--- a/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp
+++ b/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp
@@ -2057,16 +2057,18 @@ JsepSessionImpl::SetupDefaultCodecs()
   red->UpdateRedundantEncodings(mSupportedCodecs.values);
 }
 
 void
 JsepSessionImpl::SetupDefaultRtpExtensions()
 {
   AddAudioRtpExtension(webrtc::RtpExtension::kAudioLevelUri,
                        SdpDirectionAttribute::Direction::kSendrecv);
+  AddAudioRtpExtension(webrtc::RtpExtension::kCsrcAudioLevelUri,
+                       SdpDirectionAttribute::Direction::kRecvonly);
   AddAudioRtpExtension(webrtc::RtpExtension::kMIdUri,
                        SdpDirectionAttribute::Direction::kSendrecv);
   AddVideoRtpExtension(webrtc::RtpExtension::kAbsSendTimeUri,
                        SdpDirectionAttribute::Direction::kSendrecv);
   AddVideoRtpExtension(webrtc::RtpExtension::kTimestampOffsetUri,
                        SdpDirectionAttribute::Direction::kSendrecv);
   AddVideoRtpExtension(webrtc::RtpExtension::kMIdUri,
                        SdpDirectionAttribute::Direction::kSendrecv);

--- a/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
+++ b/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
@@ -609,30 +609,38 @@ WebrtcAudioConduit::ConfigureRecvMediaCo
 
   DumpCodecDB();
   return kMediaConduitNoError;
 }
 
 MediaConduitErrorCode
 WebrtcAudioConduit::EnableAudioLevelExtension(bool aEnabled,
                                               uint8_t aId,
-                                              bool aDirectionIsSend)
+                                              bool aDirectionIsSend,
+                                              bool aLevelIsSsrc)
 {
   CSFLogDebug(LOGTAG,  "%s %d %d %d", __FUNCTION__, aEnabled, aId,
               aDirectionIsSend);
 
   bool ret;
   if (aDirectionIsSend) {
+    if (!aLevelIsSsrc) {
+      CSFLogError(LOGTAG,
+                  "%s SetSendAudioLevelIndicationStatus Failed"
+                  " can not send CSRC audio levels.", __FUNCTION__);
+      return kMediaConduitMalformedArgument;
+    }
     ret = mPtrVoERTP_RTCP->SetSendAudioLevelIndicationStatus(mChannel,
                                                              aEnabled,
                                                              aId) == -1;
   } else {
     ret = mPtrRTP->SetReceiveAudioLevelIndicationStatus(mChannel,
                                                         aEnabled,
-                                                        aId) == -1;
+                                                        aId,
+                                                        aLevelIsSsrc) == -1;
   }
   if (ret) {
     CSFLogError(LOGTAG, "%s SetSendAudioLevelIndicationStatus Failed", __FUNCTION__);
     return kMediaConduitUnknownError;
   }
   return kMediaConduitNoError;
 }
 

--- a/media/webrtc/signaling/src/media-conduit/AudioConduit.h
+++ b/media/webrtc/signaling/src/media-conduit/AudioConduit.h
@@ -94,17 +94,18 @@ public:
    *        transmission sub-system on the engine.
    */
   virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
     const std::vector<AudioCodecConfig* >& codecConfigList) override;
 
   MediaConduitErrorCode
   EnableAudioLevelExtension(bool aEnabled,
                             uint8_t aId,
-                            bool aDirectionIsSend) override;
+                            bool aDirectionIsSend,
+                            bool aLevelIsSsrc = true) override;
 
   virtual MediaConduitErrorCode EnableMIDExtension(bool enabled, uint8_t id) override;
 
   /**
    * Register External Transport to this Conduit. RTP and RTCP frames from the VoiceEngine
    * shall be passed to the registered transport for transporting externally.
    */
   virtual MediaConduitErrorCode SetTransmitterTransport(RefPtr<TransportInterface> aTransport) override;

--- a/media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h
+++ b/media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h
@@ -537,17 +537,18 @@ public:
     * @param aId: the RTP extension header ID to use
     * @param aDirectionIsSend: indicates whether to set the extension on the
     *                          sender or the receiver side
     * returns an error if the extension could not be set
     */
   virtual MediaConduitErrorCode
   EnableAudioLevelExtension(bool aEnabled,
                             uint8_t aId,
-                            bool aDirectionIsSend) = 0;
+                            bool aDirectionIsSend,
+                            bool aLevelIsSsrc = true) = 0;
   virtual MediaConduitErrorCode
   EnableMIDExtension(bool enabled, uint8_t id) = 0;
 
   virtual bool SetDtmfPayloadType(unsigned char type, int freq) = 0;
 
   virtual bool InsertDTMFTone(int channel, int eventCode, bool outOfBand,
                               int lengthMs, int attenuationDb) = 0;
 

