[gst-devel] Problem with RTP?
zool at zool.dk
Thu Jul 19 09:18:11 CEST 2007
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
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
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