[Asterisk-Dev] Input and testing.

[Kevin P. Fleming]

Brian West wrote:

> I agree this would be something that would be nice to set per peer.  But
> first you have to know the codec then you have to know a valid frame size.
> The way asterisk is built you have no access to any channel structure at the
> point where this has to be set.  I was thinking of a mod to ast_rtp_write
> which is used in rtp channels...

Well, there is actually a bigger issue... if you don't expose this to 
the channel drivers, then they can't request that the peer act in the 
same way. The result of that would be Asterisk sending 60ms per frame 
(for example), but the peer continuing to send 20ms per frame. Also, 
it's possible (but not highly likely) that some RTP peers will not be 
able to accept frames with more audio than they are expecting, if the 
SDP did not tell them we planned on doing that.