[asterisk-dev] directrtpsetup broken in 1.8.3.2 (since at least 1.8.1) ?

[Marcelo Pacheco]

Hello,

My central Asterisk system has been running with directrtpsetup=yes for 
3 years now.
Early 2011 I migrated from 1.6.2.0 to 1.8.2-rc1.
Recently I noticed that directrtpsetup is broken.
The initial INVITE is going out with the central Asterisks IP on SDP c=, 
instead of the originating Asterisk IP on SDP c=
Then once the call is answered, a re-INVITE is sent out changing the c= 
to the originating leg.

All SIP peers/friends/users have nat=never and directmedia=yes. Until 
today I had canreinvite=yes, but since directmedia is superseeding 
canreinvite, I just changed canreinvite=yes with directmedia=yes everywhere.

All peers have a VALID IP with no NAT.
This defeats half of the purpose of directrtpsetup.

I downgraded to Asterisk 1.6.2.17.2, and got back to the expected 
behavior, where a simple DIAL between two SIP endpoints (both with 
NAT=NO and DIRECTMEDIA=YES) causes the SDP to be passed through keeping 
Asterisk 100% outside the media path.

Was this a mistake (bug) or an intentional change ?

Any thoughts would be appreciated.
Hopefully this can be fixed in Asterisk 1.8.4 or 1.10.
Just need some guidance before creating a bug on Bugzilla.

Regards,

Marcelo Pacheco
Running Asterisk since the early days prior to Asterisk 1.0.