[Asterisk-Dev] Identifying associated channel in rtp.c

[Kevin P. Fleming]

Alf Thomas Nilsen wrote:

> I am hoping to be able to define the amount of voice data in each packet
> through extensions.conf. That way several conversations at the same time
> will be able to operate at different voice payloads.

You cannot do this in rtp.c. By the time the RTP layer gets the 
information, the audio has already been packed into a frame with an 
appropriate number of samples.

If you want the audio frames to contain a different number of samples, 
you have to handle that during the call setup (negotiation). However, 
given your stated lack of experience in dealing with this kind of code, 
I'd honestly say that you'd be better off hiring someone to do this for 
you... Trying to figure out how to do this by asking one question at a 
time via this list is going to take forever :-)

Why don't you start by explaining _exactly_ what you want Asterisk to 
do, including giving details of the SIP (or other) clients involved, 
codecs in use, etc.