[gstreamer-bugs] [Bug 482541] Broken RTP timestamps

[GStreamer (bugzilla.gnome.org)]

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  GStreamer | gst-plugins-base | Ver: HEAD CVS

Wim Taymans changed:

           What    |Removed                     |Added
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                 CC|                            |wim.taymans at gmail.com




------- Comment #1 from Wim Taymans  2007-10-03 14:29 UTC -------
The timing rules for RTP as outlined in
http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/gst/rtp/README?revision=1.13&view=markup
state that the RTP timestamp in the payloaded packet equals the capture
timestamp of the packet at the sender side (which is the GStreamer timestamp).
We need this to reconstruct the timing at the sender in the normal "alsasrc !
amrnbenc ! rtpamrpay ! uspsink" case.

There is not really any way to get this right currently unless we start
creating the RTP timestamp from the buffer OFFSET field or so.


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