--- a/media/webrtc/signaling/src/peerconnection/TransceiverImpl.cpp
+++ b/media/webrtc/signaling/src/peerconnection/TransceiverImpl.cpp
@@ -737,16 +737,31 @@ TransceiverImpl::UpdateAudioConduit()
                                                  false);
 
       if (error) {
         MOZ_MTLOG(ML_ERROR, mPCHandle << "[" << mMid << "]: " << __FUNCTION__ <<
                             " EnableAudioLevelExtension failed: " << error);
         return NS_ERROR_FAILURE;
       }
     }
+
+    const SdpExtmapAttributeList::Extmap* csrcAudioLevelExt =
+        details.GetExt(webrtc::RtpExtension::kCsrcAudioLevelUri);
+    if (csrcAudioLevelExt) {
+      MOZ_MTLOG(ML_DEBUG, "Calling EnableAudioLevelExtension for CSRCs");
+      error = conduit->EnableAudioLevelExtension(true,
+                                                 csrcAudioLevelExt->entry,
+                                                 false,
+                                                 false);
+      if (error) {
+        MOZ_MTLOG(ML_ERROR, mPCHandle << "[" << mMid << "]: " << __FUNCTION__ <<
+                  " EnableAudioLevelExtension for CSRCs failed: " << error);
+        return NS_ERROR_FAILURE;
+      }
+    }
   }
 
   if (mJsepTransceiver->mSendTrack.GetNegotiatedDetails() &&
       mJsepTransceiver->mSendTrack.GetActive()) {
     const auto& details(*mJsepTransceiver->mSendTrack.GetNegotiatedDetails());
     PtrVector<AudioCodecConfig> configs;
     nsresult rv = NegotiatedDetailsToAudioCodecConfigs(details, &configs);
 

--- a/media/webrtc/trunk/webrtc/voice_engine/channel.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel.cc
@@ -2610,31 +2610,33 @@ int Channel::GetLocalSSRC(unsigned int& 
   return 0;
 }
 
 int Channel::GetRemoteSSRC(unsigned int& ssrc) {
   ssrc = rtp_receiver_->SSRC();
   return 0;
 }
 
-int Channel::SetSendAudioLevelIndicationStatus(bool enable, unsigned char id) {
+int Channel::SetSendAudioLevelIndicationStatus(bool enable,
+                                               unsigned char id) {
   _includeAudioLevelIndication = enable;
   return SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
 }
 
 int Channel::SetSendMIDStatus(bool enable, unsigned char id) {
   return SetSendRtpHeaderExtension(enable, kRtpExtensionMId, id);
 }
 
 int Channel::SetReceiveAudioLevelIndicationStatus(bool enable,
-                                                  unsigned char id) {
-  rtp_header_parser_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel);
-  if (enable &&
-      !rtp_header_parser_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
-                                                      id)) {
+                                                  unsigned char id,
+                                                  bool isLevelSsrc) {
+  const webrtc::RTPExtensionType& rtpExt = isLevelSsrc ?
+      kRtpExtensionAudioLevel : kRtpExtensionCsrcAudioLevel;
+  rtp_header_parser_->DeregisterRtpHeaderExtension(rtpExt);
+  if (enable && !rtp_header_parser_->RegisterRtpHeaderExtension(rtpExt, id)) {
     return -1;
   }
   return 0;
 }
 
 void Channel::EnableSendTransportSequenceNumber(int id) {
   int ret =
       SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id);

--- a/media/webrtc/trunk/webrtc/voice_engine/channel.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel.h
@@ -297,17 +297,17 @@ class Channel
 
   // VoERTP_RTCP
   int SetLocalMID(const char* mid);
   int SetLocalSSRC(unsigned int ssrc);
   int GetLocalSSRC(unsigned int& ssrc);
   int GetRemoteSSRC(unsigned int& ssrc);
   int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
   int SetSendMIDStatus(bool enable, unsigned char id);
-  int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
+  int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id, bool isLevelSsrc);
   int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
   int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
   void EnableSendTransportSequenceNumber(int id);
   void EnableReceiveTransportSequenceNumber(int id);
 
     void RegisterSenderCongestionControlObjects(
         RtpPacketSender* rtp_packet_sender,
         TransportFeedbackObserver* transport_feedback_observer,

--- a/media/webrtc/trunk/webrtc/voice_engine/channel_proxy.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel_proxy.cc
@@ -51,19 +51,21 @@ void ChannelProxy::SetNACKStatus(bool en
 }
 
 void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   int error = channel()->SetSendAudioLevelIndicationStatus(enable, id);
   RTC_DCHECK_EQ(0, error);
 }
 
