diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index ebd46b02e..a5b42ab3e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -10,6 +10,7 @@
 
 #include <string.h>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/h264_sps_parser.h"
@@ -316,6 +317,11 @@ bool RtpDepacketizerH264::Parse(ParsedPayload* parsed_payload,
                                 const uint8_t* payload_data,
                                 size_t payload_data_length) {
   assert(parsed_payload != NULL);
+  if (payload_data_length == 0) {
+    LOG(LS_ERROR) << "Empty payload.";
+    return false;
+  }
+
   uint8_t nal_type = payload_data[0] & kTypeMask;
   size_t offset = 0;
   if (nal_type == kFuA) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index 66a19ddf5..8fa83010f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -537,4 +537,12 @@ TEST_F(RtpDepacketizerH264Test, TestFuA) {
   EXPECT_EQ(kH264FuA, payload.type.Video.codecHeader.H264.packetization_type);
   EXPECT_EQ(kIdr, payload.type.Video.codecHeader.H264.nalu_type);
 }
+
+TEST_F(RtpDepacketizerH264Test, TestEmptyPayload) {
+  // Using a wild pointer to crash on accesses from inside the depacketizer.
+  uint8_t* garbage_ptr = reinterpret_cast<uint8_t*>(0x4711);
+  RtpDepacketizer::ParsedPayload payload;
+  EXPECT_FALSE(depacketizer_->Parse(&payload, garbage_ptr, 0));
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
index 1fa288aca..39b64c6ff 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
@@ -10,6 +10,7 @@
 
 #include <string>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 
@@ -90,6 +91,10 @@ bool RtpDepacketizerGeneric::Parse(ParsedPayload* parsed_payload,
                                    const uint8_t* payload_data,
                                    size_t payload_data_length) {
   assert(parsed_payload != NULL);
+  if (payload_data_length == 0) {
+    LOG(LS_ERROR) << "Empty payload.";
+    return false;
+  }
 
   uint8_t generic_header = *payload_data++;
   --payload_data_length;