[gst-devel] gstrtpjitterbuffer packet lost

[gather bzbz]

Hi, 

I am developing a RTSP/RTP/UDP decoder based on GST. The decoder can work with the encoder (a hardware box) well (lip sync, etc). But randomly we will get the followings on the decoder, and whenever it happens, the audio stream stops. The encoder feeds one stream with one video sub-stream and one audio sub-stream (not a transport stream) in it. All the "Packet lost" always happens on the audio sub-stream side. 

This only happens when sync=true for alsasink. In the sync=false case, the audio can survive the packet lost case, but there is always a 2 seconds delay.

Also, the decoder and the encoder connects to a really dumb linksys switch, so the connection itself is really simple. The packet lost is probably not caused by network jitters, then who did it?

I searched the mailing lists for quite some time and realized that the timestamp management is tricky on the receiver side. But can someone please shed some light? Thanks!!

BTW, VLC and other softwares can play the streams just fine no matter what.

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0:04:44.624343259   892    0x5b9b8 WARN      gstrtpjitterbuffer gstrtpjitterbuffer.c:1730:gst_rtp_jitter_buffer_loop:<rtpjit
terbuffer0> Packet #30712 lost
0:04:44.625804074   892    0x5b9b8 WARN      gstrtpjitterbuffer gstrtpjitterbuffer.c:1730:gst_rtp_jitter_buffer_loop:<rtpjit
terbuffer0> Packet #30713 lost
0:04:44.627219667   892    0x5b9b8 WARN      gstrtpjitterbuffer gstrtpjitterbuffer.c:1730:gst_rtp_jitter_buffer_loop:<rtpjit
terbuffer0> Packet #30714 lost
0:04:44.628635296   892    0x5b9b8 WARN      gstrtpjitterbuffer gstrtpjitterbuffer.c:1730:gst_rtp_jitter_buffer_loop:<rtpjit
terbuffer0> Packet #30715 lost
0:04:44.630364445   892    0x5b9b8 WARN      gstrtpjitterbuffer gstrtpjitterbuffer.c:1730:gst_rtp_jitter_buffer_loop:<rtpjit
terbuffer0> Packet #30716 lost
0:04:44.632029259   892    0x5b9b8 WARN      gstrtpjitterbuffer gstrtpjitterbuffer.c:1730:gst_rtp_jitter_buffer_loop:<rtpjit
terbuffer0> Packet #30717 lost