diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc
index 80fd75b5e8..f6568c3c18 100644
--- a/webrtc/pc/channel.cc
+++ b/webrtc/pc/channel.cc
@@ -748,7 +748,7 @@ bool BaseChannel::HandlesPayloadType(int packet_type) const {
 }
 
 void BaseChannel::OnPacketReceived(bool rtcp,
-                                   rtc::CopyOnWriteBuffer& packet,
+                                   rtc::CopyOnWriteBuffer* packet,
                                    const rtc::PacketTime& packet_time) {
   if (!has_received_packet_ && !rtcp) {
     has_received_packet_ = true;
@@ -758,8 +758,8 @@ void BaseChannel::OnPacketReceived(bool rtcp,
   // Unprotect the packet, if needed.
   if (srtp_filter_.IsActive()) {
     TRACE_EVENT0("webrtc", "SRTP Decode");
-    char* data = packet.data<char>();
-    int len = static_cast<int>(packet.size());
+    char* data = packet->data<char>();
+    int len = static_cast<int>(packet->size());
     bool res;
     if (!rtcp) {
       res = srtp_filter_.UnprotectRtp(data, len, &len);
@@ -784,7 +784,7 @@ void BaseChannel::OnPacketReceived(bool rtcp,
       }
     }
 
-    packet.SetSize(len);
+    packet->SetSize(len);
   } else if (srtp_required_) {
     // Our session description indicates that SRTP is required, but we got a
     // packet before our SRTP filter is active. This means either that
@@ -804,7 +804,7 @@ void BaseChannel::OnPacketReceived(bool rtcp,
 
   invoker_.AsyncInvoke<void>(
       RTC_FROM_HERE, worker_thread_,
-      Bind(&BaseChannel::ProcessPacket, this, rtcp, packet, packet_time));
+      Bind(&BaseChannel::ProcessPacket, this, rtcp, *packet, packet_time));
 }
 
 void BaseChannel::ProcessPacket(bool rtcp,
@@ -1678,7 +1678,7 @@ void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) {
 }
 
 void VoiceChannel::OnPacketReceived(bool rtcp,
-                                    rtc::CopyOnWriteBuffer& packet,
+                                    rtc::CopyOnWriteBuffer* packet,
                                     const rtc::PacketTime& packet_time) {
   BaseChannel::OnPacketReceived(rtcp, packet, packet_time);
   // Set a flag when we've received an RTP packet. If we're waiting for early
diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
index ae3a29a2ff..11d21212f8 100644
--- a/webrtc/pc/channel.h
+++ b/webrtc/pc/channel.h
@@ -271,7 +271,7 @@ class BaseChannel
                     const rtc::PacketTime& packet_time);
   // TODO(zstein): packet can be const once the RtpTransport handles protection.
   virtual void OnPacketReceived(bool rtcp,
-                                rtc::CopyOnWriteBuffer& packet,
+                                rtc::CopyOnWriteBuffer* packet,
                                 const rtc::PacketTime& packet_time);
   void ProcessPacket(bool rtcp,
                      const rtc::CopyOnWriteBuffer& packet,
@@ -505,7 +505,7 @@ class VoiceChannel : public BaseChannel {
  private:
   // overrides from BaseChannel
   void OnPacketReceived(bool rtcp,
-                        rtc::CopyOnWriteBuffer& packet,
+                        rtc::CopyOnWriteBuffer* packet,
                         const rtc::PacketTime& packet_time) override;
   void UpdateMediaSendRecvState_w() override;
   const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override;
diff --git a/webrtc/pc/rtptransport.cc b/webrtc/pc/rtptransport.cc
index 6b0141fc8e..d530c73c50 100644
--- a/webrtc/pc/rtptransport.cc
+++ b/webrtc/pc/rtptransport.cc
@@ -190,7 +190,7 @@ void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
   }
 
   // This mutates |packet| if it is protected.
-  SignalPacketReceived(rtcp, packet, packet_time);
+  SignalPacketReceived(rtcp, &packet, packet_time);
 }
 
 bool RtpTransport::WantsPacket(bool rtcp,
diff --git a/webrtc/pc/rtptransport.h b/webrtc/pc/rtptransport.h
index 87f29bac2a..a86fa126bb 100644
--- a/webrtc/pc/rtptransport.h
+++ b/webrtc/pc/rtptransport.h
@@ -73,7 +73,7 @@ class RtpTransport : public RtpTransportInterface, public sigslot::has_slots<> {
   // TODO(zstein): Consider having two signals - RtcPacketReceived and
   // RtcpPacketReceived.
   // The first argument is true for RTCP packets and false for RTP packets.
-  sigslot::signal3<bool, rtc::CopyOnWriteBuffer&, const rtc::PacketTime&>
+  sigslot::signal3<bool, rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
       SignalPacketReceived;
 
  protected:
diff --git a/webrtc/pc/rtptransport_unittest.cc b/webrtc/pc/rtptransport_unittest.cc
index 4b75a05c80..9b9e6cf377 100644
--- a/webrtc/pc/rtptransport_unittest.cc
+++ b/webrtc/pc/rtptransport_unittest.cc
@@ -162,7 +162,7 @@ class SignalPacketReceivedCounter : public sigslot::has_slots<> {
 
  private:
   void OnPacketReceived(bool rtcp,
-                        rtc::CopyOnWriteBuffer&,
+                        rtc::CopyOnWriteBuffer*,
                         const rtc::PacketTime&) {
     if (rtcp) {
       ++rtcp_count_;