diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 6954447256..7a4ceb289b 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -1099,8 +1099,6 @@ int64_t ChannelReceive::GetRTT() const {
     return associated_send_channel_->GetRTT();
   }
 
-  // TODO(nisse): This method computes RTT based on sender reports, even though
-  // a receive stream is not supposed to do that.
   for (const ReportBlockData& data : report_blocks) {
     if (data.report_block().sender_ssrc == remote_ssrc_) {
       return data.last_rtt_ms();
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 3665b60c8e..0a24481762 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -669,13 +669,6 @@ void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block,
   // If no SR has been received yet, the field is set to zero.
   // Receiver rtp_rtcp module is not expected to calculate rtt using
   // Sender Reports even if it accidentally can.
-
-  // TODO(nisse): Use this way to determine the RTT only when `receiver_only_`
-  // is false. However, that currently breaks the tests of the
-  // googCaptureStartNtpTimeMs stat for audio receive streams. To fix, either
-  // delete all dependencies on RTT measurements for audio receive streams, or
-  // ensure that audio receive streams that need RTT and stats that depend on it
-  // are configured with an associated audio send stream.
   if (send_time_ntp != 0) {
     uint32_t delay_ntp = report_block.delay_since_last_sr();
     // Local NTP time.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index 6c199af407..b1729c0c3f 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -274,9 +274,6 @@ RTCPSender::FeedbackState ModuleRtpRtcpImpl2::GetFeedbackState() {
   return state;
 }
 
-// TODO(nisse): This method shouldn't be called for a receive-only
-// stream. Delete rtp_sender_ check as soon as all applications are
-// updated.
 int32_t ModuleRtpRtcpImpl2::SetSendingStatus(const bool sending) {
   if (rtcp_sender_.Sending() != sending) {
     // Sends RTCP BYE when going from true to false