[asterisk-dev] 80ms RTP packets and G.729

[Alistair Cunningham]

We have a customer who wants to run SIP with G.729 and 80ms RTP packets. 
They want to do this as they've been told by a 3rd party that this will 
allow them to keep bandwidth within 16Kbps per call, which is all they 
have available.

I see that there's a "packetization" option in Asterisk subversion. 
Alas, we can't use it as the G.729 codec doesn't run on this version:

  [codec_g729a.so]May  3 11:56:17 WARNING[31336]: loader.c:731 
__load_resource: misstng mod_data for codec_g729a.so

Would any of the Digium developers have an expected date for a G.729 
codec for an Asterisk version with the packetization code? Will it be 
before the next stable version?

How much effort would it be to port the packetization code to Asterisk 
1.2? If we did so, would the G.729 codec work with it?


(BTW: There's a trivial typo "misstng" in the above error message in SVN 
trunk revision 24378)

-- 
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/