diff --git a/modules/audio_device/audio_device_generic.cc b/modules/audio_device/audio_device_generic.cc
index 7b8cfd1734..f6f079d074 100644
--- a/modules/audio_device/audio_device_generic.cc
+++ b/modules/audio_device/audio_device_generic.cc
@@ -63,4 +63,15 @@ int AudioDeviceGeneric::GetRecordAudioParameters(
 }
 #endif  // WEBRTC_IOS
 
+// MARK: Nicegram NCG-5828 call recording
+
+void AudioDeviceGeneric::StartNicegramRecording(const CompletionRecorderCallback& callback, const RecorderErrorCallback &errorCallback) {
+    RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+}
+
+void AudioDeviceGeneric::StopNicegramRecording(bool synchronous) {
+    RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+}
+//
+
 }  // namespace webrtc
diff --git a/modules/audio_device/audio_device_generic.h b/modules/audio_device/audio_device_generic.h
index 41e24eb3b0..5fee887d18 100644
--- a/modules/audio_device/audio_device_generic.h
+++ b/modules/audio_device/audio_device_generic.h
@@ -21,6 +21,13 @@ namespace webrtc {
 
 class AudioDeviceGeneric {
  public:
+  // MARK: Nicegram NCG-5828 call recording
+  using CompletionRecorderCallback = std::function<void(const std::string& outputFilePath,
+                                                        double durationInSeconds,
+                                                        size_t rawDataSize)>;
+    
+  using RecorderErrorCallback = std::function<void(const std::string& error)>;
+  //
   // For use with UMA logging. Must be kept in sync with histograms.xml in
   // Chrome, located at
   // https://cs.chromium.org/chromium/src/tools/metrics/histograms/histograms.xml
@@ -135,6 +142,11 @@ class AudioDeviceGeneric {
   virtual int GetRecordAudioParameters(AudioParameters* params) const;
 #endif  // WEBRTC_IOS
 
+  // MARK: Nicegram NCG-5828 call recording
+  virtual void StartNicegramRecording(const CompletionRecorderCallback& callback, const RecorderErrorCallback &errorCallback);
+  virtual void StopNicegramRecording(bool synchronous = false);
+  //
+    
   virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
 
   virtual ~AudioDeviceGeneric() {}
diff --git a/modules/audio_device/include/audio_device.h b/modules/audio_device/include/audio_device.h
index 47d2aecfa7..2874ff996b 100644
--- a/modules/audio_device/include/audio_device.h
+++ b/modules/audio_device/include/audio_device.h
@@ -23,6 +23,13 @@ class AudioDeviceModuleForTest;
 
 class AudioDeviceModule : public webrtc::RefCountInterface {
  public:
+  // MARK: Nicegram NCG-5828 call recording
+  using CompletionRecorderCallback = std::function<void(const std::string& outputFilePath,
+                                                        double durationInSeconds,
+                                                        size_t rawDataSize)>;
+    
+  using RecorderErrorCallback = std::function<void(const std::string& error)>;
+  //
   enum AudioLayer {
     kPlatformDefaultAudio = 0,
     kWindowsCoreAudio,
@@ -171,6 +178,11 @@ class AudioDeviceModule : public webrtc::RefCountInterface {
   virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
 #endif  // WEBRTC_IOS
 
+  // MARK: Nicegram NCG-5828 call recording
+  virtual void StartNicegramRecording(const CompletionRecorderCallback& callback, const RecorderErrorCallback &errorCallback) {}
+  virtual void StopNicegramRecording(bool synchronous = false) {}
+  //
+    
  protected:
   ~AudioDeviceModule() override {}
 };
diff --git a/modules/video_coding/codecs/h264/h264.cc b/modules/video_coding/codecs/h264/h264.cc
index 5b9f0338a9..c1e63ec785 100644
--- a/modules/video_coding/codecs/h264/h264.cc
+++ b/modules/video_coding/codecs/h264/h264.cc
@@ -153,7 +153,7 @@ bool H264Encoder::SupportsScalabilityMode(ScalabilityMode scalability_mode) {
 
