Fix oversized rtp header extension parsing.

webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
@@ skipping 205 matching lines @@
 
   // Extension should be ignored.
   int32_t time_offset;
   EXPECT_FALSE(packet.GetExtension<TransmissionOffset>(&time_offset));
 
   // But shouldn't prevent reading payload.
   EXPECT_THAT(make_tuple(packet.payload(), packet.payload_size()),
               ElementsAreArray(kPayload));
 }
 
+TEST(RtpPacketTest, ParseWithOverSizedExtension) {
+  // clang-format off
+  const uint8_t bad_packet[] = {
+      0x90, kPayloadType, 0x00, kSeqNum,
+      0x65, 0x43, 0x12, 0x78,  // kTimestamp.
+      0x12, 0x34, 0x56, 0x78,  // kSsrc.
+      0xbe, 0xde, 0x00, 0x01,  // Extension of size 1x32bit word.
+      0x00,  // Add a byte of padding.
+            0x12,  // Extension id 1 size (2+1).
+                  0xda, 0x1a  // Only 2 bytes of extension payload.
+  };
+  // clang-format on
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(TransmissionOffset::kId, 1);
+  RtpPacketReceived packet(&extensions);
+
+  // Parse should ignore bad extension and proceed.
+  EXPECT_TRUE(packet.Parse(bad_packet, sizeof(bad_packet)));
+  int32_t time_offset;
+  // But extracting extension should fail.
+  EXPECT_FALSE(packet.GetExtension<TransmissionOffset>(&time_offset));
+}
+
 TEST(RtpPacketTest, ParseWith2Extensions) {
   RtpPacketToSend::ExtensionManager extensions;
   extensions.Register(kRtpExtensionTransmissionTimeOffset,
                       kTransmissionOffsetExtensionId);
   extensions.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
   RtpPacketReceived packet(&extensions);
   EXPECT_TRUE(packet.Parse(kPacketWithTOAndAL, sizeof(kPacketWithTOAndAL)));
   int32_t time_offset;
   EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
   EXPECT_EQ(kTimeOffset, time_offset);
@@ skipping 37 matching lines @@
   int32_t time_offset;
   EXPECT_FALSE(packet.GetExtension<TransmissionOffset>(&time_offset));
   packet.IdentifyExtensions(&extensions);
   EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
   EXPECT_EQ(kTimeOffset, time_offset);
   EXPECT_EQ(0u, packet.payload_size());
   EXPECT_EQ(0u, packet.padding_size());
 }
 
 }  // namespace webrtc