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RTC: Fix null pointer crash in RTC2RTMP when start packet is missing. v6.0.175 (#4451)

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Try to fix #4450 

## Cause

The SRS transcode rtp packets, whose sequence number in range [start,
end], to one rtmp packet, but when the first rtp packet is empty, then
this crash happens.

check #4450 for details.

## Impact

5.0release and 6.0release branch.
develop branch already has its own solution.

So this PR is targeting to **6.0release**.

## Solution

find the first not empty rtp packet in seq range [start, end].

---------

Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Co-authored-by: winlin <winlinvip@gmail.com>
This commit is contained in:
Jacob Su 2025-08-27 06:42:50 +08:00 committed by GitHub
parent 2ab3937a68
commit c1a8a5f753
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GPG Key ID: B5690EEEBB952194
5 changed files with 144 additions and 4 deletions
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5
trunk/auto/depends.sh
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@ -671,6 +671,11 @@ fi
if [[ $SRS_SRT == YES && $SRS_USE_SYS_SRT == NO ]]; then
# Always disable c++11 for libsrt, because only the srt-app requres it.
LIBSRT_OPTIONS="--enable-apps=0 --enable-static=1 --enable-c++11=0"
CMAKE_VERSION=$(cmake --version | head -n1 | cut -d' ' -f3)
CMAKE_MAJOR=$(echo $CMAKE_VERSION | cut -d'.' -f1)
if [[ $CMAKE_MAJOR -ge 4 ]]; then
LIBSRT_OPTIONS="$LIBSRT_OPTIONS --CMAKE_POLICY_VERSION_MINIMUM=3.5"
fi
if [[ $SRS_SHARED_SRT == YES ]]; then
LIBSRT_OPTIONS="$LIBSRT_OPTIONS --enable-shared=1"
else

1
trunk/doc/CHANGELOG.md
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@ -7,6 +7,7 @@ The changelog for SRS.
<a name="v6-changes"></a>
## SRS 6.0 Changelog
* v6.0, 2025-08-26, Merge [#4451](https://github.com/ossrs/srs/pull/4451): RTC: Fix null pointer crash in RTC2RTMP when start packet is missing. v6.0.175 (#4451)
* v6.0, 2025-08-16, Merge [#4441](https://github.com/ossrs/srs/pull/4441): fix err memory leak in rtc to rtmp bridge. v6.0.174 (#4441)
* v6.0, 2025-08-14, Merge [#4161](https://github.com/ossrs/srs/pull/4161): fix hls & dash segments cleanup. v6.0.173 (#4161)
* v6.0, 2025-08-12, Merge [#4230](https://github.com/ossrs/srs/pull/4230): MP4 DVR: Fix audio/video synchronization issues in WebRTC recordings. v6.0.172 (#4230)

19
trunk/src/app/srs_app_rtc_source.cpp
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@ -1798,13 +1798,26 @@ srs_error_t SrsRtcFrameBuilder::packet_video_rtmp(const uint16_t start, const ui
//type_codec1 + avc_type + composition time + nalu size + nalu
nb_payload += 1 + 1 + 3;
// find the first not emtpy rtp packet, which is always exist in seq range [start, end],
// because nb_payload check above.
// @see https://github.com/ossrs/srs/issues/4450
SrsRtpPacket* first_pkt = NULL;
for (uint16_t i = 0; i < (uint16_t)cnt; ++i) {
uint16_t index = cache_index(start + i);
SrsRtpPacket* pkt = cache_video_pkts_[index].pkt;
if (pkt) {
first_pkt = pkt;
break;
}
}
SrsCommonMessage rtmp;
SrsRtpPacket* pkt = cache_video_pkts_[cache_index(start)].pkt;
rtmp.header.initialize_video(nb_payload, pkt->get_avsync_time(), 1);
rtmp.header.initialize_video(nb_payload, first_pkt->get_avsync_time(), 1);
rtmp.create_payload(nb_payload);
rtmp.size = nb_payload;
SrsBuffer payload(rtmp.payload, rtmp.size);
if (pkt->is_keyframe()) {
if (first_pkt->is_keyframe()) {
payload.write_1bytes(0x17); // type(4 bits): key frame; code(4bits): avc
rtp_key_frame_ts_ = -1;
} else {

2
trunk/src/core/srs_core_version6.hpp
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@ -9,6 +9,6 @@
#define VERSION_MAJOR 6
#define VERSION_MINOR 0
#define VERSION_REVISION 174
#define VERSION_REVISION 175
#endif

