fix: web bridge low-latency config — disable silence suppression, reduce jitter buffer

PTT mode was causing delayed/lost audio because:
1. Silence suppression ate the start of speech after PTT release
2. Jitter buffer target depth was too high for interactive use

Web bridge now uses:
- suppression_enabled: false (PTT handles silence at browser level)
- jitter_target: 3 (60ms vs ~1s default)
- jitter_max: 20 (400ms cap)
- jitter_min: 1 (start playing after 20ms)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>

crates/wzp-web/src/main.rs

@@ -279,7 +279,15 @@ async fn handle_ws(socket: WebSocket, room: String, state: AppState) {
         }
     }
 
-    let config = CallConfig::default();
+    // Web bridge config: low latency for PTT, disable silence suppression
+    // (PTT handles silence at the browser level, no need to suppress here)
+    let config = CallConfig {
+        suppression_enabled: false,
+        jitter_target: 3,   // 60ms instead of default (~1s)
+        jitter_max: 20,     // 400ms cap
+        jitter_min: 1,      // start playing after 20ms
+        ..CallConfig::default()
+    };
     let encoder = Arc::new(Mutex::new(CallEncoder::new(&config)));
     let decoder = Arc::new(Mutex::new(CallDecoder::new(&config)));