diff --git a/audio_capture/launch/capture.launch b/audio_capture/launch/capture.launch
index 6495bb21..245501f0 100644
--- a/audio_capture/launch/capture.launch
+++ b/audio_capture/launch/capture.launch
@@ -12,6 +12,8 @@
   <arg name="sample_format" default="S16LE"/>
   <arg name="ns" default="audio"/>
   <arg name="audio_topic" default="audio"/>
+  <arg name="capture_volume" default="50"/>
+
 
   <group ns="$(arg ns)">
     <node name="audio_capture" pkg="audio_capture" type="audio_capture" output="screen">
@@ -24,6 +26,7 @@
       <param name="depth" value="$(arg depth)"/>
       <param name="sample_rate" value="$(arg sample_rate)"/>
       <param name="sample_format" value="$(arg sample_format)"/>
+      <param name="capture_volume" value="$(arg capture_volume)"/>
     </node>
   </group>
 
diff --git a/audio_capture/src/audio_capture.cpp b/audio_capture/src/audio_capture.cpp
index 0ef67a87..2bbdd7ee 100644
--- a/audio_capture/src/audio_capture.cpp
+++ b/audio_capture/src/audio_capture.cpp
@@ -17,6 +17,7 @@ namespace audio_transport
       RosGstCapture()
       {
         _bitrate = 192;
+        _capture_volume = 50; // Default volume 50%
 
         std::string dst_type;
 
@@ -31,6 +32,11 @@ namespace audio_transport
         ros::param::param<int>("~channels", _channels, 1);
         ros::param::param<int>("~depth", _depth, 16);
         ros::param::param<int>("~sample_rate", _sample_rate, 16000);
+        
+        // The volume level for audio capture (0-100%)
+        ros::param::param<int>("~capture_volume", _capture_volume, 50);
+        // Ensure volume is within valid range
+        _capture_volume = std::max(0, std::min(_capture_volume, 100));
 
         // The destination of the audio
         ros::param::param<std::string>("~dst", dst_type, "appsink");
@@ -83,6 +89,18 @@ namespace audio_transport
           // ghcar *gst_device = device.c_str();
           g_object_set(G_OBJECT(_source), "device", device.c_str(), NULL);
         }
+        
+        // Create volume control element
+        _volume = gst_element_factory_make("volume", "volume");
+        if (!_volume) {
+          ROS_ERROR_STREAM("Failed to create volume element");
+          exitOnMainThread(1);
+        }
+        
+        // Set volume level (0-100% mapped to 0.0-1.0)
+        double volume_value = static_cast<double>(_capture_volume) / 100.0;
+        g_object_set(G_OBJECT(_volume), "volume", volume_value, NULL);
+        ROS_INFO("Setting capture volume to %d%% (%.2f)", _capture_volume, volume_value);
 
         GstCaps *caps;
         caps = gst_caps_new_simple("audio/x-raw",
@@ -114,18 +132,21 @@ namespace audio_transport
           g_object_set( G_OBJECT(_encode), "target", 1, NULL);
           g_object_set( G_OBJECT(_encode), "bitrate", _bitrate, NULL);
 
