diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index cc8b1f7842..e3d34727f6 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -333,6 +333,7 @@ int32_t ChannelSend::SendRtpAudio(FrameType frameType,
   // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
   if (!_rtpRtcpModule->SendOutgoingData((FrameType&)frameType, payloadType,
                                         timeStamp,
+                                        // TODO(https://bugs.webrtc.org/9905):
                                         // Leaving the time when this frame was
                                         // received from the capture device as
                                         // undefined for voice for now.
diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index 67fdf34d6f..b2aad1e7cb 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -258,6 +258,15 @@ void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
                                 int64_t capture_time_ms,
                                 int8_t payload_type) {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
+  // Workaround for https://bugs.webrtc.org/9905
+  // Only very first SetLastRtpTime for audio should update
+  // last_frame_capture_time_ms_ and last_payload_type_.
+  // This eliminates jitter between last rtp and capture timestamps.
+  // TODO(https://bugs.webrtc.org/9905): remove once the bug is fixed.
+  if (capture_time_ms < 0 && last_frame_capture_time_ms_ > 0 &&
+      payload_type != -1 && last_payload_type_ == payload_type) {
+    return;
+  }
   // For compatibility with clients who don't set payload type correctly on all
   // calls.
   if (payload_type != -1) {