[Asterisk-Dev] Input and testing.

[John Todd]

At 1:29 AM -0500 on 9/5/05, Brian West wrote:
>[patch] http://www.bkw.org/rtp.diff
>
>Before you ask why please compare some various options:
>http://www.packetizer.com/voip/diagnostics/bandcalc.html
>
>This is something asterisk should have had from day one along with a jitter
>buffer in rtp.  Please test and provide feedback before I post it to the bug
>tracker.
>
>/b


This is an excellent idea, and has been needed for a long time. 
(though I haven't tested the patch yet...)

However, I will be perhaps ungrateful and say that this would be 
optimally configured within each channel (and within each channel, 
for each peer.)  In the same way that choice of codec is quite 
important on a per-connection basis, I would also suggest that RTP 
frame interval is also important on a per-connection basis.  Some 
customers will work well at 40ms, while others need 20ms...  Setting 
a global value for this seems to not be quite as useful as being able 
to set it on a per-peer basis.  How difficult would it be to extend 
this very handy settings functionality across to the channel drivers 
that support RTP?

JT