-void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int id) {
+void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int id,
+                                                        bool isLevelSsrc) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  int error = channel()->SetReceiveAudioLevelIndicationStatus(enable, id);
+  int error = channel()->SetReceiveAudioLevelIndicationStatus(enable, id,
+                                                              isLevelSsrc);
   RTC_DCHECK_EQ(0, error);
 }
 
 void ChannelProxy::EnableSendTransportSequenceNumber(int id) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel()->EnableSendTransportSequenceNumber(id);
 }
 

--- a/media/webrtc/trunk/webrtc/voice_engine/channel_proxy.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel_proxy.h
@@ -50,17 +50,18 @@ class ChannelProxy {
   explicit ChannelProxy(const ChannelOwner& channel_owner);
   virtual ~ChannelProxy();
 
   virtual void SetRTCPStatus(bool enable);
   virtual void SetLocalSSRC(uint32_t ssrc);
   virtual void SetRTCP_CNAME(const std::string& c_name);
   virtual void SetNACKStatus(bool enable, int max_packets);
   virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
-  virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
+  virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id,
+                                                    bool isLevelSsrc = true);
   virtual void EnableSendTransportSequenceNumber(int id);
   virtual void EnableReceiveTransportSequenceNumber(int id);
   virtual void RegisterSenderCongestionControlObjects(
       RtpPacketSender* rtp_packet_sender,
       TransportFeedbackObserver* transport_feedback_observer,
       PacketRouter* packet_router);
   virtual void RegisterReceiverCongestionControlObjects(
       PacketRouter* packet_router);

--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_rtp_rtcp.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_rtp_rtcp.h
@@ -129,17 +129,18 @@ class WEBRTC_DLLEXPORT VoERTP_RTCP {
 
   // Sets the status of sending MID on a specific |channel|.
   virtual int SetSendMIDStatus(int channel, bool enable, unsigned char id = 1) = 0;
 
   // Sets the status of receiving rtp-audio-level-indication on a specific
   // |channel|.
   virtual int SetReceiveAudioLevelIndicationStatus(int channel,
                                                    bool enable,
-                                                   unsigned char id = 1) {
+                                                   unsigned char id = 1,
+                                                   bool isLevelSsrc = true) {
     // TODO(wu): Remove default implementation once talk is updated.
     return 0;
   }
 
   // Sets the RTCP status on a specific |channel|.
   virtual int SetRTCPStatus(int channel, bool enable) = 0;
 
   // Gets the RTCP status on a specific |channel|.

--- a/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
@@ -166,17 +166,18 @@ int VoERTP_RTCPImpl::SetSendMIDStatus(in
     return -1;
   }
   return channelPtr->SetSendMIDStatus(enable, id);
 }
 
 
 int VoERTP_RTCPImpl::SetReceiveAudioLevelIndicationStatus(int channel,
                                                           bool enable,
-                                                          unsigned char id) {
+                                                          unsigned char id,
+                                                          bool isLevelSsrc) {
   WEBRTC_TRACE(
       kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
       "SetReceiveAudioLevelIndicationStatus(channel=%d, enable=%d, id=%u)",
       channel, enable, id);
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
   }
@@ -193,17 +194,18 @@ int VoERTP_RTCPImpl::SetReceiveAudioLeve
   voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
   voe::Channel* channel_ptr = ch.channel();
   if (channel_ptr == NULL) {
     _shared->SetLastError(
         VE_CHANNEL_NOT_VALID, kTraceError,
         "SetReceiveAudioLevelIndicationStatus() failed to locate channel");
     return -1;
   }
-  return channel_ptr->SetReceiveAudioLevelIndicationStatus(enable, id);
+  return channel_ptr->SetReceiveAudioLevelIndicationStatus(enable, id,
+                                                           isLevelSsrc);
 }
 
 int VoERTP_RTCPImpl::SetRTCPStatus(int channel, bool enable) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetRTCPStatus(channel=%d, enable=%d)", channel, enable);
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;

--- a/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.h
@@ -39,17 +39,18 @@ class VoERTP_RTCPImpl : public VoERTP_RT
 
   // RTP Header Extension for Client-to-Mixer Audio Level Indication
   int SetSendAudioLevelIndicationStatus(int channel,
                                         bool enable,
                                         unsigned char id) override;
   int SetSendMIDStatus(int channel, bool enable, unsigned char id = 1) override;
   int SetReceiveAudioLevelIndicationStatus(int channel,
                                            bool enable,
-                                           unsigned char id) override;
+                                           unsigned char id,
+                                           bool isLevelSsrc = true) override;
 
   // Statistics
   int GetRTPStatistics(int channel,
                        unsigned int& averageJitterMs,
                        unsigned int& maxJitterMs,
                        unsigned int& discardedPackets,
                        unsigned int& cumulativeLost) override;