RTC2RTMP: Fix screen sharing stutter caused by packet loss. v5.0.216 (#4160)

1. Refer this commit, which contains the web demo to capture screen as
video stream through RTC.
2. Copy the `trunk/research/players/whip.html` and
`trunk/research/players/js/srs.sdk.js` to replace the `develop` branch
source code.
3. `./configure && make`
4. `./objs/srs -c conf/rtc2rtmp.conf`
5. open `http://localhost:8080/players/whip.html?schema=http`
6. check `Screen` radio option.
7. click `publish`, then check the screen to share.
8. play the rtmp live stream: `rtmp://localhost/live/livestream`
9. check the video stuttering.

When capture screen by the chrome web browser, which send RTP packet
with empty payload frequently, then all the cached RTP packets are
dropped before next key frame arrive in this case.

The OBS screen stream and camera stream do not have such problem.

><img width="581" alt="Screenshot 2024-08-28 at 2 49 46 PM"
src="https://github.com/user-attachments/assets/9557dbd2-c799-4dfd-b336-5bbf2e4f8fb8">

---------

Co-authored-by: winlin <winlinvip@gmail.com>

trunk/doc/CHANGELOG.md

@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v5-changes"></a>
 
 ## SRS 5.0 Changelog
+* v5.0, 2024-10-15, Merge [#4160](https://github.com/ossrs/srs/pull/4160): RTC2RTMP: Fix screen sharing stutter caused by packet loss. v5.0.216 (#4160)
 * v5.0, 2024-09-09, Merge [#4171](https://github.com/ossrs/srs/pull/4171): Heartbeat: Report ports for proxy server. v5.0.215 (#4171)
 * v5.0, 2024-07-24, Merge [#4126](https://github.com/ossrs/srs/pull/4126): Edge: Improve stability for state and fd closing. v5.0.214 (#4126)
 * v5.0, 2024-06-03, Merge [#4057](https://github.com/ossrs/srs/pull/4057): RTC: Support dropping h.264 SEI from NALUs. v5.0.213 (#4057)

trunk/src/app/srs_app_rtc_source.cpp

@@ -1705,6 +1705,23 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video_rtmp(const uint16_t start, const
 
     if (0 == nb_payload) {
         srs_warn("empty nalu");
+
+        // The chrome web browser send RTP packet with empty payload frequently,
+        // reset header_sn_, lost_sn_ and continue to found next frame in this case,
+        // otherwise, all the cached RTP packets are dropped before next key frame arrive.
+        header_sn_ = end + 1;
+        uint16_t tail_sn = 0;
+        int sn = find_next_lost_sn(header_sn_, tail_sn);
+        if (-1 == sn) {
+            if (check_frame_complete(header_sn_, tail_sn)) {
+                err = packet_video_rtmp(header_sn_, tail_sn);
+            }
+        } else if (-2 == sn) {
+            return srs_error_new(ERROR_RTC_RTP_MUXER, "video cache is overflow");
+        } else {
+            lost_sn_ = sn;
+        }
+
         return err;
     }
 	

trunk/src/core/srs_core_version5.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       5
 #define VERSION_MINOR       0
-#define VERSION_REVISION    215
+#define VERSION_REVISION    216
 
 #endif