[Live-devel] regarding codec standards which live media supports

[Ross Finlayson]

>1. I want to send a WAV file thru RTP packets. So, I used 
>"testWAVAudioStreamer" application to send a WAV file thru RTP. But, the 
>application internally uses RTSP session for handshake process and then it 
>uses the RTP and RTCP sessions for transfering the data. But, I want to 
>bypass the RTSP session over there. May I know the solution for the same?

Because "testWAVAudioStreamer" streams via multicast, it is possible for a 
client to play the stream without using RTSP, if it instead reads a 
SDP-format (".sdp") file that describes the stream.  (E.g., using VLC, you 
can open a ".sdp" file.)

To get the SDP description for the stream, use the "openRTSP" tool (in the 
"testProgs" directory) to 'play' the stream (from the "rtsp://" URL).  When 
it starts out, it will print the SDP description (starting with "v=").  If 
you copy this to a ".sdp" file, you should be able to use that file, in the 
future, to play the stream, without using RTSP.

>  2. Whether the applications "testMP3Streamer" and "testMP3Receiver" uses 
> purely the RTP packets to send and receive during the RTP session?

Yes, by default, they use RTP (and RTCP) only.   ("testMP3Streamer" has an 
optional RTSP server built in, but this is disabled by default.  See the 
source code.)

>  3. What are all the codec standards does the current 
> applications  support from the following list? ( G.711 (mu)Law, G.726-32, 
> GSM, G.723, G.711 aLaw, G.7 22, CD-quality audio, G.728, and G.729)?

The current "LIVE.COM Streaming Media" code can send or receive RTP streams 
that use pretty much any of the above codecs, and more.  (Note, though, 
that there is no actual decoding or encoding functionality present in the 
libraries.)


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