I'm currently working on an application that receives PCM audio from a TDM source as G711 A-Law. When the application plays out the audio over RTP the speech is asexpected.
The problem arises when I try to use the USC codec so that G729 compression codec can encode the G711 samples. The application will decode G729 frames thatare receivedover RTP and encodewith G729 for transmitted frames.
For simple testing, in the function that willencode the frame,I decode the encodedframe and playout over RTP (much like the example in the USC API). The payload that is played out is unrecognizable from the original G711 sample.
Could anybody explain what exactly has to be done to the decoded data so that the payload can beplayed out over RTP?
Any help would be greatly appreciated.