[gst-devel] RE : Problem with RTP?

[SP GLE]

Hi,

Some tips :
- replace alsasrc by audiotestsrc
- replace alsasink by fakesink dump=true

This way you will have information about effective path of buffers. You
can also add an identity with dump=true before each rtpbin to see
incoming/outgoing buffer and see where the problem comes from (some
gst-plugin-farsight elements are buggy).

Using lsof you can look at opened udp sockets by rtpbin (and see if
it's the right port number).

For more details you will have to use GST_DEBUG=4 or (worst!)
GST_DEBUG=5, in our application some buffers was lost inside rtpbin and
only a debug level of 5 helped us.

Regards.


--- Steffen Larsen <zool at zool.dk> a écrit :

> 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
> 
> 
> 
> 
> 
> 
> 
>
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