diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
index fc288b741..14ca821f7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -254,12 +254,10 @@ bool RTPPacketHistory::GetPacketAndSetSendTime(uint16_t sequence_number,
       ((now - stored_send_times_.at(index)) < min_elapsed_time_ms)) {
     WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, 
         "Skip getting packet %u, packet recently resent.", sequence_number);
-    *packet_length = 0;
-    return true;
+    return false;
   }
 
-  if (length == 0 ||
-      (retransmit && stored_types_.at(index) == kDontRetransmit)) {
+  if (retransmit && stored_types_.at(index) == kDontRetransmit) {
     // No bytes copied since this packet shouldn't be retransmitted or is
     // of zero size.
     return false;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index d0bbf9222..1682b7c33 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -249,8 +249,7 @@ TEST_F(RtpPacketHistoryTest, MinResendTime) {
 
   // Time has not elapsed. Packet should be found, but no bytes copied.
   len = kMaxPacketLength;
-  EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, 101, false, packet_, &len,
-                                             &time));
-  EXPECT_EQ(0, len);
+  EXPECT_FALSE(hist_->GetPacketAndSetSendTime(kSeqNum, 101, false, packet_,
+                                              &len, &time));
 }
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 60be77474..0f08573ef 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -560,7 +560,12 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) {
 
   ModuleRTPUtility::RTPHeaderParser rtp_parser(data_buffer, length);
   RTPHeader header;
-  rtp_parser.Parse(header);
+  if (!rtp_parser.Parse(header)) {
+    assert(false);
+    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
+                 "Failed to parse RTP header of packet to be retransmitted.");
+    return -1;
+  }
   TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::ReSendPacket",
                        "timestamp", header.timestamp,
                        "seqnum", header.sequenceNumber);