[gst-devel] RTP_EOS

[Wim Taymans]

use gstrtpbin, for example, or use some other mechanism to signal the
end of an RTP stream.

Wim

On Wed, Apr 22, 2009 at 7:18 AM, vaisakh.n <vaisakh.n at nestgroup.net> wrote:
> Hi,
>
> I was able to stream mp3file through network  using rtp
> pipeline is this
>
> Seder
> /gst-launch-0.10 -v  filesrc location=1.mp3  ! mad  ! audioconvert !
> rtpL16pay  ! udpsink port=10000
>
>
> Receiver
> ./gst-launch-0.10 udpsrc port=10000 caps ="application/x-rtp,
> media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16,
> channels=(int)2, rate=(int)44100, payload=(int)96"  !   rtpL16depay !
> audioconvert ! osssink sync=false
>
>
> And I was able to hear sound clearly
>
> But problem is that Receiver pipeline not  ending...
> that is receiver not identifying the End Of Stream (EOS)
> How can i do that?
> help me?
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