diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index f447b3e889..2818f95c0e 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -15,6 +15,7 @@ rtc_source_set("rtp_rtcp_format") {
     "include/rtcp_statistics.h",
     "include/rtp_cvo.h",
     "include/rtp_header_extension_map.h",
+    "include/rtp_packet_pacer.h",
     "include/rtp_rtcp_defines.h",
     "source/byte_io.h",
     "source/rtcp_packet.h",
diff --git a/modules/rtp_rtcp/include/rtp_packet_pacer.h b/modules/rtp_rtcp/include/rtp_packet_pacer.h
new file mode 100644
index 0000000000..9820fc2a46
--- /dev/null
+++ b/modules/rtp_rtcp/include/rtp_packet_pacer.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_PACER_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_PACER_H_
+
+#include <memory>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+namespace webrtc {
+
+// Interface for a paced sender, as implemented in the pacing module.
+// This intended to replace the RtpPacketSender interface defined in
+// rtp_rtcp_defines.h
+// TODO(bugs.webrtc.org/10633): Add things missing to this interface so that we
+// can use multiple different pacer implementations, and stop inheriting from
+// RtpPacketSender.
+class RtpPacketPacer : RtpPacketSender {
+ public:
+  RtpPacketPacer() = default;
+  ~RtpPacketPacer() override;
+
+  // Insert packet into queue, for eventual transmission. Based on the type of
+  // the packet, it will prioritized and scheduled relative to other packets and
+  // the current target send rate.
+  virtual void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_PACER_H_
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 5d906126f9..d666ab4239 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -340,21 +340,21 @@ class RtcpRttStats {
   virtual ~RtcpRttStats() {}
 };
 
+// This class will be deprecated and replaced with RtpPacketPacer.
 class RtpPacketSender {
  public:
   RtpPacketSender() {}
   virtual ~RtpPacketSender() {}
 
+  // These are part of the legacy PacedSender interface and will be removed.
   enum Priority {
     kHighPriority = 0,    // Pass through; will be sent immediately.
     kNormalPriority = 2,  // Put in back of the line.
     kLowPriority = 3,     // Put in back of the low priority line.
   };
-  // Low priority packets are mixed with the normal priority packets
-  // while we are paused.
 
-  // Returns true if we send the packet now, else it will add the packet
-  // information to the queue and call TimeToSendPacket when it's time to send.
+  // Adds the packet information to the queue and call TimeToSendPacket when
+  // it's time to send.
   virtual void InsertPacket(Priority priority,
                             uint32_t ssrc,
                             uint16_t sequence_number,
@@ -367,7 +367,7 @@ class RtpPacketSender {
   // the pacer budget calculation. The audio traffic still will be injected
   // at high priority.
   // TODO(alexnarest): Make it pure virtual after rtp_sender_unittest will be
-  // updated to support it
+  // updated to support it.
   virtual void SetAccountForAudioPackets(bool account_for_audio) {}
 };
 
diff --git a/modules/rtp_rtcp/source/flexfec_sender.cc b/modules/rtp_rtcp/source/flexfec_sender.cc
index 02fee86aee..d7b8eca91c 100644
--- a/modules/rtp_rtcp/source/flexfec_sender.cc
+++ b/modules/rtp_rtcp/source/flexfec_sender.cc
@@ -126,7 +126,8 @@ std::vector<std::unique_ptr<RtpPacketToSend>> FlexfecSender::GetFecPackets() {
   for (const auto* fec_packet : ulpfec_generator_.generated_fec_packets_) {
     std::unique_ptr<RtpPacketToSend> fec_packet_to_send(
         new RtpPacketToSend(&rtp_header_extension_map_));
-    fec_packet_to_send->set_is_fec(true);
+    fec_packet_to_send->set_packet_type(
+        RtpPacketToSend::Type::kForwardErrorCorrection);
 
     // RTP header.
     fec_packet_to_send->SetMarker(false);
diff --git a/modules/rtp_rtcp/source/rtp_packet_to_send.h b/modules/rtp_rtcp/source/rtp_packet_to_send.h
index 21c9c86a4b..89612841e7 100644
--- a/modules/rtp_rtcp/source/rtp_packet_to_send.h
+++ b/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -14,6 +14,7 @@
 #include <stdint.h>
 #include <vector>
 
+#include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/video/video_timing.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
@@ -23,6 +24,14 @@ namespace webrtc {
 // Class to hold rtp packet with metadata for sender side.
 class RtpPacketToSend : public RtpPacket {
  public:
+  enum class Type {
+    kAudio,                   // Audio media packets.
+    kVideo,                   // Video media packets.
+    kRetransmission,          // RTX (usually) packets send as response to NACK.
+    kForwardErrorCorrection,  // FEC packets.
+    kPadding                  // RTX or plain padding sent to maintain BWE.
+  };
+
   explicit RtpPacketToSend(const ExtensionManager* extensions);
   RtpPacketToSend(const ExtensionManager* extensions, size_t capacity);
   RtpPacketToSend(const RtpPacketToSend& packet);
@@ -38,9 +47,8 @@ class RtpPacketToSend : public RtpPacket {
 
   void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }
 
-  bool is_fec() const { return is_fec_; }
-
-  void set_is_fec(bool fec) { is_fec_ = fec; }
+  void set_packet_type(Type type) { packet_type_ = type; }
+  absl::optional<Type> packet_type() const { return packet_type_; }
 
   // Additional data bound to the RTP packet for use in application code,
   // outside of WebRTC.
@@ -78,8 +86,7 @@ class RtpPacketToSend : public RtpPacket {
 
  private:
   int64_t capture_time_ms_ = 0;
-  // Used for accounting purposes
-  bool is_fec_ = false;
+  absl::optional<Type> packet_type_;
   std::vector<uint8_t> application_data_;
 };
 
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 3c80cae0f1..b7838592df 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -661,8 +661,9 @@ void RTPSender::UpdateRtpStats(const RtpPacketToSend& packet,
   if (counters->first_packet_time_ms == -1)
     counters->first_packet_time_ms = now_ms;
 
-  if (packet.is_fec())
+  if (packet.packet_type() == RtpPacketToSend::Type::kForwardErrorCorrection) {
     counters->fec.AddPacket(packet);
+  }
 
   if (is_retransmit) {
     counters->retransmitted.AddPacket(packet);
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 03110f784f..419e938e3c 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -307,7 +307,7 @@ void RTPSenderVideo::SendVideoPacketAsRedMaybeWithUlpfec(
         new RtpPacketToSend(*media_packet));
     RTC_CHECK(rtp_packet->Parse(fec_packet->data(), fec_packet->length()));
     rtp_packet->set_capture_time_ms(media_packet->capture_time_ms());
-    rtp_packet->set_is_fec(true);
+    rtp_packet->set_packet_type(RtpPacketToSend::Type::kForwardErrorCorrection);
     uint16_t fec_sequence_number = rtp_packet->SequenceNumber();
     if (LogAndSendToNetwork(std::move(rtp_packet), kDontRetransmit,
                             RtpPacketSender::kLowPriority)) {