diff --git a/webrtc/android/app/src/main/jni/Android.mk b/webrtc/android/app/src/main/jni/Android.mk
index c0ab0f81d2..326781d6af 100644
--- a/webrtc/android/app/src/main/jni/Android.mk
+++ b/webrtc/android/app/src/main/jni/Android.mk
@@ -31,7 +31,7 @@ GSTREAMER_NDK_BUILD_PATH  := $(GSTREAMER_ROOT)/share/gst-android/ndk-build/
 
 include $(GSTREAMER_NDK_BUILD_PATH)/plugins.mk
 
-GSTREAMER_PLUGINS_CORE_CUSTOM := coreelements app audioconvert audiorate audioresample videoconvert videorate videoscale videotestsrc volume 
+GSTREAMER_PLUGINS_CORE_CUSTOM := coreelements app audioconvert audiorate audioresample videoconvert videorate videoscale videotestsrc volume
 GSTREAMER_PLUGINS_CODECS_CUSTOM := videoparsersbad vpx opus audioparsers opusparse androidmedia
 GSTREAMER_PLUGINS_NET_CUSTOM := tcp rtsp rtp rtpmanager udp srtp webrtc dtls nice
 GSTREAMER_PLUGINS         := $(GSTREAMER_PLUGINS_CORE_CUSTOM) $(GSTREAMER_PLUGINS_CODECS_CUSTOM) $(GSTREAMER_PLUGINS_NET_CUSTOM) \
diff --git a/webrtc/android/app/src/main/jni/webrtc.c b/webrtc/android/app/src/main/jni/webrtc.c
index e5b47fc5fc..7a30dd957a 100644
--- a/webrtc/android/app/src/main/jni/webrtc.c
+++ b/webrtc/android/app/src/main/jni/webrtc.c
@@ -822,6 +822,8 @@ native_class_init (JNIEnv * env, jclass klass)
     __android_log_print (ANDROID_LOG_ERROR, "GstPlayer", "%s", message);
     (*env)->ThrowNew (env, exception_class, message);
   }
+  GST_DEBUG_CATEGORY_INIT (debug_category, "webrtc", 0,
+      "GStreamer Android WebRTC");
   //gst_debug_set_threshold_from_string ("gl*:7", FALSE);
 }
 
@@ -905,7 +907,7 @@ JNI_OnLoad (JavaVM * vm, void *reserved)
   java_vm = vm;
 
   if ((*vm)->GetEnv (vm, (void **) &env, JNI_VERSION_1_4) != JNI_OK) {
-    __android_log_print (ANDROID_LOG_ERROR, "GstPlayer",
+    __android_log_print (ANDROID_LOG_ERROR, "GstWebRTC",
         "Could not retrieve JNIEnv");
     return 0;
   }