(Auto)update libjingle 69568113-> 69587333
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6500 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/media/base/rtputils.cc b/media/base/rtputils.cc
index 5215c3b..221d949 100644
--- a/media/base/rtputils.cc
+++ b/media/base/rtputils.cc
@@ -223,4 +223,15 @@
SetRtpSsrc(data, len, header.ssrc));
}
+bool IsRtpPacket(const void* data, size_t len) {
+ if (len < kMinRtpPacketLen)
+ return false;
+
+ int version = 0;
+ if (!GetRtpVersion(data, len, &version))
+ return false;
+
+ return version == kRtpVersion;
+}
+
} // namespace cricket
diff --git a/media/base/rtputils.h b/media/base/rtputils.h
index 6f76866..f653e42 100644
--- a/media/base/rtputils.h
+++ b/media/base/rtputils.h
@@ -74,6 +74,7 @@
// Assumes version 2, no padding, no extensions, no csrcs.
bool SetRtpHeader(void* data, size_t len, const RtpHeader& header);
+bool IsRtpPacket(const void* data, size_t len);
} // namespace cricket
#endif // TALK_MEDIA_BASE_RTPUTILS_H_
diff --git a/session/media/bundlefilter.cc b/session/media/bundlefilter.cc
index 0d7927c..d3b51c4 100755
--- a/session/media/bundlefilter.cc
+++ b/session/media/bundlefilter.cc
@@ -47,6 +47,10 @@
// |streams_| is empty, we will allow all rtcp packets pass through provided
// that they are valid rtcp packets in case that they are for early media.
if (!rtcp) {
+ // It may not be a RTP packet (e.g. SCTP).
+ if (!IsRtpPacket(data, len))
+ return false;
+
int payload_type = 0;
if (!GetRtpPayloadType(data, len, &payload_type)) {
return false;
diff --git a/session/media/bundlefilter_unittest.cc b/session/media/bundlefilter_unittest.cc
index 0386666..a3e58c1 100755
--- a/session/media/bundlefilter_unittest.cc
+++ b/session/media/bundlefilter_unittest.cc
@@ -105,6 +105,15 @@
0x81, 0xCE, 0x00, 0x0C, 0x00, 0x00, 0x11, 0x11, 0x00, 0x00, 0x11, 0x11,
};
+// An SCTP packet.
+static const unsigned char kSctpPacket[] = {
+ 0x00, 0x01, 0x00, 0x01,
+ 0xff, 0xff, 0xff, 0xff,
+ 0x00, 0x00, 0x00, 0x00,
+ 0x03, 0x00, 0x00, 0x04,
+ 0x00, 0x00, 0x00, 0x00,
+};
+
TEST(BundleFilterTest, AddRemoveStreamTest) {
cricket::BundleFilter bundle_filter;
EXPECT_FALSE(bundle_filter.HasStreams());
@@ -194,3 +203,11 @@
reinterpret_cast<const char*>(kRtcpPacketSrSsrc2),
sizeof(kRtcpPacketSrSsrc2), true));
}
+
+TEST(BundleFilterTest, InvalidRtpPacket) {
+ cricket::BundleFilter bundle_filter;
+ EXPECT_TRUE(bundle_filter.AddStream(StreamParams::CreateLegacy(kSsrc1)));
+ EXPECT_FALSE(bundle_filter.DemuxPacket(
+ reinterpret_cast<const char*>(kSctpPacket),
+ sizeof(kSctpPacket), false));
+}
diff --git a/session/media/channel.cc b/session/media/channel.cc
index 575c759..7bf853e 100644
--- a/session/media/channel.cc
+++ b/session/media/channel.cc
@@ -2168,21 +2168,11 @@
return GetFirstDataContent(sdesc);
}
-
-static bool IsRtpPacket(const talk_base::Buffer* packet) {
- int version;
- if (!GetRtpVersion(packet->data(), packet->length(), &version)) {
- return false;
- }
-
- return version == 2;
-}
-
bool DataChannel::WantsPacket(bool rtcp, talk_base::Buffer* packet) {
if (data_channel_type_ == DCT_SCTP) {
// TODO(pthatcher): Do this in a more robust way by checking for
// SCTP or DTLS.
- return !IsRtpPacket(packet);
+ return !IsRtpPacket(packet->data(), packet->length());
} else if (data_channel_type_ == DCT_RTP) {
return BaseChannel::WantsPacket(rtcp, packet);
}
