diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 517f0deb60..190693cbc3 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -191,8 +191,6 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   stats.payload_bytes_rcvd = call_stats.payload_bytes_rcvd;
   stats.header_and_padding_bytes_rcvd =
       call_stats.header_and_padding_bytes_rcvd;
-  stats.bytes_rcvd =
-      stats.payload_bytes_rcvd + stats.header_and_padding_bytes_rcvd;
   stats.packets_rcvd = call_stats.packetsReceived;
   stats.packets_lost = call_stats.cumulativeLost;
   stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 3fd8a8afe6..7ef9852838 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -443,8 +443,6 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
   stats.payload_bytes_sent = call_stats.payload_bytes_sent;
   stats.header_and_padding_bytes_sent =
       call_stats.header_and_padding_bytes_sent;
-  stats.bytes_sent =
-      stats.payload_bytes_sent + stats.header_and_padding_bytes_sent;
   stats.retransmitted_bytes_sent = call_stats.retransmitted_bytes_sent;
   stats.packets_sent = call_stats.packetsSent;
   stats.retransmitted_packets_sent = call_stats.retransmitted_packets_sent;
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 2999c3cb87..1f8ad1090e 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -36,9 +36,6 @@ class AudioReceiveStream {
     Stats();
     ~Stats();
     uint32_t remote_ssrc = 0;
-    // TODO(nisse): Sum of below two values. Deprecated, delete as soon as
-    // downstream applications are updated.
-    int64_t bytes_rcvd;
     int64_t payload_bytes_rcvd = 0;
     int64_t header_and_padding_bytes_rcvd = 0;
     uint32_t packets_rcvd = 0;
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index f2dab9a3d3..f3730551dc 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -43,9 +43,6 @@ class AudioSendStream {
 
     // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
     uint32_t local_ssrc = 0;
-    // TODO(nisse): Sum of below two values. Deprecated, delete as soon as
-    // downstream applications are updated.
-    int64_t bytes_sent;
     int64_t payload_bytes_sent = 0;
     int64_t header_and_padding_bytes_sent = 0;
     // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index c3e8be57a3..c20e03950e 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -393,8 +393,8 @@ struct MediaSenderInfo {
       return 0;
     }
   }
-  // TODO(nisse): Sum of below two values. Deprecated, delete as soon as
-  // downstream applications are updated.
+  // TODO(bugs.webrtc.org/10525): Unused, delete as soon as downstream code is
+  // updated.
   int64_t bytes_sent;
   // https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-bytessent
   int64_t payload_bytes_sent = 0;
@@ -453,8 +453,8 @@ struct MediaReceiverInfo {
     }
   }
 
-  // TODO(nisse): Sum of below two values. Deprecated, delete as soon as
-  // downstream applications are updated.
+  // TODO(bugs.webrtc.org/10525): Unused, delete as soon as downstream code is
+  // updated.
   int64_t bytes_rcvd;
   // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-bytesreceived
   int64_t payload_bytes_rcvd = 0;
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 294e3be6f6..7bd7b49f3b 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -2405,9 +2405,6 @@ VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo(
       info.report_block_datas.push_back(stream_stats.report_block_data.value());
     }
   }
-  info.bytes_sent =
-      info.payload_bytes_sent + info.header_and_padding_bytes_sent;
-
   if (!stats.substreams.empty()) {
     // TODO(pbos): Report fraction lost per SSRC.
     webrtc::VideoSendStream::StreamStats first_stream_stats =
@@ -2799,8 +2796,6 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
   info.header_and_padding_bytes_rcvd =
       stats.rtp_stats.packet_counter.header_bytes +
       stats.rtp_stats.packet_counter.padding_bytes;
-  info.bytes_rcvd =
-      info.payload_bytes_rcvd + info.header_and_padding_bytes_rcvd;
   info.packets_rcvd = stats.rtp_stats.packet_counter.packets;
   info.packets_lost = stats.rtp_stats.packets_lost;
 
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index a7d40b5059..08fe73fb37 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2173,8 +2173,6 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     sinfo.add_ssrc(stats.local_ssrc);
     sinfo.payload_bytes_sent = stats.payload_bytes_sent;
     sinfo.header_and_padding_bytes_sent = stats.header_and_padding_bytes_sent;
-    sinfo.bytes_sent =
-        sinfo.payload_bytes_sent + sinfo.header_and_padding_bytes_sent;
     sinfo.retransmitted_bytes_sent = stats.retransmitted_bytes_sent;
     sinfo.packets_sent = stats.packets_sent;
     sinfo.retransmitted_packets_sent = stats.retransmitted_packets_sent;
@@ -2219,8 +2217,6 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     rinfo.add_ssrc(stats.remote_ssrc);
     rinfo.payload_bytes_rcvd = stats.payload_bytes_rcvd;
     rinfo.header_and_padding_bytes_rcvd = stats.header_and_padding_bytes_rcvd;
-    rinfo.bytes_rcvd =
-        rinfo.payload_bytes_rcvd + rinfo.header_and_padding_bytes_rcvd;
     rinfo.packets_rcvd = stats.packets_rcvd;
     rinfo.fec_packets_received = stats.fec_packets_received;
     rinfo.fec_packets_discarded = stats.fec_packets_discarded;