[Live-devel] live.com streaming libraries and mobile phones

[Ross Finlayson]

> >As far as I know, '3GPP' uses the open standard RTSP/RTP/RTCP
> >protocols for
> >streaming, and uses the MPEG-4 video and AAC audio RTP payload
> >formats,
> >which the LIVE.COM code also supports.
>
>I thought that the free version of LIVE.COM code didn't support AAC audio.

When I wrote "AAC audio" above, I actually meant to say "AMR audio" - 
sorry.  However, the LIVE.COM code *does* support the sending and receiving 
of AAC (and aacPlus) audio RTP streams (the MIME type 
"audio/MPEG4-GENERIC") - using the "MPEG4GenericRTPSink" and 
"MPEG4GenericRTPSource" classes.  (There's even a third-part product out 
there <http://www.orban.com/orban/products/stream/1010_overview.html> that 
uses this.)

For example, you can run "openRTSP rtsp://mpeg4.sunsite.utk.edu/wuot.sdp" 
to make a recording of an AAC audio stream.  (The incoming audio data will 
be recorded as a raw data file.)  Or you can play this stream using the 
"VLC" or "MPlayer" media players (which both use the LIVE.COM code for 
RTSP/RTP client support).

Note, however, that ".m4a" (MPEG-4 audio) format *files* are not yet 
supported in the code, which is why there's currently no 
"testAACAudioStreamer" test program.

>Is it possible to stream some MPEG-4 video also?

Yes, you should be able to use "testMPEG4VideoStreamer" (for multicast) or 
"testOnDemandRTSPServer" (for unicast) as the server.  (Note that your 
"test.m4v" file must be a *raw* MPEG-4 video file.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>