[gst-devel] gst-launch rtp problem with filesink and packet loss

[Marco Ballesio]

Hi,

On Mon, May 24, 2010 at 11:03 PM, Luca Gaggero
<luca_gaggero at fastwebnet.it> wrote:
>
> But I need to introduce in the comunication a packet loss.

How (at which ISO/OSI level) are you introducing this packet loss? If
you simply removed a portion of the original file this is the expected
behaviour ;).

> Now if I repeate the test with a packet loss the file is created, but
> the duration of the audio stream is less.
> I want to write a silence frame when the packet are missing, as like as
> a listener listen in real time the audio stream...

If timestamps are preserved correctly and the loss occurred in the
network you should have a silence.. Can you try with a pipeline like
the examples in:

http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-gstrtpbin.html

Regards

>
> Someone have a solution?
> I also write a programm with java media framework but I have the same
> problem...
>
>
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