[Asterisk-Users] Asterisk codecs and packet size

[Michael Manousos]

Luis Vazquez wrote:
> Does anybody knows if it's posible or if there is some develoment in 
> course to be able to use longer transmit packet sizes (as long as I know 
> this is fixed in 20ms now) with the compressed voip codecs in asterisk 
> (g729, g726, gsm, etc).
> I need to use asterisk to connect remote sip clients with 24kb bandwidth 
> lines and I'm using a licences g729 codec but because I can't increase 
> the packet size to 40 or 60 ms in asterisk the connection is useless.

The quick and dirty way:
------------------------

In rtp.c, function "ast_rtp_write", in the "switch" statement,
"AST_FORMAT_G729A" case, change the smoother creation to something
larger. E.g.:

     rtp->smoother = ast_smoother_new(40);

Keep in mind that you must set this into something valid
(45 obviously is not valid). Recompile and you should be fine.


The right (but longer) way:
---------------------------

The ability to packetize variable number of frames per RTP
packet for various codecs should be configurable from within
the rtp.conf file. This requires some coding of course. Currently,
I don't have time available to do it, but I could do it as soon
as I find some free time.


> Thanks very much
> Luis
> 

Michael.