[Asterisk-Dev] Codec not negotiating

[Kevin P. Fleming]

Jerris, Michael MI wrote:
> http://bugs.digium.com/bug_view_page.php?bug_id=0003346 should address 
> this issue, but there is not yet a patch with the implementation that 
> was decided upon yet.

While the patch for 3346 will possibly help with this situation, it 
won't solve the problem completely. The issue is that according the 
SIP/SDP RFCs, the receiver of an INVITE is free to choose any format 
offered in the INVITE, regardless of the order in which they are presented.

With that said, _most_ devices will honor the order in the SDP, and will 
use the first offered format if they are able to do so. That is why 
Asterisk will move the format of the calling channel to the top of the 
list in the outgoing INVITE.

The only way that this problem can be handled with 100% control is to 
only offer the codec you want to use in the outgoing INVITE. For now, 
that means using two different SIP peers for the outgoing calls.