I’ve been facing lot of issues with webrtcbin with regards to artifcating (packet-loss) as I’ve posted on a previous thread and have created an issue. I assumed it was something I was doing wrong and decided to switch the webrtcsrc from gst-plugins-rs.
I realized it’s better to run the webrtc-precise-sync-recv example before fully committing to the rewrite.
I modified the source code of gstwebrtc-api to use getDisplayMedia instead of getUserMedia.
I opened 3 tabs on my terminal. I ran the following in each tab from the root of the repository
cargo r --example webrtc-precise-sync-recv --no-default-features
cargo run --bin gst-webrtc-signalling-server --no-default-features
Navigated to gstwebrtc-api. pnpm i && pnpm start
I’m seeing lot of artifacting immediately. Is there something I’m missing?
OS: Fedora Linux 41 (Workstation Edition)
Output of gst-launch-1.0 --version
gst-launch-1.0 version 1.24.10
GStreamer 1.24.10
http://download.fedoraproject.org
The webrtc-precise-sync recv / send examples handle configurations suitable for use cases where the producer & consumer honor specific synchronisation properties.
When you use the gstwebrtc-api example as a producer, you should use the default configuration you get from webrtcsrc. ex.:
It seems like you experience connectivity issues: for each screenshot, local time and sent packets increase while remote time and received packets remain the same.
Yup you’re right. But I don’t think it’s a network issue as I’m not facing this issue when the receiver is my browser.
To provide more details and a proper comparison, I took screenshots of GStreamer as receiver and browser as receiver (If you need any more details let me know)