121
trunk/src/utest/srs_utest_rtc.cpp
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@ -1365,3 +1365,124 @@ VOID TEST(KernelRTCTest, JitterSequence)
EXPECT_EQ((uint32_t)11, jitter.correct(11));
}
// Helper function to create a test RTP packet
SrsRtpPacket *mock_create_test_rtp_packet(uint16_t sequence_number, uint32_t timestamp, bool marker = false)
{
SrsRtpPacket *pkt = new SrsRtpPacket();
pkt->header.set_sequence(sequence_number);
pkt->header.set_timestamp(timestamp);
pkt->header.set_marker(marker);
pkt->header.set_ssrc(12345);
return pkt;
}
// Mock bridge for testing SrsRtcFrameBuilder
class MockStreamBridge : public ISrsStreamBridge
{
public:
srs_error_t last_error;
int frame_count;
MockStreamBridge() {
last_error = srs_success;
frame_count = 0;
}
virtual ~MockStreamBridge() {
srs_freep(last_error);
}
virtual srs_error_t initialize(SrsRequest *r) {
return srs_success;
}
virtual srs_error_t on_publish() {
return srs_success;
}
virtual srs_error_t on_frame(SrsSharedPtrMessage *frame) {
frame_count++;
return srs_success;
}
virtual void on_unpublish() {
}
};
VOID TEST(KernelRTC2Test, SrsRtcFrameBuilderPacketVideoRtmpNullPointerCrash)
{
srs_error_t err;
// Test reproducing the null pointer crash from issue #4450 fixed by PR #4451
//
// ISSUE BACKGROUND:
// Before PR 4451, the packet_video_rtmp() function assumed that the packet at the
// 'start' sequence number would always be available in the cache. However, due to
// network packet loss or reordering, the packet at the start position could be missing.
//
// THE CRASH:
// The original code did: pkt = cache_video_pkts_[cache_index(start)].pkt;
// When pkt was NULL, calling pkt->get_avsync_time() caused a segmentation fault.
//
// THE FIX:
// PR #4451 added a loop to find the first non-null packet in the sequence range
// instead of blindly using the packet at the start position.
//
// This test simulates the crash scenario: packets exist at positions 101, 102, 103
// but the start packet (100) is missing. With the fix, the function should use
// packet 101 (first available) instead of crashing on the missing packet 100.
if (true) {
MockStreamBridge bridge;
SrsRtcFrameBuilder frame_builder(&bridge);
// Skip initialization and directly set up the test scenario
// We only need to test the packet_video_rtmp function, not the full initialization
// Manually populate the video cache to simulate the crash scenario
// We'll store packets at positions 101, 102, 103 but NOT at position 100 (start)
// This simulates network packet loss where the first packet is missing
// Create test packets with payload to ensure nb_payload > 0
SrsRtpPacket *pkt101 = mock_create_test_rtp_packet(101, 1000, false);
SrsRtpPacket *pkt102 = mock_create_test_rtp_packet(102, 1000, false);
SrsRtpPacket *pkt103 = mock_create_test_rtp_packet(103, 1000, true); // marker bit
// Add some payload to ensure packets are not empty
char payload_data[] = "test_payload_data";
SrsRtpRawPayload *payload101 = new SrsRtpRawPayload();
payload101->payload = (char*)payload_data;
payload101->nn_payload = strlen(payload_data);
pkt101->set_payload(payload101, SrsRtspPacketPayloadTypeRaw);
SrsRtpRawPayload *payload102 = new SrsRtpRawPayload();
payload102->payload = (char*)payload_data;
payload102->nn_payload = strlen(payload_data);
pkt102->set_payload(payload102, SrsRtspPacketPayloadTypeRaw);
SrsRtpRawPayload *payload103 = new SrsRtpRawPayload();
payload103->payload = (char*)payload_data;
payload103->nn_payload = strlen(payload_data);
pkt103->set_payload(payload103, SrsRtspPacketPayloadTypeRaw);
// Set the avsync time for the packets to avoid other null pointer issues
pkt101->set_avsync_time(1000);
pkt102->set_avsync_time(1000);
pkt103->set_avsync_time(1000);
// Store packets in cache (but skip sequence 100 - this is the missing start packet)
frame_builder.cache_video_pkts_[frame_builder.cache_index(pkt101->header.get_sequence())].pkt = pkt101;
frame_builder.cache_video_pkts_[frame_builder.cache_index(pkt102->header.get_sequence())].pkt = pkt102;
frame_builder.cache_video_pkts_[frame_builder.cache_index(pkt103->header.get_sequence())].pkt = pkt103;
// Before the fix in PR #4451, calling packet_video_rtmp(100, 103) would crash
// because it would try to access cache_video_pkts_[cache_index(100)].pkt which is NULL
// and then call pkt->get_avsync_time() causing a null pointer dereference
// The fix ensures we find the first non-null packet (101) instead of assuming
// the start packet (100) exists
HELPER_EXPECT_SUCCESS(frame_builder.packet_video_rtmp(100, 103));
// Verify that a frame was successfully processed
EXPECT_EQ(1, bridge.frame_count);
}
}