Some questions about gstreamer-rtsp-server

[Emmanuel Pacaud]

Hi,

In one of the software I'm developing, I'm using gst-rtsp 0.10.8 for the
streaming of a video coming from a GigEVision camera.

On the server side, the pipeline is built with a stream created with the
following code:

launch = g_strdup_printf ("( "
		  " appsrc name=source is-live=true stream-type=stream do-timestamp=true !"
		  " videorate skip-to-first=true ! video/x-raw-gray,framerate=%u/1 !"
		  " ffmpegcolorspace !"
		  " videoscale ! video/x-raw-yuv,width=%u,height=%u !"
		  " %s"
		  " x264enc speed-preset=ultrafast tune=zerolatency bitrate=800"
		  "   threads=0 sliced-threads=true intra-refresh=true"
		  "   option-string=\"slice-max-size=8192:vbv-maxrate=80:vbv-bufsize=10\" !"
		  " rtph264pay name=pay0 pt=96 "
		  " )",
		  server->frame_rate, server->width, server->height,
		  show_time ? "cairotimeoverlay !" : "");

On the client, I'm using the following pipeline:

gst-launch rtspsrc location="rtsp://server:8554/cam1" latency=10 buffer-mode=auto ! decodebin ! autovideosink

I'm trying to have the minimum latency for the video output.

After a day or two of use, I start to see an increased latency, below 1
second, but greater than what I get when I start the client pipeline
(1-2 frames).

At this point, if I connect a second client, the latency is the same as
the first client.

If I kill all the clients, wait for a while (until the server pipeline
is automatically destroyed), and connect a new client, the latency is
back to the good value.

Hence my questions:

- Do you have an hint on where the latency may increase in the server
pipeline ?
- How do I have to proceed in order to debug the latency in the server
pipeline ?

Two somewhat related questions:

- Is the buffer timestamp transmitted over the rtsp stream ? 
- What is the best way to get the buffer timestamp in a receiving
pipeline ?

Thanks in advance,

	Cheers,

		Emmanuel.