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Bug 750281 - Fatal error on failed preroll of one of streams
Fatal error on failed preroll of one of streams
Status: RESOLVED OBSOLETE
Product: GStreamer
Classification: Platform
Component: gst-rtsp-server
git master
Other Linux
: Normal major
: git master
Assigned To: GStreamer Maintainers
GStreamer Maintainers
Depends on:
Blocks:
 
 
Reported: 2015-06-02 13:56 UTC by Yuuki
Modified: 2018-11-03 15:38 UTC
See Also:
GNOME target: ---
GNOME version: ---

Attachments
pcap dump (12.39 KB, application/x-pcapng)
2015-06-02 13:56 UTC, Yuuki 		Details

Description Yuuki 2015-06-02 13:56:41 UTC 
Created attachment 304429 [details]
pcap dump

I have camera which tells on DESCRIBE that it has audio stream, but in fact it is do not send any RTP packets of that SSRC. Using RTSP server to relay stream fails due to "preroll failed" error and shuts down client connection.
I guess it is not correct behavior - instead of fatal fault it needs to continue stream successfully prerolled streams.
Comment 1 Tim-Philipp Müller 2015-06-02 14:08:10 UTC 
This is an issue with rtspsrc not gst-rtsp-server, right?
Comment 2 Yuuki 2015-06-02 14:15:13 UTC 
gst-launch uridecodebin uri=rtsp://... ! autovideosink works pretty well.
gst-launch rtspsrc location=rtsp:// ! rtph264depay ! ... works good too.

As I see in rtsp-media.c:2120 status of media sets to ERROR state and returns FALSE in wait_preroll. If I hack it to not return FALSE, rtsp-client.c:2108 can't create sdp.

I do not think it is issue in rtspsrc.
Comment 3 Yuuki 2015-06-02 14:29:27 UTC 
If I hack gst-rtsp-server in rtsp-media.c to return GST_RTSP_MEDIA_STATUS_PREPARED instead of GST_RTSP_MEDIA_STATUS_ERROR in timeout case, then rtsp-server drops client connection.
Comment 4 Edward Hervey 2018-01-15 10:50:09 UTC 
Yuuki, do you still have this issue with current gst-rtsp-server (1.12 or master) ?
Comment 5 GStreamer system administrator 2018-11-03 15:38:50 UTC 
-- GitLab Migration Automatic Message --

This bug has been migrated to freedesktop.org's GitLab instance and has been closed from further activity.

You can subscribe and participate further through the new bug through this link to our GitLab instance: https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/12.