Hi! I'm using Gstreamer to save RTP audio streams to wav files. This works in general, however, I need to deal with packet loss and in this case the pipeline I'm currently using fails: filesrc location=foobar.pcap ! pcapparse ! "application/x-rtp, payload=0, clock-rate=8000" ! gstrtpjitterbuffer ! rtppcmudepay ! mulawdec ! audioconvert ! audioresample ! wavenc ! filesink location=foobar.wav This produces audio files that are shorter than the original RTP stream, e.g. for a certain amount of consecutively lost packets a second of audio is missing from the wav file. For a real audio sink this doesn't happen as it plays audio when something is available, the filesink however -- I assume -- writes data sequentially and only when data is actually available. By the way: the effect is the same for udpsrc as it is for filesrc+pcapparse. I tried the above pipeline with filesink's sync property set to true ([1]), this didn't make any difference. I read somewhere in the Gstreamer docs that the global clock is [usually] dependent on the sink element. Do I need to reverse this and make the sink's clock dependent on the source's clock? 1. <<a href="http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer-libs/html/GstBaseSink.html#gst-base-sink-set-sync">http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer-libs/html/GstBaseSink.html#gst-base-sink-set-sync</a>> Maybe I'm totally off in this regard and there is no implicit way to achieve what I need. Ultimately some element in the pipeline needs to produce "silence/empty/zero data" if its input buffer is empty. Is there an explicit way (helper element/attribute) to do this? Thorsten