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