Hello,
I need to add support for a new audio format in Asterisk. It will be
actually a very limited support : I just need to take the audio from one
side and transmit it to the other side. No decoding / transcoding will
be involved. One side will be a DAHDI endpoint, the other one a SIP
endpoint.
Let's put aside for the moment the modifications which may have to be
done on the channels drivers to handle the codec negotiation. If I focus
on Asterisk's core and RTP handling, it seems I could achieve this
simply by registering a new format in codec_builtin.c. Two questions then :
- firstly, is this really the only thing to do, or am I missing something ?
- secondly, is there a more "pluggable" way to do this ? Maybe with a
shared object which would be loaded on startup ?
Best regards
Jean Aunis
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