diff --git a/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp b/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
index e95ff3ee957c..01ba45ed8a2a 100755
--- a/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
+++ b/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp
@@ -578,6 +578,7 @@ WebrtcVideoConduit::ConfigureSendMediaCodec(const VideoCodecConfig* codecConfig)
     video_stream.max_qp = kQpMax;
     simulcast_config.jsScaleDownBy = simulcastEncoding.constraints.scaleDownBy;
     simulcast_config.jsMaxBitrate = simulcastEncoding.constraints.maxBr; // bps
+    mSendStreamConfig.rtp.rids.push_back(simulcastEncoding.rid);
 
     if (codecConfig->mName == "H264") {
       if (codecConfig->mEncodingConstraints.maxMbps > 0) {
diff --git a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 3072a512ab97..01cf0a6bee22 100644
--- a/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -204,6 +204,10 @@ RTPSender::~RTPSender() {
     delete it->second;
     payload_type_map_.erase(it);
   }
+
+  if (rid_) {
+    delete[] rid_;
+  }
 }
 
 void RTPSender::SetTargetBitrate(uint32_t bitrate) {
diff --git a/media/webrtc/trunk/webrtc/video/video_send_stream.cc b/media/webrtc/trunk/webrtc/video/video_send_stream.cc
index fd7c8079ef43..ae75d38b0464 100644
--- a/media/webrtc/trunk/webrtc/video/video_send_stream.cc
+++ b/media/webrtc/trunk/webrtc/video/video_send_stream.cc
@@ -185,7 +185,8 @@ VideoSendStream::VideoSendStream(
     } else if (extension == RtpExtension::kTransportSequenceNumber) {
       RTC_CHECK_EQ(0, vie_channel_->SetSendTransportSequenceNumber(true, id));
     } else if (extension == RtpExtension::kRtpStreamId) {
-      RTC_CHECK_EQ(0, vie_channel_->SetSendRtpStreamId(true,id));
+      RTC_CHECK_EQ(ssrcs.size(), config_.rtp.rids.size());
+      RTC_CHECK_EQ(0, vie_channel_->SetSendRtpStreamId(true,id,config_.rtp.rids));
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
diff --git a/media/webrtc/trunk/webrtc/video/vie_channel.cc b/media/webrtc/trunk/webrtc/video/vie_channel.cc
index 03527a2f07b9..4b7d72e54f56 100644
--- a/media/webrtc/trunk/webrtc/video/vie_channel.cc
+++ b/media/webrtc/trunk/webrtc/video/vie_channel.cc
@@ -655,23 +655,24 @@ int ViEChannel::SetReceiveTransportSequenceNumber(bool enable, int id) {
   return vie_receiver_.SetReceiveTransportSequenceNumber(enable, id) ? 0 : -1;
 }
 
-int ViEChannel::SetSendRtpStreamId(bool enable, int id) { //}, const char *rid)
-  CriticalSectionScoped cs(crit_.get());
+int ViEChannel::SetSendRtpStreamId(bool enable, int id,
+                                   std::vector<std::string> rids) {
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    rtp_rtcp->DeregisterSendRtpHeaderExtension(
+      kRtpExtensionRtpStreamId);
+  }
+  if (!enable) {
+    return 0;
+  }
   int error = 0;
-  if (enable) {
-    // Enable the extension, but disable possible old id to avoid errors.
-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-      rtp_rtcp->DeregisterSendRtpHeaderExtension(
-        kRtpExtensionRtpStreamId);
-      error = rtp_rtcp->RegisterSendRtpHeaderExtension(
-        kRtpExtensionRtpStreamId, id);
-    }
-    // NOTE: simulcast streams must be set via the SetSendCodec() API
-  } else {
-    // Disable the extension.
-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    	rtp_rtcp->DeregisterSendRtpHeaderExtension(
-        kRtpExtensionRtpStreamId);
+  unsigned long i = 0;
+  // Enable the extension
+  // NOTE: simulcast streams must be set via the SetSendCodec() API
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
+      kRtpExtensionRtpStreamId, id);
+    if (rids.size() > i) {
+      rtp_rtcp->SetRID(rids[i++].c_str());
     }
   }
   return error;
diff --git a/media/webrtc/trunk/webrtc/video/vie_channel.h b/media/webrtc/trunk/webrtc/video/vie_channel.h
index a1102760851c..c75a8d5c8edd 100644
--- a/media/webrtc/trunk/webrtc/video/vie_channel.h
+++ b/media/webrtc/trunk/webrtc/video/vie_channel.h
@@ -113,8 +113,8 @@ class ViEChannel : public VCMFrameTypeCallback,
   int SetReceiveVideoRotationStatus(bool enable, int id);
   int SetSendTransportSequenceNumber(bool enable, int id);
   int SetReceiveTransportSequenceNumber(bool enable, int id);
-  int SetSendRtpStreamId(bool enable, int id); // RtpStreamId (RID)
-	int SetReceiveRtpStreamId(bool enable, int id); // RtpStreamId (RID)
+  int SetSendRtpStreamId(bool enable, int id, std::vector<std::string> rids);
+  int SetReceiveRtpStreamId(bool enable, int id);
   void SetRtcpXrRrtrStatus(bool enable);
   void EnableTMMBR(bool enable);
 
diff --git a/media/webrtc/trunk/webrtc/video_send_stream.h b/media/webrtc/trunk/webrtc/video_send_stream.h
index 0d9fa57bef13..03c0c716f5fe 100644
--- a/media/webrtc/trunk/webrtc/video_send_stream.h
+++ b/media/webrtc/trunk/webrtc/video_send_stream.h
@@ -130,6 +130,8 @@ class VideoSendStream : public SendStream {
 
       // RTCP CNAME, see RFC 3550.
       std::string c_name;
+
+      std::vector<std::string> rids;
     } rtp;
 
     // Transport for outgoing packets.