- From: Venkatesh Kuppan via GitHub <sysbot+gh@w3.org>
- Date: Tue, 06 Nov 2018 10:18:02 +0000
- To: public-webrtc-logs@w3.org
@ilyanikolaevskiy So far, I haven't used WebRTC in my application. I was looking into WebRTC to analyse if it can be used here. My application uses GStreamer framework, which provides RTP plugins and timestamps, video rendering plugins, encode/decode plugins, etc. So there are no browser involved as of now, its just plain video rendering using the GStreamer plugins. Here are the pipelines: Tx(iMX6 device): `v4l2src fps-n=30 -> h264encode -> rtph264pay -> rtpbin -> udpsink(port=5000) -> rtpbin.send_rtcp(port=5001) -> rtpbin.recv_rtcp(port=5002) ` Rx(Ubuntu PC) `udpsrc(port=5000) -> rtpbin -> rtph264depay -> avdec_h264 -> rtpbin.recv_rtcp(port=5000) -> rtpbin.send_rtcp(port=5000) -> custom IMU frame insertion plugin -> videosink ` [Here](http://gstreamer-devel.966125.n4.nabble.com/Does-it-make-sense-to-consider-timestamps-of-each-individual-video-frames-RTP-buffers-applied-at-Tx--td4688910.html), I have explained the issue in a more detailed way. Please let me know if you need more information. Regards -- GitHub Notification of comment by venkatesh-kuppan Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/229#issuecomment-436201966 using your GitHub account
Received on Tuesday, 6 November 2018 10:18:04 UTC