[gst-devel] RTP sync and sender report ntp timestamps

[Jon Burgess]

Hi all,

I'm currently looking at syncing of two independent RTP sources (one video,
one audio), both of which are recorded by our application to be played back
later, in sync.

I understand that part of the RTP/RTCP spec is that RTCP sender reports sent
periodically (e.g. every few seconds) contain a ntp timestamp which
correlates ntp time to RTP timestamps, in theory allowing accurate
synchronisation between multiple sources.

My question is: do any of the gstreamer rtp related elements take account of
the ntp timestamps in sender reports and adjust buffer timestamps
accordingly?  Which element(s) do RTCP processing?

My video source is a RTSP/mpeg4 part2 network encoder, and the relevant
gst-launch equivalent of my pipeline looks something like:

> gst-launch uridecodebin uri="rtsp://someurl" ! queue ! decodebin2 ....

in case that can help somebody direct me...

The audio source is received by something other than a g-streamer pipeline
btw.

Thanks in advance,
Jono
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