 std::unique_ptr<H264Decoder> H264Decoder::Create() {
   RTC_DCHECK(H264Decoder::IsSupported());
-#if defined(WEBRTC_USE_H264)
+#if defined(WEBRTC_USE_H264) && defined(WEBRTC_USE_H264_DECODER)
   RTC_CHECK(g_rtc_use_h264);
   RTC_LOG(LS_INFO) << "Creating H264DecoderImpl.";
   return std::make_unique<H264DecoderImpl>();
diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/modules/video_coding/codecs/h264/h264_decoder_impl.cc
index c6446c25ce..b90a592327 100644
--- a/modules/video_coding/codecs/h264/h264_decoder_impl.cc
+++ b/modules/video_coding/codecs/h264/h264_decoder_impl.cc
@@ -13,6 +13,7 @@
 // build with H264 support, please do not move anything out of the
 // #ifdef unless needed and tested.
 #ifdef WEBRTC_USE_H264
+#ifdef WEBRTC_USE_H264_DECODER
 
 #include "modules/video_coding/codecs/h264/h264_decoder_impl.h"
 
@@ -658,4 +659,5 @@ void H264DecoderImpl::ReportError() {
 
 }  // namespace webrtc
 
+#endif // WEBRTC_USE_H264_DECODER
 #endif  // WEBRTC_USE_H264
diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.h b/modules/video_coding/codecs/h264/h264_decoder_impl.h
index 97d091cf4b..7c1f1aac9b 100644
--- a/modules/video_coding/codecs/h264/h264_decoder_impl.h
+++ b/modules/video_coding/codecs/h264/h264_decoder_impl.h
@@ -16,6 +16,7 @@
 // build with H264 support, please do not move anything out of the
 // #ifdef unless needed and tested.
 #ifdef WEBRTC_USE_H264
+#ifdef WEBRTC_USE_H264_DECODER
 
 #if defined(WEBRTC_WIN) && !defined(__clang__)
 #error "See: bugs.webrtc.org/9213#c13."
@@ -104,6 +105,7 @@ class H264DecoderImpl : public H264Decoder {
 
 }  // namespace webrtc
 
+#endif  // WEBRTC_USE_H264_DECODER
 #endif  // WEBRTC_USE_H264
 
 #endif  // MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
diff --git a/p2p/base/ice_transport_internal.cc b/p2p/base/ice_transport_internal.cc
index 0dc7e50af2..4619ef5898 100644
--- a/p2p/base/ice_transport_internal.cc
+++ b/p2p/base/ice_transport_internal.cc
@@ -35,9 +35,12 @@ RTCError VerifyCandidate(const Candidate& cand) {
     return RTCError::OK();
   }
   if (port < 1024) {
-    if ((port != 80) && (port != 443)) {
-      return RTCError(RTCErrorType::INVALID_PARAMETER,
-                      "candidate has port below 1024, but not 80 or 443");
+    if (absl::EndsWith(cand.address().hostname(), ".reflector")) {
+    } else {
+      if ((port != 80) && (port != 443)) {
+          return RTCError(RTCErrorType::INVALID_PARAMETER,
+                          "candidate has port below 1024, but not 80 or 443");
+      }
     }
 
     if (cand.address().IsPrivateIP()) {
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index f66a2feace..43ca8a9298 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -328,7 +328,7 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
   }
 
   // Only construct the encrypted receiver if frame encryption is enabled.
-  if (config_.crypto_options.sframe.require_frame_encryption) {
+  if (config_.crypto_options.sframe.require_frame_encryption || frame_decryptor) {
     buffered_frame_decryptor_ =
         std::make_unique<BufferedFrameDecryptor>(this, this, field_trials_);
     if (frame_decryptor != nullptr) {