Hi,

I don't really have time to look at your stuff but you should do some
basic gstreamer debugging. Run your program with :
GST_DEBUG="rtpbin:5, rtpsend:5, rtprecv:5" and look for any suspicious
messages.

Regards,
Philippe

On Wed, 10 Jan 2007 00:22:03 -0700
Scott Zuk <[EMAIL PROTECTED]> wrote:

> Hi,
> 
> I'm trying to use the rtpbin element in a pipeline that I want to set up 
> initially to receive rtp data but change to send and receive when a 
> destination and send codec is known.  I have attached a small test program 
> that shows what I am doing.
> 
> Basically, I create a pipeline and add rtpbin and rtpdemux elements and set 
> the pipeline to playing to start receiving data.  After a few seconds I add 
> an audio source, encoder and payloader to the pipeline, link them up and set 
> a destination ip and port.  The problem though is that no rtp data ever gets 
> to the network and hence the rtp payload callbacks are never called (I'm 
> sending to the listen port).  Not sure why this is happening.
> 
> However, if I add a fakesrc element to the pipeline and link it to the sink 
> pad of the rtpbin and remove it immediately after setting the pipeline to 
> playing, sending rtp audio will then be successful.  It seems like a hack to 
> me though.  You can activate this behavior in my test program by passing 
> the --use-fakesrc argument.  It seems that linking the fakesrc to the rtpbin 
> sink causes something to be initialized differently but I don't know what.  
> Any ideas?
> 
> BTW, I'm using a copy of the latest rtpbin from darcs compiled without jrtp 
> and jingle and gstreamer 0.10.10
> 
> Thanks,
> ~Scott

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