[Asterisk-Dev] T.38 pass-through / T.38 support
Kevin P. Fleming
kpfleming at digium.com
Sat Jun 4 15:52:55 MST 2005
Steve Underwood wrote:
> through *. What I assume happens is the ATA calls into *. * gives the
> above message. A G.711 call goes through to the Cisco. A re-invite
> connects those 2 directly, bypassing *. They chat amongst themselves,
> renegotiate, and talk UDPTL directly.
Jumping in late...
Even if this scenario is true, they cannot 'chat amongst themselves'. A
SIP reinvite moves _only_ the media path, not the signaling. Any
renegotiation of the media stream would still go through Asterisk, if
that's how the original call was placed.
I too am wondering just how Matthew Boehm's system is working; as of
this point, a T.38 call of any form (RTP or UDPTL) can't be handled by
chan_sip, and an existing call can't turn into T.38 either.
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