[gst-devel] Problem with RTP?

[Steffen Larsen]

Hi,

I already tried to use the audiotestsrc and fakesink, with the same  
outcome. I have under here showed what I get back from the console  
(debug) when I am testing it with alsasink and alsatestsrc.

the "server" side looks like this:

<SNIP>
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
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock

/pipeline0/rtpbin0/rtprecv0.datasrc: caps = application/x-rtp, media= 
(string)audio, payload=(int)110, clock-rate=(int)8000
^^^^^^^^^^^^^^^^^^^^
</SNIP>

Here I can see that I connects with my client, but no sound comes out  
of the speakers.. and yes they are working! ;-)


Here is the "client" side:

<SNIP>
hansson at ubuntu-laptop:~$ gst-launch-0.10 -v audiotestsrc ! speexenc !  
rtpspeexpay ! rtpbin localport=10000 destinations=127.0.0.1:7078
Setting pipeline to PAUSED ...
/pipeline0/audiotestsrc0.src: caps = audio/x-raw-int, endianness=(int) 
1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int) 
44100, channels=(int)1
Pipeline is PREROLLING ...
/pipeline0/speexenc0: last-message = "Warning: speex is optimized for  
8, 16 and 32 KHz"
/pipeline0/speexenc0.sink: caps = audio/x-raw-int, endianness=(int) 
1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int) 
44100, channels=(int)1
/pipeline0/speexenc0.src: caps = audio/x-speex, streamheader=(buffer) 
<  
5370656578202020312e312e313200000000000000000000000000000100000050000000 
44ac0000020000000400000001000000ffffffff80020000000000000100000000000000 
0000000000000000,  
1f000000456e636f6465642077697468204753747265616d6572205370656578656e6301 
0000001a0000004445534352495054494f4e3d617564696f74657374207761766501 >
/pipeline0/rtpspeexpay0.src: caps = application/x-rtp, media=(string) 
audio, payload=(int)110, clock-rate=(int)8000, encoding-name=(string) 
speex, ssrc=(guint)1089854986, clock-base=(guint)4275184238, seqnum- 
base=(guint)12973
/pipeline0/rtpspeexpay0.sink: caps = audio/x-speex, streamheader= 
(buffer)<  
5370656578202020312e312e313200000000000000000000000000000100000050000000 
44ac0000020000000400000001000000ffffffff80020000000000000100000000000000 
0000000000000000,  
1f000000456e636f6465642077697468204753747265616d6572205370656578656e6301 
0000001a0000004445534352495054494f4e3d617564696f74657374207761766501 >
/pipeline0/rtpbin0/rtpsend0.datasink: caps = application/x-rtp, media= 
(string)audio, payload=(int)110, clock-rate=(int)8000, encoding-name= 
(string)speex, ssrc=(guint)1089854986, clock-base=(guint)4275184238,  
seqnum-base=(guint)12973
/pipeline0/rtpbin0.sink: caps = application/x-rtp, media=(string) 
audio, payload=(int)110, clock-rate=(int)8000, encoding-name=(string) 
speex, ssrc=(guint)1089854986, clock-base=(guint)4275184238, seqnum- 
base=(guint)12973
/pipeline0/rtpbin0.sink: caps = application/x-rtp, media=(string) 
audio, payload=(int)110, clock-rate=(int)8000, encoding-name=(string) 
speex, ssrc=(guint)1089854986, clock-base=(guint)4275184238, seqnum- 
base=(guint)12973
/pipeline0/rtpbin0.sink.proxypad0: caps = application/x-rtp, media= 
(string)audio, payload=(int)110, clock-rate=(int)8000, encoding-name= 
(string)speex, ssrc=(guint)1089854986, clock-base=(guint)4275184238,  
seqnum-base=(guint)12973
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
</SNIP>


Hope some of you can understand what is wrong with my pipeline.. I  
has worked before (for 3-4 month ago), but now No sound comes out! :-(


-Cheers!
/Steffen





On Jul 19, 2007, at 9:18 AM, Steffen Larsen wrote:

> 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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