-          gst_bin_add_many( GST_BIN(_pipeline), _source, _filter, _convert, _encode, _sink, NULL);
-          link_ok = gst_element_link_many(_source, _filter, _convert, _encode, _sink, NULL);
+          // Add volume element to pipeline
+          gst_bin_add_many( GST_BIN(_pipeline), _source, _filter, _volume, _convert, _encode, _sink, NULL);
+          link_ok = gst_element_link_many(_source, _filter, _volume, _convert, _encode, _sink, NULL);
         } else if (_format == "wave") {
           if (dst_type == "appsink") {
             g_object_set( G_OBJECT(_sink), "caps", caps, NULL);
             gst_caps_unref(caps);
-            gst_bin_add_many( GST_BIN(_pipeline), _source, _sink, NULL);
-            link_ok = gst_element_link_many( _source, _sink, NULL);
+            // Add volume element to pipeline
+            gst_bin_add_many( GST_BIN(_pipeline), _source, _volume, _sink, NULL);
+            link_ok = gst_element_link_many( _source, _volume, _sink, NULL);
           } else {
             _filter = gst_element_factory_make("wavenc", "filter");
-            gst_bin_add_many( GST_BIN(_pipeline), _source, _filter, _sink, NULL);
-            link_ok = gst_element_link_many( _source, _filter, _sink, NULL);
+            // Add volume element to pipeline
+            gst_bin_add_many( GST_BIN(_pipeline), _source, _volume, _filter, _sink, NULL);
+            link_ok = gst_element_link_many( _source, _volume, _filter, _sink, NULL);
           }
         } else {
           ROS_ERROR_STREAM("format must be \"wave\" or \"mp3\"");
@@ -159,6 +180,9 @@ namespace audio_transport
         info_msg.bitrate = _bitrate;
         info_msg.coding_format = _format;
         _pub_info.publish(info_msg);
+        
+        // Start timer to check for parameter updates
+        _timer = _nh.createTimer(ros::Duration(1.0), &RosGstCapture::parameterCallback, this);
       }
 
       ~RosGstCapture()
@@ -168,6 +192,30 @@ namespace audio_transport
         gst_object_unref(_pipeline);
         g_main_loop_unref(_loop);
       }
+      
+      // Check for volume parameter updates
+      void parameterCallback(const ros::TimerEvent& event)
+      {
+        int current_volume;
+        if(ros::param::get("~capture_volume", current_volume))
+        {
+          if(current_volume != _capture_volume)
+          {
+            _capture_volume = std::max(0, std::min(current_volume, 100));
+            updateVolume();
+          }
+        }
+      }
+      
+      // Update volume in GStreamer pipeline
+      void updateVolume()
+      {
+        if(_volume) {
+          double volume_value = static_cast<double>(_capture_volume) / 100.0;
+          ROS_INFO("Updating capture volume to %d%% (%.2f)", _capture_volume, volume_value);
+          g_object_set(G_OBJECT(_volume), "volume", volume_value, NULL);
+        }
+      }
 
       void exitOnMainThread(int code)
       {
@@ -242,12 +290,14 @@ namespace audio_transport
       ros::Publisher _pub;
       ros::Publisher _pub_stamped;
       ros::Publisher _pub_info;
+      ros::Subscriber _volume_sub;
+      ros::Timer _timer;
 
       boost::thread _gst_thread;
 
-      GstElement *_pipeline, *_source, *_filter, *_sink, *_convert, *_encode;
+      GstElement *_pipeline, *_source, *_filter, *_sink, *_convert, *_encode, *_volume;
       GstBus *_bus;
-      int _bitrate, _channels, _depth, _sample_rate;
+      int _bitrate, _channels, _depth, _sample_rate, _capture_volume;
       GMainLoop *_loop;
       std::string _format, _sample_format;
   };
diff --git a/audio_play/launch/play.launch b/audio_play/launch/play.launch
index 2a11ead4..739fe9fa 100644
--- a/audio_play/launch/play.launch
+++ b/audio_play/launch/play.launch
@@ -12,6 +12,8 @@
   <arg name="sample_format" default="S16LE"/>
   <arg name="ns" default="audio"/>
   <arg name="audio_topic" default="audio" />
+  <arg name="play_volume" default="50"/>
+
 
   <group ns="$(arg ns)">
     <node name="audio_play" pkg="audio_play" type="audio_play" output="screen">
@@ -24,6 +26,7 @@
       <param name="depth" value="$(arg depth)"/>
       <param name="sample_rate" value="$(arg sample_rate)"/>
       <param name="sample_format" value="$(arg sample_format)"/>
+      <param name="play_volume" value="$(arg play_volume)"/>
     </node>
   </group>
 </launch>
diff --git a/audio_play/src/audio_play.cpp b/audio_play/src/audio_play.cpp
index fe127c2a..dd932747 100644
--- a/audio_play/src/audio_play.cpp
+++ b/audio_play/src/audio_play.cpp
@@ -33,6 +33,10 @@ namespace audio_transport
         ros::param::param<int>("~depth", depth, 16);
         ros::param::param<int>("~sample_rate", sample_rate, 16000);
         ros::param::param<std::string>("~sample_format", sample_format, "S16LE");
+        
+        // The volume level for audio playback (0-100%)
+        ros::param::param<int>("~play_volume", _play_volume, 50);
+        _play_volume = std::max(0, std::min(_play_volume, 100));
 
