[gst-devel] RTP Payload Status
robtaylor at floopily.org
Thu Apr 7 13:43:34 CEST 2005
[forgot to reply on list =)]
On Thu, 2005-04-07 at 12:00 -0300, Flavio Oliveira wrote:
> > This doesn't make senseto me; LiveMedia is another plugin framework
> > (vaguely) similar to gstreamer.
> Maybe I have not been clear, but I can be wrong.
> I understand that Live.com isn't similar to gstreamer. Live.com is
> to streaming applications, using open standard protocols (RTP/RTCP,
> RTSP, SIP). It is used to support RTP/RTPS in Players like MPlayer and
> Multimedia Streaming Servers like LiveCaster.
> GStreamer doens't support to RTP and RTSP, but I believe that it will
> have with Farsight RTP Plugin. Live.com implements so many RTP
> in LiveMedia library, so I intent to get the Payload implementatios
> uses it over Farsight RTP Plugin. Example, Farsight has a Plugin
> r263enc, and I think that you use it with RTP Plugin.
> So, we could have a LiveMedia Plugin where you can do MP3 Streaming
> using the RTP Plugin. I am thinking to create a LiveMedia Interface,
> something like you did with r263enc plugin that use the libr263.
> Really, I can need wrap livemedia objects as gstreamer objects to I/O
Right, I understand now. This is certainly an avenue worth
investigating. I've looked a little at this, and it seems like it might
be a large task to wrap these things, and may require a lot of
rewriting, but if we can, its a big bonus. I expect you wont need to use
the planned payloader base class here, so all thats needed is for
encoders to output application/x-rtp-noheaders and encoders to take
application/x-rtp. If you find encoders want to output full RTP packets,
let me know =)
Best of luck,
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