[asterisk-users] SIp Signalling

[Alex Balashov]

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/
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