[asterisk-dev] RTP packetization (bugid 5162)

[Dan Austin]

The test branch oej opened for this feature has languished for quite
some time. (no offense to oej, he's got bigger fish to fry)

I ported the most recent patch in 5162 to SVN revision 37291 today, in
case anyone is still interested in this feature.  The new patch is
attached to 5162.

I also confirmed that the last patch for chan_ooh323 still applies to
the latest asterisk-addons svn trunk version.

Chan_sip and chan_ooh323 have been lightly tested here with no issues.

Lightly tested means calling into a meetme bridge with each channel and
confirming with rtp debug, and on the phones statistics page, that the
choosen payload was in use.

Dan