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Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust.

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BUG=

Review URL: https://webrtc-codereview.appspot.com/1232005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3726 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
solenberg@webrtc.org 2013-03-26 14:02:30 +00:00
parent 0633cccb4f
commit d8a6e72057
1 changed files with 15 additions and 3 deletions
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18
webrtc/modules/rtp_rtcp/source/rtp_utility.cc
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@ -45,6 +45,11 @@ namespace webrtc {
namespace ModuleRTPUtility {
enum {
kRtcpMinHeaderLength = 4,
kRtcpExpectedVersion = 2
};
/*
* Time routines.
*/
@ -269,11 +274,18 @@ bool RTPHeaderParser::RTCP() const {
* FMT 15: Application layer FB message
*/
const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
if (length < kRtcpMinHeaderLength) {
return false;
}
const WebRtc_UWord8 V = _ptrRTPDataBegin[0] >> 6;
if (V != kRtcpExpectedVersion) {
return false;
}
const WebRtc_UWord8 payloadType = _ptrRTPDataBegin[1];
bool RTCP = false;
// check if this is a RTCP packet
switch (payloadType) {
case 192:
RTCP = true;