On 11/30/2012 12:00 PM, deepak wrote:
> Hi all,
>
> I have developed a client that receives RTP and converts it to vorbis and
> then plays it.
> I have used udpsrc, rtpvorbisdepay, vorbisdec, audioconvert and alsasink in
> my pipeline.
> I have provided various properties such as caps for udpsrc element.
>
> The audio is being received by the client but there are breaks in the audio.
> The frequency of breaks is not fixed, they occur randomly. Can you suggest
> where I may be wrong.
You need to put a rtpjitterbuffer element after udpsrc to add some
buffering and to
compensate for packet reordering and packet loss.
Wim
_______________________________________________
gstreamer-devel mailing list
[hidden email]
http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
Locked