         _sub = _nh.subscribe("audio", 10, &RosGstPlay::onAudio, this);
 
@@ -61,14 +65,28 @@ namespace audio_transport
           _convert = gst_element_factory_make("audioconvert", "convert");
           audiopad = gst_element_get_static_pad(_convert, "sink");
           _resample = gst_element_factory_make("audioresample", "resample");
+          
+          // Create volume control element
+          _volume = gst_element_factory_make("volume", "volume");
+          if (!_volume) {
+            ROS_ERROR_STREAM("Failed to create volume element");
+            return;
+          }
+          
+          // Set volume level (0-100% mapped to 0.0-1.0)
+          double volume_value = static_cast<double>(_play_volume) / 100.0;
+          g_object_set(G_OBJECT(_volume), "volume", volume_value, NULL);
+          ROS_INFO("Setting play volume to %d%% (%.2f)", _play_volume, volume_value);
 
           _sink = gst_element_factory_make("alsasink", "sink");
           g_object_set(G_OBJECT(_sink), "sync", FALSE, NULL);
           if (!device.empty()) {
             g_object_set(G_OBJECT(_sink), "device", device.c_str(), NULL);
           }
-          gst_bin_add_many( GST_BIN(_audio), _convert, _resample, _sink, NULL);
-          gst_element_link_many(_convert, _resample, _sink, NULL);
+          
+          // Add volume element to pipeline
+          gst_bin_add_many( GST_BIN(_audio), _convert, _resample, _volume, _sink, NULL);
+          gst_element_link_many(_convert, _resample, _volume, _sink, NULL);
           gst_element_add_pad(_audio, gst_ghost_pad_new("sink", audiopad));
         }
         else
@@ -126,6 +144,41 @@ namespace audio_transport
         //gst_element_set_state(GST_ELEMENT(_playbin), GST_STATE_PLAYING);
 
         _gst_thread = boost::thread( boost::bind(g_main_loop_run, _loop) );
+        
+        // Start timer to check for parameter updates
+        _timer = _nh.createTimer(ros::Duration(1.0), &RosGstPlay::parameterCallback, this);
+      }
+      
+      ~RosGstPlay()
+      {
+        g_main_loop_quit(_loop);
+        gst_element_set_state(_pipeline, GST_STATE_NULL);
+        gst_object_unref(_pipeline);
+        g_main_loop_unref(_loop);
+      }
+      
+      // Check for volume parameter updates
+      void parameterCallback(const ros::TimerEvent& event)
+      {
+        int current_volume;
+        if(ros::param::get("~play_volume", current_volume))
+        {
+          if(current_volume != _play_volume)
+          {
+            _play_volume = std::max(0, std::min(current_volume, 100));
+            updateVolume();
+          }
+        }
+      }
+      
+      // Update volume in GStreamer pipeline
+      void updateVolume()
+      {
+        if(_volume) {
+          double volume_value = static_cast<double>(_play_volume) / 100.0;
+          ROS_INFO("Updating play volume to %d%% (%.2f)", _play_volume, volume_value);
+          g_object_set(G_OBJECT(_volume), "volume", volume_value, NULL);
+        }
       }
 
     private:
@@ -176,11 +229,14 @@ namespace audio_transport
 
       ros::NodeHandle _nh;
       ros::Subscriber _sub;
+      ros::Subscriber _volume_sub;
+      ros::Timer _timer;
       boost::thread _gst_thread;
 
-      GstElement *_pipeline, *_source, *_sink, *_decoder, *_convert, *_audio, *_resample, *_filter;
+      GstElement *_pipeline, *_source, *_sink, *_decoder, *_convert, *_audio, *_resample, *_filter, *_volume;
       GstElement *_playbin;
       GMainLoop *_loop;
+      int _play_volume; // Volume control (0-100%)
   };
 }