[gst-devel] Problem with RTP?

[Steffen Larsen]

HI all,

Me and my friend are implementing a larger multimedia implementation  
to Pidgin. In this one we have chosen to implement the voice / cam  
stream through Gstreamer. I know we could have chosen Farsight, but  
it is far to un-documented and we would like to have complete control  
over the streams.. But to the problem.
First of all, our microphones and speakers are working, but when we  
are trying to carry the sound or video over a RTP stream it does not  
play any sound!. We can see that the client connects to the server,  
but no sound comes out. It has earlier been working (I think it was  
back in april).

We have also tried to use gst-launch, which also worked earlier:

gst-launch-0.10 -v rtpbin localport=7078 pt-caps="application/x-rtp,  
media=(string)audio, payload=(int)110, clock-rate=(int)8000" !  
rtpspeexdepay ! speexdec ! alsasink sync=false

gst-launch-0.10 -v alsasrc ! speexenc ! rtpspeexpay ! rtpbin  
localport=10000 destinations=127.0.0.1:7078

Of course this is for a localhost, but we also tried with a remote  
host, and it does'nt work. All our other not-network stuff work, like  
playing a ogg or mp3 file through gst-launch:

gst-launch-0.10 filesrc location=sample.ogg ! oggdemux ! vorbisdec !  
audioconvert ! alsasink

And it is not any network firewall or router problems, because it is  
a local network (10.0.0.x addresses).

Do any of you have a clue of what it can be???? Have there been any  
updates that have been changing gst-launch, so it works in a  
different way? or is it the RTP that is fucked up at the moment? I  
know that is a bit experimental.
Any repons would be GREAT! :-)

  We are using gstreamer0.10.12 and is running it on a top of ubuntu  
linux 7.10 (feisty). I can't remember the kernel (I think it is  
2.20.x something, but i cant check it right now, I am on my mac. he he).


-Thanks for nice open source development for GStreamer.. I would  
personally be more involved in the project soon..

/Steffen & Jesper