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

[Jeff Brower]

Vikram-

> 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 ?

This would be 'native bridging' mode as I've seen it described a few places on the web, correct?  If Asterisk is "out
of the RTP loop", then what can it still do?  Only billing?  It would not detect DTMF, no RTP record or announcement
playout, etc.

I'm not clear on whether anyone actually uses Asterisk in this mode and if so, for what reason.

-Jeff