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

[A. Anand Krishnan]

Hi,

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?

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

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)?

Kindly assist me regarding the same.

Thanks,
A. Anand Krishnan



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