[pulseaudio-discuss] Listening to the RTP stream

Lennart Poettering lennart at poettering.net
Tue Sep 4 06:35:08 PDT 2007


On Tue, 04.09.07 13:36, Jan-Benedict Glaw (jbglaw at lug-owl.de) wrote:

> On Mon, 2007-09-03 19:11:47 +0200, Lennart Poettering <lennart at poettering.net> wrote:
> > On Mon, 03.09.07 13:49, Jan-Benedict Glaw (jbglaw at lug-owl.de) wrote:
> > 
> > > I played with the RTP sending module a bit. Seems to work, at least
> > > there's a RTP stream coming out of the box :)
> > > 
> > > However, are there alternative ways to listen to this/these streams
> > > except using the receiver module? Eg. in theory, mplayer should be
> > > able to listen to this stream, but it seems to have problems
> > > recognizing the contents. How do I actually tell it to use a specific
> > > PCM format?
> > 
> > Hmm. Good question. This used to work out of the box. We use the
> > standard type descriptors for S16BE uncompressed audio. Nothing
> > special, very basic stuff. 
> 
> "descriptors are in rtsp:// not in rtp://" is the answer[1] from the
> mplayer mailing list.  But the rtp module only send RTP, not RTSP,
> right?

Hmm, you forgot to add the footnote for [1]. 

We implement RTP and SAP, we don't implement RTSP. Since the primary
usecase for our RTP stuff is multicasting audio RTP+SAP makes far more sense
then RTP+RTSP.

OK, I looked this up in the archives now. Mr. Sabbi is plain
wrong. rtsp:// is for RTSP, rtp:// is for RTP.

> > From looking at the RTP RFCs, the "payload type" was probably ment,
> > instead of "type descriptors"?

Yes, you're right. It has been quite a while since I implemented
this... ;-)

> 
> > Another client for RTP that works fine is "dumprtp" from the dvbstream
> > package. And I heard some people had success with VLC. 
> 
> This indeed works, but is less convenient :)

Yes, that's right.

But still, it's a limitation/regression in MPlayer. On RTP they should
be using the payload type and not some kind of magic for detecting the
content type. Raw PCM data is very difficult to detect with
magic. Please file a bug to them.

BTW: afaik gst now has some rtp support
too:
  
   http://gstreamer.freedesktop.org/documentation/rtp.html 

Might be worth checking this out.

Lennart

-- 
Lennart Poettering                        Red Hat, Inc.
lennart [at] poettering [dot] net         ICQ# 11060553
http://0pointer.net/lennart/           GnuPG 0x1A015CC4



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