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Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out.

Browse Source 
BUG=2954
R=mflodman@webrtc.org, stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5786 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
solenberg@webrtc.org 2014-03-25 19:57:07 +00:00
parent 6cd201cf31
commit 440fa23553
2 changed files with 77 additions and 74 deletions
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145
webrtc/modules/rtp_rtcp/source/rtp_utility.cc
View File

@ -457,6 +457,8 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
// | ID | len |
// +-+-+-+-+-+-+-+-+
// Note that 'len' is the header extension element length, which is the
// number of bytes - 1.
const uint8_t id = (*ptr & 0xf0) >> 4;
const uint8_t len = (*ptr & 0x0f);
ptr++;
@ -469,84 +471,85 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
RTPExtensionType type;
if (ptrExtensionMap->GetType(id, &type) != 0) {
// If we encounter an unknown extension, just skip over it.
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
"Failed to find extension id: %d", id);
return;
}
} else {
switch (type) {
case kRtpExtensionTransmissionTimeOffset: {
if (len != 2) {
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
"Incorrect transmission time offset len: %d", len);
return;
}
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | transmission offset |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
switch (type) {
case kRtpExtensionTransmissionTimeOffset: {
if (len != 2) {
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
"Incorrect transmission time offset len: %d", len);
int32_t transmissionTimeOffset = ptr[0] << 16;
transmissionTimeOffset += ptr[1] << 8;
transmissionTimeOffset += ptr[2];
header.extension.transmissionTimeOffset =
transmissionTimeOffset;
if (transmissionTimeOffset & 0x800000) {
// Negative offset, correct sign for Word24 to Word32.
header.extension.transmissionTimeOffset |= 0xFF000000;
}
header.extension.hasTransmissionTimeOffset = true;
break;
}
case kRtpExtensionAudioLevel: {
if (len != 0) {
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
"Incorrect audio level len: %d", len);
return;
}
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=0 |V| level | 0x00 | 0x00 |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
// Parse out the fields but only use it for debugging for now.
// const uint8_t V = (*ptr & 0x80) >> 7;
// const uint8_t level = (*ptr & 0x7f);
// DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u,
// level=%u", ID, len, V, level);
header.extension.audioLevel = ptr[0];
header.extension.hasAudioLevel = true;
break;
}
case kRtpExtensionAbsoluteSendTime: {
if (len != 2) {
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
"Incorrect absolute send time len: %d", len);
return;
}
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | absolute send time |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
uint32_t absoluteSendTime = ptr[0] << 16;
absoluteSendTime += ptr[1] << 8;
absoluteSendTime += ptr[2];
header.extension.absoluteSendTime = absoluteSendTime;
header.extension.hasAbsoluteSendTime = true;
break;
}
default: {
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
"Extension type not implemented.");
return;
}
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | transmission offset |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
int32_t transmissionTimeOffset = *ptr++ << 16;
transmissionTimeOffset += *ptr++ << 8;
transmissionTimeOffset += *ptr++;
header.extension.transmissionTimeOffset =
transmissionTimeOffset;
if (transmissionTimeOffset & 0x800000) {
// Negative offset, correct sign for Word24 to Word32.
header.extension.transmissionTimeOffset |= 0xFF000000;
}
header.extension.hasTransmissionTimeOffset = true;
break;
}
case kRtpExtensionAudioLevel: {
if (len != 0) {
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
"Incorrect audio level len: %d", len);
return;
}
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=0 |V| level | 0x00 | 0x00 |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
// Parse out the fields but only use it for debugging for now.
// const uint8_t V = (*ptr & 0x80) >> 7;
// const uint8_t level = (*ptr & 0x7f);
// DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u,
// level=%u", ID, len, V, level);
header.extension.audioLevel = *ptr++;
header.extension.hasAudioLevel = true;
break;
}
case kRtpExtensionAbsoluteSendTime: {
if (len != 2) {
WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
"Incorrect absolute send time len: %d", len);
return;
}
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | absolute send time |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
uint32_t absoluteSendTime = *ptr++ << 16;
absoluteSendTime += *ptr++ << 8;
absoluteSendTime += *ptr++;
header.extension.absoluteSendTime = absoluteSendTime;
header.extension.hasAbsoluteSendTime = true;
break;
}
default: {
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
"Extension type not implemented.");
return;
}
}
ptr += (len + 1);
uint8_t num_bytes = ParsePaddingBytes(ptrRTPDataExtensionEnd, ptr);
ptr += num_bytes;
}

6
webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
View File

@ -27,8 +27,7 @@ class ExtensionVerifyTransport : public webrtc::Transport {
ok_packets_(0),
parser_(webrtc::RtpHeaderParser::Create()),
audio_level_id_(-1),
absolute_sender_time_id_(-1) {
}
absolute_sender_time_id_(-1) {}
virtual int SendPacket(int channel, const void* data, int len) {
++received_packets_;
@ -128,7 +127,8 @@ TEST_F(SendRtpRtcpHeaderExtensionsTest, SentPacketsIncludeAllExtensions2) {
EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true,
9));
verifying_transport_.SetAbsoluteSenderTimeId(3);
verifying_transport_.SetAudioLevelId(9);
// Don't register audio level with header parser - unknown extensions should
// be ignored when parsing.
ResumePlaying();
EXPECT_TRUE(verifying_transport_.WaitForNPackets(10));
}