[asterisk-users] Setting up RTP to flow between endpoints directly bypassing Asterisk

[Vikram Ragukumar]

Hello,

http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly

The link above indicates that it is possible to setup RTP streams to 
directly flow between endpoints and completely bypass Asterisk. I would 
like to know if this configuration would work when,

a) both endpoints are behind NAT, and/or
b) both endpoints don't support same codecs

with media flowing through a SIP+rtpproxy server that can do
transcoding ?

Thanks and Regards,
Vikram.