--- a/media/webrtc/signaling/src/media-conduit/AudioConduit.h
+++ b/media/webrtc/signaling/src/media-conduit/AudioConduit.h
@@ -263,30 +263,30 @@ public:
                    const int64_t aTimestamp,
                    const uint32_t aJitter) override;
 
   // test-only: inserts fake CSRCs and audio level data
   void InsertAudioLevelForContributingSource(uint32_t aSource,
                                              int64_t aTimestamp,
                                              bool aHasLevel,
                                              uint8_t aLevel);
+
+  bool IsSamplingFreqSupported(int freq) const override;
+
 private:
   WebrtcAudioConduit(const WebrtcAudioConduit& other) = delete;
   void operator=(const WebrtcAudioConduit& other) = delete;
 
   //Local database of currently applied receive codecs
   typedef std::vector<AudioCodecConfig* > RecvCodecList;
 
   //Function to convert between WebRTC and Conduit codec structures
   bool CodecConfigToWebRTCCodec(const AudioCodecConfig* codecInfo,
                                 webrtc::CodecInst& cinst);
 
-  //Checks if given sampling frequency is supported
-  bool IsSamplingFreqSupported(int freq) const;
-
   //Generate block size in sample lenght for a given sampling frequency
   unsigned int GetNum10msSamplesForFrequency(int samplingFreqHz) const;
 
   // Function to copy a codec structure to Conduit's database
   bool CopyCodecToDB(const AudioCodecConfig* codecInfo);
 
   // Functions to verify if the codec passed is already in
   // conduits database

--- a/media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h
+++ b/media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h
@@ -514,16 +514,22 @@ public:
    *       This ensures the decoded samples are ready for reading.
    *
    */
   virtual MediaConduitErrorCode GetAudioFrame(int16_t speechData[],
                                               int32_t samplingFreqHz,
                                               int32_t capture_delay,
                                               int& lengthSamples) = 0;
 
+  /**
+   * Checks if given sampling frequency is supported
+   * @param freq: Sampling rate (in Hz) to check
+   */
+  virtual bool IsSamplingFreqSupported(int freq) const = 0;
+
    /**
     * Function to configure send codec for the audio session
     * @param sendSessionConfig: CodecConfiguration
     * NOTE: See VideoConduit for more information
     */
 
   virtual MediaConduitErrorCode ConfigureSendMediaCodec(const AudioCodecConfig* sendCodecConfig) = 0;
 

--- a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
@@ -2167,17 +2167,25 @@ class MediaPipelineReceiveAudio::Pipelin
 {
 public:
   PipelineListener(dom::MediaStreamTrack* aTrack,
                    const RefPtr<MediaSessionConduit>& aConduit)
     : GenericReceiveListener(aTrack)
     , mConduit(aConduit)
     , mSource(mTrack->GetInputStream()->AsSourceStream())
     , mTrackId(mTrack->GetInputTrackId())
-    , mRate(mSource ? mSource->GraphRate() : 0)
+    // AudioSession conduit only supports 16, 32, 44.1 and 48kHz
+    // This is an artificial limitation, it would however require more changes
+    // to support any rates.
+    // If the sampling rate is not-supported, we will use 48kHz instead.
+    , mRate(mSource ? (static_cast<AudioSessionConduit*>(mConduit.get())
+                           ->IsSamplingFreqSupported(mSource->GraphRate())
+                         ? mSource->GraphRate()
+                         : WEBRTC_MAX_SAMPLE_RATE)
+                    : WEBRTC_MAX_SAMPLE_RATE)
     , mTaskQueue(
         new AutoTaskQueue(GetMediaThreadPool(MediaThreadType::WEBRTC_DECODER),
                           "AudioPipelineListener"))
     , mLastLog(0)
   {
     MOZ_ASSERT(mSource);
   }