Il Neofita wrote:
> Is there a way to force asterisk to take care only of sip signaling
> without forcing it to take care of rtp traffic?
Yes. The canonical way is to enable "canreinvite=yes" on both SIP
peers
(incoming and outgoing legs), which will cause Asterisk to send a new
INVITE within the dialog that has updated SDP information corresponding
to both endpoints.
The more interesting option is newer -- "directrtpsetup=yes" in
sip.conf. This will cause Asterisk to behave more like a proxy does
with respect to media and simply pass the SDP payloads as received to
both endpoints without pivoting the media stream toward itself at any
time, unless explicitly forced to do so (i.e. generating music on hold
or IVR messages).
Both approaches come with the caveat that the endpoints must be able to
address each other directly, so it can't be that one endpoint is behind
NAT on a private network that only Asterisk can see and the other
endpoint cannot. But if that's taken care of, or you have a far-end NAT
traversal solution in place to go with it, then you can do media release
on Asterisk.
-- Alex
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599