RTC2RTMP: Fix screen sharing stutter caused by packet loss. v5.0.216 v6.0.157 v7.0.18 #4160
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winlinvip
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winlinvip
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…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>
winlinvip
added a commit
that referenced
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Oct 15, 2024
…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>
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How to reproduce?
trunk/research/players/whip.htmlandtrunk/research/players/js/srs.sdk.jsto replace thedevelopbranch source code../configure && make./objs/srs -c conf/rtc2rtmp.confhttp://localhost:8080/players/whip.html?schema=httpScreenradio option.publish, then check the screen to share.rtmp://localhost/live/livestreamCause
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.
Add screen stream to WHIP demo