diff --git a/plugins/rtp/CMakeLists.txt b/plugins/rtp/CMakeLists.txt
index b19c8a8f..52419425 100644
--- a/plugins/rtp/CMakeLists.txt
+++ b/plugins/rtp/CMakeLists.txt
@@ -17,7 +17,7 @@ if(Gst_VERSION VERSION_GREATER "1.16")
 endif()
 
 if(WebRTCAudioProcessing_VERSION GREATER "0.4")
-    message(WARNING "Ignoring WebRTCAudioProcessing, only versions < 0.4 supported so far")
+    message(STATUS "Ignoring WebRTCAudioProcessing, only versions < 0.4 supported so far")
     unset(WebRTCAudioProcessing_FOUND)
 endif()
 
@@ -25,8 +25,9 @@ if(WebRTCAudioProcessing_FOUND)
     set(RTP_DEFINITIONS ${RTP_DEFINITIONS} WITH_VOICE_PROCESSOR)
     set(RTP_VOICE_PROCESSOR_VALA src/voice_processor.vala)
     set(RTP_VOICE_PROCESSOR_CXX src/voice_processor_native.cpp)
+    set(RTP_VOICE_PROCESSOR_LIB webrtc-audio-processing)
 else()
-    message(WARNING "WebRTCAudioProcessing not found, build without voice pre-processing!")
+    message(STATUS "WebRTCAudioProcessing not found, build without voice pre-processing!")
 endif()
 
 vala_precompile(RTP_VALA_C
@@ -53,7 +54,7 @@ DEFINITIONS
 
 add_definitions(${VALA_CFLAGS} -DG_LOG_DOMAIN="rtp" -I${CMAKE_CURRENT_SOURCE_DIR}/src)
 add_library(rtp SHARED ${RTP_VALA_C} ${RTP_VOICE_PROCESSOR_CXX})
-target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES} gstreamer-rtp-1.0 webrtc-audio-processing)
+target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES} gstreamer-rtp-1.0 ${RTP_VOICE_PROCESSOR_LIB})
 set_target_properties(rtp PROPERTIES PREFIX "")
 set_target_properties(rtp PROPERTIES LIBRARY_OUTPUT_DIRECTORY ${CMAKE_BINARY_DIR}/plugins/)
 
diff --git a/plugins/rtp/src/voice_processor.vala b/plugins/rtp/src/voice_processor.vala
index e6dc7e8f..66e95d72 100644
--- a/plugins/rtp/src/voice_processor.vala
+++ b/plugins/rtp/src/voice_processor.vala
@@ -123,7 +123,7 @@ public class Dino.Plugins.Rtp.VoiceProcessor : Audio.Filter {
         }
         analyze_reverse_stream(native, echo_probe.audio_info, buffer);
         if (adjust_delay_timeout_id == 0 && echo_probe != null) {
-            adjust_delay_timeout_id = Timeout.add(5000, adjust_delay);
+            adjust_delay_timeout_id = Timeout.add(1000, adjust_delay);
         }
     }
 
diff --git a/plugins/rtp/src/voice_processor_native.cpp b/plugins/rtp/src/voice_processor_native.cpp
index 00f719e1..8a052cf8 100644
--- a/plugins/rtp/src/voice_processor_native.cpp
+++ b/plugins/rtp/src/voice_processor_native.cpp
@@ -11,6 +11,8 @@
 struct _DinoPluginsRtpVoiceProcessorNative {
     webrtc::AudioProcessing *apm;
     gint stream_delay;
+    gint last_median;
+    gint last_poor_delays;
 };
 
 extern "C" void *dino_plugins_rtp_adjust_to_running_time(GstBaseTransform *transform, GstBuffer *buffer) {
@@ -26,6 +28,8 @@ extern "C" void *dino_plugins_rtp_voice_processor_init_native(gint stream_delay)
     config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, 85));
     native->apm = webrtc::AudioProcessing::Create(config);
     native->stream_delay = stream_delay;
+    native->last_median = 0;
+    native->last_poor_delays = 0;
     return native;
 }
 
@@ -65,19 +69,19 @@ dino_plugins_rtp_voice_processor_analyze_reverse_stream(void *native_ptr, GstAud
     webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false);
     webrtc::AudioProcessing *apm = native->apm;
 
