caller stream blocksizes from 4096/2048 to 1024
- Replace librosa.resample with numpy interpolation in send_audio_to_caller

Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>

frontend/call-in.html

@@ -150,6 +150,6 @@
         </div>
     </div>
 
-    <script src="/js/call-in.js"></script>
+    <script src="/js/call-in.js?v=2"></script>
 </body>
 </html>