[gst-devel] Dinamically managing the buffer size in the RTP client side

Tue Mar 31 23:36:04 CEST 2009

Hi,

In order to provide with a seamless video streaming service to the user when
performing a handover between two different networks, I would like to
increase the buffer size in the client side while receiving RTSP/RTP data,
so the period of time that the client device is not attached to any network
can be "hidden" to the user while playing the buffered content. Once the new
connection has successfully been established, the streaming session could be
continued through this new link and the buffer size configured back to the
previous value.

So, it is possible to dinamically change the buffer size of the
corresponding element or it must be configured prior to the streaming
session? Any suggestion will be welcome.

Regards,
Javi
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