-    GstAudioBuffer audio_buffer;
-    gst_audio_buffer_map(&audio_buffer, info, buffer, GST_MAP_READ);
+    GstMapInfo map;
+    gst_buffer_map(buffer, &map, GST_MAP_READ);
 
     webrtc::AudioFrame frame;
     frame.num_channels_ = info->channels;
     frame.sample_rate_hz_ = info->rate;
     frame.samples_per_channel_ = gst_buffer_get_size(buffer) / info->bpf;
-    memcpy(frame.data_, audio_buffer.planes[0], frame.samples_per_channel_ * info->bpf);
+    memcpy(frame.data_, map.data, frame.samples_per_channel_ * info->bpf);
 
     int err = apm->AnalyzeReverseStream(&frame);
     if (err < 0) g_warning("voice_processor_native.cpp: ProcessReverseStream %i", err);
 
-    gst_audio_buffer_unmap(&audio_buffer);
+    gst_buffer_unmap(buffer, &map);
 }
 
 extern "C" void dino_plugins_rtp_voice_processor_notify_gain_level(void *native_ptr, gint gain_level) {
@@ -101,14 +105,17 @@ extern "C" bool dino_plugins_rtp_voice_processor_get_stream_has_voice(void *nati
 extern "C" void dino_plugins_rtp_voice_processor_adjust_stream_delay(void *native_ptr) {
     _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
     webrtc::AudioProcessing *apm = native->apm;
-    int median, std;
+    int median, std, poor_delays;
     float fraction_poor_delays;
     apm->echo_cancellation()->GetDelayMetrics(&median, &std, &fraction_poor_delays);
-    if (fraction_poor_delays < 0) return;
-    g_debug("voice_processor_native.cpp: Stream delay metrics: %i %i %f", median, std, fraction_poor_delays);
-    if (fraction_poor_delays > 0.5) {
-        native->stream_delay = std::max(0, native->stream_delay + std::min(-10, std::max(median, 10)));
-        g_debug("voice_processor_native.cpp: Adjusted stream delay %i", native->stream_delay);
+    poor_delays = (int)(fraction_poor_delays * 100.0);
+    if (fraction_poor_delays < 0 || (native->last_median == median && native->last_poor_delays == poor_delays)) return;
+    g_debug("voice_processor_native.cpp: Stream delay metrics: median=%i std=%i poor_delays=%i%%", median, std, poor_delays);
+    native->last_median = median;
+    native->last_poor_delays = poor_delays;
+    if (poor_delays > 90) {
+        native->stream_delay = std::min(std::max(0, native->stream_delay + std::min(48, std::max(median, -48))), 384);
+        g_debug("voice_processor_native.cpp: set stream_delay=%i", native->stream_delay);
     }
 }
 
@@ -118,21 +125,21 @@ dino_plugins_rtp_voice_processor_process_stream(void *native_ptr, GstAudioInfo *
     webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false);
     webrtc::AudioProcessing *apm = native->apm;
 
-    GstAudioBuffer audio_buffer;
-    gst_audio_buffer_map(&audio_buffer, info, buffer, GST_MAP_READWRITE);
+    GstMapInfo map;
+    gst_buffer_map(buffer, &map, GST_MAP_READWRITE);
 
     webrtc::AudioFrame frame;
     frame.num_channels_ = info->channels;
     frame.sample_rate_hz_ = info->rate;
     frame.samples_per_channel_ = info->rate / 100;
-    memcpy(frame.data_, audio_buffer.planes[0], frame.samples_per_channel_ * info->bpf);
+    memcpy(frame.data_, map.data, frame.samples_per_channel_ * info->bpf);
 
     apm->set_stream_delay_ms(native->stream_delay);
     int err = apm->ProcessStream(&frame);
-    if (err >= 0) memcpy(audio_buffer.planes[0], frame.data_, frame.samples_per_channel_ * info->bpf);
+    if (err >= 0) memcpy(map.data, frame.data_, frame.samples_per_channel_ * info->bpf);
     if (err < 0) g_warning("voice_processor_native.cpp: ProcessStream %i", err);
 
-    gst_audio_buffer_unmap(&audio_buffer);
+    gst_buffer_unmap(buffer, &map);
 }
 
 extern "C" void dino_plugins_rtp_voice_processor_destroy_native(void *native_ptr) {