diff --git a/src/engine/engine_config.h b/src/engine/engine_config.h
index 3433e1e1..9f5be0b6 100644
--- a/src/engine/engine_config.h
+++ b/src/engine/engine_config.h
@@ -27,8 +27,10 @@
 #define AUDIO_MIX_CHANNEL_COUNT 16
 #define AUDIO_DEVICEPAIR_INPUTBUFFER 16384
 
-// Avoid too high resampler quality - it can take many CPU and cause gaps in playing
-#define AUDIO_RESAMPLER_QUALITY 1
+// Resampler quality on the Speex 0..10 scale. Quality 1 upsampling (e.g. G.711 8k -> 48k)
+// adds audible aliasing and hurts PVQA/AQuA scores; 7 is a good fidelity/CPU balance for a
+// single mono stream on modern ARM64. Raise toward 10 for max fidelity if CPU allows.
+#define AUDIO_RESAMPLER_QUALITY 7
 #define AEC_FRAME_TIME 10
 #define AEC_TAIL_TIME 160
 
diff --git a/src/engine/media/MT_AudioReceiver.cpp b/src/engine/media/MT_AudioReceiver.cpp
index fd7d435f..86cfc2b7 100644
--- a/src/engine/media/MT_AudioReceiver.cpp
+++ b/src/engine/media/MT_AudioReceiver.cpp
@@ -381,6 +381,12 @@ AudioReceiver::AudioReceiver(const CodecList::Settings& settings, MT::Statistics
     // Avoid collecting too much data
     mRtpBuffer.setHigh(240ms);
 
+    // Keep a non-zero low-water cushion. With low-water at 0 the buffer drains all the way to
+    // empty before reacting, so ordinary clock drift / jitter periodically starves the decoder
+    // and forces silence/PLC insertion (heard as periodic artifacts, uncounted by rtp_lost /
+    // rtp_dropped). Holding ~60ms lets a late packet arrive before the buffer underruns.
+    mRtpBuffer.setLow(60ms);
+
     // Resamplers are lazy inside; there is no actual memory allocation
     mResampler8.start(AUDIO_CHANNELS, 8000, AUDIO_SAMPLERATE);
     mResampler16.start(AUDIO_CHANNELS, 16000, AUDIO_SAMPLERATE);